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From tfrazee at gmail.

com Wed Jan 2 09:50:26 2013


From: tfrazee at gmail.com (Tim Frazee)
Date: Wed, 2 Jan 2013 08:50:26 -0600
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
Message-ID: <CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>

I didnt test to see if the 9.1 from CCO is bootable. In the past they
havent been.

attached is a screenshoot of the error I received when I tried to feed the


9.1 via CCO during a booted-from-nfr9.0 media

On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at ldschurch.org>wrote:

> I had lots of problems doing upgrade during installs with 9.0 ESs. The
> ESs are usually bootable so I just gave up and installed fresh. Is the 9.1
> download bootable?****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Tim Frazee
> *Sent:* Monday, December 31, 2012 3:50 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] upgrade during install from 9.0 to 9.1****
>
> ** **
>
> This was for UCM and Unity Connection. didnt try anything else.
>
>
> ****
>
> On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com> wrote:****
>
> I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.
>
> I'm getting an error if I feed the 9.1 iso to the 9.0 install process that
> i want to upgrade during the install process. I've been able to do this
> many times in the past with never a problem like this.
>
> Anyone have any ideas?****
>
> ** **
>
>
>
> NOTICE: This email message is for the sole use of the intended
> recipient(s) and may contain confidential and privileged information. Any
> unauthorized review, use, disclosure or distribution is prohibited. If you
> are not the intended recipient, please contact the sender by reply email
> and destroy all copies of the original message.****
>
>
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From VanMarenNP at ldschurch.org Wed Jan 2 10:07:26 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Wed, 2 Jan 2013 15:07:26 +0000
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E3F97A@W12112.ldschurch.org>

Well that's different than what I was seeing, mine would crash during the install,
instead of saying you can't do it.

And I tried to boot from the CCO version of 9.1, it doesn't.

So if you really need to be able to install straight to 9.1, it looks like you'll
need to have TAC post the bootable iso for you.

-Nate

From: Tim Frazee [mailto:tfrazee at gmail.com]


Sent: Wednesday, January 02, 2013 7:50 AM
To: Nate VanMaren
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

I didnt test to see if the 9.1 from CCO is bootable. In the past they havent been.

attached is a screenshoot of the error I received when I tried to feed the 9.1 via
CCO during a booted-from-nfr9.0 media
On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at
ldschurch.org<mailto:VanMarenNP at ldschurch.org>> wrote:
I had lots of problems doing upgrade during installs with 9.0 ESs. The ESs are
usually bootable so I just gave up and installed fresh. Is the 9.1 download
bootable?

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-
bounces at puck.nether.net>] On Behalf Of Tim Frazee
Sent: Monday, December 31, 2012 3:50 PM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

This was for UCM and Unity Connection. didnt try anything else.

On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com<mailto:tfrazee at
gmail.com>> wrote:
I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.

I'm getting an error if I feed the 9.1 iso to the 9.0 install process that i want
to upgrade during the install process. I've been able to do this many times in the
past with never a problem like this.

Anyone have any ideas?

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

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From rratliff at cisco.com Wed Jan 2 10:47:42 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 2 Jan 2013 10:47:42 -0500
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
Message-ID: <A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>

9.0.0.99101-22 is not a 9.0 ES, it's a pre-release build of 9.1. Throw it in the
trash and try with a real 9.0 build (I'm going to start this now).

-Ryan

On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com> wrote:

I didnt test to see if the 9.1 from CCO is bootable. In the past they havent been.

attached is a screenshoot of the error I received when I tried to feed the 9.1 via
CCO during a booted-from-nfr9.0 media
On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at ldschurch.org> wrote:
I had lots of problems doing upgrade during installs with 9.0 ESs. The ESs are
usually bootable so I just gave up and installed fresh. Is the 9.1 download
bootable?

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Tim Frazee
Sent: Monday, December 31, 2012 3:50 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

This was for UCM and Unity Connection. didnt try anything else.

On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com> wrote:

I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.

I'm getting an error if I feed the 9.1 iso to the 9.0 install process that i want
to upgrade during the install process. I've been able to do this many times in the
past with never a problem like this.

Anyone have any ideas?

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

<temp.png>_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From tfrazee at gmail.com Wed Jan 2 11:18:21 2013


From: tfrazee at gmail.com (Tim Frazee)
Date: Wed, 2 Jan 2013 10:18:21 -0600
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
Message-ID: <CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>

I did just that. after I tried with the pre-release, I used my NFR iso.
Same result.

I only used the pre-release because it was already on my datastore and i


was feeling a bit lazy over vacation. After I attempted the same procedure
with 9.0(1) -37 iso, I received the exact same error.

Ryan, should I be able to boot off of 9.0 and upgrade-during-install with


9.1?

On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com> wrote:

> 9.0.0.99101-22 is not a 9.0 ES, it's a pre-release build of 9.1. Throw
> it in the trash and try with a real 9.0 build (I'm going to start this
> now).
>
> -Ryan
>
> On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com> wrote:
>
> I didnt test to see if the 9.1 from CCO is bootable. In the past they
> havent been.
>
> attached is a screenshoot of the error I received when I tried to feed the
> 9.1 via CCO during a booted-from-nfr9.0 media
>
> On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at
ldschurch.org>wrote:
>
>> I had lots of problems doing upgrade during installs with 9.0 ESs. The
>> ESs are usually bootable so I just gave up and installed fresh. Is the 9.1
>> download bootable?****
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> *From:* cisco-voip-bounces at puck.nether.net [mailto:
>> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Tim Frazee
>> *Sent:* Monday, December 31, 2012 3:50 PM
>> *To:* cisco-voip at puck.nether.net
>> *Subject:* Re: [cisco-voip] upgrade during install from 9.0 to 9.1****
>>
>> ** **
>>
>> This was for UCM and Unity Connection. didnt try anything else.
>>
>>
>> ****
>>
>> On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com> wrote:***
>> *
>>
>> I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.
>>
>> I'm getting an error if I feed the 9.1 iso to the 9.0 install process
>> that i want to upgrade during the install process. I've been able to do
>> this many times in the past with never a problem like this.
>>
>> Anyone have any ideas?****
>>
>> ** **
>>
>>
>>
>> NOTICE: This email message is for the sole use of the intended
>> recipient(s) and may contain confidential and privileged information. Any
>> unauthorized review, use, disclosure or distribution is prohibited. If you
>> are not the intended recipient, please contact the sender by reply email
>> and destroy all copies of the original message.****
>>
>>
> <temp.png>_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From rratliff at cisco.com Wed Jan 2 13:21:46 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 2 Jan 2013 13:21:46 -0500
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
<CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
Message-ID: <00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>

Confirmed I see it here in the lab and it looks to be intentional, though I'm still
digging.
Initial word is for a while now upgrade-during-install is only supported to the
same major/minor version.

Anything beyond that requires a separate upgrade after install.

-Ryan

On Jan 2, 2013, at 11:18 AM, Tim Frazee <tfrazee at gmail.com> wrote:

I did just that. after I tried with the pre-release, I used my NFR iso. Same
result.

I only used the pre-release because it was already on my datastore and i was
feeling a bit lazy over vacation. After I attempted the same procedure with 9.0(1)
-37 iso, I received the exact same error.

Ryan, should I be able to boot off of 9.0 and upgrade-during-install with 9.1?

On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com> wrote:
9.0.0.99101-22 is not a 9.0 ES, it's a pre-release build of 9.1. Throw it in the
trash and try with a real 9.0 build (I'm going to start this now).

-Ryan

On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com> wrote:

I didnt test to see if the 9.1 from CCO is bootable. In the past they havent been.

attached is a screenshoot of the error I received when I tried to feed the 9.1 via
CCO during a booted-from-nfr9.0 media

On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at ldschurch.org> wrote:
I had lots of problems doing upgrade during installs with 9.0 ESs. The ESs are
usually bootable so I just gave up and installed fresh. Is the 9.1 download
bootable?

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Tim Frazee
Sent: Monday, December 31, 2012 3:50 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

This was for UCM and Unity Connection. didnt try anything else.

On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com> wrote:

I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.

I'm getting an error if I feed the 9.1 iso to the 9.0 install process that i want
to upgrade during the install process. I've been able to do this many times in the
past with never a problem like this.

Anyone have any ideas?

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

<temp.png>_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From ewellnitzvoip at gmail.com Wed Jan 2 14:15:26 2013


From: ewellnitzvoip at gmail.com (Erick Wellnitz)
Date: Wed, 2 Jan 2013 13:15:26 -0600
Subject: [cisco-voip] (no subject)
Message-ID: <CAK0wOsBfkmgyRKKBTgCB0BVLQs5FZEdMjo54taRs8S7SgG4Gyg@mail.gmail.com>

Hello all and happy new year!

I'm having a database connection issue with CUEAC 8.6

At intermittent intervals the client is unable to connect with an error


saying it can not connect to the database. Tried repairing the DB from the
database management section of the web gui with no difference. The only
thing that seems to resolve the issue temporarily is to restart the
attendant console server service.

I see a ton of the following in the log.

2/1/2013
7:50:23.034|
M|"AmendContactNumber","Request","IPA:10.12.52.46","IPP:49986","NUR:N00100161"
2/1/2013
7:50:23.034|M|"AmendContactNumber","Fail(DB-
>ICDEC_INVALIDCONTACTNOREF)","IPA:10.12.52.46","IPP:49986","NUR:N00100161","CUR:CTH
100081"
2/1/2013
7:50:33.093|
M|"AmendContactNumber","Request","IPA:10.12.52.46","IPP:49986","NUR:N00100161"

Anyone seen this before? TAC case 624321291 is open but I'm not having
much success getting more than silence from them.

Thanks!
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From tfrazee at gmail.com Wed Jan 2 16:12:56 2013


From: tfrazee at gmail.com (Tim Frazee)
Date: Wed, 2 Jan 2013 15:12:56 -0600
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
<CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
<00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>
Message-ID: <CABzsfHz0Km6uPom_mhLkU9iMiZMawubm2JXAG1+uaJ+_wKACug@mail.gmail.com>

I could see that.

But tell that to my 8.5 upgrade from an 8.0 boot disk I was able to do back
in the day......

In short, you say that the only way currently to get to 9.1 is upgrade from
an already installed support version, not during the install process.

for the record and I know its not supported, I did try the hack of grabbing
the boot info file from 9.0 and pushing it into the 9.1 iso. The install
process failed post installing everything.

Thanks for digging Ryan.

On Wed, Jan 2, 2013 at 12:21 PM, Ryan Ratliff <rratliff at cisco.com> wrote:

> Confirmed I see it here in the lab and it looks to be intentional, though
> I'm still digging.
> Initial word is for a while now upgrade-during-install is only supported
> to the same major/minor version.
>
> Anything beyond that requires a separate upgrade after install.
>
> -Ryan
>
> On Jan 2, 2013, at 11:18 AM, Tim Frazee <tfrazee at gmail.com> wrote:
>
> I did just that. after I tried with the pre-release, I used my NFR iso.
> Same result.
>
> I only used the pre-release because it was already on my datastore and i
> was feeling a bit lazy over vacation. After I attempted the same procedure
> with 9.0(1) -37 iso, I received the exact same error.
>
> Ryan, should I be able to boot off of 9.0 and upgrade-during-install with
> 9.1?
>
> On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com> wrote:
>
>> 9.0.0.99101-22 is not a 9.0 ES, it's a pre-release build of 9.1. Throw
>> it in the trash and try with a real 9.0 build (I'm going to start this
>> now).
>>
>> -Ryan
>>
>> On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com> wrote:
>>
>> I didnt test to see if the 9.1 from CCO is bootable. In the past they
>> havent been.
>>
>> attached is a screenshoot of the error I received when I tried to feed
>> the 9.1 via CCO during a booted-from-nfr9.0 media
>>
>> On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at
ldschurch.org>wrote:
>>
>>> I had lots of problems doing upgrade during installs with 9.0 ESs.
>>> The ESs are usually bootable so I just gave up and installed fresh. Is the
>>> 9.1 download bootable?****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> *From:* cisco-voip-bounces at puck.nether.net [mailto:
>>> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Tim Frazee
>>> *Sent:* Monday, December 31, 2012 3:50 PM
>>> *To:* cisco-voip at puck.nether.net
>>> *Subject:* Re: [cisco-voip] upgrade during install from 9.0 to 9.1****
>>>
>>> ** **
>>>
>>> This was for UCM and Unity Connection. didnt try anything else.
>>>
>>>
>>> ****
>>>
>>> On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com> wrote:**
>>> **
>>>
>>> I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.
>>>
>>> I'm getting an error if I feed the 9.1 iso to the 9.0 install process
>>> that i want to upgrade during the install process. I've been able to do
>>> this many times in the past with never a problem like this.
>>>
>>> Anyone have any ideas?****
>>>
>>> ** **
>>>
>>>
>>>
>>> NOTICE: This email message is for the sole use of the intended
>>> recipient(s) and may contain confidential and privileged information. Any
>>> unauthorized review, use, disclosure or distribution is prohibited. If you
>>> are not the intended recipient, please contact the sender by reply email
>>> and destroy all copies of the original message.****
>>>
>>>
>> <temp.png>_______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
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From rratliff at cisco.com Wed Jan 2 16:22:07 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 2 Jan 2013 16:22:07 -0500
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <CABzsfHz0Km6uPom_mhLkU9iMiZMawubm2JXAG1+uaJ+_wKACug@mail.gmail.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
<CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
<00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>
<CABzsfHz0Km6uPom_mhLkU9iMiZMawubm2JXAG1+uaJ+_wKACug@mail.gmail.com>
Message-ID: <5992A434-99D1-406D-894A-02F51A195AA7@cisco.com>

I was told this restriction was added around 8.5 but I'm still waiting on some
other folks to comment.

To get to 9.1 you either do a fresh install or you upgrade, same as any other
version. I understand the release of 9.1 has immediately replaced 9.0 on new 9.x
orders (much like 8.6 did for 8.5) so any 9.x media kit ordered today will be sent
9.1 bootable media.

-Ryan

On Jan 2, 2013, at 4:12 PM, Tim Frazee <tfrazee at gmail.com> wrote:

I could see that.

But tell that to my 8.5 upgrade from an 8.0 boot disk I was able to do back in the
day......

In short, you say that the only way currently to get to 9.1 is upgrade from an
already installed support version, not during the install process.

for the record and I know its not supported, I did try the hack of grabbing the
boot info file from 9.0 and pushing it into the 9.1 iso. The install process failed
post installing everything.

Thanks for digging Ryan.

On Wed, Jan 2, 2013 at 12:21 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
Confirmed I see it here in the lab and it looks to be intentional, though I'm still
digging.
Initial word is for a while now upgrade-during-install is only supported to the
same major/minor version.

Anything beyond that requires a separate upgrade after install.

-Ryan

On Jan 2, 2013, at 11:18 AM, Tim Frazee <tfrazee at gmail.com> wrote:

I did just that. after I tried with the pre-release, I used my NFR iso. Same
result.

I only used the pre-release because it was already on my datastore and i was
feeling a bit lazy over vacation. After I attempted the same procedure with 9.0(1)
-37 iso, I received the exact same error.

Ryan, should I be able to boot off of 9.0 and upgrade-during-install with 9.1?

On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com> wrote:
9.0.0.99101-22 is not a 9.0 ES, it's a pre-release build of 9.1. Throw it in the
trash and try with a real 9.0 build (I'm going to start this now).

-Ryan

On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com> wrote:

I didnt test to see if the 9.1 from CCO is bootable. In the past they havent been.

attached is a screenshoot of the error I received when I tried to feed the 9.1 via
CCO during a booted-from-nfr9.0 media

On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at ldschurch.org> wrote:
I had lots of problems doing upgrade during installs with 9.0 ESs. The ESs are
usually bootable so I just gave up and installed fresh. Is the 9.1 download
bootable?

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Tim Frazee
Sent: Monday, December 31, 2012 3:50 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

This was for UCM and Unity Connection. didnt try anything else.

On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com> wrote:

I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.
I'm getting an error if I feed the 9.1 iso to the 9.0 install process that i want
to upgrade during the install process. I've been able to do this many times in the
past with never a problem like this.

Anyone have any ideas?

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

<temp.png>_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From MLoraditch at heliontechnologies.com Wed Jan 2 16:34:00 2013


From: MLoraditch at heliontechnologies.com (Matthew Loraditch)
Date: Wed, 2 Jan 2013 21:34:00 +0000
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <5992A434-99D1-406D-894A-02F51A195AA7@cisco.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
<CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
<00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>
<CABzsfHz0Km6uPom_mhLkU9iMiZMawubm2JXAG1+uaJ+_wKACug@mail.gmail.com>
<5992A434-99D1-406D-894A-02F51A195AA7@cisco.com>
Message-ID: <C75AF2AD9308C246AFBDDB994E3E298311081C47@PHANES.helion.local>

Well that's good, I can just put a PUT order in edelivery and get it. Let's see if
it works.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284
Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, January 02, 2013 4:22 PM
To: Tim Frazee
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

I was told this restriction was added around 8.5 but I'm still waiting on some
other folks to comment.

To get to 9.1 you either do a fresh install or you upgrade, same as any other
version. I understand the release of 9.1 has immediately replaced 9.0 on new 9.x
orders (much like 8.6 did for 8.5) so any 9.x media kit ordered today will be sent
9.1 bootable media.

-Ryan

On Jan 2, 2013, at 4:12 PM, Tim Frazee <tfrazee at gmail.com<mailto:tfrazee at


gmail.com>> wrote:

I could see that.

But tell that to my 8.5 upgrade from an 8.0 boot disk I was able to do back in the
day......

In short, you say that the only way currently to get to 9.1 is upgrade from an
already installed support version, not during the install process.

for the record and I know its not supported, I did try the hack of grabbing the
boot info file from 9.0 and pushing it into the 9.1 iso. The install process failed
post installing everything.

Thanks for digging Ryan.

On Wed, Jan 2, 2013 at 12:21 PM, Ryan Ratliff <rratliff at


cisco.com<mailto:rratliff at cisco.com>> wrote:
Confirmed I see it here in the lab and it looks to be intentional, though I'm still
digging.
Initial word is for a while now upgrade-during-install is only supported to the
same major/minor version.

Anything beyond that requires a separate upgrade after install.

-Ryan

On Jan 2, 2013, at 11:18 AM, Tim Frazee <tfrazee at gmail.com<mailto:tfrazee at


gmail.com>> wrote:

I did just that. after I tried with the pre-release, I used my NFR iso. Same
result.

I only used the pre-release because it was already on my datastore and i was
feeling a bit lazy over vacation. After I attempted the same procedure with 9.0(1)
-37 iso, I received the exact same error.

Ryan, should I be able to boot off of 9.0 and upgrade-during-install with 9.1?
On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com<mailto:rratliff
at cisco.com>> wrote:
9.0.0.99101-22<tel:9.0.0.99101-22> is not a 9.0 ES, it's a pre-release build of
9.1. Throw it in the trash and try with a real 9.0 build (I'm going to start this
now).

-Ryan

On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com<mailto:tfrazee at


gmail.com>> wrote:

I didnt test to see if the 9.1 from CCO is bootable. In the past they havent been.

attached is a screenshoot of the error I received when I tried to feed the 9.1 via
CCO during a booted-from-nfr9.0 media
On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at
ldschurch.org<mailto:VanMarenNP at ldschurch.org>> wrote:
I had lots of problems doing upgrade during installs with 9.0 ESs. The ESs are
usually bootable so I just gave up and installed fresh. Is the 9.1 download
bootable?

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-
bounces at puck.nether.net>] On Behalf Of Tim Frazee
Sent: Monday, December 31, 2012 3:50 PM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

This was for UCM and Unity Connection. didnt try anything else.

On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com<mailto:tfrazee at
gmail.com>> wrote:
I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.

I'm getting an error if I feed the 9.1 iso to the 9.0 install process that i want
to upgrade during the install process. I've been able to do this many times in the
past with never a problem like this.

Anyone have any ideas?

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

<temp.png>_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From robhass at gmail.com Wed Jan 2 20:47:04 2013


From: robhass at gmail.com (Robert Hass)
Date: Thu, 3 Jan 2013 02:47:04 +0100
Subject: [cisco-voip] Call Recording on CUCM
Message-ID: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>

Hi
My boss asked if we can enable call recording on our CUCM 8.6 (just CUCM
without Contact Center).
We considering two options of call recording
a) record all voice calls
b) record voice calls on demand - user can turn on/off recording via xml
application of softkey on the phone

My question : Are above scenarios of call recording are possible on CUCM ?


What else I need - probably server for call-recording with big amount of
storange and some additional software (Zoom ? Cisco ?)

thanks for help

Rob
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From JOrr at parknationalbank.com Wed Jan 2 21:05:01 2013


From: JOrr at parknationalbank.com (Orr, Jeff B.)
Date: Thu, 3 Jan 2013 02:05:01 +0000
Subject: [cisco-voip] Call Recording on CUCM
In-Reply-To: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
References: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
Message-ID: <72FE638DB23C1049AA265B28B5F87F3364A56940@prk-alford-mbx1.PRK.LOCAL>

Hi Rob,

I just went through this for our environment. Call manager will provide the backend
requirements to do recordings. However, you will need a 3rd party software to
actually record and store the calls.

We evaluated several options and went with Zoom. It is a nice, Linux based
recording software. It fully supports spanless recordings and can function as
record all the time or on-demand recording. It does this by actually recording
every call, and then allowing a user to press a service button to record a call
that occurred earlier.
Jeff

____________________________________
Jeff Orr
Technical Services - Network Engineer
Park National Corporation (AMEX: PRK)

This message is confidential and is intended only for the named recipients, and may
contain information that is privileged, or exempt from disclosure under applicable
law. If you are not the intended recipients of the email, you are hereby notified
that the dissemination, distribution or copying of this e-mail or its contents is
strictly prohibited. If you received this e-mail in error, please notify the sender
at either the e-mail address or the phone number above and delete this e-mail from
your computer.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Robert Hass
Sent: Wednesday, January 02, 2013 8:47 PM
To: cisco-voip
Subject: [cisco-voip] Call Recording on CUCM

Hi
My boss asked if we can enable call recording on our CUCM 8.6 (just CUCM without
Contact Center).
We considering two options of call recording
a) record all voice calls
b) record voice calls on demand - user can turn on/off recording via xml
application of softkey on the phone

My question : Are above scenarios of call recording are possible on CUCM ? What
else I need - probably server for call-recording with big amount of storange and
some additional software (Zoom ? Cisco ?)

thanks for help

Rob

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From erickbee at gmail.com Wed Jan 2 22:32:24 2013


From: erickbee at gmail.com (Erick B)
Date: Wed, 2 Jan 2013 21:32:24 -0600
Subject: [cisco-voip] Call Recording on CUCM
In-Reply-To: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
References: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
Message-ID: <44547C16-7070-4462-9213-8661B44DBD8B@gmail.com>

Yes, you'll need zoom or another 3rd party recording application.

On recent cucm versions, you enable built in bridge on newer model phones then
assign a recording profile to the DN on the phone you want to record. The recording
has the IP address of recording server (sip trunk) the phone will send the audio
to.
You can do it the old way with span ports to on switches, depends on recording
application you are using and where the phones are if span works easily or not.

Sent from my iPhone

On Jan 2, 2013, at 7:47 PM, Robert Hass <robhass at gmail.com> wrote:

> Hi
> My boss asked if we can enable call recording on our CUCM 8.6 (just CUCM without
Contact Center).
> We considering two options of call recording
> a) record all voice calls
> b) record voice calls on demand - user can turn on/off recording via xml
application of softkey on the phone
>
> My question : Are above scenarios of call recording are possible on CUCM ? What
else I need - probably server for call-recording with big amount of storange and
some additional software (Zoom ? Cisco ?)
>
> thanks for help
>
> Rob
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From ckos1976 at hotmail.com Thu Jan 3 05:43:26 2013


From: ckos1976 at hotmail.com (costas georgiou)
Date: Thu, 3 Jan 2013 10:43:26 +0000
Subject: [cisco-voip] No access to Publisher
In-Reply-To: <CAKav0XS31BFuyEjFykKRx5r5fWr2cT-f5d67KpwMK1MO5PfMOg@mail.gmail.com>
References: <BLU174-W3DD4536E1CF5AD7768B1FD2370@phx.gbl>
<006901cdde95$ddc4ad80$994e0880$@com>,
<BLU174-W11BBB4BF5838982980836ED2370@phx.gbl>,
<CAKav0XS31BFuyEjFykKRx5r5fWr2cT-f5d67KpwMK1MO5PfMOg@mail.gmail.com>
Message-ID: <BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>

Hi All,

I have been informed by Cisco that I have to rebuild my subscriber (version 8.5),
are there any good Cisco docs out there on rebuilds?

Regards

Cos

From: salamka at gmail.com


Date: Thu, 20 Dec 2012 16:05:35 +0530
Subject: Re: [cisco-voip] No access to Publisher
To: ckos1976 at hotmail.com
CC: davidytk at netvigator.com; cisco-voip at puck.nether.net
You got a remote console , like iLO or Vsphere ?

---AS

On Thu, Dec 20, 2012 at 3:15 PM, costas georgiou <ckos1976 at hotmail.com> wrote:

Hi,

Thanks for getting back to me. I tried restarting Tomcat on the pub and I can
access it for a while then I can't. Tomcat service on the Sub, i cannot restart
yet as I cannot access the server. DO you think these problems are due to the Sub
being down? I think this server has been down for a few days.

From: davidytk at netvigator.com


To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No access to Publisher
Date: Thu, 20 Dec 2012 17:39:11 +0800

Try to restart the Tomcat service in Pub & Sub

Util service restart Cisco Tomcat

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of costas georgiou
Sent: Thursday, December 20, 2012 5:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No access to Publisher

Hi All,

I was wondering whether anyone has come across this before. I have just started at
a new company and they have a CUCM cluster running 8.5.1, they have one pub and two
subs. I downloaded RTMT and noticed that one of the subs was not accessible, I can
ping the IP address, but cannot access it via SSH or URL, someone should be going
to the site today to re-boot. The day after, I could no longer access the
Publisher, this server I can access via SSH, but cannot access via URL or RMTM, I
stopped and started the Tomcat service and it came back for a while, but after a
while i cannot access again.

Any Ideas.

Regards

Costas

__________ Information from ESET NOD32 Antivirus, version of virus signature


database 7819 (20121220) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

__________ Information from ESET NOD32 Antivirus, version of virus signature


database 7819 (20121220) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From salamka at gmail.com Thu Jan 3 05:49:40 2013


From: salamka at gmail.com (Abdul Salam .)
Date: Thu, 3 Jan 2013 16:19:40 +0530
Subject: [cisco-voip] No access to Publisher
In-Reply-To: <BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>
References: <BLU174-W3DD4536E1CF5AD7768B1FD2370@phx.gbl>
<006901cdde95$ddc4ad80$994e0880$@com>
<BLU174-W11BBB4BF5838982980836ED2370@phx.gbl>
<CAKav0XS31BFuyEjFykKRx5r5fWr2cT-f5d67KpwMK1MO5PfMOg@mail.gmail.com>
<BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>
Message-ID: <CAKav0XTZLu6M6Uzp4EYDTg-0LymQSnYqiPhZTDXbc8OBiuh10w@mail.gmail.com>

I think DRS admin guide would be helpful

*---AS*

On Thu, Jan 3, 2013 at 4:13 PM, costas georgiou <ckos1976 at hotmail.com>wrote:

> Hi All,
>
> I have been informed by Cisco that I have to rebuild my subscriber
> (version 8.5), are there any good Cisco docs out there on rebuilds?
>
> Regards
>
> Cos
>
> ------------------------------
> From: salamka at gmail.com
> Date: Thu, 20 Dec 2012 16:05:35 +0530
> Subject: Re: [cisco-voip] No access to Publisher
> To: ckos1976 at hotmail.com
> CC: davidytk at netvigator.com; cisco-voip at puck.nether.net
>
>
> You got a remote console , like iLO or Vsphere ?
>
>
>
> *---AS*
>
>
>
> On Thu, Dec 20, 2012 at 3:15 PM, costas georgiou <ckos1976 at hotmail.com>wrote:
>
> Hi,
>
> Thanks for getting back to me. I tried restarting Tomcat on the pub and I
> can access it for a while then I can't. Tomcat service on the Sub, i
> cannot restart yet as I cannot access the server. DO you think these
> problems are due to the Sub being down? I think this server has been down
> for a few days.
>
> ------------------------------
> From: davidytk at netvigator.com
> To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] No access to Publisher
> Date: Thu, 20 Dec 2012 17:39:11 +0800
>
>
> Try to restart the Tomcat service in Pub & Sub
>
> Util service restart Cisco Tomcat
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *costas georgiou
> *Sent:* Thursday, December 20, 2012 5:32 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] No access to Publisher
>
> Hi All,
>
> I was wondering whether anyone has come across this before. I have just
> started at a new company and they have a CUCM cluster running 8.5.1, they
> have one pub and two subs. I downloaded RTMT and noticed that one of the
> subs was not accessible, I can ping the IP address, but cannot access it
> via SSH or URL, someone should be going to the site today to re-boot. The
> day after, I could no longer access the Publisher, this server I can access
> via SSH, but cannot access via URL or RMTM, I stopped and started the
> Tomcat service and it came back for a while, but after a while i cannot
> access again.
>
> Any Ideas.
>
> Regards
>
> Costas
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 7819 (20121220) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 7819 (20121220) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From terry.cheema at gmail.com Thu Jan 3 06:54:02 2013


From: terry.cheema at gmail.com (Terry Cheema)
Date: Thu, 3 Jan 2013 22:54:02 +1100
Subject: [cisco-voip] No access to Publisher
In-Reply-To: <BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>
References: <BLU174-W3DD4536E1CF5AD7768B1FD2370@phx.gbl>
<006901cdde95$ddc4ad80$994e0880$@com>
<BLU174-W11BBB4BF5838982980836ED2370@phx.gbl>
<CAKav0XS31BFuyEjFykKRx5r5fWr2cT-f5d67KpwMK1MO5PfMOg@mail.gmail.com>
<BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>
Message-ID: <81490114-72D1-4F33-AD06-292E8C5516A6@gmail.com>

Hi Costas,

While rebuiding servers most critical thing is you need to record info from old
servers and enter the same information in new servers.

Please refer to the below document and read carefully. It has all the pre-
checklists and post check lists. The pre-checklist has all the information you
need to gather before you start the rebuild, which is very critical.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/install/8_5_1/cluster/clstr851.h
tml

I have done this few times, and in one case eventually I had to rebuild the whole
cluster. Whats the hardware? I had done this on MCS servers, but process will be
almost same even if you have UCS, where you can simply create a new VM. But you
need to gather and record all information before you start the rebuild. When you
rebuild, all the information on server being rebuilt should be exactly same as per
original server.

I will try to quickly summarize the info you would need to collect before you
start, but still i will highly encourage you to go through the above link, its best
resource. Below information you will be asked when you are rebuilding the server:

1) Get your security password for the cluster (very important, reqd for
dbreplication, you have to be 200% sure, if you are not sure, go ahead and first
change it on all servers - i think you can do from recovery disk, if you dont know
the security pwd, if i correctly remember, and you need to restart all servers
after changing)

2) Record your administrator login/pwd

3) Record application login/pwd

4) run and record output from cli - show network eth0


It will give you ip address, subnet, default gateway, duplex, dns etc all network
related info

5) run and record output from cli - utils ntp status. Will give all ntp servers

6) run and record output command show status from CLI, will show hostname, license
mac etc.

7) Record all device information etc from RTMT device summary


prior and match the same post rebuilt.

8) After rebuilt make sure dbreplication is good, may take abt 15-20 mins to
syncronize

Hope that helps and let us know if you have any other query.

Terry

PS : excuse fonts from iphone notes :)

Sent from my iPhone

On 03/01/2013, at 9:43 PM, costas georgiou <ckos1976 at hotmail.com> wrote:

> Hi All,
>
> I have been informed by Cisco that I have to rebuild my subscriber (version 8.5),
are there any good Cisco docs out there on rebuilds?
>
> Regards
>
> Cos
>
> From: salamka at gmail.com
> Date: Thu, 20 Dec 2012 16:05:35 +0530
> Subject: Re: [cisco-voip] No access to Publisher
> To: ckos1976 at hotmail.com
> CC: davidytk at netvigator.com; cisco-voip at puck.nether.net
>
> You got a remote console , like iLO or Vsphere ?
>
>
>
> ---AS
>
>
>
> On Thu, Dec 20, 2012 at 3:15 PM, costas georgiou <ckos1976 at hotmail.com> wrote:
> Hi,
>
> Thanks for getting back to me. I tried restarting Tomcat on the pub and I can
access it for a while then I can't. Tomcat service on the Sub, i cannot restart
yet as I cannot access the server. DO you think these problems are due to the Sub
being down? I think this server has been down for a few days.
>
> From: davidytk at netvigator.com
> To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] No access to Publisher
> Date: Thu, 20 Dec 2012 17:39:11 +0800
>
>
> Try to restart the Tomcat service in Pub & Sub
>
> Util service restart Cisco Tomcat
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of costas georgiou
> Sent: Thursday, December 20, 2012 5:32 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] No access to Publisher
>
> Hi All,
>
> I was wondering whether anyone has come across this before. I have just started
at a new company and they have a CUCM cluster running 8.5.1, they have one pub and
two subs. I downloaded RTMT and noticed that one of the subs was not accessible, I
can ping the IP address, but cannot access it via SSH or URL, someone should be
going to the site today to re-boot. The day after, I could no longer access the
Publisher, this server I can access via SSH, but cannot access via URL or RMTM, I
stopped and started the Tomcat service and it came back for a while, but after a
while i cannot access again.
>
> Any Ideas.
>
> Regards
>
> Costas
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature
database 7819 (20121220) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature
database 7819 (20121220) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From terry.cheema at gmail.com Thu Jan 3 07:15:04 2013


From: terry.cheema at gmail.com (Terry Cheema)
Date: Thu, 3 Jan 2013 23:15:04 +1100
Subject: [cisco-voip] No access to Publisher
In-Reply-To: <81490114-72D1-4F33-AD06-292E8C5516A6@gmail.com>
References: <BLU174-W3DD4536E1CF5AD7768B1FD2370@phx.gbl>
<006901cdde95$ddc4ad80$994e0880$@com>
<BLU174-W11BBB4BF5838982980836ED2370@phx.gbl>
<CAKav0XS31BFuyEjFykKRx5r5fWr2cT-f5d67KpwMK1MO5PfMOg@mail.gmail.com>
<BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>
<81490114-72D1-4F33-AD06-292E8C5516A6@gmail.com>
Message-ID: <5FB65336-EDDF-46A4-AFB2-DDD95D0A2D3F@gmail.com>

To add further to below mail:

Once you record all information. Take a backup of your system.


Shut down the server and rebuild the new server with information at your hand.
In the end restore back up data to this node.

Terry

Sent from my iPhone

On 03/01/2013, at 10:54 PM, Terry Cheema <terry.cheema at gmail.com> wrote:

> Hi Costas,
>
> While rebuiding servers most critical thing is you need to record info from old
servers and enter the same information in new servers.
>
> Please refer to the below document and read carefully. It has all the pre-
checklists and post check lists. The pre-checklist has all the information you
need to gather before you start the rebuild, which is very critical.
>
>
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/install/8_5_1/cluster/clstr851.h
tml
>
> I have done this few times, and in one case eventually I had to rebuild the whole
cluster. Whats the hardware? I had done this on MCS servers, but process will be
almost same even if you have UCS, where you can simply create a new VM. But you
need to gather and record all information before you start the rebuild. When you
rebuild, all the information on server being rebuilt should be exactly same as per
original server.
>
> I will try to quickly summarize the info you would need to collect before you
start, but still i will highly encourage you to go through the above link, its best
resource. Below information you will be asked when you are rebuilding the server:
>
> 1) Get your security password for the cluster (very important, reqd for
dbreplication, you have to be 200% sure, if you are not sure, go ahead and first
change it on all servers - i think you can do from recovery disk, if you dont know
the security pwd, if i correctly remember, and you need to restart all servers
after changing)
>
> 2) Record your administrator login/pwd
>
> 3) Record application login/pwd
>
> 4) run and record output from cli - show network eth0
> It will give you ip address, subnet, default gateway, duplex, dns etc all network
related info
>
> 5) run and record output from cli - utils ntp status. Will give all ntp servers
>
> 6) run and record output command show status from CLI, will show hostname,
license mac etc.
>
> 7) Record all device information etc from RTMT device summary
> prior and match the same post rebuilt.
>
> 8) After rebuilt make sure dbreplication is good, may take abt 15-20 mins to
syncronize
>
> Hope that helps and let us know if you have any other query.
>
> Terry
>
> PS : excuse fonts from iphone notes :)
>
> Sent from my iPhone
>
> On 03/01/2013, at 9:43 PM, costas georgiou <ckos1976 at hotmail.com> wrote:
>
>> Hi All,
>>
>> I have been informed by Cisco that I have to rebuild my subscriber (version
8.5), are there any good Cisco docs out there on rebuilds?
>>
>> Regards
>>
>> Cos
>>
>> From: salamka at gmail.com
>> Date: Thu, 20 Dec 2012 16:05:35 +0530
>> Subject: Re: [cisco-voip] No access to Publisher
>> To: ckos1976 at hotmail.com
>> CC: davidytk at netvigator.com; cisco-voip at puck.nether.net
>>
>> You got a remote console , like iLO or Vsphere ?
>>
>>
>>
>> ---AS
>>
>>
>>
>> On Thu, Dec 20, 2012 at 3:15 PM, costas georgiou <ckos1976 at hotmail.com>
wrote:
>> Hi,
>>
>> Thanks for getting back to me. I tried restarting Tomcat on the pub and I can
access it for a while then I can't. Tomcat service on the Sub, i cannot restart
yet as I cannot access the server. DO you think these problems are due to the Sub
being down? I think this server has been down for a few days.
>>
>> From: davidytk at netvigator.com
>> To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net
>> Subject: RE: [cisco-voip] No access to Publisher
>> Date: Thu, 20 Dec 2012 17:39:11 +0800
>>
>>
>> Try to restart the Tomcat service in Pub & Sub
>>
>> Util service restart Cisco Tomcat
>>
>> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of costas georgiou
>> Sent: Thursday, December 20, 2012 5:32 PM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] No access to Publisher
>>
>> Hi All,
>>
>> I was wondering whether anyone has come across this before. I have just started
at a new company and they have a CUCM cluster running 8.5.1, they have one pub and
two subs. I downloaded RTMT and noticed that one of the subs was not accessible, I
can ping the IP address, but cannot access it via SSH or URL, someone should be
going to the site today to re-boot. The day after, I could no longer access the
Publisher, this server I can access via SSH, but cannot access via URL or RMTM, I
stopped and started the Tomcat service and it came back for a while, but after a
while i cannot access again.
>>
>> Any Ideas.
>>
>> Regards
>>
>> Costas
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus signature
database 7819 (20121220) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus signature
database 7819 (20121220) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
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From Zoltan.Kelemen at emerson.com Thu Jan 3 07:38:55 2013


From: Zoltan.Kelemen at emerson.com (Zoltan.Kelemen at emerson.com)
Date: Thu, 3 Jan 2013 12:38:55 +0000
Subject: [cisco-voip] Calling Party Transformation Patterns on CUCM 8.x
Message-ID: <F8E0CC3253A10C4CB137F12F568DAD061A96F325B9@GBLONZ-PMSGEM02.emrsn.org>

Hi and a Happy New Year!

CUCM 8.5.1 and I'm trying to globalize calling numbers of outgoing calls on a
specific SIP trunk.

My problem is, there are more than one DID ranges, i.e.:
1XXX numbers would have +40 345 671 XXX
2XXX numbers would have +40 341 232 XXX

I want to make sure to set the proper caller ID/calling number on outgoing calls.
(I can do that since it's an internal SIP trunk, so any callerID is ok)

So I've created a partition and a CSS for transformations and added a Calling Party
Transformation Pattern (Call Routing > Transformation > Transformation Pattern >
Calling Party Transformation Pattern), applied it properly to the SIP trunk etc.

For testing I have created a single test pattern, with my own extension: 2356
This matched and applied the transformations I was expecting. I tested it with
changing the transformations, it kept working.

However, when I rewrote the pattern to 2XXX it stopped matching. Basically it seems
that I'm unable to use any non-specific pattern to match the calling party number.
(neither 2!, nor 235X nor anything else that I've tried seems to match)
Any ideas?

Thanks,
Zoltan Kelemen
Emerson

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From terry.cheema at gmail.com Thu Jan 3 08:13:42 2013


From: terry.cheema at gmail.com (Terry Cheema)
Date: Fri, 4 Jan 2013 00:13:42 +1100
Subject: [cisco-voip] No access to Publisher
In-Reply-To: <38B85D43-C2BF-41B9-9432-F4AAB6E18898@gmail.com>
References: <BLU174-W3DD4536E1CF5AD7768B1FD2370@phx.gbl>
<006901cdde95$ddc4ad80$994e0880$@com>
<BLU174-W11BBB4BF5838982980836ED2370@phx.gbl>
<CAKav0XS31BFuyEjFykKRx5r5fWr2cT-f5d67KpwMK1MO5PfMOg@mail.gmail.com>
<BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>
<81490114-72D1-4F33-AD06-292E8C5516A6@gmail.com>
<5FB65336-EDDF-46A4-AFB2-DDD95D0A2D3F@gmail.com>
<BLU174-W8123B4C672C665F3898B6D2210@phx.gbl>
<38B85D43-C2BF-41B9-9432-F4AAB6E18898@gmail.com>
Message-ID: <C673498F-E86C-49DB-AEAF-64317E717A2D@gmail.com>

Including the list, in case anyone has other ideas/experiences...

Sent from my iPhone

On 03/01/2013, at 11:56 PM, Terry Cheema <terry.cheema at gmail.com> wrote:

> Hi Costas,
>
> Not a problem. Few things i would suggest.
>
> 1) First of all try if you can recover the servers using recovery disk. Once you
can access the servers and verify dbreplication (repair, reboot if you nned to) is
correct then take a good backup and move ahead with rebuild. Now if you question,
why to rebuild if everything is good - because it will again go into read only mode
or start having dbreplication issues or file system errors in a week or two.
>
> 2) Regards to your question of backup restore, if possible best approach would be
to restore from backup. Thats how Cisco recommends, in the doc, if you go to
replacing a subscriber section. That works fine.
>
> 3) And for your server accesibilty and errors, I would suggest you to run a
recovery disk and recover your server first. It would require a reboot.
>
> 4) In the end, if you are not able to recover your server by all means, then
consult with TAC and rebuild the server and let dbreplication do the work, if you
dont have a recent good backup. But again I would say if you must do this way get
full consultation from TAC first.
>
> When I ran into this first time, TAC recommended to first recover the server to
normal to minimize any risk. Everything went fine with that approach.
>
>
>
> Terry
>
>
>
> Sent from my iPhone
>
> On 03/01/2013, at 11:21 PM, costas georgiou <ckos1976 at hotmail.com> wrote:
>
>> Hi Terry,
>>
>> Thanks for the info, appreciate it. I was going to rebuild then let replication
do its thing, do you suggest restoring with a backup? Currently replication is not
working, I was going to rebuild the subscriber then do a reboot on all servers
probably on Monday to sort out replication. The reason for this is because I have
top raise a change request, I have only joined this company and found the server
down when I downloaded RTMT. Also, just to let you know, I cannot access the
faulty sub, i get to the CLI enter username and password then get lots of error
messages and it hangs. Cisco recommended the rebuild.
>>
>> Regards
>> Costas
>>
>> Subject: Re: [cisco-voip] No access to Publisher
>> From: terry.cheema at gmail.com
>> Date: Thu, 3 Jan 2013 23:15:04 +1100
>> To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net
>>
>> To add further to below mail:
>>
>> Once you record all information. Take a backup of your system.
>> Shut down the server and rebuild the new server with information at your hand.
>> In the end restore back up data to this node.
>>
>> Terry
>>
>> Sent from my iPhone
>>
>> On 03/01/2013, at 10:54 PM, Terry Cheema <terry.cheema at gmail.com> wrote:
>>
>> Hi Costas,
>>
>> While rebuiding servers most critical thing is you need to record info from old
servers and enter the same information in new servers.
>>
>> Please refer to the below document and read carefully. It has all the pre-
checklists and post check lists. The pre-checklist has all the information you
need to gather before you start the rebuild, which is very critical.
>>
>>
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/install/8_5_1/cluster/clstr851.h
tml
>>
>> I have done this few times, and in one case eventually I had to rebuild the
whole cluster. Whats the hardware? I had done this on MCS servers, but process will
be almost same even if you have UCS, where you can simply create a new VM. But you
need to gather and record all information before you start the rebuild. When you
rebuild, all the information on server being rebuilt should be exactly same as per
original server.
>>
>> I will try to quickly summarize the info you would need to collect before you
start, but still i will highly encourage you to go through the above link, its best
resource. Below information you will be asked when you are rebuilding the server:
>>
>> 1) Get your security password for the cluster (very important, reqd for
dbreplication, you have to be 200% sure, if you are not sure, go ahead and first
change it on all servers - i think you can do from recovery disk, if you dont know
the security pwd, if i correctly remember, and you need to restart all servers
after changing)
>>
>> 2) Record your administrator login/pwd
>>
>> 3) Record application login/pwd
>>
>> 4) run and record output from cli - show network eth0
>> It will give you ip address, subnet, default gateway, duplex, dns etc all
network related info
>>
>> 5) run and record output from cli - utils ntp status. Will give all ntp servers
>>
>> 6) run and record output command show status from CLI, will show hostname,
license mac etc.
>>
>> 7) Record all device information etc from RTMT device summary
>> prior and match the same post rebuilt.
>>
>> 8) After rebuilt make sure dbreplication is good, may take abt 15-20 mins to
syncronize
>>
>> Hope that helps and let us know if you have any other query.
>>
>> Terry
>>
>> PS : excuse fonts from iphone notes :)
>>
>> Sent from my iPhone
>>
>> On 03/01/2013, at 9:43 PM, costas georgiou <ckos1976 at hotmail.com> wrote:
>>
>> Hi All,
>>
>> I have been informed by Cisco that I have to rebuild my subscriber (version
8.5), are there any good Cisco docs out there on rebuilds?
>>
>> Regards
>>
>> Cos
>>
>> From: salamka at gmail.com
>> Date: Thu, 20 Dec 2012 16:05:35 +0530
>> Subject: Re: [cisco-voip] No access to Publisher
>> To: ckos1976 at hotmail.com
>> CC: davidytk at netvigator.com; cisco-voip at puck.nether.net
>>
>> You got a remote console , like iLO or Vsphere ?
>>
>>
>>
>> ---AS
>>
>>
>>
>> On Thu, Dec 20, 2012 at 3:15 PM, costas georgiou <ckos1976 at hotmail.com>
wrote:
>> Hi,
>>
>> Thanks for getting back to me. I tried restarting Tomcat on the pub and I can
access it for a while then I can't. Tomcat service on the Sub, i cannot restart
yet as I cannot access the server. DO you think these problems are due to the Sub
being down? I think this server has been down for a few days.
>>
>> From: davidytk at netvigator.com
>> To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net
>> Subject: RE: [cisco-voip] No access to Publisher
>> Date: Thu, 20 Dec 2012 17:39:11 +0800
>>
>>
>> Try to restart the Tomcat service in Pub & Sub
>>
>> Util service restart Cisco Tomcat
>>
>> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of costas georgiou
>> Sent: Thursday, December 20, 2012 5:32 PM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] No access to Publisher
>>
>> Hi All,
>>
>> I was wondering whether anyone has come across this before. I have just started
at a new company and they have a CUCM cluster running 8.5.1, they have one pub and
two subs. I downloaded RTMT and noticed that one of the subs was not accessible, I
can ping the IP address, but cannot access it via SSH or URL, someone should be
going to the site today to re-boot. The day after, I could no longer access the
Publisher, this server I can access via SSH, but cannot access via URL or RMTM, I
stopped and started the Tomcat service and it came back for a while, but after a
while i cannot access again.
>>
>> Any Ideas.
>>
>> Regards
>>
>> Costas
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus signature
database 7819 (20121220) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus signature
database 7819 (20121220) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
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From robhass at gmail.com Thu Jan 3 08:14:12 2013


From: robhass at gmail.com (Robert Hass)
Date: Thu, 3 Jan 2013 14:14:12 +0100
Subject: [cisco-voip] Call Recording on CUCM
In-Reply-To: <44547C16-7070-4462-9213-8661B44DBD8B@gmail.com>
References: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
<44547C16-7070-4462-9213-8661B44DBD8B@gmail.com>
Message-ID: <CALNfFTE=hH3mQroMKhWqDpd6Ytbg7xgwO451NCwmXArew3qU3A@mail.gmail.com>

We mostly have 7941 and 7945 ip phones. Are these both models have build-in
bridge ?

Rob

On Thursday, January 3, 2013, Erick B wrote:

> Yes, you'll need zoom or another 3rd party recording application.
>
> On recent cucm versions, you enable built in bridge on newer model phones
> then assign a recording profile to the DN on the phone you want to record.
> The recording has the IP address of recording server (sip trunk) the phone
> will send the audio to.
>
> You can do it the old way with span ports to on switches, depends on
> recording application you are using and where the phones are if span works
> easily or not.
>
> Sent from my iPhone
>
> On Jan 2, 2013, at 7:47 PM, Robert Hass <robhass at gmail.com <javascript:;>>
> wrote:
>
> > Hi
> > My boss asked if we can enable call recording on our CUCM 8.6 (just CUCM
> without Contact Center).
> > We considering two options of call recording
> > a) record all voice calls
> > b) record voice calls on demand - user can turn on/off recording via xml
> application of softkey on the phone
> >
> > My question : Are above scenarios of call recording are possible on CUCM
> ? What else I need - probably server for call-recording with big amount of
> storange and some additional software (Zoom ? Cisco ?)
> >
> > thanks for help
> >
> > Rob
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net <javascript:;>
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From robhass at gmail.com Thu Jan 3 08:15:55 2013


From: robhass at gmail.com (Robert Hass)
Date: Thu, 3 Jan 2013 14:15:55 +0100
Subject: [cisco-voip] Call Recording on CUCM
In-Reply-To: <72FE638DB23C1049AA265B28B5F87F3364A56940@prk-alford-mbx1.PRK.LOCAL>
References: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
<72FE638DB23C1049AA265B28B5F87F3364A56940@prk-alford-mbx1.PRK.LOCAL>
Message-ID: <CALNfFTHLmMJeiUTcSyWBP9PfmReg4WWKOMRa2HMsz1E-_4q14w@mail.gmail.com>

Thanks for info


Are you recording allcalls or only selected DNs ?
Is it possible to on/off recording by user using some XML application or
soft key on user's phone ?

Rob

On Thursday, January 3, 2013, Orr, Jeff B. wrote:

> Hi Rob,****
>
> ** **
>
> I just went through this for our environment. Call manager will provide
> the backend requirements to do recordings. However, you will need a 3rdparty
software to actually record and store the calls.
> ****
>
> ** **
>
> We evaluated several options and went with Zoom. It is a nice, Linux based
> recording software. It fully supports spanless recordings and can function
> as record all the time or on-demand recording. It does this by actually
> recording every call, and then allowing a user to press a service button to
> record a call that occurred earlier. ****
>
> ** **
>
> Jeff ****
>
> ** **
>
> ** **
>
> ____________________________________****
>
> Jeff Orr****
>
> Technical Services - Network Engineer****
>
> Park National Corporation (AMEX: PRK)****
>
> ** **
>
> This message is confidential and is intended only for the named
> recipients, and may contain information that is privileged, or exempt from
> disclosure under applicable law. If you are not the intended recipients of
> the email, you are hereby notified that the dissemination, distribution or
> copying of this e-mail or its contents is strictly prohibited. If you
> received this e-mail in error, please notify the sender at either the
> e-mail address or the phone number above and delete this e-mail from your
> computer.****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* cisco-voip-bounces at puck.nether.net <javascript:_e({}, 'cvml',
> 'cisco-voip-bounces at puck.nether.net');> [mailto:
> cisco-voip-bounces at puck.nether.net <javascript:_e({}, 'cvml',
> 'cisco-voip-bounces at puck.nether.net');>] *On Behalf Of *Robert Hass
> *Sent:* Wednesday, January 02, 2013 8:47 PM
> *To:* cisco-voip
> *Subject:* [cisco-voip] Call Recording on CUCM****
>
> ** **
>
> Hi****
>
> My boss asked if we can enable call recording on our CUCM 8.6 (just CUCM
> without Contact Center).****
>
> We considering two options of call recording****
>
> a) record all voice calls****
>
> b) record voice calls on demand - user can turn on/off recording via xml
> application of softkey on the phone****
>
> ** **
>
> My question : Are above scenarios of call recording are possible on CUCM ?
> What else I need - probably server for call-recording with big amount of
> storange and some additional software (Zoom ? Cisco ?)****
>
> ** **
>
> thanks for help****
>
> ** **
>
> Rob****
>
> ** **
>
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From terry.cheema at gmail.com Thu Jan 3 08:28:15 2013


From: terry.cheema at gmail.com (Terry Cheema)
Date: Fri, 4 Jan 2013 00:28:15 +1100
Subject: [cisco-voip] No access to Publisher
In-Reply-To: <BLU174-W135B28071F9C3411896663D2210@phx.gbl>
References: <BLU174-W3DD4536E1CF5AD7768B1FD2370@phx.gbl>
<006901cdde95$ddc4ad80$994e0880$@com>
<BLU174-W11BBB4BF5838982980836ED2370@phx.gbl>
<CAKav0XS31BFuyEjFykKRx5r5fWr2cT-f5d67KpwMK1MO5PfMOg@mail.gmail.com>
<BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>
<81490114-72D1-4F33-AD06-292E8C5516A6@gmail.com>
<5FB65336-EDDF-46A4-AFB2-DDD95D0A2D3F@gmail.com>
<BLU174-W8123B4C672C665F3898B6D2210@phx.gbl>
<38B85D43-C2BF-41B9-9432-F4AAB6E18898@gmail.com>
<BLU174-W135B28071F9C3411896663D2210@phx.gbl>
Message-ID: <BF7A14DC-23EE-48F8-8BF6-D77DE8BBD9E9@gmail.com>

If you have tried the recovery disk already then I think you may not have much
options left. You would probably try rebuilding the server with the procedure TAC
is recommending.

Since you are restoring on subscriber only, even if you restore from 6th December
backup, dbreplication should take care for the rest. Is TAC recommending this way?
How bad is your dbreplication, its just bad on this server, rest two have good
status??

If not already given, ask TAC to give you a precise action plan and follow that
closely. That may be your best bet.

Good luck with that, keep us posted !!

Terry

Sent from my iPhone

On 04/01/2013, at 12:09 AM, costas georgiou <ckos1976 at hotmail.com> wrote:

> Cheers Terry,


>
> We tried the recovery disk yesterday, it stated that it was successful and then
we were getting the same errors. I sent the errors over to TAC and they came back
with rebuilding from backup, but the earliest backup we have is Dec 6th.
>
> Regards
>
> Costas
>
> Subject: Re: [cisco-voip] No access to Publisher
> From: terry.cheema at gmail.com
> Date: Thu, 3 Jan 2013 23:56:36 +1100
> To: ckos1976 at hotmail.com
>
> Hi Costas,
>
> Not a problem. Few things i would suggest.
>
> 1) First of all try if you can recover the servers using recovery disk. Once you
can access the servers and verify dbreplication (repair, reboot if you nned to) is
correct then take a good backup and move ahead with rebuild. Now if you question,
why to rebuild if everything is good - because it will again go into read only mode
or start having dbreplication issues or file system errors in a week or two.
>
> 2) Regards to your question of backup restore, if possible best approach would be
to restore from backup. Thats how Cisco recommends, in the doc, if you go to
replacing a subscriber section. That works fine.
>
> 3) And for your server accesibilty and errors, I would suggest you to run a
recovery disk and recover your server first. It would require a reboot.
>
> 4) In the end, if you are not able to recover your server by all means, then
consult with TAC and rebuild the server and let dbreplication do the work, if you
dont have a recent good backup. But again I would say if you must do this way get
full consultation from TAC first.
>
> When I ran into this first time, TAC recommended to first recover the server to
normal to minimize any risk. Everything went fine with that approach.
>
>
>
> Terry
>
>
>
> Sent from my iPhone
>
> On 03/01/2013, at 11:21 PM, costas georgiou <ckos1976 at hotmail.com> wrote:
>
> Hi Terry,
>
> Thanks for the info, appreciate it. I was going to rebuild then let replication
do its thing, do you suggest restoring with a backup? Currently replication is not
working, I was going to rebuild the subscriber then do a reboot on all servers
probably on Monday to sort out replication. The reason for this is because I have
top raise a change request, I have only joined this company and found the server
down when I downloaded RTMT. Also, just to let you know, I cannot access the
faulty sub, i get to the CLI enter username and password then get lots of error
messages and it hangs. Cisco recommended the rebuild.
>
> Regards
> Costas
>
> Subject: Re: [cisco-voip] No access to Publisher
> From: terry.cheema at gmail.com
> Date: Thu, 3 Jan 2013 23:15:04 +1100
> To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net
>
> To add further to below mail:
>
> Once you record all information. Take a backup of your system.
> Shut down the server and rebuild the new server with information at your hand.
> In the end restore back up data to this node.
>
> Terry
>
> Sent from my iPhone
>
> On 03/01/2013, at 10:54 PM, Terry Cheema <terry.cheema at gmail.com> wrote:
>
> Hi Costas,
>
> While rebuiding servers most critical thing is you need to record info from old
servers and enter the same information in new servers.
>
> Please refer to the below document and read carefully. It has all the pre-
checklists and post check lists. The pre-checklist has all the information you
need to gather before you start the rebuild, which is very critical.
>
>
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/install/8_5_1/cluster/clstr851.h
tml
>
> I have done this few times, and in one case eventually I had to rebuild the whole
cluster. Whats the hardware? I had done this on MCS servers, but process will be
almost same even if you have UCS, where you can simply create a new VM. But you
need to gather and record all information before you start the rebuild. When you
rebuild, all the information on server being rebuilt should be exactly same as per
original server.
>
> I will try to quickly summarize the info you would need to collect before you
start, but still i will highly encourage you to go through the above link, its best
resource. Below information you will be asked when you are rebuilding the server:
>
> 1) Get your security password for the cluster (very important, reqd for
dbreplication, you have to be 200% sure, if you are not sure, go ahead and first
change it on all servers - i think you can do from recovery disk, if you dont know
the security pwd, if i correctly remember, and you need to restart all servers
after changing)
>
> 2) Record your administrator login/pwd
>
> 3) Record application login/pwd
>
> 4) run and record output from cli - show network eth0
> It will give you ip address, subnet, default gateway, duplex, dns etc all network
related info
>
> 5) run and record output from cli - utils ntp status. Will give all ntp servers
>
> 6) run and record output command show status from CLI, will show hostname,
license mac etc.
>
> 7) Record all device information etc from RTMT device summary
> prior and match the same post rebuilt.
>
> 8) After rebuilt make sure dbreplication is good, may take abt 15-20 mins to
syncronize
>
> Hope that helps and let us know if you have any other query.
>
> Terry
>
> PS : excuse fonts from iphone notes :)
>
> Sent from my iPhone
>
> On 03/01/2013, at 9:43 PM, costas georgiou <ckos1976 at hotmail.com> wrote:
>
> Hi All,
>
> I have been informed by Cisco that I have to rebuild my subscriber (version 8.5),
are there any good Cisco docs out there on rebuilds?
>
> Regards
>
> Cos
>
> From: salamka at gmail.com
> Date: Thu, 20 Dec 2012 16:05:35 +0530
> Subject: Re: [cisco-voip] No access to Publisher
> To: ckos1976 at hotmail.com
> CC: davidytk at netvigator.com; cisco-voip at puck.nether.net
>
> You got a remote console , like iLO or Vsphere ?
>
>
>
> ---AS
>
>
>
> On Thu, Dec 20, 2012 at 3:15 PM, costas georgiou <ckos1976 at hotmail.com> wrote:
> Hi,
>
> Thanks for getting back to me. I tried restarting Tomcat on the pub and I can
access it for a while then I can't. Tomcat service on the Sub, i cannot restart
yet as I cannot access the server. DO you think these problems are due to the Sub
being down? I think this server has been down for a few days.
>
> From: davidytk at netvigator.com
> To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] No access to Publisher
> Date: Thu, 20 Dec 2012 17:39:11 +0800
>
>
> Try to restart the Tomcat service in Pub & Sub
>
> Util service restart Cisco Tomcat
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of costas georgiou
> Sent: Thursday, December 20, 2012 5:32 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] No access to Publisher
>
> Hi All,
>
> I was wondering whether anyone has come across this before. I have just started
at a new company and they have a CUCM cluster running 8.5.1, they have one pub and
two subs. I downloaded RTMT and noticed that one of the subs was not accessible, I
can ping the IP address, but cannot access it via SSH or URL, someone should be
going to the site today to re-boot. The day after, I could no longer access the
Publisher, this server I can access via SSH, but cannot access via URL or RMTM, I
stopped and started the Tomcat service and it came back for a while, but after a
while i cannot access again.
>
> Any Ideas.
>
> Regards
>
> Costas
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature
database 7819 (20121220) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature
database 7819 (20121220) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From VanMarenNP at ldschurch.org Thu Jan 3 09:45:01 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Thu, 3 Jan 2013 14:45:01 +0000
Subject: [cisco-voip] Call Recording on CUCM
In-Reply-To: <CALNfFTE=hH3mQroMKhWqDpd6Ytbg7xgwO451NCwmXArew3qU3A@mail.gmail.com>
References: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
<44547C16-7070-4462-9213-8661B44DBD8B@gmail.com>
<CALNfFTE=hH3mQroMKhWqDpd6Ytbg7xgwO451NCwmXArew3qU3A@mail.gmail.com>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E42E71@W12112.ldschurch.org>

Yes they do. Old school 7940/7960 do not.


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of Robert Hass
Sent: Thursday, January 03, 2013 6:14 AM
To: Erick B
Cc: cisco-voip
Subject: Re: [cisco-voip] Call Recording on CUCM

We mostly have 7941 and 7945 ip phones. Are these both models have build-in
bridge ?

Rob

On Thursday, January 3, 2013, Erick B wrote:


Yes, you'll need zoom or another 3rd party recording application.

On recent cucm versions, you enable built in bridge on newer model phones then
assign a recording profile to the DN on the phone you want to record. The recording
has the IP address of recording server (sip trunk) the phone will send the audio
to.

You can do it the old way with span ports to on switches, depends on recording
application you are using and where the phones are if span works easily or not.

Sent from my iPhone

On Jan 2, 2013, at 7:47 PM, Robert Hass <robhass at gmail.com<javascript:;>> wrote:

> Hi
> My boss asked if we can enable call recording on our CUCM 8.6 (just CUCM without
Contact Center).
> We considering two options of call recording
> a) record all voice calls
> b) record voice calls on demand - user can turn on/off recording via xml
application of softkey on the phone
>
> My question : Are above scenarios of call recording are possible on CUCM ? What
else I need - probably server for call-recording with big amount of storange and
some additional software (Zoom ? Cisco ?)
>
> thanks for help
>
> Rob
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net<javascript:;>
> https://puck.nether.net/mailman/listinfo/cisco-voip

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

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From Edward.Countryman at presencehealth.org Thu Jan 3 10:15:24 2013


From: Edward.Countryman at presencehealth.org (Countryman, Edward)
Date: Thu, 3 Jan 2013 09:15:24 -0600
Subject: [cisco-voip] wireless phone network authentication
Message-ID: <3E508304295FC04988DFEEA1A2F177C347085E@AEXMV22.phnet.phroot.local>

Just curious how other folks handle deployment and subsequent device
management of wireless phone authentication on your networks?

It appears Cisco publishes a best practice of using 802.1x


authentication rather than PSK's. Given this, do most folks use A/D
accounts? Do you use individual accounts per telephone of share an
account across many devices?

Does anyone use the BD utility and have an open provisioning network
setup? How is that working out?

I am potentially looking at over a thousand devices and want to


establish our internal process for managing them. Any thoughts or
input is appreciated.

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From ckos1976 at hotmail.com Thu Jan 3 10:53:57 2013


From: ckos1976 at hotmail.com (costas georgiou)
Date: Thu, 3 Jan 2013 15:53:57 +0000
Subject: [cisco-voip] No access to Publisher
In-Reply-To: <BF7A14DC-23EE-48F8-8BF6-D77DE8BBD9E9@gmail.com>
References: <BLU174-W3DD4536E1CF5AD7768B1FD2370@phx.gbl>
<006901cdde95$ddc4ad80$994e0880$@com>
<BLU174-W11BBB4BF5838982980836ED2370@phx.gbl>
<CAKav0XS31BFuyEjFykKRx5r5fWr2cT-f5d67KpwMK1MO5PfMOg@mail.gmail.com>
<BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>
<81490114-72D1-4F33-AD06-292E8C5516A6@gmail.com>
<5FB65336-EDDF-46A4-AFB2-DDD95D0A2D3F@gmail.com>
<BLU174-W8123B4C672C665F3898B6D2210@phx.gbl>
<38B85D43-C2BF-41B9-9432-F4AAB6E18898@gmail.com>
<BLU174-W135B28071F9C3411896663D2210@phx.gbl>,
<BF7A14DC-23EE-48F8-8BF6-D77DE8BBD9E9@gmail.com>
Message-ID: <BLU174-W422A2557D0879D4AE967E9D2210@phx.gbl>

Hi Terry,

Thanks for the info. Cisco are recommending I rebuild the server as if it were a
fresh install on CUCM, the Publisher will then push the configuration to the
Subscriber once added to the cluster again.
Thanks

Cos

CC: cisco-voip at puck.nether.net


From: terry.cheema at gmail.com
Subject: Re: [cisco-voip] No access to Publisher
Date: Fri, 4 Jan 2013 00:28:15 +1100
To: ckos1976 at hotmail.com

If you have tried the recovery disk already then I think you may not have much
options left. You would probably try rebuilding the server with the procedure TAC
is recommending.

Since you are restoring on subscriber only, even if you restore from 6th December
backup, dbreplication should take care for the rest. Is TAC recommending this way?
How bad is your dbreplication, its just bad on this server, rest two have good
status??

If not already given, ask TAC to give you a precise action plan and follow that
closely. That may be your best bet.

Good luck with that, keep us posted !!

Terry

Sent from my iPhone

On 04/01/2013, at 12:09 AM, costas georgiou <ckos1976 at hotmail.com> wrote:

Cheers Terry,

We tried the recovery disk yesterday, it stated that it was successful and then we
were getting the same errors. I sent the errors over to TAC and they came back
with rebuilding from backup, but the earliest backup we have is Dec 6th.

Regards

Costas
Subject: Re: [cisco-voip] No access to Publisher
From: terry.cheema at gmail.com
Date: Thu, 3 Jan 2013 23:56:36 +1100
To: ckos1976 at hotmail.com

Hi Costas,

Not a problem. Few things i would suggest.

1) First of all try if you can recover the servers using recovery disk. Once you
can access the servers and verify dbreplication (repair, reboot if you nned to) is
correct then take a good backup and move ahead with rebuild. Now if you question,
why to rebuild if everything is good - because it will again go into read only mode
or start having dbreplication issues or file system errors in a week or two.

2) Regards to your question of backup restore, if possible best approach would be


to restore from backup. Thats how Cisco recommends, in the doc, if you go to
replacing a subscriber section. That works fine.

3) And for your server accesibilty and errors, I would suggest you to run a
recovery disk and recover your server first. It would require a reboot.

4) In the end, if you are not able to recover your server by all means, then
consult with TAC and rebuild the server and let dbreplication do the work, if you
dont have a recent good backup. But again I would say if you must do this way get
full consultation from TAC first.

When I ran into this first time, TAC recommended to first recover the server to
normal to minimize any risk. Everything went fine with that approach.

Terry

Sent from my iPhone

On 03/01/2013, at 11:21 PM, costas georgiou <ckos1976 at hotmail.com> wrote:


Hi Terry,

Thanks for the info, appreciate it. I was going to rebuild then let replication do
its thing, do you suggest restoring with a backup? Currently replication is not
working, I was going to rebuild the subscriber then do a reboot on all servers
probably on Monday to sort out replication. The reason for this is because I have
top raise a change request, I have only joined this company and found the server
down when I downloaded RTMT. Also, just to let you know, I cannot access the
faulty sub, i get to the CLI enter username and password then get lots of error
messages and it hangs. Cisco recommended the rebuild.

Regards
Costas

Subject: Re: [cisco-voip] No access to Publisher


From: terry.cheema at gmail.com
Date: Thu, 3 Jan 2013 23:15:04 +1100
To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net

To add further to below mail:

Once you record all information. Take a backup of your system.


Shut down the server and rebuild the new server with information at your hand.
In the end restore back up data to this node.

Terry

Sent from my iPhone

On 03/01/2013, at 10:54 PM, Terry Cheema <terry.cheema at gmail.com> wrote:

Hi Costas,

While rebuiding servers most critical thing is you need to record info from old
servers and enter the same information in new servers.

Please refer to the below document and read carefully. It has all the pre-
checklists and post check lists. The pre-checklist has all the information you
need to gather before you start the rebuild, which is very critical.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/install/8_5_1/cluster/clstr851.h
tml
I have done this few times, and in one case eventually I had to rebuild the whole
cluster. Whats the hardware? I had done this on MCS servers, but process will be
almost same even if you have UCS, where you can simply create a new VM. But you
need to gather and record all information before you start the rebuild. When you
rebuild, all the information on server being rebuilt should be exactly same as per
original server.

I will try to quickly summarize the info you would need to collect before you
start, but still i will highly encourage you to go through the above link, its best
resource. Below information you will be asked when you are rebuilding the server:

1) Get your security password for the cluster (very important, reqd for
dbreplication, you have to be 200% sure, if you are not sure, go ahead and first
change it on all servers - i think you can do from recovery disk, if you dont know
the security pwd, if i correctly remember, and you need to restart all servers
after changing)

2) Record your administrator login/pwd

3) Record application login/pwd

4) run and record output from cli - show network eth0


It will give you ip address, subnet, default gateway, duplex, dns etc all network
related info

5) run and record output from cli - utils ntp status. Will give all ntp servers

6) run and record output command show status from CLI, will show hostname, license
mac etc.

7) Record all device information etc from RTMT device summary


prior and match the same post rebuilt.

8) After rebuilt make sure dbreplication is good, may take abt 15-20 mins to
syncronize

Hope that helps and let us know if you have any other query.

Terry

PS : excuse fonts from iphone notes :)


Sent from my iPhone

On 03/01/2013, at 9:43 PM, costas georgiou <ckos1976 at hotmail.com> wrote:


Hi All,

I have been informed by Cisco that I have to rebuild my subscriber (version 8.5),
are there any good Cisco docs out there on rebuilds?

Regards

Cos

From: salamka at gmail.com


Date: Thu, 20 Dec 2012 16:05:35 +0530
Subject: Re: [cisco-voip] No access to Publisher
To: ckos1976 at hotmail.com
CC: davidytk at netvigator.com; cisco-voip at puck.nether.net

You got a remote console , like iLO or Vsphere ?

---AS

On Thu, Dec 20, 2012 at 3:15 PM, costas georgiou <ckos1976 at hotmail.com> wrote:

Hi,

Thanks for getting back to me. I tried restarting Tomcat on the pub and I can
access it for a while then I can't. Tomcat service on the Sub, i cannot restart
yet as I cannot access the server. DO you think these problems are due to the Sub
being down? I think this server has been down for a few days.

From: davidytk at netvigator.com


To: ckos1976 at hotmail.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No access to Publisher
Date: Thu, 20 Dec 2012 17:39:11 +0800
Try to restart the Tomcat service in Pub & Sub

Util service restart Cisco Tomcat

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of costas georgiou
Sent: Thursday, December 20, 2012 5:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No access to Publisher

Hi All,

I was wondering whether anyone has come across this before. I have just started at
a new company and they have a CUCM cluster running 8.5.1, they have one pub and two
subs. I downloaded RTMT and noticed that one of the subs was not accessible, I can
ping the IP address, but cannot access it via SSH or URL, someone should be going
to the site today to re-boot. The day after, I could no longer access the
Publisher, this server I can access via SSH, but cannot access via URL or RMTM, I
stopped and started the Tomcat service and it came back for a while, but after a
while i cannot access again.

Any Ideas.

Regards

Costas

__________ Information from ESET NOD32 Antivirus, version of virus signature


database 7819 (20121220) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

__________ Information from ESET NOD32 Antivirus, version of virus signature


database 7819 (20121220) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From chrward at cisco.com Thu Jan 3 11:15:17 2013


From: chrward at cisco.com (Chris Ward (chrward))
Date: Thu, 3 Jan 2013 16:15:17 +0000
Subject: [cisco-voip] No access to Publisher
In-Reply-To: <BLU174-W422A2557D0879D4AE967E9D2210@phx.gbl>
References: <BLU174-W3DD4536E1CF5AD7768B1FD2370@phx.gbl>
<006901cdde95$ddc4ad80$994e0880$@com>
<BLU174-W11BBB4BF5838982980836ED2370@phx.gbl>
<CAKav0XS31BFuyEjFykKRx5r5fWr2cT-f5d67KpwMK1MO5PfMOg@mail.gmail.com>
<BLU174-W132F70007E327A7F9A38F4D2210@phx.gbl>
<81490114-72D1-4F33-AD06-292E8C5516A6@gmail.com>
<5FB65336-EDDF-46A4-AFB2-DDD95D0A2D3F@gmail.com>
<BLU174-W8123B4C672C665F3898B6D2210@phx.gbl>
<38B85D43-C2BF-41B9-9432-F4AAB6E18898@gmail.com>
<BLU174-W135B28071F9C3411896663D2210@phx.gbl>,
<BF7A14DC-23EE-48F8-8BF6-D77DE8BBD9E9@gmail.com>
<BLU174-W422A2557D0879D4AE967E9D2210@phx.gbl>
Message-ID: <C3D1FCA271936B48839E081F898E17AA1CB1DF@xmb-rcd-x13.cisco.com>

Without a DRS restore there are certain things that won't be automatically pushed
back to a rebuilt subscriber:

1) Service Activations

2) TFTP Files

3) Non-standard or COP installed phone firmware

4) MOH Files

5) Server certificates
There may be some other but just be aware of these things that you will have to
manually correct.

+Chris
Unity Connection TME

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of costas georgiou
Sent: Thursday, January 03, 2013 10:54 AM
To: terry.cheema at gmail.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No access to Publisher

Hi Terry,

Thanks for the info. Cisco are recommending I rebuild the server as if it were a
fresh install on CUCM, the Publisher will then push the configuration to the
Subscriber once added to the cluster again.

Thanks
Cos

________________________________
CC: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
From: terry.cheema at gmail.com<mailto:terry.cheema at gmail.com>
Subject: Re: [cisco-voip] No access to Publisher
Date: Fri, 4 Jan 2013 00:28:15 +1100
To: ckos1976 at hotmail.com<mailto:ckos1976 at hotmail.com>
If you have tried the recovery disk already then I think you may not have much
options left. You would probably try rebuilding the server with the procedure TAC
is recommending.

Since you are restoring on subscriber only, even if you restore from 6th December
backup, dbreplication should take care for the rest. Is TAC recommending this way?
How bad is your dbreplication, its just bad on this server, rest two have good
status??

If not already given, ask TAC to give you a precise action plan and follow that
closely. That may be your best bet.

Good luck with that, keep us posted !!

Terry

Sent from my iPhone

On 04/01/2013, at 12:09 AM, costas georgiou <ckos1976 at


hotmail.com<mailto:ckos1976 at hotmail.com>> wrote:
Cheers Terry,

We tried the recovery disk yesterday, it stated that it was successful and then we
were getting the same errors. I sent the errors over to TAC and they came back
with rebuilding from backup, but the earliest backup we have is Dec 6th.

Regards

Costas

________________________________
Subject: Re: [cisco-voip] No access to Publisher
From: terry.cheema at gmail.com<mailto:terry.cheema at gmail.com>
Date: Thu, 3 Jan 2013 23:56:36 +1100
To: ckos1976 at hotmail.com<mailto:ckos1976 at hotmail.com>
Hi Costas,

Not a problem. Few things i would suggest.

1) First of all try if you can recover the servers using recovery disk. Once you
can access the servers and verify dbreplication (repair, reboot if you nned to) is
correct then take a good backup and move ahead with rebuild. Now if you question,
why to rebuild if everything is good - because it will again go into read only mode
or start having dbreplication issues or file system errors in a week or two.

2) Regards to your question of backup restore, if possible best approach would be


to restore from backup. Thats how Cisco recommends, in the doc, if you go to
replacing a subscriber section. That works fine.

3) And for your server accesibilty and errors, I would suggest you to run a
recovery disk and recover your server first. It would require a reboot.

4) In the end, if you are not able to recover your server by all means, then
consult with TAC and rebuild the server and let dbreplication do the work, if you
dont have a recent good backup. But again I would say if you must do this way get
full consultation from TAC first.

When I ran into this first time, TAC recommended to first recover the server to
normal to minimize any risk. Everything went fine with that approach.

Terry

Sent from my iPhone

On 03/01/2013, at 11:21 PM, costas georgiou <ckos1976 at


hotmail.com<mailto:ckos1976 at hotmail.com>> wrote:
Hi Terry,

Thanks for the info, appreciate it. I was going to rebuild then let replication do
its thing, do you suggest restoring with a backup? Currently replication is not
working, I was going to rebuild the subscriber then do a reboot on all servers
probably on Monday to sort out replication. The reason for this is because I have
top raise a change request, I have only joined this company and found the server
down when I downloaded RTMT. Also, just to let you know, I cannot access the
faulty sub, i get to the CLI enter username and password then get lots of error
messages and it hangs. Cisco recommended the rebuild.

Regards
Costas

________________________________
Subject: Re: [cisco-voip] No access to Publisher
From: terry.cheema at gmail.com<mailto:terry.cheema at gmail.com>
Date: Thu, 3 Jan 2013 23:15:04 +1100
To: ckos1976 at hotmail.com<mailto:ckos1976 at hotmail.com>; cisco-voip at
puck.nether.net<mailto:cisco-voip at puck.nether.net>
To add further to below mail:

Once you record all information. Take a backup of your system.


Shut down the server and rebuild the new server with information at your hand.
In the end restore back up data to this node.

Terry

Sent from my iPhone

On 03/01/2013, at 10:54 PM, Terry Cheema <terry.cheema at


gmail.com<mailto:terry.cheema at gmail.com>> wrote:
Hi Costas,

While rebuiding servers most critical thing is you need to record info from old
servers and enter the same information in new servers.

Please refer to the below document and read carefully. It has all the pre-
checklists and post check lists. The pre-checklist has all the information you
need to gather before you start the rebuild, which is very critical.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/install/8_5_1/cluster/clstr851.h
tml

I have done this few times, and in one case eventually I had to rebuild the whole
cluster. Whats the hardware? I had done this on MCS servers, but process will be
almost same even if you have UCS, where you can simply create a new VM. But you
need to gather and record all information before you start the rebuild. When you
rebuild, all the information on server being rebuilt should be exactly same as per
original server.

I will try to quickly summarize the info you would need to collect before you
start, but still i will highly encourage you to go through the above link, its best
resource. Below information you will be asked when you are rebuilding the server:

1) Get your security password for the cluster (very important, reqd for
dbreplication, you have to be 200% sure, if you are not sure, go ahead and first
change it on all servers - i think you can do from recovery disk, if you dont know
the security pwd, if i correctly remember, and you need to restart all servers
after changing)

2) Record your administrator login/pwd

3) Record application login/pwd

4) run and record output from cli - show network eth0


It will give you ip address, subnet, default gateway, duplex, dns etc all network
related info

5) run and record output from cli - utils ntp status. Will give all ntp servers

6) run and record output command show status from CLI, will show hostname, license
mac etc.

7) Record all device information etc from RTMT device summary


prior and match the same post rebuilt.

8) After rebuilt make sure dbreplication is good, may take abt 15-20 mins to
syncronize

Hope that helps and let us know if you have any other query.

Terry

PS : excuse fonts from iphone notes :)

Sent from my iPhone

On 03/01/2013, at 9:43 PM, costas georgiou <ckos1976 at hotmail.com<mailto:ckos1976


at hotmail.com>> wrote:
Hi All,

I have been informed by Cisco that I have to rebuild my subscriber (version 8.5),
are there any good Cisco docs out there on rebuilds?

Regards

Cos
________________________________
From: salamka at gmail.com<mailto:salamka at gmail.com>
Date: Thu, 20 Dec 2012 16:05:35 +0530
Subject: Re: [cisco-voip] No access to Publisher
To: ckos1976 at hotmail.com<mailto:ckos1976 at hotmail.com>
CC: davidytk at netvigator.com<mailto:davidytk at netvigator.com>; cisco-voip at
puck.nether.net<mailto:cisco-voip at puck.nether.net>
You got a remote console , like iLO or Vsphere ?

---AS

On Thu, Dec 20, 2012 at 3:15 PM, costas georgiou <ckos1976 at


hotmail.com<mailto:ckos1976 at hotmail.com>> wrote:
Hi,

Thanks for getting back to me. I tried restarting Tomcat on the pub and I can
access it for a while then I can't. Tomcat service on the Sub, i cannot restart
yet as I cannot access the server. DO you think these problems are due to the Sub
being down? I think this server has been down for a few days.

________________________________
From: davidytk at netvigator.com<mailto:davidytk at netvigator.com>
To: ckos1976 at hotmail.com<mailto:ckos1976 at hotmail.com>; cisco-voip at
puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] No access to Publisher
Date: Thu, 20 Dec 2012 17:39:11 +0800

Try to restart the Tomcat service in Pub & Sub

Util service restart Cisco Tomcat

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-
bounces at puck.nether.net>] On Behalf Of costas georgiou
Sent: Thursday, December 20, 2012 5:32 PM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] No access to Publisher

Hi All,

I was wondering whether anyone has come across this before. I have just started at
a new company and they have a CUCM cluster running 8.5.1, they have one pub and two
subs. I downloaded RTMT and noticed that one of the subs was not accessible, I can
ping the IP address, but cannot access it via SSH or URL, someone should be going
to the site today to re-boot. The day after, I could no longer access the
Publisher, this server I can access via SSH, but cannot access via URL or RMTM, I
stopped and started the Tomcat service and it came back for a while, but after a
while i cannot access again.

Any Ideas.

Regards

Costas
__________ Information from ESET NOD32 Antivirus, version of virus signature
database 7819 (20121220) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com<http://www.eset.com/>

__________ Information from ESET NOD32 Antivirus, version of virus signature


database 7819 (20121220) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com<http://www.eset.com/>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip
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From svoll.voip at gmail.com Thu Jan 3 14:03:23 2013


From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 3 Jan 2013 11:03:23 -0800
Subject: [cisco-voip] Call Recording on CUCM
In-Reply-To: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
References: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
Message-ID: <CAHgd+3_mp35-QojV+-RZFeiNfCb20XQ3EJMat9XgvUPy4Q4p_g@mail.gmail.com>

As others have already stated. you will need a 3rd party app. We ended up
with CallRex. It has a Great Web interface for the end users and API's if
you want to build a app to interface with it.

As a added benefit, if you end up using Extension Mobility you don't have
to associated phones to the Call Recording application. you just associate
extMob profiles. That way if your not recording All phone in your org, you
don't have to worry about associating the right phones.

YMMV

Scott

On Wed, Jan 2, 2013 at 5:47 PM, Robert Hass <robhass at gmail.com> wrote:

> Hi
> My boss asked if we can enable call recording on our CUCM 8.6 (just CUCM
> without Contact Center).
> We considering two options of call recording
> a) record all voice calls
> b) record voice calls on demand - user can turn on/off recording via xml
> application of softkey on the phone
>
> My question : Are above scenarios of call recording are possible on CUCM ?
> What else I need - probably server for call-recording with big amount of
> storange and some additional software (Zoom ? Cisco ?)
>
> thanks for help
>
> Rob
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From svoll.voip at gmail.com Thu Jan 3 14:05:22 2013


From: svoll.voip at gmail.com (Scott Voll)
Date: Thu, 3 Jan 2013 11:05:22 -0800
Subject: [cisco-voip] Calling Party Transformation Patterns on CUCM 8.x
In-Reply-To: <F8E0CC3253A10C4CB137F12F568DAD061A96F325B9@GBLONZ-PMSGEM02.emrsn.org>
References: <F8E0CC3253A10C4CB137F12F568DAD061A96F325B9@GBLONZ-PMSGEM02.emrsn.org>
Message-ID: <CAHgd+3-bMYHATsZMmf3GsyX9LRvPwwfUcUa_FJBPhaVzbUxu3Q@mail.gmail.com>

Can you just set it on the line and pass it through to the sip trunk?

Scott

On Thu, Jan 3, 2013 at 4:38 AM, <Zoltan.Kelemen at emerson.com> wrote:

> Hi and a Happy New Year!****


>
> ** **
>
> CUCM 8.5.1 and I?m trying to globalize calling numbers of outgoing calls
> on a specific SIP trunk.****
>
> ** **
>
> My problem is, there are more than one DID ranges, i.e.:****
>
> 1XXX numbers would have +40 345 671 XXX****
>
> 2XXX numbers would have +40 341 232 XXX****
>
> ** **
>
> I want to make sure to set the proper caller ID/calling number on outgoing
> calls. (I can do that since it?s an internal SIP trunk, so any callerID is
> ok)****
>
> ** **
>
> So I?ve created a partition and a CSS for transformations and added a
> Calling Party Transformation Pattern (Call Routing > Transformation >
> Transformation Pattern > Calling Party Transformation Pattern), applied it
> properly to the SIP trunk etc.****
>
> ** **
>
> For testing I have created a single test pattern, with my own extension:
> 2356****
>
> This matched and applied the transformations I was expecting. I tested it
> with changing the transformations, it kept working.****
>
> ** **
>
> However, when I rewrote the pattern to 2XXX it stopped matching. Basically
> it seems that I?m unable to use any non-specific pattern to match the
> calling party number. (neither 2!, nor 235X nor anything else that I?ve
> tried seems to match)****
>
> ** **
>
> Any ideas?****
>
> ** **
>
> Thanks,****
>
> *Zoltan Kelemen**
> *Emerson****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From erickbee at gmail.com Thu Jan 3 14:25:09 2013


From: erickbee at gmail.com (Erick B.)
Date: Thu, 3 Jan 2013 13:25:09 -0600
Subject: [cisco-voip] Call Recording on CUCM
In-Reply-To: <CAHgd+3_mp35-QojV+-RZFeiNfCb20XQ3EJMat9XgvUPy4Q4p_g@mail.gmail.com>
References: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
<CAHgd+3_mp35-QojV+-RZFeiNfCb20XQ3EJMat9XgvUPy4Q4p_g@mail.gmail.com>
Message-ID: <CAHSnBQymi0iW4ieZbkQ9UH=vtki8_iJ19QHyoCCqavo8tC0xRQ@mail.gmail.com>

To chime in on what others have said, with the built in bridge you enable
recording per DN. There is a option to record call calls or have it user or
application controlled on the CUCM side also. I haven't done it that way
yet. I tried user controlled on a 9971 but had problems with the record
button working.
The 3rd party applications all vary in way they operate and features, etc.
With some you can tell the 3rd party app what extensions to record, what
time, and just inbound calls or outbound, etc.

On Thu, Jan 3, 2013 at 1:03 PM, Scott Voll <svoll.voip at gmail.com> wrote:

> As others have already stated. you will need a 3rd party app. We ended up
> with CallRex. It has a Great Web interface for the end users and API's if
> you want to build a app to interface with it.
>
> As a added benefit, if you end up using Extension Mobility you don't have
> to associated phones to the Call Recording application. you just associate
> extMob profiles. That way if your not recording All phone in your org, you
> don't have to worry about associating the right phones.
>
> YMMV
>
> Scott
>
>
> On Wed, Jan 2, 2013 at 5:47 PM, Robert Hass <robhass at gmail.com> wrote:
>
>> Hi
>> My boss asked if we can enable call recording on our CUCM 8.6 (just CUCM
>> without Contact Center).
>> We considering two options of call recording
>> a) record all voice calls
>> b) record voice calls on demand - user can turn on/off recording via xml
>> application of softkey on the phone
>>
>> My question : Are above scenarios of call recording are possible on CUCM
>> ? What else I need - probably server for call-recording with big amount of
>> storange and some additional software (Zoom ? Cisco ?)
>>
>> thanks for help
>>
>> Rob
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From jsteinberg at gmail.com Thu Jan 3 15:30:00 2013
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 3 Jan 2013 15:30:00 -0500
Subject: [cisco-voip] VG350 support in 8.5 ?
Message-ID: <CACCAghbsbxC4mtObcUNh8EJo3-T-w4MzfbVVRgcdJCNfoHh=wQ@mail.gmail.com>

Does anyone know if CM 8.5.1su5 will support the VG350 ?

The VG350 release notes say 8.62a and 9.0.1, but in the 8.5.1su5 release
notes it mentions a resolved caveat around the VG350.

CSCtu07982 : QED changes to add support for VG350 gateways and service
modules

I am curious whether that means full support for the new gateway on 8.5.

Thanks.

Justin
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From rratliff at cisco.com Thu Jan 3 15:41:20 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 3 Jan 2013 15:41:20 -0500
Subject: [cisco-voip] VG350 support in 8.5 ?
In-Reply-To: <CACCAghbsbxC4mtObcUNh8EJo3-T-w4MzfbVVRgcdJCNfoHh=wQ@mail.gmail.com>
References: <CACCAghbsbxC4mtObcUNh8EJo3-T-w4MzfbVVRgcdJCNfoHh=wQ@mail.gmail.com>
Message-ID: <E79BBD61-3E38-4D9A-8A02-E31F9E61290A@cisco.com>

QED is what we call the stuff that adds device support to CCMAdmin. A device pack
or SU that's later than any version that bug is fixed in will get you VG350
support.

-Ryan

On Jan 3, 2013, at 3:30 PM, Justin Steinberg <jsteinberg at gmail.com> wrote:

Does anyone know if CM 8.5.1su5 will support the VG350 ?

The VG350 release notes say 8.62a and 9.0.1, but in the 8.5.1su5 release notes it
mentions a resolved caveat around the VG350.

CSCtu07982 : QED changes to add support for VG350 gateways and service modules

I am curious whether that means full support for the new gateway on 8.5.

Thanks.

Justin
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From jsteinberg at gmail.com Thu Jan 3 15:56:22 2013


From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 3 Jan 2013 15:56:22 -0500
Subject: [cisco-voip] VG350 support in 8.5 ?
In-Reply-To: <E79BBD61-3E38-4D9A-8A02-E31F9E61290A@cisco.com>
References: <CACCAghbsbxC4mtObcUNh8EJo3-T-w4MzfbVVRgcdJCNfoHh=wQ@mail.gmail.com>
<E79BBD61-3E38-4D9A-8A02-E31F9E61290A@cisco.com>
Message-ID: <CACCAghYCnqdgF=fFEj5euFxaPy_ZxVmnh16CpDO5Wrg1tt5MLw@mail.gmail.com>

Great, thanks.

On Thu, Jan 3, 2013 at 3:41 PM, Ryan Ratliff <rratliff at cisco.com> wrote:

> QED is what we call the stuff that adds device support to CCMAdmin. A
> device pack or SU that's later than any version that bug is fixed in will
> get you VG350 support.
>
> -Ryan
>
> On Jan 3, 2013, at 3:30 PM, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> Does anyone know if CM 8.5.1su5 will support the VG350 ?
>
> The VG350 release notes say 8.62a and 9.0.1, but in the 8.5.1su5 release
> notes it mentions a resolved caveat around the VG350.
>
> CSCtu07982 : QED changes to add support for VG350 gateways and service
> modules
>
> I am curious whether that means full support for the new gateway on 8.5.
>
> Thanks.
>
> Justin
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From Dennis.Heim at wwt.com Thu Jan 3 16:55:04 2013


From: Dennis.Heim at wwt.com (Heim, Dennis)
Date: Thu, 3 Jan 2013 15:55:04 -0600
Subject: [cisco-voip] SIP SRST Configuration
Message-ID: <0CC57FCAB07CEB4595526952471493D316F234407F@PRODCMS1.wwt.local>

Does anyone have a sip srst configuration I could look at?


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From VanMarenNP at ldschurch.org Thu Jan 3 18:20:59 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Thu, 3 Jan 2013 23:20:59 +0000
Subject: [cisco-voip] SIP SRST Configuration
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F234407F@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F234407F@PRODCMS1.wwt.local>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E44421@W12112.ldschurch.org>

SIP SRST

voice service voip

sip

registrar server

voice register global

Mode SRST

system message Blah

max-dn 100

max-pool 1

source-address 10.10.32.254 port 5060

voice register pool 1

translation-profile incoming SRST-IN

id network 0.0.0.0 mask 0.0.0.0

voice translation-rule 30

rule 1 /^\(1..\)/ /62144\1/

rule 1 /^\(19..\)/ /+15115115138\1/

voice translation-profile SRST-IN

translate called 30

interface Loopback10
ip address 192.168.255.255 255.255.255.255

call-manager-fallback

max-conferences 8 gain -6

transfer-system full-consult

ip source-address 10.19.13.5 port 2000

max-ephones 58

max-dn 10 octo-line

moh flash0:/MOH-MOTAB.wav

multicast moh 239.1.1.1 port 16384 route 192.168.255.255 10.1.68.268

ccm-manager music-on-hold

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Heim, Dennis
Sent: Thursday, January 03, 2013 2:55 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SIP SRST Configuration

Does anyone have a sip srst configuration I could look at?

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
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From JOrr at parknationalbank.com Thu Jan 3 20:54:49 2013


From: JOrr at parknationalbank.com (Orr, Jeff B.)
Date: Fri, 4 Jan 2013 01:54:49 +0000
Subject: [cisco-voip] Call Recording on CUCM
In-Reply-To: <CALNfFTHLmMJeiUTcSyWBP9PfmReg4WWKOMRa2HMsz1E-_4q14w@mail.gmail.com>
References: <CALNfFTG7BKtQ6dQqoLpp4Va8WP+=qsC5-094GP9_Vws6TzTvPQ@mail.gmail.com>
<72FE638DB23C1049AA265B28B5F87F3364A56940@prk-alford-mbx1.PRK.LOCAL>,
<CALNfFTHLmMJeiUTcSyWBP9PfmReg4WWKOMRa2HMsz1E-_4q14w@mail.gmail.com>
Message-ID: <FB3FA53C-BA3E-4740-93A6-CF3EC100FCB7@parknationalbank.com>

I am only doing selected DNs in a call center, and a group that has regulatory
requirements for calls to be recorded. Zoom?s on-demand feature allows the user to
select a call to be ?recorded? either during the call, or up to 30 minutes after it
is completed. This is done by an XML service on the call-recording server accessed
by a subscribed service on those users? phones.

With the ?pre-record? feature, all calls on line are recorded, but discarded after
a time period unless the user requests the call to be saved.

Jeff

Sent from my iPad

On Jan 3, 2013, at 8:27 AM, "Robert Hass" <robhass at gmail.com<mailto:robhass at


gmail.com>> wrote:

Thanks for info


Are you recording allcalls or only selected DNs ?
Is it possible to on/off recording by user using some XML application or soft key
on user's phone ?

Rob

On Thursday, January 3, 2013, Orr, Jeff B. wrote:


Hi Rob,

I just went through this for our environment. Call manager will provide the backend
requirements to do recordings. However, you will need a 3rd party software to
actually record and store the calls.

We evaluated several options and went with Zoom. It is a nice, Linux based
recording software. It fully supports spanless recordings and can function as
record all the time or on-demand recording. It does this by actually recording
every call, and then allowing a user to press a service button to record a call
that occurred earlier.

Jeff

____________________________________
Jeff Orr
Technical Services - Network Engineer
Park National Corporation (AMEX: PRK)

This message is confidential and is intended only for the named recipients, and may
contain information that is privileged, or exempt from disclosure under applicable
law. If you are not the intended recipients of the email, you are hereby notified
that the dissemination, distribution or copying of this e-mail or its contents is
strictly prohibited. If you received this e-mail in error, please notify the sender
at either the e-mail address or the phone number above and delete this e-mail from
your computer.

From: cisco-voip-bounces at puck.nether.net<javascript:_e({},%20'cvml',%20'cisco-


voip-bounces at puck.nether.net');> [mailto:cisco-voip-bounces at
puck.nether.net<javascript:_e({},%20'cvml',%20'cisco-voip-bounces at
puck.nether.net');>] On Behalf Of Robert Hass
Sent: Wednesday, January 02, 2013 8:47 PM
To: cisco-voip
Subject: [cisco-voip] Call Recording on CUCM
Hi
My boss asked if we can enable call recording on our CUCM 8.6 (just CUCM without
Contact Center).
We considering two options of call recording
a) record all voice calls
b) record voice calls on demand - user can turn on/off recording via xml
application of softkey on the phone

My question : Are above scenarios of call recording are possible on CUCM ? What
else I need - probably server for call-recording with big amount of storange and
some additional software (Zoom ? Cisco ?)

thanks for help

Rob

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From Robin.Clayton at rrfa.org.uk Fri Jan 4 05:41:17 2013


From: Robin.Clayton at rrfa.org.uk (Robin Clayton)
Date: Fri, 4 Jan 2013 10:41:17 +0000
Subject: [cisco-voip] adding multiple speed dial buttons to 7915
Message-ID: <883574BD6A83B04EAA11A5B54F4DFFE3B99B15@SV-C-EXCHMB-
01.richardrose.internal>

CCM 7.x

How can one bulk add speed dials to a 7915?

Every time I go to modify buttons I can only select SD once?

Rob

=========================
Robin Clayton

Senior I.T. Technician


Richard Rose Federation
Richard Rose Central Academy
Victoria Place
Carlisle
Cumbria
CA1 1LY

Tel: 01228 822075


www: www.rrfa.org.uk

-----------------------------------------------------------------------------------
---------------------------

Important Notice:

This e-mail and any attachment are confidentialand may be privileged


or otherwise protected from disclosure. It is solely intended for the
person(s) named above. If you are not the intended recipient, any
reading, use, disclosure, copying or distribution of all or parts of this
e-mail or associated attachments is strictly prohibited. If you are not
an intended recipient, please notify the sender immediately by
replying to this message or by telephone and delete this e-mail and
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From Robin.Clayton at rrfa.org.uk Fri Jan 4 07:12:29 2013


From: Robin.Clayton at rrfa.org.uk (Robin Clayton)
Date: Fri, 4 Jan 2013 12:12:29 +0000
Subject: [cisco-voip] adding multiple speed dial buttons to 7915
In-Reply-To: <883574BD6A83B04EAA11A5B54F4DFFE3B99B15@SV-C-EXCHMB-
01.richardrose.internal>
References: <883574BD6A83B04EAA11A5B54F4DFFE3B99B15@SV-C-EXCHMB-
01.richardrose.internal>
Message-ID: <883574BD6A83B04EAA11A5B54F4DFFE3B99B4A@SV-C-EXCHMB-
01.richardrose.internal>

Sorted....

Created a new phone template with 54 buttons.

Now to remember how to use bat...

Rob

=========================
Robin Clayton

Senior I.T. Technician


Richard Rose Federation
Richard Rose Central Academy
Victoria Place
Carlisle
Cumbria
CA1 1LY

Tel: 01228 822075


www: www.rrfa.org.uk

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Robin Clayton
Sent: 04 January 2013 10:41
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] adding multiple speed dial buttons to 7915

CCM 7.x

How can one bulk add speed dials to a 7915?

Every time I go to modify buttons I can only select SD once?


Rob

=========================
Robin Clayton

Senior I.T. Technician


Richard Rose Federation
Richard Rose Central Academy
Victoria Place
Carlisle
Cumbria
CA1 1LY

Tel: 01228 822075


www: www.rrfa.org.uk

-----------------------------------------------------------------------------------
---------------------------

Important Notice:

This e-mail and any attachment are confidentialand may be privileged


or otherwise protected from disclosure. It is solely intended for the
person(s) named above. If you are not the intended recipient, any
reading, use, disclosure, copying or distribution of all or parts of this
e-mail or associated attachments is strictly prohibited. If you are not
an intended recipient, please notify the sender immediately by
replying to this message or by telephone and delete this e-mail and
any attachments permanently from your system.
The Richard Rose Federation<http://www.rrfa.org.uk>

-----------------------------------------------------------------------------------
---------------------------

Important Notice:

This e-mail and any attachment are confidentialand may be privileged


or otherwise protected from disclosure. It is solely intended for the
person(s) named above. If you are not the intended recipient, any
reading, use, disclosure, copying or distribution of all or parts of this
e-mail or associated attachments is strictly prohibited. If you are not
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From rkulagow at gmail.com Fri Jan 4 09:35:41 2013


From: rkulagow at gmail.com (Robert Kulagowski)
Date: Fri, 4 Jan 2013 08:35:41 -0600
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
Message-ID: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
Since no one who knows anything for real is probably going to say
anything for now, are there any mitigating factors that I can start
thinking about once management sees the following article?

http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-phones-
vulnerable-to-eavesdropping-hack-researchers-say?lite

From ealeatherman at gmail.com Fri Jan 4 10:00:26 2013


From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 4 Jan 2013 10:00:26 -0500
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
Message-ID: <CAFC4dsoeSVQZ86+MnC1nxmbDzuCZAPbiPapUcU=A8ZtEdCpG4A@mail.gmail.com>

Hah i just had someone ask me about this same article this morning. There
was a article on it in IEEE Spectrum also - neither article seemed to give
enough info for customers to take specific action on.

On Fri, Jan 4, 2013 at 9:35 AM, Robert Kulagowski <rkulagow at gmail.com>wrote:

> Since no one who knows anything for real is probably going to say
> anything for now, are there any mitigating factors that I can start
> thinking about once management sees the following article?
>
>
> http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-phones-
vulnerable-to-eavesdropping-hack-researchers-say?lite
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

--
Ed Leatherman
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From svoll.voip at gmail.com Fri Jan 4 10:02:45 2013


From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 4 Jan 2013 07:02:45 -0800
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
Message-ID: <CAHgd+38U2ZKTRt-w16f2d66dnuuz6HX7pXJqbHTYDXPAufGKsQ@mail.gmail.com>

Lelio sent this out a week or two ago.


http://m.spectrum.ieee.org/computing/embedded-systems/cisco-ip-phones-vulnerable
Check out the video.

We are a closed facility, so the attacker would have to either be inside,


or take a phone off the wall in a reception area AND have SSH access.
I talked to my SE and he said:
Workaround = Restrict SSH and CLI access to trusted users only.
Administrators may consider leveraging 802.1x device authentication to
prevent unauthorized devices or systems from accessing the voice network.

Ang accomplished this by first gaining access to the device via SSH and
utilizing TFTP to pull down a malicious binary that is designed to exploit
the insufficient validation issue of the affected System Calls. He ran this
from the user context on the device which performed the exploit. The
caveats of this particular issue are that an attacker would need to have
Authenticated Access either via SSH (Which would need to be enabled, it is
not enabled by default), or local access via the Serial port. The attacker
would also need to be able to point the device at an attacker-controlled
TFTP server to retrieve the payload.

YMMV

Scott

On Fri, Jan 4, 2013 at 6:35 AM, Robert Kulagowski <rkulagow at gmail.com>wrote:

> Since no one who knows anything for real is probably going to say
> anything for now, are there any mitigating factors that I can start
> thinking about once management sees the following article?
>
>
> http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-phones-
vulnerable-to-eavesdropping-hack-researchers-say?lite
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From chrward at cisco.com Fri Jan 4 10:22:22 2013


From: chrward at cisco.com (Chris Ward (chrward))
Date: Fri, 4 Jan 2013 15:22:22 +0000
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <CAHgd+38U2ZKTRt-w16f2d66dnuuz6HX7pXJqbHTYDXPAufGKsQ@mail.gmail.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
<CAHgd+38U2ZKTRt-w16f2d66dnuuz6HX7pXJqbHTYDXPAufGKsQ@mail.gmail.com>
Message-ID: <C3D1FCA271936B48839E081F898E17AA1CC1D7@xmb-rcd-x13.cisco.com>

Also, this does NOT affect 7940s and 7960s as they don't run linux which is basis
of the exploit.

+Chris
Unity Connection TME

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Scott Voll
Sent: Friday, January 04, 2013 10:03 AM
To: Robert Kulagowski
Cc: Cisco VOIP
Subject: Re: [cisco-voip] Cisco phones vulnerable to hack / remote access?

Lelio sent this out a week or two ago.


http://m.spectrum.ieee.org/computing/embedded-systems/cisco-ip-phones-vulnerable
Check out the video.

We are a closed facility, so the attacker would have to either be inside, or take a
phone off the wall in a reception area AND have SSH access.

I talked to my SE and he said:


Workaround = Restrict SSH and CLI access to trusted users only. Administrators may
consider leveraging 802.1x device authentication to prevent unauthorized devices or
systems from accessing the voice network.

Ang accomplished this by first gaining access to the device via SSH and utilizing
TFTP to pull down a malicious binary that is designed to exploit the insufficient
validation issue of the affected System Calls. He ran this from the user context on
the device which performed the exploit. The caveats of this particular issue are
that an attacker would need to have Authenticated Access either via SSH (Which
would need to be enabled, it is not enabled by default), or local access via the
Serial port. The attacker would also need to be able to point the device at an
attacker-controlled TFTP server to retrieve the payload.
YMMV
Scott

On Fri, Jan 4, 2013 at 6:35 AM, Robert Kulagowski <rkulagow at


gmail.com<mailto:rkulagow at gmail.com>> wrote:
Since no one who knows anything for real is probably going to say
anything for now, are there any mitigating factors that I can start
thinking about once management sees the following article?

http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-phones-
vulnerable-to-eavesdropping-hack-researchers-say?lite
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From Robin.Clayton at rrfa.org.uk Fri Jan 4 10:37:53 2013


From: Robin.Clayton at rrfa.org.uk (Robin Clayton)
Date: Fri, 4 Jan 2013 15:37:53 +0000
Subject: [cisco-voip] Adding Speed Dials to 7915 using BAT ??
Message-ID: <883574BD6A83B04EAA11A5B54F4DFFE3B99C14@SV-C-EXCHMB-
01.richardrose.internal>

Guys...

CCM 7.1.5.33900-10
I have been bashing my head on this one all afternoon

I have one phone with 2 7915's

I am trying to add 50 speed dials but can't get it to work.

I followed a Cisco guide which used "insert specific phone details" and only
succeeded in wiping the phone config apart from those detail ( and some template
details).

I have tried update but that finds no records in my csv file???

Anyone got a real world working example to update BLF SD's

Cheers

Rob

=========================
Robin Clayton

Senior I.T. Technician


Richard Rose Federation
Richard Rose Central Academy
Victoria Place
Carlisle
Cumbria
CA1 1LY

Tel: 01228 822075


www: www.rrfa.org.uk

-----------------------------------------------------------------------------------
---------------------------

Important Notice:

This e-mail and any attachment are confidentialand may be privileged


or otherwise protected from disclosure. It is solely intended for the
person(s) named above. If you are not the intended recipient, any
reading, use, disclosure, copying or distribution of all or parts of this
e-mail or associated attachments is strictly prohibited. If you are not
an intended recipient, please notify the sender immediately by
replying to this message or by telephone and delete this e-mail and
any attachments permanently from your system.

The Richard Rose Federation<http://www.rrfa.org.uk>


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From Robin.Clayton at rrfa.org.uk Fri Jan 4 10:40:09 2013


From: Robin.Clayton at rrfa.org.uk (Robin Clayton)
Date: Fri, 4 Jan 2013 15:40:09 +0000
Subject: [cisco-voip] Adding Speed Dials to 7915 using BAT ??
In-Reply-To: <883574BD6A83B04EAA11A5B54F4DFFE3B99C14@SV-C-EXCHMB-
01.richardrose.internal>
References: <883574BD6A83B04EAA11A5B54F4DFFE3B99C14@SV-C-EXCHMB-
01.richardrose.internal>
Message-ID: <883574BD6A83B04EAA11A5B54F4DFFE3B99C25@SV-C-EXCHMB-
01.richardrose.internal>

Sorry

Using BAT import/update...

Rob

=========================
Robin Clayton

Senior I.T. Technician


Richard Rose Federation
Richard Rose Central Academy
Victoria Place
Carlisle
Cumbria
CA1 1LY

Tel: 01228 822075


www: www.rrfa.org.uk

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Robin Clayton
Sent: 04 January 2013 15:38
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Adding Speed Dials to 7915 using BAT ??

Guys...

CCM 7.1.5.33900-10

I have been bashing my head on this one all afternoon

I have one phone with 2 7915's

I am trying to add 50 speed dials but can't get it to work.

I followed a Cisco guide which used "insert specific phone details" and only
succeeded in wiping the phone config apart from those detail ( and some template
details).

I have tried update but that finds no records in my csv file???

Anyone got a real world working example to update BLF SD's

Cheers

Rob

=========================
Robin Clayton
Senior I.T. Technician
Richard Rose Federation
Richard Rose Central Academy
Victoria Place
Carlisle
Cumbria
CA1 1LY

Tel: 01228 822075


www: www.rrfa.org.uk

-----------------------------------------------------------------------------------
---------------------------

Important Notice:

This e-mail and any attachment are confidentialand may be privileged


or otherwise protected from disclosure. It is solely intended for the
person(s) named above. If you are not the intended recipient, any
reading, use, disclosure, copying or distribution of all or parts of this
e-mail or associated attachments is strictly prohibited. If you are not
an intended recipient, please notify the sender immediately by
replying to this message or by telephone and delete this e-mail and
any attachments permanently from your system.
The Richard Rose Federation<http://www.rrfa.org.uk>

-----------------------------------------------------------------------------------
---------------------------

Important Notice:

This e-mail and any attachment are confidentialand may be privileged


or otherwise protected from disclosure. It is solely intended for the
person(s) named above. If you are not the intended recipient, any
reading, use, disclosure, copying or distribution of all or parts of this
e-mail or associated attachments is strictly prohibited. If you are not
an intended recipient, please notify the sender immediately by
replying to this message or by telephone and delete this e-mail and
any attachments permanently from your system.

The Richard Rose Federation<http://www.rrfa.org.uk>


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From matthnick at gmail.com Fri Jan 4 10:47:55 2013


From: matthnick at gmail.com (Nick Matthews)
Date: Fri, 4 Jan 2013 10:47:55 -0500
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <CAFC4dsoeSVQZ86+MnC1nxmbDzuCZAPbiPapUcU=A8ZtEdCpG4A@mail.gmail.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
<CAFC4dsoeSVQZ86+MnC1nxmbDzuCZAPbiPapUcU=A8ZtEdCpG4A@mail.gmail.com>
Message-ID: <CAM-K-NrV9ZF8K6j6Ni9uQV=pcVW0MpgC5ChJovu6wVru++Y=EQ@mail.gmail.com>

This may help:


https://psirt.cisco.com/PSIRThot/7900KernelSysCall

Particularly that they need physical access or the user authentication


details for this attack to happen.

-nick

On Fri, Jan 4, 2013 at 10:00 AM, Ed Leatherman <ealeatherman at gmail.com>wrote:

> Hah i just had someone ask me about this same article this morning. There
> was a article on it in IEEE Spectrum also - neither article seemed to give
> enough info for customers to take specific action on.
>
>
> On Fri, Jan 4, 2013 at 9:35 AM, Robert Kulagowski <rkulagow at gmail.com>wrote:
>
>> Since no one who knows anything for real is probably going to say
>> anything for now, are there any mitigating factors that I can start
>> thinking about once management sees the following article?
>>
>>
>> http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-phones-
vulnerable-to-eavesdropping-hack-researchers-say?lite
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
>
> --
> Ed Leatherman
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From blake at pfankuch.me Fri Jan 4 13:44:24 2013


From: blake at pfankuch.me (Blake Pfankuch)
Date: Fri, 4 Jan 2013 18:44:24 +0000
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <CAM-K-NrV9ZF8K6j6Ni9uQV=pcVW0MpgC5ChJovu6wVru++Y=EQ@mail.gmail.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
<CAFC4dsoeSVQZ86+MnC1nxmbDzuCZAPbiPapUcU=A8ZtEdCpG4A@mail.gmail.com>
<CAM-K-NrV9ZF8K6j6Ni9uQV=pcVW0MpgC5ChJovu6wVru++Y=EQ@mail.gmail.com>
Message-ID: <CC75EEBF17C7374EA8309102B7B10C840109D47A4A@SHSBS.shenrons-house.local>

Uhhhhh....

[blake at shlt01-centos ~]# host psirt.cisco.com


psirt.cisco.com has address 172.18.104.137

I see a problem...

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Nick Matthews
Sent: Friday, January 04, 2013 8:48 AM
To: Ed Leatherman
Cc: Cisco VOIP
Subject: Re: [cisco-voip] Cisco phones vulnerable to hack / remote access?

This may help:


https://psirt.cisco.com/PSIRThot/7900KernelSysCall
Particularly that they need physical access or the user authentication details for
this attack to happen.
-nick

On Fri, Jan 4, 2013 at 10:00 AM, Ed Leatherman <ealeatherman at


gmail.com<mailto:ealeatherman at gmail.com>> wrote:
Hah i just had someone ask me about this same article this morning. There was a
article on it in IEEE Spectrum also - neither article seemed to give enough info
for customers to take specific action on.

On Fri, Jan 4, 2013 at 9:35 AM, Robert Kulagowski <rkulagow at


gmail.com<mailto:rkulagow at gmail.com>> wrote:
Since no one who knows anything for real is probably going to say
anything for now, are there any mitigating factors that I can start
thinking about once management sees the following article?

http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-phones-
vulnerable-to-eavesdropping-hack-researchers-say?lite
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

--
Ed Leatherman

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From lisa.notarianni at scranton.edu Fri Jan 4 14:00:52 2013


From: lisa.notarianni at scranton.edu (Lisa Notarianni)
Date: Fri, 4 Jan 2013 19:00:52 +0000
Subject: [cisco-voip] Hunt Group Question
Message-ID:
<B1D1594F698E684DB83C15828D282DD856920284@SN2PRD0310MB359.namprd03.prod.outlook.com
>
We currently do not have any groups on campus using hunt groups but have a possible
need to.
Out Financial Aid office has peak busy times when they would like to use a hunt
group. Is anyone out there familiar with the details of using them? I have some
questions I would appreciate help with:

1. Is there a tool for an administrator or office manager to use to see who


is logged into the hunt group?

2. Can we choose which phone to start sending the incoming calls to? (Main
receptionist first and then whoever else is available in the hunt group) - note -
they will all be answering the same DN.

3. If someone is logged in and a call is sent to their phone but they cannot
answer it, can you control how many times it will ring before going to the next
person in line to answer a call?

4. If someone is available - but it is not their turn to answer the call -


but they want to answer the call - can they?

5. Can you be logged into more than 1 hunt group?

6. Is there a report that shows which phone answered X number of calls,


etc...?

7. Our goal is to eventually integrate UCCX. If you have UCCX - can the
calls go from there to a hunt group?
Thank you in advance!

[LisaNotarianniSignature]

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From ealeatherman at gmail.com Fri Jan 4 14:11:24 2013


From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 4 Jan 2013 14:11:24 -0500
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <CAHgd+38U2ZKTRt-w16f2d66dnuuz6HX7pXJqbHTYDXPAufGKsQ@mail.gmail.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
<CAHgd+38U2ZKTRt-w16f2d66dnuuz6HX7pXJqbHTYDXPAufGKsQ@mail.gmail.com>
Message-ID: <CAFC4dsp8XVUYSmiTG5FQAv=DtR7p5YYuyhROhy2S41Or8F+7-Q@mail.gmail.com>

I completely missed the video at the top of the IEEE article the first time
i read it.. i think my brain saw it as an advertisement and just ignored it.

The researchers full presentation is here also:


http://www.youtube.com/watch?v=f3zUOZcewtA&feature=youtu.be

On Fri, Jan 4, 2013 at 10:02 AM, Scott Voll <svoll.voip at gmail.com> wrote:

> Lelio sent this out a week or two ago.


> http://m.spectrum.ieee.org/computing/embedded-systems/cisco-ip-phones-vulnerable
Check out the video.
>
> We are a closed facility, so the attacker would have to either be inside,
> or take a phone off the wall in a reception area AND have SSH access.
>
> I talked to my SE and he said:
> Workaround = Restrict SSH and CLI access to trusted users only.
> Administrators may consider leveraging 802.1x device authentication to
> prevent unauthorized devices or systems from accessing the voice network.
>
> Ang accomplished this by first gaining access to the device via SSH and
> utilizing TFTP to pull down a malicious binary that is designed to exploit
> the insufficient validation issue of the affected System Calls. He ran this
> from the user context on the device which performed the exploit. The
> caveats of this particular issue are that an attacker would need to have
> Authenticated Access either via SSH (Which would need to be enabled, it is
> not enabled by default), or local access via the Serial port. The attacker
> would also need to be able to point the device at an attacker-controlled
> TFTP server to retrieve the payload.
>
> YMMV
>
> Scott
>
>
>
>
>
> On Fri, Jan 4, 2013 at 6:35 AM, Robert Kulagowski <rkulagow at gmail.com>wrote:
>
>> Since no one who knows anything for real is probably going to say
>> anything for now, are there any mitigating factors that I can start
>> thinking about once management sees the following article?
>>
>>
>> http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-phones-
vulnerable-to-eavesdropping-hack-researchers-say?lite
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
--
Ed Leatherman
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From jsteinberg at gmail.com Fri Jan 4 14:21:39 2013


From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 4 Jan 2013 14:21:39 -0500
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <CAFC4dsp8XVUYSmiTG5FQAv=DtR7p5YYuyhROhy2S41Or8F+7-Q@mail.gmail.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
<CAHgd+38U2ZKTRt-w16f2d66dnuuz6HX7pXJqbHTYDXPAufGKsQ@mail.gmail.com>
<CAFC4dsp8XVUYSmiTG5FQAv=DtR7p5YYuyhROhy2S41Or8F+7-Q@mail.gmail.com>
Message-ID: <CACCAghZZzjvXbDRoiwQ6ND5NuqELg4Vcsy12i3w+RDt_ped9kQ@mail.gmail.com>

Nick's link seems like an internal site. I don't see anything on the
public psirt page.

http://tools.cisco.com/security/center/publicationListing.x#~CiscoSecurityAdvisory

On Fri, Jan 4, 2013 at 2:11 PM, Ed Leatherman <ealeatherman at gmail.com>wrote:

> I completely missed the video at the top of the IEEE article the first
> time i read it.. i think my brain saw it as an advertisement and just
> ignored it.
>
> The researchers full presentation is here also:
> http://www.youtube.com/watch?v=f3zUOZcewtA&feature=youtu.be
>
>
> On Fri, Jan 4, 2013 at 10:02 AM, Scott Voll <svoll.voip at gmail.com> wrote:
>
>> Lelio sent this out a week or two ago.
>> http://m.spectrum.ieee.org/computing/embedded-systems/cisco-ip-phones-vulnerable
Check out the video.
>>
>> We are a closed facility, so the attacker would have to either be inside,
>> or take a phone off the wall in a reception area AND have SSH access.
>>
>> I talked to my SE and he said:
>> Workaround = Restrict SSH and CLI access to trusted users only.
>> Administrators may consider leveraging 802.1x device authentication to
>> prevent unauthorized devices or systems from accessing the voice network.
>>
>> Ang accomplished this by first gaining access to the device via SSH and
>> utilizing TFTP to pull down a malicious binary that is designed to exploit
>> the insufficient validation issue of the affected System Calls. He ran this
>> from the user context on the device which performed the exploit. The
>> caveats of this particular issue are that an attacker would need to have
>> Authenticated Access either via SSH (Which would need to be enabled, it is
>> not enabled by default), or local access via the Serial port. The attacker
>> would also need to be able to point the device at an attacker-controlled
>> TFTP server to retrieve the payload.
>>
>> YMMV
>>
>> Scott
>>
>>
>>
>>
>>
>> On Fri, Jan 4, 2013 at 6:35 AM, Robert Kulagowski <rkulagow at gmail.com>wrote:
>>
>>> Since no one who knows anything for real is probably going to say
>>> anything for now, are there any mitigating factors that I can start
>>> thinking about once management sees the following article?
>>>
>>>
>>> http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-phones-
vulnerable-to-eavesdropping-hack-researchers-say?lite
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
> --
> Ed Leatherman
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From afrankel at cisco.com Fri Jan 4 14:24:57 2013


From: afrankel at cisco.com (Adam Frankel)
Date: Fri, 04 Jan 2013 14:24:57 -0500
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <CAFC4dsp8XVUYSmiTG5FQAv=DtR7p5YYuyhROhy2S41Or8F+7-Q@mail.gmail.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
<CAHgd+38U2ZKTRt-w16f2d66dnuuz6HX7pXJqbHTYDXPAufGKsQ@mail.gmail.com>
<CAFC4dsp8XVUYSmiTG5FQAv=DtR7p5YYuyhROhy2S41Or8F+7-Q@mail.gmail.com>
Message-ID: <50E72C89.9050400@cisco.com>

PSIRT will be including all updated information related to this on the


defect, CSCuc83860.

Adam
------------------------------------------------------------------------
*From:* Ed Leatherman <ealeatherman at gmail.com>
*Sent:* Fri, Jan 04, 2013 2:11:24 PM
*To:* Scott Voll <svoll.voip at gmail.com>
*CC:* Cisco VOIP <cisco-voip at puck.nether.net>
*Subject:* Re: [cisco-voip] Cisco phones vulnerable to hack / remote access?

> I completely missed the video at the top of the IEEE article the first
> time i read it.. i think my brain saw it as an advertisement and just
> ignored it.
>
> The researchers full presentation is here also:
> http://www.youtube.com/watch?v=f3zUOZcewtA&feature=youtu.be
>
>
> On Fri, Jan 4, 2013 at 10:02 AM, Scott Voll <svoll.voip at gmail.com
> <mailto:svoll.voip at gmail.com>> wrote:
>
> Lelio sent this out a week or two ago.
> http://m.spectrum.ieee.org/computing/embedded-systems/cisco-ip-phones-
vulnerable
> Check out the video.
>
> We are a closed facility, so the attacker would have to either be
> inside, or take a phone off the wall in a reception area AND have
> SSH access.
>
> I talked to my SE and he said:
> Workaround = Restrict SSH and CLI access to trusted users only.
> Administrators may consider leveraging 802.1x device
> authentication to prevent unauthorized devices or systems from
> accessing the voice network.
>
> Ang accomplished this by first gaining access to the device via
> SSH and utilizing TFTP to pull down a malicious binary that is
> designed to exploit the insufficient validation issue of the
> affected System Calls. He ran this from the user context on the
> device which performed the exploit. The caveats of this particular
> issue are that an attacker would need to have Authenticated Access
> either via SSH (Which would need to be enabled, it is not enabled
> by default), or local access via the Serial port. The attacker
> would also need to be able to point the device at an
> attacker-controlled TFTP server to retrieve the payload.
>
> YMMV
>
> Scott
>
>
>
>
> On Fri, Jan 4, 2013 at 6:35 AM, Robert Kulagowski
> <rkulagow at gmail.com <mailto:rkulagow at gmail.com>> wrote:
>
> Since no one who knows anything for real is probably going to say
> anything for now, are there any mitigating factors that I can
> start
> thinking about once management sees the following article?
>
> http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-
phones-vulnerable-to-eavesdropping-hack-researchers-say?lite
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> --
> Ed Leatherman
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From thomaslemay at comcast.net Fri Jan 4 14:40:21 2013


From: thomaslemay at comcast.net (Thomas LeMay)
Date: Fri, 4 Jan 2013 14:40:21 -0500
Subject: [cisco-voip] 7962 Phone Sets Failing Authentication
Message-ID: <002401cdeab3$5566d2e0$003478a0$@comcast.net>

We have recently received several boxes of new 7962 phone sets from the
Cisco factory. We are running call manager version 7.1.5.34900-7 on MCS
servers (Linux). When we take several of these new 7962 phone sets and plug
them into the network port they will pull a TAPS number successfully but
fail the authentication and software upgrade portion for phone load
SCCP4.2.9-2-1S.. Is there a known work around or middle firmware version we
need to load on a TFTP server to update the phone sets? Thank you in
advance.

Tom

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From rratliff at cisco.com Fri Jan 4 14:54:22 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 4 Jan 2013 14:54:22 -0500
Subject: [cisco-voip] 7962 Phone Sets Failing Authentication
In-Reply-To: <002401cdeab3$5566d2e0$003478a0$@comcast.net>
References: <002401cdeab3$5566d2e0$003478a0$@comcast.net>
Message-ID: <9588778D-E912-4CB5-9B9E-C34C27D67941@cisco.com>

What load is coming on them? If it's old enough it can't go straight to 9-2-1 and
has to hit an interim first.
See
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/firmware/9_2_1/english/release
/notes/7900_921.html#wp42493.

-Ryan

On Jan 4, 2013, at 2:40 PM, Thomas LeMay <thomaslemay at comcast.net> wrote:

We have recently received several boxes of new 7962 phone sets from the Cisco
factory. We are running call manager version 7.1.5.34900-7 on MCS servers (Linux).
When we take several of these new 7962 phone sets and plug them into the network
port they will pull a TAPS number successfully but fail the authentication and
software upgrade portion for phone load SCCP4.2.9-2-1S.. Is there a known work
around or middle firmware version we need to load on a TFTP server to update the
phone sets? Thank you in advance.

Tom
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From florian.kroessbacher at gmail.com Fri Jan 4 15:01:34 2013


From: florian.kroessbacher at gmail.com (Florian Kroessbacher)
Date: Fri, 4 Jan 2013 21:01:34 +0100
Subject: [cisco-voip] 7962 Phone Sets Failing Authentication
In-Reply-To: <002401cdeab3$5566d2e0$003478a0$@comcast.net>
References: <002401cdeab3$5566d2e0$003478a0$@comcast.net>
Message-ID: <BA0F7185-481F-4C44-B912-50B1C240BC73@gmail.com>

Hy

mabe your hardware is very very new than u can hit this

read here for

http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7900_series/firmware/9_3_1SR1/
release_notes/P790_BK_R4E1E768_00_rn-9_3_1_sr1-7900-
series_chapter_00.html#P790_RF_F4EA96A6_00

Features Available with Firmware Release


The following sections describe the features available in the firmware.
Hardware Updates
Hardware Updates
The hardware updates improve the compatibility of internal phone components.
The following table lists the updated hardware versions that require this release.
Phone
Hardware Version
Cisco Unified Wireless IP Phone 7942G
15.0 and higher
Cisco Unified Wireless IP Phone 7962G
15.0 and higher
Cisco Unified Wireless IP Phone 7945G
13.0 and higher
Cisco Unified Wireless IP Phone 7965G
13.0 and higher
Cisco Unified Wireless IP Phone 7975G
12.0 and higher
Phones manufactured with these hardware versions must run Firmware Release
9.3(1)SR1 or later. The phone firmware does not allow the phone to be downgraded to
releases earlier than Release 9.3(1)SR1.

--
Florian Kroessbacher
gmail: florian.kroessbacher at gmail.com

Am 04.01.2013 um 20:40 schrieb "Thomas LeMay" <thomaslemay at comcast.net>:

> We have recently received several boxes of new 7962 phone sets from the Cisco
factory. We are running call manager version 7.1.5.34900-7 on MCS servers (Linux).
When we take several of these new 7962 phone sets and plug them into the network
port they will pull a TAPS number successfully but fail the authentication and
software upgrade portion for phone load SCCP4.2.9-2-1S.. Is there a known work
around or middle firmware version we need to load on a TFTP server to update the
phone sets? Thank you in advance.
>
> Tom
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From chrward at cisco.com Fri Jan 4 15:06:27 2013


From: chrward at cisco.com (Chris Ward (chrward))
Date: Fri, 4 Jan 2013 20:06:27 +0000
Subject: [cisco-voip] 7962 Phone Sets Failing Authentication
In-Reply-To: <002401cdeab3$5566d2e0$003478a0$@comcast.net>
References: <002401cdeab3$5566d2e0$003478a0$@comcast.net>
Message-ID: <C3D1FCA271936B48839E081F898E17AA1CC6FF@xmb-rcd-x13.cisco.com>

What's the current load on the phone? There was an old issue where 8.3.2 and
earlier couldn't upgrade directly to 8.5.1 or later without an interim release. If
you move them to something like an 8.4.4 first or 8.3.3 first, that would work
around this issue.

https://supportforums.cisco.com/docs/DOC-24326

+Chris
Unity Connection TME

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Thomas LeMay
Sent: Friday, January 04, 2013 2:40 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7962 Phone Sets Failing Authentication

We have recently received several boxes of new 7962 phone sets from the Cisco
factory. We are running call manager version 7.1.5.34900-7 on MCS servers (Linux).
When we take several of these new 7962 phone sets and plug them into the network
port they will pull a TAPS number successfully but fail the authentication and
software upgrade portion for phone load SCCP4.2.9-2-1S.. Is there a known work
around or middle firmware version we need to load on a TFTP server to update the
phone sets? Thank you in advance.

Tom
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From thomaslemay at comcast.net Fri Jan 4 15:18:15 2013


From: thomaslemay at comcast.net (Thomas LeMay)
Date: Fri, 4 Jan 2013 15:18:15 -0500
Subject: [cisco-voip] 7962 Phone Sets Failing Authentication
In-Reply-To: <C3D1FCA271936B48839E081F898E17AA1CC6FF@xmb-rcd-x13.cisco.com>
References: <002401cdeab3$5566d2e0$003478a0$@comcast.net>
<C3D1FCA271936B48839E081F898E17AA1CC6FF@xmb-rcd-x13.cisco.com>
Message-ID: <003e01cdeab8$a11c7050$e35550f0$@comcast.net>

Hi, Chris,

The current load for SCCP 7962 is SCCP42.9-2-1S.

Tom

From: Chris Ward (chrward) [mailto:chrward at cisco.com]


Sent: Friday, January 04, 2013 3:06 PM
To: thomaslemay at comcast.net; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] 7962 Phone Sets Failing Authentication

What's the current load on the phone? There was an old issue where 8.3.2 and
earlier couldn't upgrade directly to 8.5.1 or later without an interim
release. If you move them to something like an 8.4.4 first or 8.3.3 first,
that would work around this issue.

https://supportforums.cisco.com/docs/DOC-24326

+Chris
Unity Connection TME

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Thomas LeMay
Sent: Friday, January 04, 2013 2:40 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7962 Phone Sets Failing Authentication

We have recently received several boxes of new 7962 phone sets from the
Cisco factory. We are running call manager version 7.1.5.34900-7 on MCS
servers (Linux). When we take several of these new 7962 phone sets and plug
them into the network port they will pull a TAPS number successfully but
fail the authentication and software upgrade portion for phone load
SCCP4.2.9-2-1S.. Is there a known work around or middle firmware version we
need to load on a TFTP server to update the phone sets? Thank you in
advance.

Tom

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From rratliff at cisco.com Fri Jan 4 15:26:18 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 4 Jan 2013 15:26:18 -0500
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <C75AF2AD9308C246AFBDDB994E3E298311081C47@PHANES.helion.local>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
<CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
<00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>
<CABzsfHz0Km6uPom_mhLkU9iMiZMawubm2JXAG1+uaJ+_wKACug@mail.gmail.com>
<5992A434-99D1-406D-894A-02F51A195AA7@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E298311081C47@PHANES.helion.local>
Message-ID: <01373FBF-B7E2-4803-89CB-4F47569A181E@cisco.com>

To close the loop on this the restriction was added in 8.6 (when refresh upgrade
came in) because we don't test this upgrade between major versions and the fact
that we started having to do OS reinstalls (refresh) for some combinations made the
likelihood of failure too high.

It is not documented, and that will remedied in the Release Notes for 9.1 shortly
and in the Upgrade/Install docs at some point in the future (they can't be changed
as fast as release notes).

-Ryan
On Jan 2, 2013, at 4:34 PM, Matthew Loraditch <MLoraditch at
heliontechnologies.com> wrote:

Well that?s good, I can just put a PUT order in edelivery and get it. Let?s see if
it works.

Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter | Facebook | Website | Email Support

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, January 02, 2013 4:22 PM
To: Tim Frazee
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

I was told this restriction was added around 8.5 but I'm still waiting on some
other folks to comment.

To get to 9.1 you either do a fresh install or you upgrade, same as any other
version. I understand the release of 9.1 has immediately replaced 9.0 on new 9.x
orders (much like 8.6 did for 8.5) so any 9.x media kit ordered today will be sent
9.1 bootable media.

-Ryan

On Jan 2, 2013, at 4:12 PM, Tim Frazee <tfrazee at gmail.com> wrote:

I could see that.

But tell that to my 8.5 upgrade from an 8.0 boot disk I was able to do back in the
day......

In short, you say that the only way currently to get to 9.1 is upgrade from an
already installed support version, not during the install process.

for the record and I know its not supported, I did try the hack of grabbing the
boot info file from 9.0 and pushing it into the 9.1 iso. The install process failed
post installing everything.

Thanks for digging Ryan.

On Wed, Jan 2, 2013 at 12:21 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
Confirmed I see it here in the lab and it looks to be intentional, though I'm still
digging.
Initial word is for a while now upgrade-during-install is only supported to the
same major/minor version.

Anything beyond that requires a separate upgrade after install.

-Ryan

On Jan 2, 2013, at 11:18 AM, Tim Frazee <tfrazee at gmail.com> wrote:

I did just that. after I tried with the pre-release, I used my NFR iso. Same
result.

I only used the pre-release because it was already on my datastore and i was
feeling a bit lazy over vacation. After I attempted the same procedure with 9.0(1)
-37 iso, I received the exact same error.

Ryan, should I be able to boot off of 9.0 and upgrade-during-install with 9.1?

On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com> wrote:
9.0.0.99101-22 is not a 9.0 ES, it's a pre-release build of 9.1. Throw it in the
trash and try with a real 9.0 build (I'm going to start this now).

-Ryan

On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com> wrote:

I didnt test to see if the 9.1 from CCO is bootable. In the past they havent been.

attached is a screenshoot of the error I received when I tried to feed the 9.1 via
CCO during a booted-from-nfr9.0 media

On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at ldschurch.org> wrote:
I had lots of problems doing upgrade during installs with 9.0 ESs. The ESs are
usually bootable so I just gave up and installed fresh. Is the 9.1 download
bootable?

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Tim Frazee
Sent: Monday, December 31, 2012 3:50 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

This was for UCM and Unity Connection. didnt try anything else.

On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com> wrote:
I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.

I'm getting an error if I feed the 9.1 iso to the 9.0 install process that i want
to upgrade during the install process. I've been able to do this many times in the
past with never a problem like this.

Anyone have any ideas?

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

<temp.png>_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From matthnick at gmail.com Fri Jan 4 15:47:08 2013


From: matthnick at gmail.com (Nick Matthews)
Date: Fri, 4 Jan 2013 15:47:08 -0500
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <CACCAghZZzjvXbDRoiwQ6ND5NuqELg4Vcsy12i3w+RDt_ped9kQ@mail.gmail.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
<CAHgd+38U2ZKTRt-w16f2d66dnuuz6HX7pXJqbHTYDXPAufGKsQ@mail.gmail.com>
<CAFC4dsp8XVUYSmiTG5FQAv=DtR7p5YYuyhROhy2S41Or8F+7-Q@mail.gmail.com>
<CACCAghZZzjvXbDRoiwQ6ND5NuqELg4Vcsy12i3w+RDt_ped9kQ@mail.gmail.com>
Message-ID: <CAM-K-Nq5__KvT-dg7w3m1-t+Xr-SicF0S3z0bme1Y7GAPYtVjg@mail.gmail.com>

Apologies for that, thought it was a public PSIRT. Looks like these release
notes are about the same as what I was looking at:
http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?
method=fetchBugDetails&bugId=CSCuc83860

On Fri, Jan 4, 2013 at 2:21 PM, Justin Steinberg <jsteinberg at gmail.com>wrote:

> Nick's link seems like an internal site. I don't see anything on the
> public psirt page.
>
>
>
http://tools.cisco.com/security/center/publicationListing.x#~CiscoSecurityAdvisory
>
>
>
> On Fri, Jan 4, 2013 at 2:11 PM, Ed Leatherman <ealeatherman at gmail.com>wrote:
>
>> I completely missed the video at the top of the IEEE article the first
>> time i read it.. i think my brain saw it as an advertisement and just
>> ignored it.
>>
>> The researchers full presentation is here also:
>> http://www.youtube.com/watch?v=f3zUOZcewtA&feature=youtu.be
>>
>>
>> On Fri, Jan 4, 2013 at 10:02 AM, Scott Voll <svoll.voip at gmail.com> wrote:
>>
>>> Lelio sent this out a week or two ago.
>>> http://m.spectrum.ieee.org/computing/embedded-systems/cisco-ip-phones-
vulnerable Check out the video.
>>>
>>> We are a closed facility, so the attacker would have to either be
>>> inside, or take a phone off the wall in a reception area AND have SSH
>>> access.
>>>
>>> I talked to my SE and he said:
>>> Workaround = Restrict SSH and CLI access to trusted users only.
>>> Administrators may consider leveraging 802.1x device authentication to
>>> prevent unauthorized devices or systems from accessing the voice network.
>>>
>>> Ang accomplished this by first gaining access to the device via SSH and
>>> utilizing TFTP to pull down a malicious binary that is designed to exploit
>>> the insufficient validation issue of the affected System Calls. He ran this
>>> from the user context on the device which performed the exploit. The
>>> caveats of this particular issue are that an attacker would need to have
>>> Authenticated Access either via SSH (Which would need to be enabled, it is
>>> not enabled by default), or local access via the Serial port. The attacker
>>> would also need to be able to point the device at an attacker-controlled
>>> TFTP server to retrieve the payload.
>>>
>>> YMMV
>>>
>>> Scott
>>>
>>>
>>>
>>>
>>>
>>> On Fri, Jan 4, 2013 at 6:35 AM, Robert Kulagowski <rkulagow at gmail.com>wrote:
>>>
>>>> Since no one who knows anything for real is probably going to say
>>>> anything for now, are there any mitigating factors that I can start
>>>> thinking about once management sees the following article?
>>>>
>>>>
>>>> http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-phones-
vulnerable-to-eavesdropping-hack-researchers-say?lite
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>>
>> --
>> Ed Leatherman
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From jason.aarons at dimensiondata.com Fri Jan 4 15:55:47 2013


From: jason.aarons at dimensiondata.com (Jason Aarons (AM))
Date: Fri, 4 Jan 2013 15:55:47 -0500
Subject: [cisco-voip] 7962 Phone Sets Failing Authentication
In-Reply-To: <002401cdeab3$5566d2e0$003478a0$@comcast.net>
References: <002401cdeab3$5566d2e0$003478a0$@comcast.net>
Message-ID:
<4E38DB0A1959B04C8C83EDCF069B53ED0D2C5EFC39@USISPCLEXDB01.na.didata.local>

What load do they have out of the box? Below SCCP 8.5.x you need to upgrade to
8.5.x before you can upgrade to 9.2.1S.

Check the release notes for SCCP 8.5 loads.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Thomas LeMay
Sent: Friday, January 04, 2013 2:40 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7962 Phone Sets Failing Authentication

We have recently received several boxes of new 7962 phone sets from the Cisco
factory. We are running call manager version 7.1.5.34900-7 on MCS servers (Linux).
When we take several of these new 7962 phone sets and plug them into the network
port they will pull a TAPS number successfully but fail the authentication and
software upgrade portion for phone load SCCP4.2.9-2-1S.. Is there a known work
around or middle firmware version we need to load on a TFTP server to update the
phone sets? Thank you in advance.

Tom

itevomcid
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From mail at darioquiroz.com Fri Jan 4 16:08:22 2013


From: mail at darioquiroz.com (Dario Quiroz)
Date: Fri, 4 Jan 2013 18:08:22 -0300
Subject: [cisco-voip] MWI blinking but voicemail service is not activated on
DN
Message-ID: <CAHuYCEEcD6SnJhYEQrDX9+JwR95ib+Hv=xDvgOSnc1OmggYYxw@mail.gmail.com>

HI all! We have a little problem with a CUCM 8.5.1 and Unity connection.
The MWI is blinking in some 7911 phones, but these DN doesn't have the
voice mail service activated.
Someone know why are they blinking and what is the solution for that?
Thanks in advance

--
Atenciosamente,
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From VanMarenNP at ldschurch.org Fri Jan 4 17:10:14 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Fri, 4 Jan 2013 22:10:14 +0000
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <01373FBF-B7E2-4803-89CB-4F47569A181E@cisco.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
<CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
<00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>
<CABzsfHz0Km6uPom_mhLkU9iMiZMawubm2JXAG1+uaJ+_wKACug@mail.gmail.com>
<5992A434-99D1-406D-894A-02F51A195AA7@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E298311081C47@PHANES.helion.local>
<01373FBF-B7E2-4803-89CB-4F47569A181E@cisco.com>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E45C37@W12112.ldschurch.org>

This just causes trouble for rebuilding/ adding new servers to an existing cluster.
Because you have to install the same version that is running on the cluster.

-Nate
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Friday, January 04, 2013 1:26 PM
To: Matthew Loraditch
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

To close the loop on this the restriction was added in 8.6 (when refresh upgrade
came in) because we don't test this upgrade between major versions and the fact
that we started having to do OS reinstalls (refresh) for some combinations made the
likelihood of failure too high.

It is not documented, and that will remedied in the Release Notes for 9.1 shortly
and in the Upgrade/Install docs at some point in the future (they can't be changed
as fast as release notes).

-Ryan

On Jan 2, 2013, at 4:34 PM, Matthew Loraditch <MLoraditch at


heliontechnologies.com<mailto:MLoraditch at heliontechnologies.com>> wrote:

Well that's good, I can just put a PUT order in edelivery and get it. Let's see if
it works.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net<mailto:voip-bounces
at puck.nether.net>] On Behalf Of Ryan Ratliff
Sent: Wednesday, January 02, 2013 4:22 PM
To: Tim Frazee
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

I was told this restriction was added around 8.5 but I'm still waiting on some
other folks to comment.

To get to 9.1 you either do a fresh install or you upgrade, same as any other
version. I understand the release of 9.1 has immediately replaced 9.0 on new 9.x
orders (much like 8.6 did for 8.5) so any 9.x media kit ordered today will be sent
9.1 bootable media.

-Ryan

On Jan 2, 2013, at 4:12 PM, Tim Frazee <tfrazee at gmail.com<mailto:tfrazee at


gmail.com>> wrote:

I could see that.

But tell that to my 8.5 upgrade from an 8.0 boot disk I was able to do back in the
day......

In short, you say that the only way currently to get to 9.1 is upgrade from an
already installed support version, not during the install process.

for the record and I know its not supported, I did try the hack of grabbing the
boot info file from 9.0 and pushing it into the 9.1 iso. The install process failed
post installing everything.

Thanks for digging Ryan.


On Wed, Jan 2, 2013 at 12:21 PM, Ryan Ratliff <rratliff at
cisco.com<mailto:rratliff at cisco.com>> wrote:
Confirmed I see it here in the lab and it looks to be intentional, though I'm still
digging.
Initial word is for a while now upgrade-during-install is only supported to the
same major/minor version.

Anything beyond that requires a separate upgrade after install.

-Ryan

On Jan 2, 2013, at 11:18 AM, Tim Frazee <tfrazee at gmail.com<mailto:tfrazee at


gmail.com>> wrote:

I did just that. after I tried with the pre-release, I used my NFR iso. Same
result.

I only used the pre-release because it was already on my datastore and i was
feeling a bit lazy over vacation. After I attempted the same procedure with 9.0(1)
-37 iso, I received the exact same error.

Ryan, should I be able to boot off of 9.0 and upgrade-during-install with 9.1?
On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com<mailto:rratliff
at cisco.com>> wrote:
9.0.0.99101-22<tel:9.0.0.99101-22> is not a 9.0 ES, it's a pre-release build of
9.1. Throw it in the trash and try with a real 9.0 build (I'm going to start this
now).

-Ryan

On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com<mailto:tfrazee at


gmail.com>> wrote:

I didnt test to see if the 9.1 from CCO is bootable. In the past they havent been.

attached is a screenshoot of the error I received when I tried to feed the 9.1 via
CCO during a booted-from-nfr9.0 media
On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at
ldschurch.org<mailto:VanMarenNP at ldschurch.org>> wrote:
I had lots of problems doing upgrade during installs with 9.0 ESs. The ESs are
usually bootable so I just gave up and installed fresh. Is the 9.1 download
bootable?

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-
bounces at puck.nether.net>] On Behalf Of Tim Frazee
Sent: Monday, December 31, 2012 3:50 PM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

This was for UCM and Unity Connection. didnt try anything else.

On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com<mailto:tfrazee at
gmail.com>> wrote:
I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.
I'm getting an error if I feed the 9.1 iso to the 9.0 install process that i want
to upgrade during the install process. I've been able to do this many times in the
past with never a problem like this.

Anyone have any ideas?

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

<temp.png>_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

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From svoll.voip at gmail.com Fri Jan 4 18:34:33 2013


From: svoll.voip at gmail.com (Scott Voll)
Date: Fri, 4 Jan 2013 15:34:33 -0800
Subject: [cisco-voip] Jabber 9.1 non-domain computer
Message-ID: <CAHgd+3_vys=A8J5_DXzXrtsUe0QJ4fHUb_r27krPSKwDm9SstA@mail.gmail.com>

How do you setup Jabber to see the Domain contacts when it's not a Domain
PC. (example. Home end user PC).

TIA

Scott
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From drucker.mark at gmail.com Fri Jan 4 18:39:00 2013


From: drucker.mark at gmail.com (Mark Drucker)
Date: Fri, 4 Jan 2013 15:39:00 -0800
Subject: [cisco-voip] Jabber 9.1 non-domain computer
In-Reply-To: <CAHgd+3_vys=A8J5_DXzXrtsUe0QJ4fHUb_r27krPSKwDm9SstA@mail.gmail.com>
References: <CAHgd+3_vys=A8J5_DXzXrtsUe0QJ4fHUb_r27krPSKwDm9SstA@mail.gmail.com>
Message-ID: <CABUeUByvWfo-1kBZVfzpO4+e5oDeYy1_0W9Ohx43mr8J+heFfg@mail.gmail.com>

Hi Scott,

You can install the app with the below command string:

msiexec.exe /i CiscoJabberSetup.msi TYPE=WEBEX SSO_ORG_DOMAIN=DOMAIN.com


/quiet

Mark

On Fri, Jan 4, 2013 at 3:34 PM, Scott Voll <svoll.voip at gmail.com> wrote:

> How do you setup Jabber to see the Domain contacts when it's not a Domain
> PC. (example. Home end user PC).
>
> TIA
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

--
Mark Drucker
(925) 321-5791
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From jsteinberg at gmail.com Sat Jan 5 10:31:35 2013


From: jsteinberg at gmail.com (Justin Steinberg)
Date: Sat, 5 Jan 2013 10:31:35 -0500
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <2F143E71016CA34C924BF4C33AEF211056E45C37@W12112.ldschurch.org>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
<CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
<00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>
<CABzsfHz0Km6uPom_mhLkU9iMiZMawubm2JXAG1+uaJ+_wKACug@mail.gmail.com>
<5992A434-99D1-406D-894A-02F51A195AA7@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E298311081C47@PHANES.helion.local>
<01373FBF-B7E2-4803-89CB-4F47569A181E@cisco.com>
<2F143E71016CA34C924BF4C33AEF211056E45C37@W12112.ldschurch.org>
Message-ID: <CACCAghbG-_Md1iWCNLgBS3LrqVzHKU6k0ZZ8TKu1sAbo4s+vOg@mail.gmail.com>

Agreed. Especially when you can download the non-bootable upgrade ISO to
go from 9.0 to 9.1. Then if a subscriber fails, you have to get with TAC
to get a bootable 9.1 (hours), or go through the edelivery (days) all over
again. This is a time consuming process.

On Fri, Jan 4, 2013 at 5:10 PM, Nate VanMaren <VanMarenNP at ldschurch.org>wrote:

> ** **
>
> This just causes trouble for rebuilding/ adding new servers to an existing
> cluster. Because you have to install the same version that is running on
> the cluster.****
>
> ** **
>
> -Nate****
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Ryan Ratliff
> *Sent:* Friday, January 04, 2013 1:26 PM
> *To:* Matthew Loraditch
>
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] upgrade during install from 9.0 to 9.1****
>
> ** **
>
> To close the loop on this the restriction was added in 8.6 (when refresh
> upgrade came in) because we don't test this upgrade between major versions
> and the fact that we started having to do OS reinstalls (refresh) for some
> combinations made the likelihood of failure too high.****
>
> ** **
>
> It is not documented, and that will remedied in the Release Notes for 9.1
> shortly and in the Upgrade/Install docs at some point in the future (they
> can't be changed as fast as release notes).****
>
> ** **
>
> -Ryan ****
>
> ** **
>
> On Jan 2, 2013, at 4:34 PM, Matthew Loraditch <
> MLoraditch at heliontechnologies.com> wrote:****
>
> ** **
>
> Well that?s good, I can just put a PUT order in edelivery and get it.
> Let?s see if it works.****
>
> ****
>
> ****
>
> Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA
>
> 1965 Greenspring Drive
> Timonium, MD 21093
>
> voice. 410.252.8830
> fax. 410.252.9284
>
> Twitter <http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296>
> | Website <http://www.heliontechnologies.com/> | Email Support<support at
heliontechnologies.com?subject=Technical%20Support%20Request>
> ****
>
> ****
>
> ****
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:cisco-
> voip-bounces at puck.nether.net] *On Behalf Of *Ryan Ratliff
> *Sent:* Wednesday, January 02, 2013 4:22 PM
> *To:* Tim Frazee
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] upgrade during install from 9.0 to 9.1****
>
> ****
>
> I was told this restriction was added around 8.5 but I'm still waiting on
> some other folks to comment.****
>
> ****
>
> To get to 9.1 you either do a fresh install or you upgrade, same as any
> other version. I understand the release of 9.1 has immediately replaced
> 9.0 on new 9.x orders (much like 8.6 did for 8.5) so any 9.x media kit
> ordered today will be sent 9.1 bootable media.****
>
> ****
>
> -Ryan****
>
> ****
>
> On Jan 2, 2013, at 4:12 PM, Tim Frazee <tfrazee at gmail.com> wrote:****
>
>
> I could see that.
>
> But tell that to my 8.5 upgrade from an 8.0 boot disk I was able to do
> back in the day......
>
> In short, you say that the only way currently to get to 9.1 is upgrade
> from an already installed support version, not during the install process.
>
>
>
> for the record and I know its not supported, I did try the hack of
> grabbing the boot info file from 9.0 and pushing it into the 9.1 iso. The
> install process failed post installing everything.
>
> Thanks for digging Ryan.
>
>
>
> ****
>
> On Wed, Jan 2, 2013 at 12:21 PM, Ryan Ratliff <rratliff at cisco.com> wrote:*
> ***
>
> Confirmed I see it here in the lab and it looks to be intentional, though
> I'm still digging. ****
>
> Initial word is for a while now upgrade-during-install is only supported
> to the same major/minor version. ****
>
> ****
>
> Anything beyond that requires a separate upgrade after install.****
>
> ****
>
> -Ryan****
>
> ****
>
> On Jan 2, 2013, at 11:18 AM, Tim Frazee <tfrazee at gmail.com> wrote:****
>
>
> I did just that. after I tried with the pre-release, I used my NFR iso.
> Same result.
>
> I only used the pre-release because it was already on my datastore and i
> was feeling a bit lazy over vacation. After I attempted the same procedure
> with 9.0(1) -37 iso, I received the exact same error.
>
> Ryan, should I be able to boot off of 9.0 and upgrade-during-install with
> 9.1?****
>
> On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com> wrote:**
> **
>
> 9.0.0.99101-22 is not a 9.0 ES, it's a pre-release build of 9.1. Throw
> it in the trash and try with a real 9.0 build (I'm going to start this
> now). ****
>
> ****
>
> -Ryan****
>
> ****
>
> On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com> wrote:****
>
>
> I didnt test to see if the 9.1 from CCO is bootable. In the past they
> havent been.
>
> attached is a screenshoot of the error I received when I tried to feed the
> 9.1 via CCO during a booted-from-nfr9.0 media****
>
> On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at ldschurch.org>
> wrote:****
>
> I had lots of problems doing upgrade during installs with 9.0 ESs. The
> ESs are usually bootable so I just gave up and installed fresh. Is the 9.1
> download bootable?****
>
> ****
>
> ****
>
> ****
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Tim Frazee
> *Sent:* Monday, December 31, 2012 3:50 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] upgrade during install from 9.0 to 9.1****
>
> ****
>
> This was for UCM and Unity Connection. didnt try anything else.
>
>
> ****
>
> On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com> wrote:****
>
> I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.
>
> I'm getting an error if I feed the 9.1 iso to the 9.0 install process that
> i want to upgrade during the install process. I've been able to do this
> many times in the past with never a problem like this.
>
> Anyone have any ideas?****
>
> ****
>
>
>
> NOTICE: This email message is for the sole use of the intended
> recipient(s) and may contain confidential and privileged information. Any
> unauthorized review, use, disclosure or distribution is prohibited. If you
> are not the intended recipient, please contact the sender by reply email
> and destroy all copies of the original message.****
>
> ****
>
> ****
>
> <temp.png>_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ****
>
> ****
>
> ****
>
> ****
>
> ** **
>
>
>
> NOTICE: This email message is for the sole use of the intended
> recipient(s) and may contain confidential and privileged information. Any
> unauthorized review, use, disclosure or distribution is prohibited. If you
> are not the intended recipient, please contact the sender by reply email
> and destroy all copies of the original message.****
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From jsteinberg at gmail.com Sat Jan 5 10:34:10 2013


From: jsteinberg at gmail.com (Justin Steinberg)
Date: Sat, 5 Jan 2013 10:34:10 -0500
Subject: [cisco-voip] Jabber 9.1 non-domain computer
In-Reply-To: <CAHgd+3_vys=A8J5_DXzXrtsUe0QJ4fHUb_r27krPSKwDm9SstA@mail.gmail.com>
References: <CAHgd+3_vys=A8J5_DXzXrtsUe0QJ4fHUb_r27krPSKwDm9SstA@mail.gmail.com>
Message-ID: <CACCAghb1qrT4RCWsiub4RSxeFnkgVVd3TAN0zU4c76b8BrRepg@mail.gmail.com>

Two options, depending on what directory method you are using:

1) UDS - this is the easiest, since Jabber queries CUCM enduser table for
directory searches. This must be enabled in the jabber-config.xml file.
UDS is a service on CUCM, available starting in 8.6
2) EDI - this is the default Jabber directory search, and as you mention,
queries AD directly. This is a problem for non domain PCs. You can edit
the jabber-config.xml file and hard code AD domain controller IPs and
credentials for Jabber to authenticate to query AD.

For a 8.6+ cluster that is LDAP integrated to your AD, I would just use the
UDS method. It is easier.

On Fri, Jan 4, 2013 at 6:34 PM, Scott Voll <svoll.voip at gmail.com> wrote:

> How do you setup Jabber to see the Domain contacts when it's not a Domain
> PC. (example. Home end user PC).
>
> TIA
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From member at linkedin.com Sun Jan 6 21:14:21 2013


From: member at linkedin.com (Thilan Jayathilake via LinkedIn)
Date: Mon, 7 Jan 2013 02:14:21 +0000 (UTC)
Subject: [cisco-voip] Invitation to connect on LinkedIn
Message-ID: <598039155.55364253.1357524861267.JavaMail.app@ela4-app2317.prod>

LinkedIn
------------

Thilan Jayathilake requested to add you as a connection on LinkedIn:

------------------------------------------

Ray,

Find out why I use LinkedIn. Stay in touch and build your professional network.

- Thilan

Accept invitation from Thilan Jayathilake


http://www.linkedin.com/e/6mrtb0-hbmz92rq-41/H2MsQd1jS6Z70hLZmjMP5c-
Q5YrI0nyXrVBZemsV/blk/I499238028_9/e39SrCAJoS5vrCAJoyRJtCVFnSRJrScJr6RBfnhv9ClRsDgZ
p6lQs6lzoQ5AomZIpn8_elYUcz0UcP8Vejh9bRhdpiRiqCYTbPsRd3wVcPAUdz4LrCBxbOYWrSlI/eml-
comm_invm-b-in_ac-inv28/?hs=false&tok=1F4qecana6f5A1

View profile of Thilan Jayathilake


http://www.linkedin.com/e/6mrtb0-hbmz92rq-
41/rso/32167167/nPEo/name/1296620_I499238028_9/?hs=false&tok=3a-4oemWe6f5A1
------------------------------------------
You are receiving Invitation emails.

This email was intended for Ray Burkholder.


Learn why this is included: http://www.linkedin.com/e/6mrtb0-hbmz92rq-41/plh/http
%3A%2F%2Fhelp%2Elinkedin%2Ecom%2Fapp%2Fanswers%2Fdetail%2Fa_id%2F4788/-GXI/?
hs=false&tok=3dGD9wmje6f5A1

(c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

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From Zoltan.Kelemen at emerson.com Mon Jan 7 04:22:38 2013


From: Zoltan.Kelemen at emerson.com (Zoltan.Kelemen at emerson.com)
Date: Mon, 7 Jan 2013 09:22:38 +0000
Subject: [cisco-voip] Calling Party Transformation Patterns on CUCM 8.x
In-Reply-To: <CAHgd+3-bMYHATsZMmf3GsyX9LRvPwwfUcUa_FJBPhaVzbUxu3Q@mail.gmail.com>
References: <F8E0CC3253A10C4CB137F12F568DAD061A96F325B9@GBLONZ-PMSGEM02.emrsn.org>
<CAHgd+3-bMYHATsZMmf3GsyX9LRvPwwfUcUa_FJBPhaVzbUxu3Q@mail.gmail.com>
Message-ID: <F8E0CC3253A10C4CB137F12F568DAD061A96F32FEB@GBLONZ-PMSGEM02.emrsn.org>

I could set external phone number mask on the line of course and send that through
the trunk, but that changes what's displayed on the phone.
Also, it may require changes to a large number of phones, whereas this meant much
less change.

Or were you referring to something else?

Cheers,

Zoltan Kelemen
Emerson

From: Scott Voll [mailto:svoll.voip at gmail.com]


Sent: Thursday, January 03, 2013 9:05 PM
To: Kelemen, Zoltan [CORP/RO]
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Calling Party Transformation Patterns on CUCM 8.x

Can you just set it on the line and pass it through to the sip trunk?

Scott

On Thu, Jan 3, 2013 at 4:38 AM, <Zoltan.Kelemen at


emerson.com<mailto:Zoltan.Kelemen at emerson.com>> wrote:
Hi and a Happy New Year!

CUCM 8.5.1 and I'm trying to globalize calling numbers of outgoing calls on a
specific SIP trunk.

My problem is, there are more than one DID ranges, i.e.:
1XXX numbers would have +40 345 671 XXX
2XXX numbers would have +40 341 232 XXX

I want to make sure to set the proper caller ID/calling number on outgoing calls.
(I can do that since it's an internal SIP trunk, so any callerID is ok)

So I've created a partition and a CSS for transformations and added a Calling Party
Transformation Pattern (Call Routing > Transformation > Transformation Pattern >
Calling Party Transformation Pattern), applied it properly to the SIP trunk etc.

For testing I have created a single test pattern, with my own extension: 2356
This matched and applied the transformations I was expecting. I tested it with
changing the transformations, it kept working.

However, when I rewrote the pattern to 2XXX it stopped matching. Basically it seems
that I'm unable to use any non-specific pattern to match the calling party number.
(neither 2!, nor 235X nor anything else that I've tried seems to match)

Any ideas?

Thanks,
Zoltan Kelemen
Emerson
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From ciscovoipuser at gmail.com Mon Jan 7 05:41:41 2013


From: ciscovoipuser at gmail.com (Boon)
Date: Mon, 7 Jan 2013 10:41:41 +0000
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
Message-ID: <CACue4GgUO92Rx9wbQMxXY-0Ft4JaFEwQ6C2ZrR+JNnTXA44U+Q@mail.gmail.com>

Hi,

We are planning to upgrade a customer to CUCM 8.6 on UCS from 6.1(3) on


MCS.

I understand the upgrade process but the customer has thrown us a curve
ball by asking that in order to mitigate downtime on their solution we use
a spare MCS server and rebuild it as a Publisher before upgrading to 6.1(4)
then 8.6 ready for the DRS backup and restore onto UCS.

They have around 50 x license files which have been added incrementally
over the last few years.

My question is whether I'll need to get each license file rehosted to the
spare MCS server's MAC or whether I can simply apply for a rollup license
from TAC when it's eventually restored on the UCS?

Has anybody been through a similar upgrade experience?

Thanks in advance
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From gwenzit at gmail.com Mon Jan 7 06:37:16 2013


From: gwenzit at gmail.com (gwenzit)
Date: Mon, 07 Jan 2013 06:37:16 -0500
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
Message-ID: <4g3c1hov4uecgod1h936by3j.1357558636395@email.android.com>

i have done most of my upgrades this way and used all licenses on the final
version.?

Sent from my Galaxy S?III

-------- Original message --------


From: Boon <ciscovoipuser at gmail.com>
Date:
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
Hi,

We are planning to upgrade a customer to CUCM 8.6 on UCS from 6.1(3) on MCS.
?
I understand the upgrade process but the customer has thrown us a curve ball by
asking that in order to mitigate downtime on their solution we use a spare MCS
server and rebuild it as a Publisher before upgrading to 6.1(4) then 8.6 ready for
the DRS backup and restore onto UCS.
?
They have around 50 x license files which have been added incrementally over the
last few years.
?
My question is whether I'll need to get each license file rehosted?to the spare?MCS
server's MAC?or whether I can simply apply for a rollup license from TAC when it's
eventually restored on the UCS?
?
Has anybody been through a similar upgrade?experience?
?
Thanks in advance
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From abbaseo at gmail.com Mon Jan 7 08:38:30 2013


From: abbaseo at gmail.com (abbas Wali)
Date: Mon, 7 Jan 2013 13:38:30 +0000
Subject: [cisco-voip] cisco phones for visually imparied user
Message-ID: <CAFdHCp7VOz3NL2xFkC-cLYtU+C92jzG5PeE5LHEjO0MQVVvuDA@mail.gmail.com>

Hi,

can we add/modify anything to help the visually impaired users i.e.


increase the font size on cisco 79XX ??

thanks

--
@bbas..
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From Chris_kauffman at eloyalty.com Mon Jan 7 07:47:09 2013


From: Chris_kauffman at eloyalty.com (Kauffman, Christopher)
Date: Mon, 7 Jan 2013 12:47:09 +0000
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
In-Reply-To: <4g3c1hov4uecgod1h936by3j.1357558636395@email.android.com>
References: <4g3c1hov4uecgod1h936by3j.1357558636395@email.android.com>
Message-ID: <929AADA2-4DA1-4789-9BBD-CE80F22B171A@eloyalty.com>

I have done these upgrades both ways. You could open a TAC case for licensing and
in cases like this they have rekeyed all the files into one file for upgrade. This
has the added benefit of streamlining your license files from the point of upgrade
since you should now only need the new license and the upgrade license file.

Chris
Sent from my mobile.

On Jan 7, 2013, at 6:40 AM, "gwenzit" <gwenzit at gmail.com<mailto:gwenzit at


gmail.com>> wrote:

i have done most of my upgrades this way and used all licenses on the final
version.

Sent from my Galaxy S?III

-------- Original message --------


From: Boon <ciscovoipuser at gmail.com<mailto:ciscovoipuser at gmail.com>>
Date:
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question

Hi,

We are planning to upgrade a customer to CUCM 8.6 on UCS from 6.1(3) on MCS.

I understand the upgrade process but the customer has thrown us a curve ball by
asking that in order to mitigate downtime on their solution we use a spare MCS
server and rebuild it as a Publisher before upgrading to 6.1(4) then 8.6 ready for
the DRS backup and restore onto UCS.

They have around 50 x license files which have been added incrementally over the
last few years.

My question is whether I'll need to get each license file rehosted to the spare MCS
server's MAC or whether I can simply apply for a rollup license from TAC when it's
eventually restored on the UCS?

Has anybody been through a similar upgrade experience?

Thanks in advance
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

From rratliff at cisco.com Mon Jan 7 11:21:47 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 7 Jan 2013 11:21:47 -0500
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <CACCAghbG-_Md1iWCNLgBS3LrqVzHKU6k0ZZ8TKu1sAbo4s+vOg@mail.gmail.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
<CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
<00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>
<CABzsfHz0Km6uPom_mhLkU9iMiZMawubm2JXAG1+uaJ+_wKACug@mail.gmail.com>
<5992A434-99D1-406D-894A-02F51A195AA7@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E298311081C47@PHANES.helion.local>
<01373FBF-B7E2-4803-89CB-4F47569A181E@cisco.com>
<2F143E71016CA34C924BF4C33AEF211056E45C37@W12112.ldschurch.org>
<CACCAghbG-_Md1iWCNLgBS3LrqVzHKU6k0ZZ8TKu1sAbo4s+vOg@mail.gmail.com>
Message-ID: <6FD3C6EF-3CE5-4D6B-AE71-287BCDD5251E@cisco.com>

Is there any way you can be entitled to the upgrade and not request media either be
mailed to you or downloaded via e-delivery?

Successful planning for DRS and an upgrade is itself a time consuming process.
Adding an extra line item to that checklist to get the correct media is going to be
much less time consuming than having to troubleshoot a failure in the middle of a
maintenance window that could be caused by any number of component interaction
issues that aren't tested by anybody.

-Ryan

On Jan 5, 2013, at 10:31 AM, Justin Steinberg <jsteinberg at gmail.com> wrote:

Agreed. Especially when you can download the non-bootable upgrade ISO to go from
9.0 to 9.1. Then if a subscriber fails, you have to get with TAC to get a
bootable 9.1 (hours), or go through the edelivery (days) all over again. This is
a time consuming process.

On Fri, Jan 4, 2013 at 5:10 PM, Nate VanMaren <VanMarenNP at ldschurch.org> wrote:

This just causes trouble for rebuilding/ adding new servers to an existing cluster.
Because you have to install the same version that is running on the cluster.

-Nate

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Friday, January 04, 2013 1:26 PM
To: Matthew Loraditch

Cc: cisco-voip at puck.nether.net


Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

To close the loop on this the restriction was added in 8.6 (when refresh upgrade
came in) because we don't test this upgrade between major versions and the fact
that we started having to do OS reinstalls (refresh) for some combinations made the
likelihood of failure too high.

It is not documented, and that will remedied in the Release Notes for 9.1 shortly
and in the Upgrade/Install docs at some point in the future (they can't be changed
as fast as release notes).
-Ryan

On Jan 2, 2013, at 4:34 PM, Matthew Loraditch <MLoraditch at


heliontechnologies.com> wrote:

Well that?s good, I can just put a PUT order in edelivery and get it. Let?s see if
it works.

Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter | Facebook | Website | Email Support

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, January 02, 2013 4:22 PM
To: Tim Frazee
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

I was told this restriction was added around 8.5 but I'm still waiting on some
other folks to comment.

To get to 9.1 you either do a fresh install or you upgrade, same as any other
version. I understand the release of 9.1 has immediately replaced 9.0 on new 9.x
orders (much like 8.6 did for 8.5) so any 9.x media kit ordered today will be sent
9.1 bootable media.

-Ryan

On Jan 2, 2013, at 4:12 PM, Tim Frazee <tfrazee at gmail.com> wrote:


I could see that.

But tell that to my 8.5 upgrade from an 8.0 boot disk I was able to do back in the
day......

In short, you say that the only way currently to get to 9.1 is upgrade from an
already installed support version, not during the install process.

for the record and I know its not supported, I did try the hack of grabbing the
boot info file from 9.0 and pushing it into the 9.1 iso. The install process failed
post installing everything.

Thanks for digging Ryan.

On Wed, Jan 2, 2013 at 12:21 PM, Ryan Ratliff <rratliff at cisco.com> wrote:

Confirmed I see it here in the lab and it looks to be intentional, though I'm still
digging.

Initial word is for a while now upgrade-during-install is only supported to the


same major/minor version.

Anything beyond that requires a separate upgrade after install.

-Ryan

On Jan 2, 2013, at 11:18 AM, Tim Frazee <tfrazee at gmail.com> wrote:

I did just that. after I tried with the pre-release, I used my NFR iso. Same
result.

I only used the pre-release because it was already on my datastore and i was
feeling a bit lazy over vacation. After I attempted the same procedure with 9.0(1)
-37 iso, I received the exact same error.

Ryan, should I be able to boot off of 9.0 and upgrade-during-install with 9.1?

On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com> wrote:

9.0.0.99101-22 is not a 9.0 ES, it's a pre-release build of 9.1. Throw it in the
trash and try with a real 9.0 build (I'm going to start this now).

-Ryan
On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com> wrote:

I didnt test to see if the 9.1 from CCO is bootable. In the past they havent been.

attached is a screenshoot of the error I received when I tried to feed the 9.1 via
CCO during a booted-from-nfr9.0 media

On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at ldschurch.org> wrote:

I had lots of problems doing upgrade during installs with 9.0 ESs. The ESs are
usually bootable so I just gave up and installed fresh. Is the 9.1 download
bootable?

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Tim Frazee
Sent: Monday, December 31, 2012 3:50 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1

This was for UCM and Unity Connection. didnt try anything else.

On Mon, Dec 31, 2012 at 1:04 PM, Tim Frazee <tfrazee at gmail.com> wrote:

I have the 9.0 NFR and the 9.1 upgrade iso pulled from CCO.

I'm getting an error if I feed the 9.1 iso to the 9.0 install process that i want
to upgrade during the install process. I've been able to do this many times in the
past with never a problem like this.

Anyone have any ideas?

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

<temp.png>_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From garrett at skjelstad.org Mon Jan 7 11:48:40 2013


From: garrett at skjelstad.org (Garrett Skjelstad)
Date: Mon, 7 Jan 2013 08:48:40 -0800
Subject: [cisco-voip] upgrade during install from 9.0 to 9.1
In-Reply-To: <6FD3C6EF-3CE5-4D6B-AE71-287BCDD5251E@cisco.com>
References: <CABzsfHzzCaVfCCGnvfC4zMOCGgbGZ0qd7aCVcFwV8MqSEAYQKQ@mail.gmail.com>
<CABzsfHwc83EgNK9VyYLmHUwi7Y+CS529Lc7JAWwB16=x8NYQDQ@mail.gmail.com>
<2F143E71016CA34C924BF4C33AEF211056E3DF45@W12112.ldschurch.org>
<CABzsfHw09OgbA+cqnRu2UhHJ4QaprXt8xm8N7uJ+RFcQB_evbA@mail.gmail.com>
<A458AAAC-818A-4A61-9D41-E2C167C8D8BB@cisco.com>
<CABzsfHz51PpnrFoorNbtNRVPLJ-7UNT9JK+t=dYPZawE7KejpQ@mail.gmail.com>
<00C7D131-EDB3-4404-A37A-02C434617D89@cisco.com>
<CABzsfHz0Km6uPom_mhLkU9iMiZMawubm2JXAG1+uaJ+_wKACug@mail.gmail.com>
<5992A434-99D1-406D-894A-02F51A195AA7@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E298311081C47@PHANES.helion.local>
<01373FBF-B7E2-4803-89CB-4F47569A181E@cisco.com>
<2F143E71016CA34C924BF4C33AEF211056E45C37@W12112.ldschurch.org>
<CACCAghbG-_Md1iWCNLgBS3LrqVzHKU6k0ZZ8TKu1sAbo4s+vOg@mail.gmail.com>
<6FD3C6EF-3CE5-4D6B-AE71-287BCDD5251E@cisco.com>
Message-ID: <96456C98-ED5A-4F51-B888-DE816FB95A19@skjelstad.org>

Back are the days of voice consultants carrying around binders of DVDs for every
single version they could run across... As they are usually called when someone
didn't do DRS planning that well...
Sent from my iPhone 5

On Jan 7, 2013, at 8:21, Ryan Ratliff <rratliff at cisco.com> wrote:

> Is there any way you can be entitled to the upgrade and not request media either
be mailed to you or downloaded via e-delivery?
>
> Successful planning for DRS and an upgrade is itself a time consuming process.
Adding an extra line item to that checklist to get the correct media is going to be
much less time consuming than having to troubleshoot a failure in the middle of a
maintenance window that could be caused by any number of component interaction
issues that aren't tested by anybody.
>
>
> -Ryan
>
> On Jan 5, 2013, at 10:31 AM, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> Agreed. Especially when you can download the non-bootable upgrade ISO to go from
9.0 to 9.1. Then if a subscriber fails, you have to get with TAC to get a
bootable 9.1 (hours), or go through the edelivery (days) all over again. This is
a time consuming process.
>
> On Fri, Jan 4, 2013 at 5:10 PM, Nate VanMaren <VanMarenNP at ldschurch.org>
wrote:
>>
>>
>> This just causes trouble for rebuilding/ adding new servers to an existing
cluster. Because you have to install the same version that is running on the
cluster.
>>
>>
>>
>> -Nate
>>
>> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of Ryan Ratliff
>> Sent: Friday, January 04, 2013 1:26 PM
>> To: Matthew Loraditch
>>
>>
>> Cc: cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1
>>
>>
>>
>> To close the loop on this the restriction was added in 8.6 (when refresh upgrade
came in) because we don't test this upgrade between major versions and the fact
that we started having to do OS reinstalls (refresh) for some combinations made the
likelihood of failure too high.
>>
>>
>>
>> It is not documented, and that will remedied in the Release Notes for 9.1
shortly and in the Upgrade/Install docs at some point in the future (they can't be
changed as fast as release notes).
>>
>>
>>
>> -Ryan
>>
>>
>>
>> On Jan 2, 2013, at 4:34 PM, Matthew Loraditch <MLoraditch at
heliontechnologies.com> wrote:
>>
>>
>>
>> Well that?s good, I can just put a PUT order in edelivery and get it. Let?s see
if it works.
>>
>>
>>
>>
>>
>> Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA
>>
>> 1965 Greenspring Drive
>> Timonium, MD 21093
>>
>> voice. 410.252.8830
>> fax. 410.252.9284
>>
>> Twitter | Facebook | Website | Email Support
>>
>>
>>
>>
>>
>> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of Ryan Ratliff
>> Sent: Wednesday, January 02, 2013 4:22 PM
>> To: Tim Frazee
>> Cc: cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] upgrade during install from 9.0 to 9.1
>>
>>
>>
>> I was told this restriction was added around 8.5 but I'm still waiting on some
other folks to comment.
>>
>>
>>
>> To get to 9.1 you either do a fresh install or you upgrade, same as any other
version. I understand the release of 9.1 has immediately replaced 9.0 on new 9.x
orders (much like 8.6 did for 8.5) so any 9.x media kit ordered today will be sent
9.1 bootable media.
>>
>>
>>
>> -Ryan
>>
>>
>>
>> On Jan 2, 2013, at 4:12 PM, Tim Frazee <tfrazee at gmail.com> wrote:
>>
>>
>> I could see that.
>>
>> But tell that to my 8.5 upgrade from an 8.0 boot disk I was able to do back in
the day......
>>
>> In short, you say that the only way currently to get to 9.1 is upgrade from an
already installed support version, not during the install process.
>>
>>
>>
>> for the record and I know its not supported, I did try the hack of grabbing the
boot info file from 9.0 and pushing it into the 9.1 iso. The install process failed
post installing everything.
>>
>> Thanks for digging Ryan.
>>
>>
>>
>>
>> On Wed, Jan 2, 2013 at 12:21 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>>
>> Confirmed I see it here in the lab and it looks to be intentional, though I'm
still digging.
>>
>> Initial word is for a while now upgrade-during-install is only supported to the
same major/minor version.
>>
>>
>>
>> Anything beyond that requires a separate upgrade after install.
>>
>>
>>
>> -Ryan
>>
>>
>>
>> On Jan 2, 2013, at 11:18 AM, Tim Frazee <tfrazee at gmail.com> wrote:
>>
>>
>> I did just that. after I tried with the pre-release, I used my NFR iso. Same
result.
>>
>> I only used the pre-release because it was already on my datastore and i was
feeling a bit lazy over vacation. After I attempted the same procedure with 9.0(1)
-37 iso, I received the exact same error.
>>
>> Ryan, should I be able to boot off of 9.0 and upgrade-during-install with 9.1?
>>
>> On Wed, Jan 2, 2013 at 9:47 AM, Ryan Ratliff <rratliff at cisco.com> wrote:
>>
>> 9.0.0.99101-22 is not a 9.0 ES, it's a pre-release build of 9.1. Throw it in
the trash and try with a real 9.0 build (I'm going to start this now).
>>
>>
>>
>> -Ryan
>>
>>
>>
>> On Jan 2, 2013, at 9:50 AM, Tim Frazee <tfrazee at gmail.com> wrote:
>>
>>
>> I didnt test to see if the 9.1 from CCO is bootable. In the past they havent
been.
>>
>> attached is a screenshoot of the error I received when I tried to feed the 9.1
via CCO during a booted-from-nfr9.0 media
>>
>> On Mon, Dec 31, 2012 at 5:02 PM, Nate VanMaren <VanMarenNP at ldschurch.org>
wrote:
>>
>> I had lots of problems doing upgrade during installs with 9.0 ESs. The ESs are
usually bootable so I just gave up and installed fresh. Is the 9.1 download
bootable?
>>
>>
>>
>>
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From erickbee at gmail.com Mon Jan 7 14:04:49 2013


From: erickbee at gmail.com (Erick B.)
Date: Mon, 7 Jan 2013 13:04:49 -0600
Subject: [cisco-voip] Check constraint error adding home/mobile phone to
personal address book
Message-ID: <CAHSnBQze_sDFvH+MgXbA2ynhYtsQiqJOC3gkqn6gh1zuKYCSUg@mail.gmail.com>

Anyone seen this before?

Not finding any bug for this at moment.

When a user tries to add a home/mobile to their personal address book via
web page they get the following error. They can do this fine on the phone
itself. This user has over 100 entries in their PAB, any limitation on
that?

Update failed. Check constraint


(informix.cc_personalphonebook_personalfastdialindex) failed

Version is 6.1.2.1000-13 (I know its old and needs upgrading).

Thanks,
Erick
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From me at go0se.com Mon Jan 7 15:07:55 2013


From: me at go0se.com (me at go0se.com)
Date: Mon, 7 Jan 2013 13:07:55 -0700
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9
Message-ID: <26214250391bb95e30f1e312dc82fa80.squirrel@go0se.com>

I've searched and don't seem to find any threads regarding this. I have
ordered my upgrades from 8.6 to 9 from the PUT tool, and downloaded the
respective ISOs for my various products. I attempted to upgrade my
callmanager publisher and I get the following message:

"Name matches a filter which indicates the name does not represent a
signed file. Upgrade requires signed files."

I do not see that I can download a signed file from Cisco. I read on the
cisco support forums someone suggested adding the "sgn" to the iso file
name but this seems unwise. I went ahead and added the sgn and reattempted
to start the upgrade. Callmanager no longer complains and it appears to be
happy with the new file. I'm afraid to proceed with the upgrade.

What have I done wrong?

Any assistance is appreciated.

Thanks,

Goose
http://atc.go0se.com

==================================
Help those less fortunate than you
http://www.hopegivers.org
==================================

From VanMarenNP at ldschurch.org Mon Jan 7 15:11:09 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Mon, 7 Jan 2013 20:11:09 +0000
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9
In-Reply-To: <26214250391bb95e30f1e312dc82fa80.squirrel@go0se.com>
References: <26214250391bb95e30f1e312dc82fa80.squirrel@go0se.com>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E4DA80@W12112.ldschurch.org>

There must have been an accidental file rename in the flow.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of me at go0se.com
Sent: Monday, January 07, 2013 1:08 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9

I've searched and don't seem to find any threads regarding this. I have
ordered my upgrades from 8.6 to 9 from the PUT tool, and downloaded the
respective ISOs for my various products. I attempted to upgrade my
callmanager publisher and I get the following message:

"Name matches a filter which indicates the name does not represent a
signed file. Upgrade requires signed files."
I do not see that I can download a signed file from Cisco. I read on the
cisco support forums someone suggested adding the "sgn" to the iso file
name but this seems unwise. I went ahead and added the sgn and reattempted
to start the upgrade. Callmanager no longer complains and it appears to be
happy with the new file. I'm afraid to proceed with the upgrade.

What have I done wrong?

Any assistance is appreciated.

Thanks,

Goose
http://atc.go0se.com

==================================
Help those less fortunate than you
http://www.hopegivers.org
==================================

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

From mikeeo at msn.com Mon Jan 7 15:24:50 2013


From: mikeeo at msn.com (Mike Olivere)
Date: Mon, 7 Jan 2013 15:24:50 -0500
Subject: [cisco-voip] VG 224 sip example conf
Message-ID: <BLU0-SMTP169D6CF7559A9447B0617E3C5250@phx.gbl>

Does anyone have a example config? I'm thinking its just like a sip router config
correct?

Thanks
Mike

Sent from my iPhone

From rratliff at cisco.com Mon Jan 7 16:02:03 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 7 Jan 2013 16:02:03 -0500
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9
In-Reply-To: <2F143E71016CA34C924BF4C33AEF211056E4DA80@W12112.ldschurch.org>
References: <26214250391bb95e30f1e312dc82fa80.squirrel@go0se.com>
<2F143E71016CA34C924BF4C33AEF211056E4DA80@W12112.ldschurch.org>
Message-ID: <21D4BE5A-8F13-4081-A0D4-B899E4EDEC32@cisco.com>

What browser did you use to download the ISOs? I seem to recall an issue a while
back where one browser (IE I think) would strip file extensions like that. If you
want to be sure make sure the filename you have on your PC matches the link you
downloaded it from and run an md5sum. If that checks out then you are good to go.

The "disk check" option at the beginning of the install also will do an md5
verification using the md5 we build into the ISO itself so you can run that as
well.

-Ryan

On Jan 7, 2013, at 3:11 PM, Nate VanMaren <VanMarenNP at ldschurch.org> wrote:

There must have been an accidental file rename in the flow.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of me at go0se.com
Sent: Monday, January 07, 2013 1:08 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9

I've searched and don't seem to find any threads regarding this. I have
ordered my upgrades from 8.6 to 9 from the PUT tool, and downloaded the
respective ISOs for my various products. I attempted to upgrade my
callmanager publisher and I get the following message:

"Name matches a filter which indicates the name does not represent a
signed file. Upgrade requires signed files."

I do not see that I can download a signed file from Cisco. I read on the
cisco support forums someone suggested adding the "sgn" to the iso file
name but this seems unwise. I went ahead and added the sgn and reattempted
to start the upgrade. Callmanager no longer complains and it appears to be
happy with the new file. I'm afraid to proceed with the upgrade.

What have I done wrong?

Any assistance is appreciated.

Thanks,

Goose
http://atc.go0se.com

==================================
Help those less fortunate than you
http://www.hopegivers.org
==================================

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From me at go0se.com Mon Jan 7 16:18:39 2013


From: me at go0se.com (me at go0se.com)
Date: Mon, 7 Jan 2013 14:18:39 -0700
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9
In-Reply-To: <21D4BE5A-8F13-4081-A0D4-B899E4EDEC32@cisco.com>
References: <26214250391bb95e30f1e312dc82fa80.squirrel@go0se.com>
<2F143E71016CA34C924BF4C33AEF211056E4DA80@W12112.ldschurch.org>
<21D4BE5A-8F13-4081-A0D4-B899E4EDEC32@cisco.com>
Message-ID: <703c47b7d6bdd7546327c2a38a0b3c7d.squirrel@go0se.com>

Oddly enough I used Firefox.

Thanks,

Goose
http://atc.go0se.com

==================================
Help those less fortunate than you
http://www.hopegivers.org
==================================

> What browser did you use to download the ISOs? I seem to recall an issue
> a while back where one browser (IE I think) would strip file extensions
> like that. If you want to be sure make sure the filename you have on
> your PC matches the link you downloaded it from and run an md5sum. If
> that checks out then you are good to go.
>
> The "disk check" option at the beginning of the install also will do an
> md5 verification using the md5 we build into the ISO itself so you can run
> that as well.
>
> -Ryan
>
> On Jan 7, 2013, at 3:11 PM, Nate VanMaren <VanMarenNP at ldschurch.org>
> wrote:
>
> There must have been an accidental file rename in the flow.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of me at go0se.com
> Sent: Monday, January 07, 2013 1:08 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] iso file for upgrading from 8.6 to 9
>
> I've searched and don't seem to find any threads regarding this. I have
> ordered my upgrades from 8.6 to 9 from the PUT tool, and downloaded the
> respective ISOs for my various products. I attempted to upgrade my
> callmanager publisher and I get the following message:
>
> "Name matches a filter which indicates the name does not represent a
> signed file. Upgrade requires signed files."
>
> I do not see that I can download a signed file from Cisco. I read on the
> cisco support forums someone suggested adding the "sgn" to the iso file
> name but this seems unwise. I went ahead and added the sgn and reattempted
> to start the upgrade. Callmanager no longer complains and it appears to be
> happy with the new file. I'm afraid to proceed with the upgrade.
>
> What have I done wrong?
>
> Any assistance is appreciated.
>
> Thanks,
>
> Goose
> http://atc.go0se.com
>
> ==================================
> Help those less fortunate than you
> http://www.hopegivers.org
> ==================================
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisc

From MLoraditch at heliontechnologies.com Mon Jan 7 16:19:49 2013


From: MLoraditch at heliontechnologies.com (Matthew Loraditch)
Date: Mon, 7 Jan 2013 21:19:49 +0000
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9
In-Reply-To: <21D4BE5A-8F13-4081-A0D4-B899E4EDEC32@cisco.com>
References: <26214250391bb95e30f1e312dc82fa80.squirrel@go0se.com>
<2F143E71016CA34C924BF4C33AEF211056E4DA80@W12112.ldschurch.org>
<21D4BE5A-8F13-4081-A0D4-B899E4EDEC32@cisco.com>
Message-ID: <C75AF2AD9308C246AFBDDB994E3E29831108D66D@PHANES.helion.local>

I can't recall 100% but I believe the bootable versions of the ISOs from edelivery
don't have sgn in the file name. I have added it in the past and it worked as you
described.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Monday, January 07, 2013 4:02 PM
To: me at go0se.com
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] iso file for upgrading from 8.6 to 9

What browser did you use to download the ISOs? I seem to recall an issue a while
back where one browser (IE I think) would strip file extensions like that. If you
want to be sure make sure the filename you have on your PC matches the link you
downloaded it from and run an md5sum. If that checks out then you are good to go.

The "disk check" option at the beginning of the install also will do an md5
verification using the md5 we build into the ISO itself so you can run that as
well.

-Ryan

On Jan 7, 2013, at 3:11 PM, Nate VanMaren <VanMarenNP at


ldschurch.org<mailto:VanMarenNP at ldschurch.org>> wrote:

There must have been an accidental file rename in the flow.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at
puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net<mailto:voip-bounces
at puck.nether.net>] On Behalf Of me at go0se.com<mailto:me at go0se.com>
Sent: Monday, January 07, 2013 1:08 PM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9

I've searched and don't seem to find any threads regarding this. I have
ordered my upgrades from 8.6 to 9 from the PUT tool, and downloaded the
respective ISOs for my various products. I attempted to upgrade my
callmanager publisher and I get the following message:

"Name matches a filter which indicates the name does not represent a
signed file. Upgrade requires signed files."

I do not see that I can download a signed file from Cisco. I read on the
cisco support forums someone suggested adding the "sgn" to the iso file
name but this seems unwise. I went ahead and added the sgn and reattempted
to start the upgrade. Callmanager no longer complains and it appears to be
happy with the new file. I'm afraid to proceed with the upgrade.

What have I done wrong?

Any assistance is appreciated.

Thanks,

Goose
http://atc.go0se.com
==================================
Help those less fortunate than you
http://www.hopegivers.org
==================================

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From terry.cheema at gmail.com Mon Jan 7 16:34:09 2013


From: terry.cheema at gmail.com (Terry Cheema)
Date: Tue, 8 Jan 2013 08:34:09 +1100
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
In-Reply-To: <CACue4GgUO92Rx9wbQMxXY-0Ft4JaFEwQ6C2ZrR+JNnTXA44U+Q@mail.gmail.com>
References: <CACue4GgUO92Rx9wbQMxXY-0Ft4JaFEwQ6C2ZrR+JNnTXA44U+Q@mail.gmail.com>
Message-ID: <7D4B06B7-0093-43A7-A801-EC2DFDE2D89B@gmail.com>

Liaise with Cisco licensing team (licensing at cisco.com). They will rehost and
consolidate all license files into one, So during the migration you will need to
upload just one license file.

Once you do a DRS restore it will also restore those old license files as well. I
have been deleting all the old license files and then upload new license file. It
cleans up the new system from obsolete license files.

You can list and delete all the old license files before you upload new from cli by
below command:

File list license *

And to delete:

file delete license * noconfirm

By using wildcard * you dont have to delete files one by one, With no confirm
option you dont have to confirm everytime.

Sent from my iPhone

On 07/01/2013, at 9:41 PM, Boon <ciscovoipuser at gmail.com> wrote:


> Hi,
>
> We are planning to upgrade a customer to CUCM 8.6 on UCS from 6.1(3) on MCS.
>
> I understand the upgrade process but the customer has thrown us a curve ball by
asking that in order to mitigate downtime on their solution we use a spare MCS
server and rebuild it as a Publisher before upgrading to 6.1(4) then 8.6 ready for
the DRS backup and restore onto UCS.
>
> They have around 50 x license files which have been added incrementally over the
last few years.
>
> My question is whether I'll need to get each license file rehosted to the spare
MCS server's MAC or whether I can simply apply for a rollup license from TAC when
it's eventually restored on the UCS?
>
> Has anybody been through a similar upgrade experience?
>
> Thanks in advance
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

-------------- next part --------------


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From rratliff at cisco.com Mon Jan 7 17:00:07 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 7 Jan 2013 17:00:07 -0500
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9
In-Reply-To: <C75AF2AD9308C246AFBDDB994E3E29831108D66D@PHANES.helion.local>
References: <26214250391bb95e30f1e312dc82fa80.squirrel@go0se.com>
<2F143E71016CA34C924BF4C33AEF211056E4DA80@W12112.ldschurch.org>
<21D4BE5A-8F13-4081-A0D4-B899E4EDEC32@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E29831108D66D@PHANES.helion.local>
Message-ID: <7EAF5181-4E1C-4589-8953-2CB4D3967A5A@cisco.com>

Correct, the bootable ISOs don't end in .sgn. The upgrade ISOs are signed files
and will end in .sgn.

I don't think you can just rename the bootable iso with a .sgn and have it work for
an upgrade. The media kit for 9.0 would have to come with a separate upgrade disk
or you'd just have to burn/mount the ISO as a disk for the upgrade to see it.

-Ryan

On Jan 7, 2013, at 4:19 PM, Matthew Loraditch <MLoraditch at


heliontechnologies.com> wrote:

I can?t recall 100% but I believe the bootable versions of the ISOs from edelivery
don?t have sgn in the file name. I have added it in the past and it worked as you
described.

Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA


1965 Greenspring Drive
Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter | Facebook | Website | Email Support

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Monday, January 07, 2013 4:02 PM
To: me at go0se.com
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] iso file for upgrading from 8.6 to 9

What browser did you use to download the ISOs? I seem to recall an issue a while
back where one browser (IE I think) would strip file extensions like that. If you
want to be sure make sure the filename you have on your PC matches the link you
downloaded it from and run an md5sum. If that checks out then you are good to go.

The "disk check" option at the beginning of the install also will do an md5
verification using the md5 we build into the ISO itself so you can run that as
well.

-Ryan

On Jan 7, 2013, at 3:11 PM, Nate VanMaren <VanMarenNP at ldschurch.org> wrote:

There must have been an accidental file rename in the flow.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Ofme at go0se.com
Sent: Monday, January 07, 2013 1:08 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] iso file for upgrading from 8.6 to 9

I've searched and don't seem to find any threads regarding this. I have
ordered my upgrades from 8.6 to 9 from the PUT tool, and downloaded the
respective ISOs for my various products. I attempted to upgrade my
callmanager publisher and I get the following message:

"Name matches a filter which indicates the name does not represent a
signed file. Upgrade requires signed files."

I do not see that I can download a signed file from Cisco. I read on the
cisco support forums someone suggested adding the "sgn" to the iso file
name but this seems unwise. I went ahead and added the sgn and reattempted
to start the upgrade. Callmanager no longer complains and it appears to be
happy with the new file. I'm afraid to proceed with the upgrade.

What have I done wrong?

Any assistance is appreciated.

Thanks,
Goose
http://atc.go0se.com

==================================
Help those less fortunate than you
http://www.hopegivers.org
==================================

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From svoll.voip at gmail.com Mon Jan 7 17:08:18 2013


From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 7 Jan 2013 14:08:18 -0800
Subject: [cisco-voip] Jabber 9.1 non-domain computer
In-Reply-To: <CACCAghb1qrT4RCWsiub4RSxeFnkgVVd3TAN0zU4c76b8BrRepg@mail.gmail.com>
References: <CAHgd+3_vys=A8J5_DXzXrtsUe0QJ4fHUb_r27krPSKwDm9SstA@mail.gmail.com>
<CACCAghb1qrT4RCWsiub4RSxeFnkgVVd3TAN0zU4c76b8BrRepg@mail.gmail.com>
Message-ID: <CAHgd+38ottVhzJ1_zaTN9iRV0GOxRfjn_Di8VfePP+3b+5OBzw@mail.gmail.com>

OK... I setup the jabber-config.xml file and uploaded it to the TFTP server
and restarted the TFTP server. I'm still not able to search for AD users
in Jabber. anyother ideas?

Scott

On Sat, Jan 5, 2013 at 7:34 AM, Justin Steinberg <jsteinberg at gmail.com>wrote:

> jabber-config.xml file


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From rratliff at cisco.com Mon Jan 7 17:16:00 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 7 Jan 2013 17:16:00 -0500
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
In-Reply-To: <7D4B06B7-0093-43A7-A801-EC2DFDE2D89B@gmail.com>
References: <CACue4GgUO92Rx9wbQMxXY-0Ft4JaFEwQ6C2ZrR+JNnTXA44U+Q@mail.gmail.com>
<7D4B06B7-0093-43A7-A801-EC2DFDE2D89B@gmail.com>
Message-ID: <3511834A-0880-41A9-983B-2930C17174F1@cisco.com>

I'd highly recommend getting up to SU2 or wherever you are going to end up on 8.6
(surely you aren't going to stick with 8.6 base...?) before deleting licenses. I
believe there were some bugs specific to early 8.6 in this area.

-Ryan

On Jan 7, 2013, at 4:34 PM, Terry Cheema <terry.cheema at gmail.com> wrote:

Liaise with Cisco licensing team (licensing at cisco.com). They will rehost and
consolidate all license files into one, So during the migration you will need to
upload just one license file.

Once you do a DRS restore it will also restore those old license files as well. I
have been deleting all the old license files and then upload new license file. It
cleans up the new system from obsolete license files.

You can list and delete all the old license files before you upload new from cli by
below command:

File list license *

And to delete:

file delete license * noconfirm

By using wildcard * you dont have to delete files one by one, With no confirm
option you dont have to confirm everytime.

Sent from my iPhone

On 07/01/2013, at 9:41 PM, Boon <ciscovoipuser at gmail.com> wrote:

> Hi,
>
> We are planning to upgrade a customer to CUCM 8.6 on UCS from 6.1(3) on MCS.
>
> I understand the upgrade process but the customer has thrown us a curve ball by
asking that in order to mitigate downtime on their solution we use a spare MCS
server and rebuild it as a Publisher before upgrading to 6.1(4) then 8.6 ready for
the DRS backup and restore onto UCS.
>
> They have around 50 x license files which have been added incrementally over the
last few years.
>
> My question is whether I'll need to get each license file rehosted to the spare
MCS server's MAC or whether I can simply apply for a rollup license from TAC when
it's eventually restored on the UCS?
>
> Has anybody been through a similar upgrade experience?
>
> Thanks in advance
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-------------- next part --------------


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From MLoraditch at heliontechnologies.com Mon Jan 7 17:20:29 2013


From: MLoraditch at heliontechnologies.com (Matthew Loraditch)
Date: Mon, 7 Jan 2013 22:20:29 +0000
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
In-Reply-To: <3511834A-0880-41A9-983B-2930C17174F1@cisco.com>
References: <CACue4GgUO92Rx9wbQMxXY-0Ft4JaFEwQ6C2ZrR+JNnTXA44U+Q@mail.gmail.com>
<7D4B06B7-0093-43A7-A801-EC2DFDE2D89B@gmail.com>
<3511834A-0880-41A9-983B-2930C17174F1@cisco.com>
Message-ID: <C75AF2AD9308C246AFBDDB994E3E29831108DA71@PHANES.helion.local>

I concur I hit these multiple times and TAC root was needed to clean things up. Not
pleasant, nor part of my plans.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Monday, January 07, 2013 5:16 PM
To: Boon
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question

I'd highly recommend getting up to SU2 or wherever you are going to end up on 8.6
(surely you aren't going to stick with 8.6 base...?) before deleting licenses. I
believe there were some bugs specific to early 8.6 in this area.

-Ryan

On Jan 7, 2013, at 4:34 PM, Terry Cheema <terry.cheema at


gmail.com<mailto:terry.cheema at gmail.com>> wrote:
Liaise with Cisco licensing team (licensing at cisco.com<mailto:licensing at
cisco.com>). They will rehost and consolidate all license files into one, So during
the migration you will need to upload just one license file.

Once you do a DRS restore it will also restore those old license files as well. I
have been deleting all the old license files and then upload new license file. It
cleans up the new system from obsolete license files.

You can list and delete all the old license files before you upload new from cli by
below command:

File list license *

And to delete:

file delete license * noconfirm

By using wildcard * you dont have to delete files one by one, With no confirm
option you dont have to confirm everytime.

Sent from my iPhone

On 07/01/2013, at 9:41 PM, Boon <ciscovoipuser at gmail.com<mailto:ciscovoipuser at


gmail.com>> wrote:

Hi,

We are planning to upgrade a customer to CUCM 8.6 on UCS from 6.1(3) on MCS.

I understand the upgrade process but the customer has thrown us a curve ball by
asking that in order to mitigate downtime on their solution we use a spare MCS
server and rebuild it as a Publisher before upgrading to 6.1(4) then 8.6 ready for
the DRS backup and restore onto UCS.

They have around 50 x license files which have been added incrementally over the
last few years.

My question is whether I'll need to get each license file rehosted to the spare MCS
server's MAC or whether I can simply apply for a rollup license from TAC when it's
eventually restored on the UCS?

Has anybody been through a similar upgrade experience?

Thanks in advance
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

-------------- next part --------------


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From terry.cheema at gmail.com Mon Jan 7 18:39:59 2013
From: terry.cheema at gmail.com (Terry Cheema)
Date: Tue, 8 Jan 2013 10:39:59 +1100
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
In-Reply-To: <C75AF2AD9308C246AFBDDB994E3E29831108DA71@PHANES.helion.local>
References: <CACue4GgUO92Rx9wbQMxXY-0Ft4JaFEwQ6C2ZrR+JNnTXA44U+Q@mail.gmail.com>
<7D4B06B7-0093-43A7-A801-EC2DFDE2D89B@gmail.com>
<3511834A-0880-41A9-983B-2930C17174F1@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E29831108DA71@PHANES.helion.local>
Message-ID: <CAPjJVx4JV4mCCN8oQQtk54-E43NsfxDiwxkX74Bbti_suQs6jw@mail.gmail.com>

Thanks Ryan and Matthew. Agree, If there's any bug you can skip
deleting the old files.

Quick one, which specific version is affected by this and does it give
any error when you try to delete the files or causes any other trouble
as well.....

On Tue, Jan 8, 2013 at 9:20 AM, Matthew Loraditch


<MLoraditch at heliontechnologies.com> wrote:
> I concur I hit these multiple times and TAC root was needed to clean things
> up. Not pleasant, nor part of my plans.
>
>
>
>
>
> Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA
>
> 1965 Greenspring Drive
> Timonium, MD 21093
>
> voice. 410.252.8830
> fax. 410.252.9284
>
> Twitter | Facebook | Website | Email Support
>
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
> Sent: Monday, January 07, 2013 5:16 PM
> To: Boon
> Cc: cisco-voip voyp list
> Subject: Re: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
>
>
>
> I'd highly recommend getting up to SU2 or wherever you are going to end up
> on 8.6 (surely you aren't going to stick with 8.6 base...?) before deleting
> licenses. I believe there were some bugs specific to early 8.6 in this
> area.
>
>
>
>
>
> -Ryan
>
>
>
> On Jan 7, 2013, at 4:34 PM, Terry Cheema <terry.cheema at gmail.com> wrote:
>
>
>
> Liaise with Cisco licensing team (licensing at cisco.com). They will rehost and
> consolidate all license files into one, So during the migration you will
> need to upload just one license file.
>
>
>
> Once you do a DRS restore it will also restore those old license files as
> well. I have been deleting all the old license files and then upload new
> license file. It cleans up the new system from obsolete license files.
>
>
>
> You can list and delete all the old license files before you upload new from
> cli by below command:
>
>
>
> File list license *
>
>
>
> And to delete:
>
>
>
> file delete license * noconfirm
>
>
>
> By using wildcard * you dont have to delete files one by one, With no
> confirm option you dont have to confirm everytime.
>
>
> Sent from my iPhone
>
> On 07/01/2013, at 9:41 PM, Boon <ciscovoipuser at gmail.com> wrote:
>
>
> Hi,
>
>
>
> We are planning to upgrade a customer to CUCM 8.6 on UCS from 6.1(3) on MCS.
>
>
>
> I understand the upgrade process but the customer has thrown us a curve ball
> by asking that in order to mitigate downtime on their solution we use a
> spare MCS server and rebuild it as a Publisher before upgrading to 6.1(4)
> then 8.6 ready for the DRS backup and restore onto UCS.
>
>
>
> They have around 50 x license files which have been added incrementally over
> the last few years.
>
>
>
> My question is whether I'll need to get each license file rehosted to the
> spare MCS server's MAC or whether I can simply apply for a rollup license
> from TAC when it's eventually restored on the UCS?
>
>
>
> Has anybody been through a similar upgrade experience?
>
>
>
> Thanks in advance
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From Vinnie at tdnetwork.com Tue Jan 8 01:28:45 2013


From: Vinnie at tdnetwork.com (Vincent)
Date: Mon, 7 Jan 2013 22:28:45 -0800
Subject: [cisco-voip] upgrade PBX to Cisco call manager
Message-ID: <B24430C6A8BC4742AF022BC8C548112A@xw8200>

HI All,

I would like to know more info about upgrading my existing BPX to Cisco Call
Manager or Unified. would someone please guide me how to achieve this goal.
I know how to configure the call manager, but upgrading is a brand new thing
for me. Also, could you please guide me the cost for what i have to pay,
like license, which is yearly for the phone and what other fee. Will Cisco
Call save me money over the PBX system?

Thank you very much for your help and very appreciated.
.........People First..........

Best Regards,

Vincent Dao

From svoll.voip at gmail.com Tue Jan 8 02:00:51 2013


From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 7 Jan 2013 23:00:51 -0800
Subject: [cisco-voip] upgrade PBX to Cisco call manager
In-Reply-To: <B24430C6A8BC4742AF022BC8C548112A@xw8200>
References: <B24430C6A8BC4742AF022BC8C548112A@xw8200>
Message-ID: <CAHgd+38QBz5G2C+k1fg9y_TFkVgzjwumD8Z45mTA3L8ce13gNQ@mail.gmail.com>

Vincent--

your leaving out a lot of info. What version CM do you currently have?
What version are you going to. The doc's should be on cisco.com. If you
have more specific questions, please post.

As for licensing, you might want to contact your Cisco Account team. this
depends on whether your setup as CUWL licensing or alacart.

As for as Cisco saving you money, there are too many questions that would
need to be asked. again talking to your account team they should be able
to help you with RIO and the in's and outs so you can figure that out.

Scott

On Mon, Jan 7, 2013 at 10:28 PM, Vincent <Vinnie at tdnetwork.com> wrote:

> HI All,
>
>
> I would like to know more info about upgrading my existing BPX to Cisco
> Call Manager or Unified. would someone please guide me how to achieve this
> goal. I know how to configure the call manager, but upgrading is a brand
> new thing for me. Also, could you please guide me the cost for what i have
> to pay, like license, which is yearly for the phone and what other fee.
> Will Cisco Call save me money over the PBX system?
>
> Thank you very much for your help and very appreciated.
>
>
> .........People First..........
>
>
> Best Regards,
>
> Vincent Dao
>
> ______________________________**_________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/**mailman/listinfo/cisco-
voip<https://puck.nether.net/mailman/listinfo/cisco-voip>
>
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From Vinnie at tdnetwork.com Tue Jan 8 03:16:17 2013


From: Vinnie at tdnetwork.com (Vincent)
Date: Tue, 8 Jan 2013 00:16:17 -0800
Subject: [cisco-voip] upgrade PBX to Cisco call manager
References: <B24430C6A8BC4742AF022BC8C548112A@xw8200>
<CAHgd+38QBz5G2C+k1fg9y_TFkVgzjwumD8Z45mTA3L8ce13gNQ@mail.gmail.com>
Message-ID: <3433C2DF00DB45EB9ACB406A54BCDBFD@xw8200>

Thanks Scott for a reply,

i have only about 20 to 30 users and right now using a PBX system with voice
mail, Couple DID with lots of extensions. basicly, should i replace the
whole PBX system with a Call Manager, or coexist with PBX. I have to keep
the same numbers though. with CM version, i am flexible on the software
side, could be 7 or 8.

.........People First..........

Best Regards,

Vincent Dao
----- Original Message -----
From: "Scott Voll" <svoll.voip at gmail.com>
To: "Vincent" <Vinnie at tdnetwork.com>
Cc: <cisco-voip at puck.nether.net>
Sent: Monday, January 07, 2013 11:00 PM
Subject: Re: [cisco-voip] upgrade PBX to Cisco call manager

> Vincent--
>
> your leaving out a lot of info. What version CM do you currently have?
> What version are you going to. The doc's should be on cisco.com. If you
> have more specific questions, please post.
>
> As for licensing, you might want to contact your Cisco Account team. this
> depends on whether your setup as CUWL licensing or alacart.
>
> As for as Cisco saving you money, there are too many questions that would
> need to be asked. again talking to your account team they should be able
> to help you with RIO and the in's and outs so you can figure that out.
>
> Scott
>
>
> On Mon, Jan 7, 2013 at 10:28 PM, Vincent <Vinnie at tdnetwork.com> wrote:
>
>> HI All,
>>
>>
>> I would like to know more info about upgrading my existing BPX to Cisco
>> Call Manager or Unified. would someone please guide me how to achieve
>> this
>> goal. I know how to configure the call manager, but upgrading is a brand
>> new thing for me. Also, could you please guide me the cost for what i
>> have
>> to pay, like license, which is yearly for the phone and what other fee.
>> Will Cisco Call save me money over the PBX system?
>>
>> Thank you very much for your help and very appreciated.
>>
>>
>> .........People First..........
>>
>>
>> Best Regards,
>>
>> Vincent Dao
>>
>> ______________________________**_________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/**mailman/listinfo/cisco-
voip<https://puck.nether.net/mailman/listinfo/cisco-voip>
>>
>

From abbaseo at gmail.com Tue Jan 8 05:41:03 2013


From: abbaseo at gmail.com (abbas Wali)
Date: Tue, 8 Jan 2013 10:41:03 +0000
Subject: [cisco-voip] RTMT alerts interpretation
Message-ID: <CAFdHCp4DLPogh1eQBJYP96=+ySwxq5araHCAw+AWnH_O1mAUBw@mail.gmail.com>

Folks,

just opened the morning emails to find weird alerts from RTMT as below -
doesn't make any sense can some one help where and how to interpret alerts
from RTMT.

At Tue Jan 08 07:24:16 GMT 2013 on node 172.30.213.75, the following


SyslogSeverityMatchFound events generated:

SeverityMatch : Alert

MatchedEvent : Jan 8 07:23:20 NCCHQ-CCM-SUB3 local7 1 ccm: 3215524:


NCCHQ-CCM-SUB3.nottinghamcity.gov.uk: Jan 08 2013 07:23:20.374 UTC :
%UC_CALLMANAGER-1-SDLLinkOOS:
%[LocalNodeId=4][LocalApplicationID=100][RemoteIPAddress=172.29.2.10]
[RemoteNodeID=1][RemoteApplicationID=100][LinkID=4:100:1:100][AppID=Cisco
CallManager][ClusterID=StandAloneCluster][NodeID=NCCHQ-CCM-SUB3]: SDL link
to remote application is out of service AppID : Cisco Syslog Agent
ClusterID :

NodeID : NCCHQ-CCM-SUB3

TimeStamp : Tue Jan 08 07:23:20 GMT 2013


SeverityMatch : Alert

MatchedEvent : Jan 8 07:23:55 NCCWG-CCM-SUB4 syslog 1 nbslogpd[6222]: 8


messages were dropped

AppID : Cisco Syslog Agent

ClusterID :

NodeID : NCCWG-CCM-SUB4

TimeStamp : Tue Jan 08 07:23:56 GMT+00:00 2013

At Tue Jan 08 07:24:45 GMT 2013 on node 172.30.213.13, the following


SyslogSeverityMatchFound events generated:

SeverityMatch : Alert

MatchedEvent : Jan 8 07:24:07 NCCHQ-CCM-TFTP local7 1 clm[14942]: 155:


NCCHQ-CCM-TFTP: Jan 08 2013 07:24:07.791 UTC :
%UC_CLUSTERMANAGER-1-CLM_MsgIntChkError:
%[NodeIP=0.0.0.0][AppID=Cisco Cluster
Manager][ClusterID=][NodeID=NCCHQ-CCM-TFTP]: ClusterMgr message integrity
check error.

AppID : Cisco Syslog Agent

ClusterID :

NodeID : NCCHQ-CCM-TFTP

TimeStamp : Tue Jan 08 07:24:07 GMT+00:00 2013

--
@bbas..
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From salamka at gmail.com Tue Jan 8 06:15:10 2013


From: salamka at gmail.com (Abdul Salam)
Date: Tue, 8 Jan 2013 16:45:10 +0530
Subject: [cisco-voip] RTMT alerts interpretation
In-Reply-To: <CAFdHCp4DLPogh1eQBJYP96=+ySwxq5araHCAw+AWnH_O1mAUBw@mail.gmail.com>
References: <CAFdHCp4DLPogh1eQBJYP96=+ySwxq5araHCAw+AWnH_O1mAUBw@mail.gmail.com>
Message-ID: <EF643CE7-125E-4477-AF75-BD98641027DB@gmail.com>

Many are documented in cucm system error message guide

Sent from my iPhone

On 08-Jan-2013, at 4:11 PM, abbas Wali <abbaseo at gmail.com> wrote:

> Folks,
>
> just opened the morning emails to find weird alerts from RTMT as below - doesn't
make any sense can some one help where and how to interpret alerts from RTMT.
>
> At Tue Jan 08 07:24:16 GMT 2013 on node 172.30.213.75, the following
SyslogSeverityMatchFound events generated:
>
> SeverityMatch : Alert
>
> MatchedEvent : Jan 8 07:23:20 NCCHQ-CCM-SUB3 local7 1 ccm: 3215524: NCCHQ-CCM-
SUB3.nottinghamcity.gov.uk: Jan 08 2013 07:23:20.374 UTC : %UC_CALLMANAGER-1-
SDLLinkOOS: %[LocalNodeId=4][LocalApplicationID=100][RemoteIPAddress=172.29.2.10]
[RemoteNodeID=1][RemoteApplicationID=100][LinkID=4:100:1:100][AppID=Cisco
CallManager][ClusterID=StandAloneCluster][NodeID=NCCHQ-CCM-SUB3]: SDL link to
remote application is out of service AppID : Cisco Syslog Agent ClusterID :
>
> NodeID : NCCHQ-CCM-SUB3
>
> TimeStamp : Tue Jan 08 07:23:20 GMT 2013
>
>
> SeverityMatch : Alert
>
> MatchedEvent : Jan 8 07:23:55 NCCWG-CCM-SUB4 syslog 1 nbslogpd[6222]: 8 messages
were dropped
>
> AppID : Cisco Syslog Agent
>
> ClusterID :
>
> NodeID : NCCWG-CCM-SUB4
>
> TimeStamp : Tue Jan 08 07:23:56 GMT+00:00 2013
>
>
>
>
> At Tue Jan 08 07:24:45 GMT 2013 on node 172.30.213.13, the following
SyslogSeverityMatchFound events generated:
>
> SeverityMatch : Alert
>
> MatchedEvent : Jan 8 07:24:07 NCCHQ-CCM-TFTP local7 1 clm[14942]: 155: NCCHQ-
CCM-TFTP: Jan 08 2013 07:24:07.791 UTC : %UC_CLUSTERMANAGER-1-CLM_MsgIntChkError:
%[NodeIP=0.0.0.0][AppID=Cisco Cluster Manager][ClusterID=][NodeID=NCCHQ-CCM-TFTP]:
ClusterMgr message integrity check error.
>
> AppID : Cisco Syslog Agent
>
> ClusterID :
>
> NodeID : NCCHQ-CCM-TFTP
>
> TimeStamp : Tue Jan 08 07:24:07 GMT+00:00 2013
>
>
>
> --
> @bbas..
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From salamka at gmail.com Tue Jan 8 06:37:05 2013


From: salamka at gmail.com (Abdul Salam .)
Date: Tue, 8 Jan 2013 17:07:05 +0530
Subject: [cisco-voip] RTMT alerts interpretation
In-Reply-To: <CAFdHCp4DLPogh1eQBJYP96=+ySwxq5araHCAw+AWnH_O1mAUBw@mail.gmail.com>
References: <CAFdHCp4DLPogh1eQBJYP96=+ySwxq5araHCAw+AWnH_O1mAUBw@mail.gmail.com>
Message-ID: <CAKav0XS9QG3HXhgOhcAby_6wq3wxyREJvZ6HZM-1+e7usCwxyA@mail.gmail.com>

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/err_msgs/8_x/ccmalarms801.html

*---AS*

On Tue, Jan 8, 2013 at 4:11 PM, abbas Wali <abbaseo at gmail.com> wrote:

> Folks,
>
> just opened the morning emails to find weird alerts from RTMT as below -
> doesn't make any sense can some one help where and how to interpret alerts
> from RTMT.
>
> At Tue Jan 08 07:24:16 GMT 2013 on node 172.30.213.75, the following
> SyslogSeverityMatchFound events generated:
>
> SeverityMatch : Alert
>
> MatchedEvent : Jan 8 07:23:20 NCCHQ-CCM-SUB3 local7 1 ccm: 3215524:
> NCCHQ-CCM-SUB3.nottinghamcity.gov.uk: Jan 08 2013 07:23:20.374 UTC :
%UC_CALLMANAGER-1-SDLLinkOOS:
> %[LocalNodeId=4][LocalApplicationID=100][RemoteIPAddress=172.29.2.10]
[RemoteNodeID=1][RemoteApplicationID=100][LinkID=4:100:1:100][AppID=Cisco
> CallManager][ClusterID=StandAloneCluster][NodeID=NCCHQ-CCM-SUB3]: SDL link
> to remote application is out of service AppID : Cisco Syslog Agent
> ClusterID :
>
> NodeID : NCCHQ-CCM-SUB3
>
> TimeStamp : Tue Jan 08 07:23:20 GMT 2013
>
> SeverityMatch : Alert
>
> MatchedEvent : Jan 8 07:23:55 NCCWG-CCM-SUB4 syslog 1 nbslogpd[6222]: 8
> messages were dropped
>
> AppID : Cisco Syslog Agent
>
> ClusterID :
>
> NodeID : NCCWG-CCM-SUB4
>
> TimeStamp : Tue Jan 08 07:23:56 GMT+00:00 2013
>
>
>
> At Tue Jan 08 07:24:45 GMT 2013 on node 172.30.213.13, the following
> SyslogSeverityMatchFound events generated:
>
> SeverityMatch : Alert
>
> MatchedEvent : Jan 8 07:24:07 NCCHQ-CCM-TFTP local7 1 clm[14942]: 155:
> NCCHQ-CCM-TFTP: Jan 08 2013 07:24:07.791 UTC : %UC_CLUSTERMANAGER-1-
CLM_MsgIntChkError:
> %[NodeIP=0.0.0.0][AppID=Cisco Cluster
> Manager][ClusterID=][NodeID=NCCHQ-CCM-TFTP]: ClusterMgr message integrity
> check error.
>
> AppID : Cisco Syslog Agent
>
> ClusterID :
>
> NodeID : NCCHQ-CCM-TFTP
>
> TimeStamp : Tue Jan 08 07:24:07 GMT+00:00 2013
>
>
> --
> @bbas..
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From ckos1976 at hotmail.com Tue Jan 8 07:22:14 2013


From: ckos1976 at hotmail.com (costas georgiou)
Date: Tue, 8 Jan 2013 12:22:14 +0000
Subject: [cisco-voip] ARC
Message-ID: <BLU174-W3211E7CADF9D21932C246FD2240@phx.gbl>

HI All,

Does any one if ARC 5.1.2 is supported with CUCM 8.5.1?

Regards

Cos
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From abbaseo at gmail.com Tue Jan 8 08:41:57 2013
From: abbaseo at gmail.com (abbas Wali)
Date: Tue, 8 Jan 2013 13:41:57 +0000
Subject: [cisco-voip] ARC
In-Reply-To: <BLU174-W3211E7CADF9D21932C246FD2240@phx.gbl>
References: <BLU174-W3211E7CADF9D21932C246FD2240@phx.gbl>
Message-ID: <CAFdHCp4GwvW3k-=HxNBR_gBKuTn5RF0twwk6MG-8-qMXfWYt+w@mail.gmail.com>

yes. we are currently using 8.5.1 with ARC 5.1.2 [connect admin, voice
server and connect ct server]

On 8 January 2013 12:22, costas georgiou <ckos1976 at hotmail.com> wrote:

> HI All,
>
> Does any one if ARC 5.1.2 is supported with CUCM 8.5.1?
>
> Regards
>
> Cos
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

--
@bbas..
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From jamgale at cisco.com Tue Jan 8 09:26:00 2013


From: jamgale at cisco.com (Jamie Gale -X (jamgale - Arc Solutions at Cisco))
Date: Tue, 8 Jan 2013 14:26:00 +0000
Subject: [cisco-voip] ARC
In-Reply-To: <BLU174-W3211E7CADF9D21932C246FD2240@phx.gbl>
References: <BLU174-W3211E7CADF9D21932C246FD2240@phx.gbl>
Message-ID: <DBBB1EC53EEAD849B38428C0342C8E310D94B290@xmb-aln-x13.cisco.com>

Hi Costas,

You can find a compatibility chart at the following link:

http://www.arcsolutions.com/north_america/services/technicaldocumententerprise.aspx

Kind Regards

Jamie Gale
Technical Marketing Engineer, Cisco Unified Attendant Consoles
Arc Solutions, onsite at Cisco
jamgale at cisco.com
D +1 919 392 4671
M +1 919 699 4910
Find our new Cisco Unified Attendant Console End User training videos at
http://www.arcsolutions.com/north_america/solutions/products/cisco_oem_consoles.asp
x or https://www.youtube.com/channel/UC3jC1gmgsRWPR4PLR2VWBhA?feature=CCQQwRs%3D

Join the Cisco Unified Attendant Console Forum at Arc Solutions!


http://forum.arcsolutions.com/forumdisplay.php?f=4

On Jan 8, 2013, at 7:22 AM, costas georgiou <ckos1976 at hotmail.com>


wrote:

HI All,

Does any one if ARC 5.1.2 is supported with CUCM 8.5.1?

Regards

Cos
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From ckos1976 at hotmail.com Tue Jan 8 09:45:15 2013


From: ckos1976 at hotmail.com (costas georgiou)
Date: Tue, 8 Jan 2013 14:45:15 +0000
Subject: [cisco-voip] ARC
In-Reply-To: <DBBB1EC53EEAD849B38428C0342C8E310D94B290@xmb-aln-x13.cisco.com>
References: <BLU174-W3211E7CADF9D21932C246FD2240@phx.gbl>,
<DBBB1EC53EEAD849B38428C0342C8E310D94B290@xmb-aln-x13.cisco.com>
Message-ID: <BLU174-W795749A98DF5659F1284AD2240@phx.gbl>

Thanks All.

From: jamgale at cisco.com


To: ckos1976 at hotmail.com
CC: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ARC
Date: Tue, 8 Jan 2013 14:26:00 +0000

Hi Costas,

You can find a compatibility chart at the following link:

http://www.arcsolutions.com/north_america/services/technicaldocumententerprise.aspx
Kind Regards

Jamie Gale
Technical Marketing Engineer, Cisco Unified Attendant Consoles
Arc Solutions, onsite at Cisco
jamgale at cisco.com
D +1 919 392 4671
M +1 919 699 4910

Find our new Cisco Unified Attendant Console End User training videos at
http://www.arcsolutions.com/north_america/solutions/products/cisco_oem_consoles.asp
x or https://www.youtube.com/channel/UC3jC1gmgsRWPR4PLR2VWBhA?feature=CCQQwRs%3D

Join the Cisco Unified Attendant Console Forum at Arc Solutions!


http://forum.arcsolutions.com/forumdisplay.php?f=4

On Jan 8, 2013, at 7:22 AM, costas georgiou <ckos1976 at hotmail.com>


wrote:

HI All,

Does any one if ARC 5.1.2 is supported with CUCM 8.5.1?

Regards

Cos
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From rratliff at cisco.com Tue Jan 8 10:02:33 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 8 Jan 2013 10:02:33 -0500
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
In-Reply-To: <CAPjJVx4JV4mCCN8oQQtk54-E43NsfxDiwxkX74Bbti_suQs6jw@mail.gmail.com>
References: <CACue4GgUO92Rx9wbQMxXY-0Ft4JaFEwQ6C2ZrR+JNnTXA44U+Q@mail.gmail.com>
<7D4B06B7-0093-43A7-A801-EC2DFDE2D89B@gmail.com>
<3511834A-0880-41A9-983B-2930C17174F1@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E29831108DA71@PHANES.helion.local>
<CAPjJVx4JV4mCCN8oQQtk54-E43NsfxDiwxkX74Bbti_suQs6jw@mail.gmail.com>
Message-ID: <7C4A7BE1-25D6-43D6-8B8B-2653DBF983D2@cisco.com>

IIRC CSCtc59039 is the bug I was thinking of but it affected 8.5 and was fixed in
8.6.2. Basically the licenses were removed from the file system but left in the
database and root access was required to clean them up.

-Ryan
On Jan 7, 2013, at 6:39 PM, Terry Cheema <terry.cheema at gmail.com> wrote:

Thanks Ryan and Matthew. Agree, If there's any bug you can skip
deleting the old files.

Quick one, which specific version is affected by this and does it give
any error when you try to delete the files or causes any other trouble
as well.....

On Tue, Jan 8, 2013 at 9:20 AM, Matthew Loraditch


<MLoraditch at heliontechnologies.com> wrote:
> I concur I hit these multiple times and TAC root was needed to clean things
> up. Not pleasant, nor part of my plans.
>
>
>
>
>
> Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA
>
> 1965 Greenspring Drive
> Timonium, MD 21093
>
> voice. 410.252.8830
> fax. 410.252.9284
>
> Twitter | Facebook | Website | Email Support
>
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
> Sent: Monday, January 07, 2013 5:16 PM
> To: Boon
> Cc: cisco-voip voyp list
> Subject: Re: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
>
>
>
> I'd highly recommend getting up to SU2 or wherever you are going to end up
> on 8.6 (surely you aren't going to stick with 8.6 base...?) before deleting
> licenses. I believe there were some bugs specific to early 8.6 in this
> area.
>
>
>
>
>
> -Ryan
>
>
>
> On Jan 7, 2013, at 4:34 PM, Terry Cheema <terry.cheema at gmail.com> wrote:
>
>
>
> Liaise with Cisco licensing team (licensing at cisco.com). They will rehost and
> consolidate all license files into one, So during the migration you will
> need to upload just one license file.
>
>
>
> Once you do a DRS restore it will also restore those old license files as
> well. I have been deleting all the old license files and then upload new
> license file. It cleans up the new system from obsolete license files.
>
>
>
> You can list and delete all the old license files before you upload new from
> cli by below command:
>
>
>
> File list license *
>
>
>
> And to delete:
>
>
>
> file delete license * noconfirm
>
>
>
> By using wildcard * you dont have to delete files one by one, With no
> confirm option you dont have to confirm everytime.
>
>
> Sent from my iPhone
>
> On 07/01/2013, at 9:41 PM, Boon <ciscovoipuser at gmail.com> wrote:
>
>
> Hi,
>
>
>
> We are planning to upgrade a customer to CUCM 8.6 on UCS from 6.1(3) on MCS.
>
>
>
> I understand the upgrade process but the customer has thrown us a curve ball
> by asking that in order to mitigate downtime on their solution we use a
> spare MCS server and rebuild it as a Publisher before upgrading to 6.1(4)
> then 8.6 ready for the DRS backup and restore onto UCS.
>
>
>
> They have around 50 x license files which have been added incrementally over
> the last few years.
>
>
>
> My question is whether I'll need to get each license file rehosted to the
> spare MCS server's MAC or whether I can simply apply for a rollup license
> from TAC when it's eventually restored on the UCS?
>
>
>
> Has anybody been through a similar upgrade experience?
>
>
>
> Thanks in advance
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

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From voip at mrga.ch Tue Jan 8 10:18:50 2013


From: voip at mrga.ch (Reto Gassmann)
Date: Tue, 8 Jan 2013 16:18:50 +0100
Subject: [cisco-voip] Cisco UCCE Outbound dialer
Message-ID: <CAL4H0Z41b3wbZHL_fLB=Pf1gLGHN68qdK7oBx59TgxSdRCH4vw@mail.gmail.com>

Hello Group

Perhaps someone knows the difference between the two outbound SIP dialer
configurations "SIP Dialer with SIP Proxy" and "SIP Dialer with no SIP
Proxy".
We have a UCCE 8.0 and plan to Upgrade to 9.0 and move to UCS. For this we
have to change our dialer from sccp to SIP. We plan to have two ISR 3945
with dsp and pri Interfaces (E1). We also have a CUBE with a SIP Trunk to
our carrier. However Cube is not supported with outbound dialer.

Tank for feedback


Regards Reto
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From ciscovoipuser at gmail.com Tue Jan 8 10:46:52 2013


From: ciscovoipuser at gmail.com (Boon)
Date: Tue, 8 Jan 2013 15:46:52 +0000
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
In-Reply-To: <7C4A7BE1-25D6-43D6-8B8B-2653DBF983D2@cisco.com>
References: <CACue4GgUO92Rx9wbQMxXY-0Ft4JaFEwQ6C2ZrR+JNnTXA44U+Q@mail.gmail.com>
<7D4B06B7-0093-43A7-A801-EC2DFDE2D89B@gmail.com>
<3511834A-0880-41A9-983B-2930C17174F1@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E29831108DA71@PHANES.helion.local>
<CAPjJVx4JV4mCCN8oQQtk54-E43NsfxDiwxkX74Bbti_suQs6jw@mail.gmail.com>
<7C4A7BE1-25D6-43D6-8B8B-2653DBF983D2@cisco.com>
Message-ID: <CACue4Ghp=HQKXJeS=V2cc2ZM8FU1LPf6Rh-p8iyZuTwG1vWhKw@mail.gmail.com>

Thanks for the info all.

One further question. The customer has multiple branch locations each with
a voice gateway containing IOS Enhanced Media Resources for transcoder,
conf bridge, MTP and also SRST.

The CUCM IOS Compatibility matrix states that the minimum supported IOS is
15.1(4)M1. This is to provide SRST version 8x and SCCP ccm version 8x.

More than half of their gateways do not have enough DRAM and Flash to
support 15.1(4)M1. The requirements is 512/128.

To upgrade all of these gateways it will cost a considerable amount of


money.

Is it crucial that they're upgraded prior to the upgrade? Will the media
resources be affected if left on the current IOS with only 6.1(3) support?

I think I know the answer but would like a 2nd opinion.

Thanks
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From PedersenE at bennettjones.com Tue Jan 8 11:16:45 2013


From: PedersenE at bennettjones.com (Eric Pedersen)
Date: Tue, 8 Jan 2013 16:16:45 +0000
Subject: [cisco-voip] ARC
In-Reply-To: <DBBB1EC53EEAD849B38428C0342C8E310D94B290@xmb-aln-x13.cisco.com>
References: <BLU174-W3211E7CADF9D21932C246FD2240@phx.gbl>
<DBBB1EC53EEAD849B38428C0342C8E310D94B290@xmb-aln-x13.cisco.com>
Message-ID: <C77DDA7FB9437841BD08707193E420330F7CD85C@cv-exsvr2.Legal.bjlocal>

Hi Jamie,
Do you have a target date for CUEAC support with CUCM 9.1?

Thanks,
Eric

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Jamie Gale -X (jamgale - Arc Solutions at Cisco)
Sent: 08 January 2013 7:26 AM
To: costas georgiou
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ARC

Hi Costas,

You can find a compatibility chart at the following link:

http://www.arcsolutions.com/north_america/services/technicaldocumententerprise.aspx

Kind Regards

Jamie Gale
Technical Marketing Engineer, Cisco Unified Attendant Consoles
Arc Solutions, onsite at Cisco
jamgale at cisco.com<mailto:jamgale at cisco.com>
D +1 919 392 4671
M +1 919 699 4910

Find our new Cisco Unified Attendant Console End User training videos at
http://www.arcsolutions.com/north_america/solutions/products/cisco_oem_consoles.asp
x or https://www.youtube.com/channel/UC3jC1gmgsRWPR4PLR2VWBhA?feature=CCQQwRs%3D

Join the Cisco Unified Attendant Console Forum at Arc Solutions!


http://forum.arcsolutions.com/forumdisplay.php?f=4

On Jan 8, 2013, at 7:22 AM, costas georgiou <ckos1976 at


hotmail.com<mailto:ckos1976 at hotmail.com>>
wrote:

HI All,

Does any one if ARC 5.1.2 is supported with CUCM 8.5.1?

Regards

Cos
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

The contents of this message may contain confidential and/or privileged


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the sender and delete all copies. Like other forms of communication,
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From rratliff at cisco.com Tue Jan 8 11:38:36 2013
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 8 Jan 2013 11:38:36 -0500
Subject: [cisco-voip] CUCM6 MCS to 8.6 UCS bridged upgrade Question
In-Reply-To: <CACue4Ghp=HQKXJeS=V2cc2ZM8FU1LPf6Rh-p8iyZuTwG1vWhKw@mail.gmail.com>
References: <CACue4GgUO92Rx9wbQMxXY-0Ft4JaFEwQ6C2ZrR+JNnTXA44U+Q@mail.gmail.com>
<7D4B06B7-0093-43A7-A801-EC2DFDE2D89B@gmail.com>
<3511834A-0880-41A9-983B-2930C17174F1@cisco.com>
<C75AF2AD9308C246AFBDDB994E3E29831108DA71@PHANES.helion.local>
<CAPjJVx4JV4mCCN8oQQtk54-E43NsfxDiwxkX74Bbti_suQs6jw@mail.gmail.com>
<7C4A7BE1-25D6-43D6-8B8B-2653DBF983D2@cisco.com>
<CACue4Ghp=HQKXJeS=V2cc2ZM8FU1LPf6Rh-p8iyZuTwG1vWhKw@mail.gmail.com>
Message-ID: <98FD6109-5D01-4DE0-9D69-5E9E7ECBFCDC@cisco.com>

> Is it crucial that they're upgraded prior to the upgrade? Will the media
resources be affected if left on the current IOS with only 6.1(3) support?

Best-case: IOS configured for 6.x DSPfarm and works with no issues.
2nd Best-case: They don't register at all.
Worst-case: They register, appear to work, but leak calls randomly due to version
incompatibility and/or bugs.

Your customer is going to be in a pinch either way. Hopefully they can re-examine
the need for local DSP resources on all of these gateways with the upgraded UCM and
find that they are no longer required for their call flow OR you/they can do some
testing to confirm that they can register successfully to the newer UCM and run
just fine.

-Ryan

On Jan 8, 2013, at 10:46 AM, Boon <ciscovoipuser at gmail.com> wrote:

Thanks for the info all.

One further question. The customer has multiple branch locations each with a voice
gateway containing IOS Enhanced Media Resources for transcoder, conf bridge, MTP
and also SRST.

The CUCM IOS Compatibility matrix states that the minimum supported IOS is
15.1(4)M1. This is to provide SRST version 8x and SCCP ccm version 8x.

More than half of their gateways do not have enough DRAM and Flash to support
15.1(4)M1. The requirements is 512/128.

To upgrade all of these gateways it will cost a considerable amount of money.

Is it crucial that they're upgraded prior to the upgrade? Will the media resources
be affected if left on the current IOS with only 6.1(3) support?

I think I know the answer but would like a 2nd opinion.

Thanks
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From msaskin at gmail.com Tue Jan 8 12:35:50 2013


From: msaskin at gmail.com (Matthew Saskin)
Date: Tue, 8 Jan 2013 12:35:50 -0500
Subject: [cisco-voip] Cisco UCCE Outbound dialer
In-Reply-To: <CAL4H0Z41b3wbZHL_fLB=Pf1gLGHN68qdK7oBx59TgxSdRCH4vw@mail.gmail.com>
References: <CAL4H0Z41b3wbZHL_fLB=Pf1gLGHN68qdK7oBx59TgxSdRCH4vw@mail.gmail.com>
Message-ID: <CAMSV-mtUH765fn2-ontWrrSoh_3N8LbzdViJ2PCNf4ctXnhcgg@mail.gmail.com>

In short, it comes down to levels of redundancy. With a SIP proxy in


place, you would configure your dialer processes to route calls outbound
via the CUSP, and use routing rules on the CUSP to distribute calls as
appropriate to your outbound gateways. Without CUSP need to send calls
direct from the dialer process to the gateways. The issue is that the
outbound dialer process only allows setting a single IP address to route
calls, so without a CUSP there's no way to set up any redundancy.

Without CUSP, the closest you can get is having the A side dialer point to
a gateway and the B side dialer point to another gateway, which is better
than nothing I suppose.

-matthew

On Tue, Jan 8, 2013 at 10:18 AM, Reto Gassmann <voip at mrga.ch> wrote:

> We have a UCCE 8.0 and plan to Upgrade to 9.0 and move to UCS. For this we
> have to change our dialer from sccp to SIP. We plan to have two ISR 3945
> with dsp and pri Interfaces (E1). We also have a CUBE with a SIP Trunk to
> our carrier. However Cube is not supported with outbound dialer.
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From Zoltan.Kelemen at emerson.com Wed Jan 9 05:45:20 2013


From: Zoltan.Kelemen at emerson.com (Zoltan.Kelemen at emerson.com)
Date: Wed, 9 Jan 2013 10:45:20 +0000
Subject: [cisco-voip] Calling Party Transformation Patterns on CUCM 8.x
Message-ID: <F8E0CC3253A10C4CB137F12F568DAD061A96F8C9D6@GBLONZ-PMSGEM02.emrsn.org>

Hi,

For future reference, if anybody would ever hit this issue:

While I don't completely understand why it behaved as it did, the root of the
problem was, that the CSS for the Transformation Patterns included other partitions
as well besides the partition dedicated for transformations.

Cheers,

Zoltan Kelemen
Emerson

From: Kelemen, Zoltan [CORP/RO]


Sent: Thursday, January 03, 2013 2:39 PM
To: 'cisco-voip at puck.nether.net'
Subject: Calling Party Transformation Patterns on CUCM 8.x

Hi and a Happy New Year!

CUCM 8.5.1 and I'm trying to globalize calling numbers of outgoing calls on a
specific SIP trunk.

My problem is, there are more than one DID ranges, i.e.:
1XXX numbers would have +40 345 671 XXX
2XXX numbers would have +40 341 232 XXX

I want to make sure to set the proper caller ID/calling number on outgoing calls.
(I can do that since it's an internal SIP trunk, so any callerID is ok)

So I've created a partition and a CSS for transformations and added a Calling Party
Transformation Pattern (Call Routing > Transformation > Transformation Pattern >
Calling Party Transformation Pattern), applied it properly to the SIP trunk etc.

For testing I have created a single test pattern, with my own extension: 2356
This matched and applied the transformations I was expecting. I tested it with
changing the transformations, it kept working.

However, when I rewrote the pattern to 2XXX it stopped matching. Basically it seems
that I'm unable to use any non-specific pattern to match the calling party number.
(neither 2!, nor 235X nor anything else that I've tried seems to match)

Any ideas?

Thanks,
Zoltan Kelemen
Emerson

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From Robert.Bellerose at analog.com Wed Jan 9 08:00:39 2013


From: Robert.Bellerose at analog.com (Bellerose, Robert)
Date: Wed, 9 Jan 2013 08:00:39 -0500
Subject: [cisco-voip] cucm rel 8.6
Message-ID: <48531C74FED6D149B9BE27996038EDFA01EFD1888536@NWD2CMBX1.ad.analog.com>

Anyone ever see issues with Phones 7965 not registering using Taps on this release.
Found out you have to apply config then save for the phones to register. TAC is
looking to this..

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From wsisk at cisco.com Wed Jan 9 11:59:10 2013


From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 9 Jan 2013 11:59:10 -0500
Subject: [cisco-voip] Check constraint error adding home/mobile phone to
personal address book
In-Reply-To: <CAHSnBQze_sDFvH+MgXbA2ynhYtsQiqJOC3gkqn6gh1zuKYCSUg@mail.gmail.com>
References: <CAHSnBQze_sDFvH+MgXbA2ynhYtsQiqJOC3gkqn6gh1zuKYCSUg@mail.gmail.com>
Message-ID: <9DD4B67A-0C44-4DCD-8163-43F3FE592C5D@cisco.com>

Cryptic but looks resolved by CSCso80710. Something about constraints on the number
of entries enforced both in PAB java code and the database.

on casual inspection it looks like use fewer entries or upgrade.

I had to update the bug so it will take 24 hours to appear in Bug Toolkit.

/wes

On Jan 7, 2013, at 2:04 PM, Erick B. wrote:

Anyone seen this before?

Not finding any bug for this at moment.

When a user tries to add a home/mobile to their personal address book via web page
they get the following error. They can do this fine on the phone itself. This user
has over 100 entries in their PAB, any limitation on that?

Update failed. Check constraint


(informix.cc_personalphonebook_personalfastdialindex) failed

Version is 6.1.2.1000-13 (I know its old and needs upgrading).

Thanks,
Erick

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From wsisk at cisco.com Wed Jan 9 13:38:21 2013


From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 9 Jan 2013 13:38:21 -0500
Subject: [cisco-voip] cucm rel 8.6
In-Reply-To: <48531C74FED6D149B9BE27996038EDFA01EFD1888536@NWD2CMBX1.ad.analog.com>
References: <48531C74FED6D149B9BE27996038EDFA01EFD1888536@NWD2CMBX1.ad.analog.com>
Message-ID: <E654D60A-A143-4235-8929-9E5C7B55B7EA@cisco.com>

Nothing offhand.

Change notification being completely down/offline/lost during the bulk insert would
cause what you describe. If it was indeed change notification then a one time
restart of ccm service should bring realtime memory in line with configuration in
the database.

/wes

On Jan 9, 2013, at 8:00 AM, Bellerose, Robert wrote:


Anyone ever see issues with Phones 7965 not registering using Taps on this release.
Found out you have to apply config then save for the phones to register. TAC is
looking to this..

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From erickbee at gmail.com Wed Jan 9 13:51:36 2013


From: erickbee at gmail.com (Erick B)
Date: Wed, 9 Jan 2013 12:51:36 -0600
Subject: [cisco-voip] Check constraint error adding home/mobile phone to
personal address book
In-Reply-To: <9DD4B67A-0C44-4DCD-8163-43F3FE592C5D@cisco.com>
References: <CAHSnBQze_sDFvH+MgXbA2ynhYtsQiqJOC3gkqn6gh1zuKYCSUg@mail.gmail.com>
<9DD4B67A-0C44-4DCD-8163-43F3FE592C5D@cisco.com>
Message-ID: <435B974B-63B9-4FD3-91DF-4B705359A9AD@gmail.com>

Thanks Wes.

Yes, I've asked them to reduce number of entries. The user id also has a hyphen in
it. I was able to add entries with another user fine but that user only had a dozen
entries and the problem user has 163. It updates via phone service on phone.

Sent from my iPhone

On Jan 9, 2013, at 10:59 AM, Wes Sisk <wsisk at cisco.com> wrote:

> Cryptic but looks resolved by CSCso80710. Something about constraints on the
number of entries enforced both in PAB java code and the database.
>
> on casual inspection it looks like use fewer entries or upgrade.
>
> I had to update the bug so it will take 24 hours to appear in Bug Toolkit.
>
> /wes
>
> On Jan 7, 2013, at 2:04 PM, Erick B. wrote:
>
> Anyone seen this before?
>
> Not finding any bug for this at moment.
>
> When a user tries to add a home/mobile to their personal address book via web
page they get the following error. They can do this fine on the phone itself. This
user has over 100 entries in their PAB, any limitation on that?
>
> Update failed. Check constraint
(informix.cc_personalphonebook_personalfastdialindex) failed
>
> Version is 6.1.2.1000-13 (I know its old and needs upgrading).
>
> Thanks,
> Erick
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Dennis.Heim at wwt.com Wed Jan 9 13:54:53 2013


From: Dennis.Heim at wwt.com (Heim, Dennis)
Date: Wed, 9 Jan 2013 12:54:53 -0600
Subject: [cisco-voip] Diversion Headers
Message-ID: <0CC57FCAB07CEB4595526952471493D316F23447D6@PRODCMS1.wwt.local>

We have Verizon sip trunks and require unique codes based on the spoke site that is
calling. For example if we dial cisco it should be sent as 777-1-800-553-2447 for
site 1 and 888-1-800-553-2447 for site 2. We were looking at setting the diversion
in cube as Verizon requires. How are other accomplishing this? One thought was to
set the diversion header based on the subnet of the calling device.

Thanks,
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From VanMarenNP at ldschurch.org Wed Jan 9 14:25:41 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Wed, 9 Jan 2013 19:25:41 +0000
Subject: [cisco-voip] Diversion Headers
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F23447D6@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F23447D6@PRODCMS1.wwt.local>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E52C78@W12112.ldschurch.org>

I would assume it would be better to set the header based upon the number being
called, if you were wanting virtual TEHO. That could be done by matching different
dial-peers?

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Heim, Dennis
Sent: Wednesday, January 09, 2013 11:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Diversion Headers

We have Verizon sip trunks and require unique codes based on the spoke site that is
calling. For example if we dial cisco it should be sent as 777-1-800-553-2447 for
site 1 and 888-1-800-553-2447 for site 2. We were looking at setting the diversion
in cube as Verizon requires. How are other accomplishing this? One thought was to
set the diversion header based on the subnet of the calling device.

Thanks,

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

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From afrankel at cisco.com Wed Jan 9 14:49:29 2013


From: afrankel at cisco.com (Adam Frankel)
Date: Wed, 09 Jan 2013 14:49:29 -0500
Subject: [cisco-voip] Cisco phones vulnerable to hack / remote access?
In-Reply-To: <50E72C89.9050400@cisco.com>
References: <CAGe0w3SV+5_nSpB01kTOB_AAta69HGWEYpYyxnVqkFEpnPpRfQ@mail.gmail.com>
<CAHgd+38U2ZKTRt-w16f2d66dnuuz6HX7pXJqbHTYDXPAufGKsQ@mail.gmail.com>
<CAFC4dsp8XVUYSmiTG5FQAv=DtR7p5YYuyhROhy2S41Or8F+7-Q@mail.gmail.com>
<50E72C89.9050400@cisco.com>
Message-ID: <50EDC9C9.8070508@cisco.com>

A public security advisory posted:

http://www.cisco.com/en/US/products/csa/cisco-sa-20130109-uipphone.html

HTH,

Adam

------------------------------------------------------------------------
*From:* Adam Frankel <afrankel at cisco.com>
*Sent:* Fri, Jan 04, 2013 2:24:57 PM
*To:* Cisco VOIP <cisco-voip at puck.nether.net>
*CC:*
*Subject:* Re: [cisco-voip] Cisco phones vulnerable to hack / remote access?

> PSIRT will be including all updated information related to this on the
> defect, CSCuc83860.
>
> Adam
> ------------------------------------------------------------------------
> *From:* Ed Leatherman <ealeatherman at gmail.com>
> *Sent:* Fri, Jan 04, 2013 2:11:24 PM
> *To:* Scott Voll <svoll.voip at gmail.com>
> *CC:* Cisco VOIP <cisco-voip at puck.nether.net>
> *Subject:* Re: [cisco-voip] Cisco phones vulnerable to hack / remote
> access?
>
>> I completely missed the video at the top of the IEEE article the
>> first time i read it.. i think my brain saw it as an advertisement
>> and just ignored it.
>>
>> The researchers full presentation is here also:
>> http://www.youtube.com/watch?v=f3zUOZcewtA&feature=youtu.be
>>
>>
>> On Fri, Jan 4, 2013 at 10:02 AM, Scott Voll <svoll.voip at gmail.com
>> <mailto:svoll.voip at gmail.com>> wrote:
>>
>> Lelio sent this out a week or two ago.
>> http://m.spectrum.ieee.org/computing/embedded-systems/cisco-ip-phones-
vulnerable
>> Check out the video.
>>
>> We are a closed facility, so the attacker would have to either be
>> inside, or take a phone off the wall in a reception area AND have
>> SSH access.
>>
>> I talked to my SE and he said:
>> Workaround = Restrict SSH and CLI access to trusted users only.
>> Administrators may consider leveraging 802.1x device
>> authentication to prevent unauthorized devices or systems from
>> accessing the voice network.
>>
>> Ang accomplished this by first gaining access to the device via
>> SSH and utilizing TFTP to pull down a malicious binary that is
>> designed to exploit the insufficient validation issue of the
>> affected System Calls. He ran this from the user context on the
>> device which performed the exploit. The caveats of this
>> particular issue are that an attacker would need to have
>> Authenticated Access either via SSH (Which would need to be
>> enabled, it is not enabled by default), or local access via the
>> Serial port. The attacker would also need to be able to point the
>> device at an attacker-controlled TFTP server to retrieve the payload.
>>
>> YMMV
>>
>> Scott
>>
>>
>>
>>
>> On Fri, Jan 4, 2013 at 6:35 AM, Robert Kulagowski
>> <rkulagow at gmail.com <mailto:rkulagow at gmail.com>> wrote:
>>
>> Since no one who knows anything for real is probably going to say
>> anything for now, are there any mitigating factors that I can
>> start
>> thinking about once management sees the following article?
>>
>> http://redtape.nbcnews.com/_news/2013/01/04/16328998-popular-office-
phones-vulnerable-to-eavesdropping-hack-researchers-say?lite
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>>
>> --
>> Ed Leatherman
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From mh at markholloway.com Wed Jan 9 14:57:07 2013


From: mh at markholloway.com (Mark Holloway)
Date: Wed, 9 Jan 2013 14:57:07 -0500
Subject: [cisco-voip] Diversion Headers
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F23447D6@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F23447D6@PRODCMS1.wwt.local>
Message-ID: <85BEC899-432E-4E17-8401-0590EFD5745B@markholloway.com>

Do you need the number in the diversion header to be the same as the CUCM calling
party or can it be any valid number that belongs to the SIP trunk (ie. like a Pilot
Number) to authenticate the call? A SIP Profile can easily add a diversion header.

On Jan 9, 2013, at 1:54 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:

> We have Verizon sip trunks and require unique codes based on the spoke site that
is calling. For example if we dial cisco it should be sent as 777-1-800-553-2447
for site 1 and 888-1-800-553-2447 for site 2. We were looking at setting the
diversion in cube as Verizon requires. How are other accomplishing this? One
thought was to set the diversion header based on the subnet of the calling device.
>
> Thanks,
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From tman701 at gmail.com Wed Jan 9 15:02:51 2013


From: tman701 at gmail.com (Joel Perez)
Date: Wed, 9 Jan 2013 15:02:51 -0500
Subject: [cisco-voip] Diversion Headers
In-Reply-To: <2F143E71016CA34C924BF4C33AEF211056E52C78@W12112.ldschurch.org>
References: <0CC57FCAB07CEB4595526952471493D316F23447D6@PRODCMS1.wwt.local>
<2F143E71016CA34C924BF4C33AEF211056E52C78@W12112.ldschurch.org>
Message-ID: <CABdWoUHaEKSw72u4Mqo1YGMLE5xK14+6DSkSQSd=ATMYVPPqcg@mail.gmail.com>
Hi Dennis,

What I have done in the past with Verizon is to use a prefix on the CM RL
and then match that prefix on the dialpeer in the CUBE and use
diversion-header manipulation based on the prefix that matched.
However I have never had to actually add unique codes to outbound dialed
numbers into VZ SIP trunks, i only had to make sure the outbound DID was in
the range that was specified for that spoke site.

Joel P

On Wed, Jan 9, 2013 at 2:25 PM, Nate VanMaren <VanMarenNP at ldschurch.org>wrote:

> I would assume it would be better to set the header based upon the
> number being called, if you were wanting virtual TEHO. That could be done
> by matching different dial-peers?****
>
> ** **
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Heim, Dennis
> *Sent:* Wednesday, January 09, 2013 11:55 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Diversion Headers****
>
> ** **
>
> We have Verizon sip trunks and require unique codes based on the spoke
> site that is calling. For example if we dial cisco it should be sent as
> 777-1-800-553-2447 for site 1 and 888-1-800-553-2447 for site 2. We were
> looking at setting the diversion in cube as Verizon requires. How are other
> accomplishing this? One thought was to set the diversion header based on
> the subnet of the calling device.****
>
> ** **
>
> Thanks,****
>
>
>
> NOTICE: This email message is for the sole use of the intended
> recipient(s) and may contain confidential and privileged information. Any
> unauthorized review, use, disclosure or distribution is prohibited. If you
> are not the intended recipient, please contact the sender by reply email
> and destroy all copies of the original message.****
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From Dennis.Heim at wwt.com Wed Jan 9 15:17:35 2013
From: Dennis.Heim at wwt.com (Heim, Dennis)
Date: Wed, 9 Jan 2013 14:17:35 -0600
Subject: [cisco-voip] Diversion Headers
In-Reply-To: <85BEC899-432E-4E17-8401-0590EFD5745B@markholloway.com>
References: <0CC57FCAB07CEB4595526952471493D316F23447D6@PRODCMS1.wwt.local>
<85BEC899-432E-4E17-8401-0590EFD5745B@markholloway.com>
Message-ID: <0CC57FCAB07CEB4595526952471493D316F247D197@PRODCMS1.wwt.local>

I need to prefix certain digits to the diversion header.

Dennis Heim | Sr. Unified Collaboration Team Lead


World Wide Technology | 314.212.1814 | dennis.heim at wwt.com<mailto:dennis.heim at
wwt.com>
"Creating Impact, Ignition & Scalability"

From: Mark Holloway [mailto:mh at markholloway.com]


Sent: Wednesday, January 09, 2013 2:57 PM
To: Heim, Dennis
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Diversion Headers

Do you need the number in the diversion header to be the same as the CUCM calling
party or can it be any valid number that belongs to the SIP trunk (ie. like a Pilot
Number) to authenticate the call? A SIP Profile can easily add a diversion header.

On Jan 9, 2013, at 1:54 PM, "Heim, Dennis" <Dennis.Heim at


wwt.com<mailto:Dennis.Heim at wwt.com>> wrote:

We have Verizon sip trunks and require unique codes based on the spoke site that is
calling. For example if we dial cisco it should be sent as 777-1-800-553-2447 for
site 1 and 888-1-800-553-2447 for site 2. We were looking at setting the diversion
in cube as Verizon requires. How are other accomplishing this? One thought was to
set the diversion header based on the subnet of the calling device.

Thanks,
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From mh at markholloway.com Wed Jan 9 15:21:45 2013


From: mh at markholloway.com (Mark Holloway)
Date: Wed, 9 Jan 2013 15:21:45 -0500
Subject: [cisco-voip] Diversion Headers
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F247D197@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F23447D6@PRODCMS1.wwt.local>
<85BEC899-432E-4E17-8401-0590EFD5745B@markholloway.com>
<0CC57FCAB07CEB4595526952471493D316F247D197@PRODCMS1.wwt.local>
Message-ID: <954A08FB-12DF-40B9-BA57-56FF2CF9E9B6@markholloway.com>
Could you match the Calling party (CUCM) number to a dial peer? Then you could
assign a SIP Profile to that dial peer to prepend the digits.

On Jan 9, 2013, at 3:17 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:

> I need to prefix certain digits to the diversion header.


>
> Dennis Heim | Sr. Unified Collaboration Team Lead
> World Wide Technology | 314.212.1814 | dennis.heim at wwt.com
> ?Creating Impact, Ignition & Scalability?
>
> From: Mark Holloway [mailto:mh at markholloway.com]
> Sent: Wednesday, January 09, 2013 2:57 PM
> To: Heim, Dennis
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Diversion Headers
>
> Do you need the number in the diversion header to be the same as the CUCM calling
party or can it be any valid number that belongs to the SIP trunk (ie. like a Pilot
Number) to authenticate the call? A SIP Profile can easily add a diversion header.
>
>
> On Jan 9, 2013, at 1:54 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:
>
>
> We have Verizon sip trunks and require unique codes based on the spoke site that
is calling. For example if we dial cisco it should be sent as 777-1-800-553-2447
for site 1 and 888-1-800-553-2447 for site 2. We were looking at setting the
diversion in cube as Verizon requires. How are other accomplishing this? One
thought was to set the diversion header based on the subnet of the calling device.
>
> Thanks,
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

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From svoll.voip at gmail.com Wed Jan 9 15:47:45 2013


From: svoll.voip at gmail.com (Scott Voll)
Date: Wed, 9 Jan 2013 12:47:45 -0800
Subject: [cisco-voip] 7936 and corporate directory
Message-ID: <CAHgd+39TCB7o3vTe_9Sf_HC4DCzy1fm=4m9oBjH1hp+exoLMnA@mail.gmail.com>

All of my 7936 conference phones, when you press the corporate directory,
come up with error condition.

they are running: cmterm_7936.3-3-21-0 and my CM is version 8.6.2.22900-9.

I'm not finding a bug when I search. Maybe my searching is bad.

Anyone know how to fix it? all my 796x and 7970 work fine with the
corporate directory.
TIA

Scott
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From chrward at cisco.com Wed Jan 9 15:56:07 2013


From: chrward at cisco.com (Chris Ward (chrward))
Date: Wed, 9 Jan 2013 20:56:07 +0000
Subject: [cisco-voip] Diversion Headers
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F247D197@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F23447D6@PRODCMS1.wwt.local>
<85BEC899-432E-4E17-8401-0590EFD5745B@markholloway.com>
<0CC57FCAB07CEB4595526952471493D316F247D197@PRODCMS1.wwt.local>
Message-ID: <C3D1FCA271936B48839E081F898E17AA1ED796@xmb-rcd-x13.cisco.com>

In CUCM, this would probably be a good use of a SIP Normalization script. I suspect
CUBE could do it do, but I know less about that.

+Chris
Unity Connection TME

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Heim, Dennis
Sent: Wednesday, January 09, 2013 3:18 PM
To: Mark Holloway
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Diversion Headers

I need to prefix certain digits to the diversion header.

Dennis Heim | Sr. Unified Collaboration Team Lead


World Wide Technology | 314.212.1814 | dennis.heim at wwt.com<mailto:dennis.heim at
wwt.com>
"Creating Impact, Ignition & Scalability"

From: Mark Holloway [mailto:mh at markholloway.com]


Sent: Wednesday, January 09, 2013 2:57 PM
To: Heim, Dennis
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Diversion Headers

Do you need the number in the diversion header to be the same as the CUCM calling
party or can it be any valid number that belongs to the SIP trunk (ie. like a Pilot
Number) to authenticate the call? A SIP Profile can easily add a diversion header.

On Jan 9, 2013, at 1:54 PM, "Heim, Dennis" <Dennis.Heim at


wwt.com<mailto:Dennis.Heim at wwt.com>> wrote:

We have Verizon sip trunks and require unique codes based on the spoke site that is
calling. For example if we dial cisco it should be sent as 777-1-800-553-2447 for
site 1 and 888-1-800-553-2447 for site 2. We were looking at setting the diversion
in cube as Verizon requires. How are other accomplishing this? One thought was to
set the diversion header based on the subnet of the calling device.

Thanks,
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From erickbee at gmail.com Wed Jan 9 18:04:56 2013


From: erickbee at gmail.com (Erick B.)
Date: Wed, 9 Jan 2013 17:04:56 -0600
Subject: [cisco-voip] 7936 and corporate directory
In-Reply-To: <CAHgd+39TCB7o3vTe_9Sf_HC4DCzy1fm=4m9oBjH1hp+exoLMnA@mail.gmail.com>
References: <CAHgd+39TCB7o3vTe_9Sf_HC4DCzy1fm=4m9oBjH1hp+exoLMnA@mail.gmail.com>
Message-ID: <CAHSnBQwDBwWqnidsHP1AmdNbwLRaQwH8qFwYfsGdtwNxF49oow@mail.gmail.com>

What is the error message?

I had similar issue with 7935 (host not found error) and 8.5 but 7935 EOL
and had worked with TAC and these 2 bug id's were provided to us.

CSCtb89353 & CSCtb96601

TAC had told us to get a 7936 and use at least 3-3-21 version firmware.
This was in Jan 2012. I don't believe the client got 7936s yet...

On Wed, Jan 9, 2013 at 2:47 PM, Scott Voll <svoll.voip at gmail.com> wrote:

> All of my 7936 conference phones, when you press the corporate directory,
> come up with error condition.
>
> they are running: cmterm_7936.3-3-21-0 and my CM is version 8.6.2.22900-9.
>
> I'm not finding a bug when I search. Maybe my searching is bad.
>
> Anyone know how to fix it? all my 796x and 7970 work fine with the
> corporate directory.
>
> TIA
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From rratliff at cisco.com Thu Jan 10 10:32:35 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 10 Jan 2013 10:32:35 -0500
Subject: [cisco-voip] 7936 and corporate directory
In-Reply-To: <CAHSnBQwDBwWqnidsHP1AmdNbwLRaQwH8qFwYfsGdtwNxF49oow@mail.gmail.com>
References: <CAHgd+39TCB7o3vTe_9Sf_HC4DCzy1fm=4m9oBjH1hp+exoLMnA@mail.gmail.com>
<CAHSnBQwDBwWqnidsHP1AmdNbwLRaQwH8qFwYfsGdtwNxF49oow@mail.gmail.com>
Message-ID: <AA749FE6-CA31-4E2F-96B7-CF4E82FD9023@cisco.com>

What's a packet capture show the phone trying to do, and what is it configured to
use for the directory URL? Unfortunately that error can cover anything from a
simple DNS resolution problem to misconfigured URLs, to HTTPS cert exchange issues.
You really need to see what the phone is trying to do (or see that it isn't
actually sending anything) to get a better clue.

-Ryan

On Jan 9, 2013, at 6:04 PM, Erick B. <erickbee at gmail.com> wrote:

What is the error message?

I had similar issue with 7935 (host not found error) and 8.5 but 7935 EOL and had
worked with TAC and these 2 bug id's were provided to us.

CSCtb89353 & CSCtb96601

TAC had told us to get a 7936 and use at least 3-3-21 version firmware. This was in
Jan 2012. I don't believe the client got 7936s yet...

On Wed, Jan 9, 2013 at 2:47 PM, Scott Voll <svoll.voip at gmail.com> wrote:
All of my 7936 conference phones, when you press the corporate directory, come up
with error condition.

they are running: cmterm_7936.3-3-21-0 and my CM is version 8.6.2.22900-9.

I'm not finding a bug when I search. Maybe my searching is bad.

Anyone know how to fix it? all my 796x and 7970 work fine with the corporate
directory.

TIA

Scott

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From Dennis.Heim at wwt.com Thu Jan 10 13:26:23 2013


From: Dennis.Heim at wwt.com (Heim, Dennis)
Date: Thu, 10 Jan 2013 12:26:23 -0600
Subject: [cisco-voip] SIP CUBE Caller ID
Message-ID: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>

I am doing a sip integration to a pbx. We are not getting the caller id name/number
when a call is placed form cisco to the pbx via the cube. We do see the caller id
name in the debugs, but not on the phone display.

Received:
SIP/2.0 200 OK
Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51<mailto:CD2B18C9-
5A8011E2-80D5F143-739C4300 at 10.98.254.51>
CSeq: 101 INVITE
From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B
To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp
Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7
Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>
Content-Type: application/sdp
Content-Length: 367
X-Siemens-Call-Type: ST-insecure
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
Date: Thu, 10 Jan 2013 17:20:27 GMT
P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>

What am I missing?
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From kennethwhayes at gmail.com Thu Jan 10 13:32:10 2013


From: kennethwhayes at gmail.com (Kenneth Hayes)
Date: Thu, 10 Jan 2013 13:32:10 -0500
Subject: [cisco-voip] SIP CUBE Caller ID
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
Message-ID: <3410625381745969557@unknownmsgid>

Do you have privacy ID enabled on your voice service voip also on your SIP
trunk to CUBE to UCM under inbound and outbound make sure you have the
header boxes checked.

Sent from my iPad

On Jan 10, 2013, at 1:28 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:

I am doing a sip integration to a pbx. We are not getting the caller id


name/number when a call is placed form cisco to the pbx via the cube. We do
see the caller id name in the debugs, but not on the phone display.

Received:
SIP/2.0 200 OK

Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51

CSeq: 101 INVITE

From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B

To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp

Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7

Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>

Content-Type: application/sdp

Content-Length: 367

X-Siemens-Call-Type: ST-insecure

Accept-Language: en;q=0.0

Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO

Date: Thu, 10 Jan 2013 17:20:27 GMT

P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>

What am I missing?

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From mh at markholloway.com Thu Jan 10 13:50:31 2013


From: mh at markholloway.com (Mark Holloway)
Date: Thu, 10 Jan 2013 13:50:31 -0500
Subject: [cisco-voip] SIP CUBE Caller ID
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
Message-ID: <3B793D75-EFA9-4601-AE9B-3BAC2F0B7628@markholloway.com>

The FROM field should have a name in front of the phone number. (Like the bottom
line of text in this capture). Can you send the SIP INVITE from CUCM > CUBE, and
from CUBE to the PBX?

On Jan 10, 2013, at 1:26 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:
> I am doing a sip integration to a pbx. We are not getting the caller id
name/number when a call is placed form cisco to the pbx via the cube. We do see the
caller id name in the debugs, but not on the phone display.
>
> Received:
> SIP/2.0 200 OK
> Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51
> CSeq: 101 INVITE
> From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B
> To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp
> Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7
> Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>
> Content-Type: application/sdp
> Content-Length: 367
> X-Siemens-Call-Type: ST-insecure
> Accept-Language: en;q=0.0
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
> Date: Thu, 10 Jan 2013 17:20:27 GMT
> P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>
>
> What am I missing?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From Dennis.Heim at wwt.com Thu Jan 10 14:55:08 2013


From: Dennis.Heim at wwt.com (Heim, Dennis)
Date: Thu, 10 Jan 2013 13:55:08 -0600
Subject: [cisco-voip] SIP CUBE Caller ID
In-Reply-To: <3B793D75-EFA9-4601-AE9B-3BAC2F0B7628@markholloway.com>
References: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
<3B793D75-EFA9-4601-AE9B-3BAC2F0B7628@markholloway.com>
Message-ID: <0CC57FCAB07CEB4595526952471493D316F247D358@PRODCMS1.wwt.local>

It is showing in the PAI, but not the from field.

Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51


CSeq: 101 INVITE
Timestamp: 1357847526
Content-Length: 0

Jan 10 19:52:06.991: //184752/427582000000/SIP/Info/sipSPICheckResponseExt: INVITE


response with no RSEQ - disable IS_REL1XX
Jan 10 19:52:06.991: //184752/427582000000/SIP/State/sipSPIChangeState: 0x2472CB8 :
State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING,
SUBSTATE_PROCEEDING_PROCEEDING)
Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg
enqueued for SPI with IP addr: [10.98.88.38]:5060, local_address:[10.98.254.51]
Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
ccsip_spi_get_msg_type returned: 2 for event 1
Jan 10 19:52:07.119: //-
1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
Checking Invite Dialog
Jan 10 19:52:07.119: //184752/427582000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51
CSeq: 101 INVITE
From: sip:XXXX7800 at 10.210.61.43;tag=802E4C74-179B
To: <sip:18077005 at 10.98.88.38>;tag=SEC11-a58620a-1e58d20a-1-zvE1E7A74E0q
Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK5C1C8C
Contact: <sip:18077005 at 10.98.88.38:5060;maddr=10.98.88.38>
Date: Thu, 10 Jan 2013 19:52:29 GMT
P-Asserted-Identity: "My Name" <sip:8077005 at 10.98.88.38>
Content-Length: 0

From: Mark Holloway [mailto:mh at markholloway.com]


Sent: Thursday, January 10, 2013 1:51 PM
To: Heim, Dennis
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP CUBE Caller ID

The FROM field should have a name in front of the phone number. (Like the bottom
line of text in this capture). Can you send the SIP INVITE from CUCM > CUBE, and
from CUBE to the PBX?

[cid:image001.png at 01CDEF42.7BBEBC30]

On Jan 10, 2013, at 1:26 PM, "Heim, Dennis" <Dennis.Heim at


wwt.com<mailto:Dennis.Heim at wwt.com>> wrote:

I am doing a sip integration to a pbx. We are not getting the caller id name/number
when a call is placed form cisco to the pbx via the cube. We do see the caller id
name in the debugs, but not on the phone display.

Received:
SIP/2.0 200 OK
Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51<mailto:CD2B18C9-
5A8011E2-80D5F143-739C4300 at 10.98.254.51>
CSeq: 101 INVITE
From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B
To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp
Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7
Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>
Content-Type: application/sdp
Content-Length: 367
X-Siemens-Call-Type: ST-insecure
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
Date: Thu, 10 Jan 2013 17:20:27 GMT
P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>

What am I missing?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From Dennis.Heim at wwt.com Thu Jan 10 15:56:41 2013


From: Dennis.Heim at wwt.com (Heim, Dennis)
Date: Thu, 10 Jan 2013 14:56:41 -0600
Subject: [cisco-voip] SIP CUBE Caller ID
References: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
<3B793D75-EFA9-4601-AE9B-3BAC2F0B7628@markholloway.com>
<0CC57FCAB07CEB4595526952471493D316F247D358@PRODCMS1.wwt.local>
<CAKBLUbC=90Qjg90c6PHeE8+6YmnaK1mZz79sS2sUP5+AoPS06w@mail.gmail.com>
Message-ID: <0CC57FCAB07CEB4595526952471493D316F247D382@PRODCMS1.wwt.local>

This is outbound from CUCM. I dial the number it connects and the display just
shows the dialed number, without the name. I did not have a display name on my
phone, when I fixed that, the caller id name is visible to the siemens pbx.

The problem still is that I dial a number and it does not put in the name

The Cisco-Siemens direction, we see Cisco set the From with the name (From: "Cisco
Phone" sip:5987800 at ip. We also see the P-Asserted-Identity set to "Cisco Phone"
sip:5987800 at ip.

On the invite coming back from the siemens we see it set the From, only contain the
number. However, the P-Asserted-Identity, contains the caller id name.

From: Heim, Dennis


Sent: Thursday, January 10, 2013 3:46 PM
To: 'Kenneth Hayes'; Voice_VT
Subject: RE: [cisco-voip] SIP CUBE Caller ID

This is outbound from CUCM. I dial the number it connects and the display just
shows the dialed number, with the name. I did not have a display name on my phone,
when I fixed that, the caller id name is visible to the siemens pbx.

The problem still is that I dial a number and it does not put in the name.

From: Kenneth Hayes [mailto:kennethwhayes at gmail.com]<mailto:


[mailto:kennethwhayes at gmail.com]>
Sent: Thursday, January 10, 2013 3:29 PM
To: Heim, Dennis
Subject: Re: [cisco-voip] SIP CUBE Caller ID

This is inbound correct?


On Thu, Jan 10, 2013 at 2:55 PM, Heim, Dennis <Dennis.Heim at
wwt.com<mailto:Dennis.Heim at wwt.com>> wrote:
It is showing in the PAI, but not the from field.

Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51<mailto:C2451EA-


5A9611E2-92D3F143-739C4300 at 10.98.254.51>
CSeq: 101 INVITE
Timestamp: 1357847526
Content-Length: 0

Jan 10 19:52:06.991: //184752/427582000000/SIP/Info/sipSPICheckResponseExt: INVITE


response with no RSEQ - disable IS_REL1XX
Jan 10 19:52:06.991: //184752/427582000000/SIP/State/sipSPIChangeState: 0x2472CB8 :
State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING,
SUBSTATE_PROCEEDING_PROCEEDING)
Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg
enqueued for SPI with IP addr: [10.98.88.38]:5060, local_address:[10.98.254.51]
Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
ccsip_spi_get_msg_type returned: 2 for event 1
Jan 10 19:52:07.119: //-
1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
Checking Invite Dialog
Jan 10 19:52:07.119: //184752/427582000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51<mailto:C2451EA-
5A9611E2-92D3F143-739C4300 at 10.98.254.51>
CSeq: 101 INVITE
From: sip:XXXX7800 at 10.210.61.43<mailto:sip%3AXXXX7800 at
10.210.61.43>;tag=802E4C74-179B
To: <sip:18077005 at 10.98.88.38<mailto:sip%3A18077005 at 10.98.88.38>>;tag=SEC11-
a58620a-1e58d20a-1-zvE1E7A74E0q
Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK5C1C8C
Contact: <sip:18077005 at 10.98.88.38:5060;maddr=10.98.88.38>
Date: Thu, 10 Jan 2013 19:52:29 GMT
P-Asserted-Identity: "My Name" <sip:8077005 at 10.98.88.38<mailto:sip%3A8077005 at
10.98.88.38>>
Content-Length: 0

From: Mark Holloway [mailto:mh at markholloway.com<mailto:mh at markholloway.com>]


Sent: Thursday, January 10, 2013 1:51 PM
To: Heim, Dennis
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] SIP CUBE Caller ID

The FROM field should have a name in front of the phone number. (Like the bottom
line of text in this capture). Can you send the SIP INVITE from CUCM > CUBE, and
from CUBE to the PBX?

[cid:image001.png at 01CDEF4A.AC6DD570]
On Jan 10, 2013, at 1:26 PM, "Heim, Dennis" <Dennis.Heim at
wwt.com<mailto:Dennis.Heim at wwt.com>> wrote:

I am doing a sip integration to a pbx. We are not getting the caller id name/number
when a call is placed form cisco to the pbx via the cube. We do see the caller id
name in the debugs, but not on the phone display.

Received:
SIP/2.0 200 OK
Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51<mailto:CD2B18C9-
5A8011E2-80D5F143-739C4300 at 10.98.254.51>
CSeq: 101 INVITE
From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B
To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp
Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7
Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>
Content-Type: application/sdp
Content-Length: 367
X-Siemens-Call-Type: ST-insecure
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
Date: Thu, 10 Jan 2013 17:20:27 GMT
P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>

What am I missing?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
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From ck-lists at cksoft.de Thu Jan 10 15:38:30 2013


From: ck-lists at cksoft.de (Christian Kratzer)
Date: Thu, 10 Jan 2013 21:38:30 +0100 (CET)
Subject: [cisco-voip] SIP CUBE Caller ID
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
Message-ID: <alpine.BSF.2.00.1301102133580.29184@pohjola.cksoft.de>

Hi,
On Thu, 10 Jan 2013, Heim, Dennis wrote:

> I am doing a sip integration to a pbx. We are not getting the caller id
name/number when a call is placed form cisco to the pbx via the cube. We do see the
caller id name in the debugs, but not on the phone display.

try

clid strip name

in the incoming sip dial-peer.

If sip gives the cube a usable although empty name for the incoming call
then your phones won't bother to lookup the number.

If you strip the name the phones will look it up again.

Greetings
Christian

>
> Received:
> SIP/2.0 200 OK
> Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51<mailto:CD2B18C9-
5A8011E2-80D5F143-739C4300 at 10.98.254.51>
> CSeq: 101 INVITE
> From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B
> To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp
> Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7
> Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>
> Content-Type: application/sdp
> Content-Length: 367
> X-Siemens-Call-Type: ST-insecure
> Accept-Language: en;q=0.0
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
> Date: Thu, 10 Jan 2013 17:20:27 GMT
> P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>
>
> What am I missing?
>

--
Christian Kratzer CK Software GmbH
Email: ck at cksoft.de Wildberger Weg 24/2
Phone: +49 7032 893 997 - 0 D-71126 Gaeufelden
Fax: +49 7032 893 997 - 9 HRB 245288, Amtsgericht Stuttgart
Web: http://www.cksoft.de/ Geschaeftsfuehrer: Christian Kratzer

From mh at markholloway.com Thu Jan 10 16:23:12 2013


From: mh at markholloway.com (Mark Holloway)
Date: Thu, 10 Jan 2013 16:23:12 -0500
Subject: [cisco-voip] SIP CUBE Caller ID
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F247D382@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
<3B793D75-EFA9-4601-AE9B-3BAC2F0B7628@markholloway.com>
<0CC57FCAB07CEB4595526952471493D316F247D358@PRODCMS1.wwt.local>
<CAKBLUbC=90Qjg90c6PHeE8+6YmnaK1mZz79sS2sUP5+AoPS06w@mail.gmail.com>
<0CC57FCAB07CEB4595526952471493D316F247D382@PRODCMS1.wwt.local>
Message-ID: <F3544AB7-071B-4494-B057-69A7796171CB@markholloway.com>

What I was looking for, but didn't see in your TXT file, is the SIP INVITE from
CUCM includes the calling party name in the FROM field to CUBE, and CUBE includes
the same FROM name to Siemens. Assuming that is true, Siemens appears to be
disregarding the name portion of the FROM field by not only failing to display it,
but completely removing it in 1XX responses. It would be nice if it processed PAI
but I think the name is being stripped and then Siemens just assumes there is no
name for the call, as if PAI didn't exist.

On Jan 10, 2013, at 3:56 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:

> This is outbound from CUCM. I dial the number it connects and the display just
shows the dialed number, without the name. I did not have a display name on my
phone, when I fixed that, the caller id name is visible to the siemens pbx.
>
> The problem still is that I dial a number and it does not put in the name
>
> The Cisco-Siemens direction, we see Cisco set the From with the name (From: ?
Cisco Phone? sip:5987800 at ip. We also see the P-Asserted-Identity set to ?Cisco
Phone? sip:5987800 at ip.
>
> On the invite coming back from the siemens we see it set the From, only contain
the number. However, the P-Asserted-Identity, contains the caller id name.
>
> From: Heim, Dennis
> Sent: Thursday, January 10, 2013 3:46 PM
> To: 'Kenneth Hayes'; Voice_VT
> Subject: RE: [cisco-voip] SIP CUBE Caller ID
>
> This is outbound from CUCM. I dial the number it connects and the display just
shows the dialed number, with the name. I did not have a display name on my phone,
when I fixed that, the caller id name is visible to the siemens pbx.
>
> The problem still is that I dial a number and it does not put in the name.
>
> From: Kenneth Hayes [mailto:kennethwhayes at gmail.com]
> Sent: Thursday, January 10, 2013 3:29 PM
> To: Heim, Dennis
> Subject: Re: [cisco-voip] SIP CUBE Caller ID
>
> This is inbound correct?
>
> On Thu, Jan 10, 2013 at 2:55 PM, Heim, Dennis <Dennis.Heim at wwt.com> wrote:
> It is showing in the PAI, but not the from field.
>
> Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51
> CSeq: 101 INVITE
> Timestamp: 1357847526
> Content-Length: 0
>
>
> Jan 10 19:52:06.991: //184752/427582000000/SIP/Info/sipSPICheckResponseExt:
INVITE response with no RSEQ - disable IS_REL1XX
> Jan 10 19:52:06.991: //184752/427582000000/SIP/State/sipSPIChangeState: 0x2472CB8
: State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING,
SUBSTATE_PROCEEDING_PROCEEDING)
> Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg
enqueued for SPI with IP addr: [10.98.88.38]:5060, local_address:[10.98.254.51]
> Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
ccsip_spi_get_msg_type returned: 2 for event 1
> Jan 10 19:52:07.119: //-
1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
> Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
Checking Invite Dialog
> Jan 10 19:52:07.119: //184752/427582000000/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 180 Ringing
> Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51
> CSeq: 101 INVITE
> From: sip:XXXX7800 at 10.210.61.43;tag=802E4C74-179B
> To: <sip:18077005 at 10.98.88.38>;tag=SEC11-a58620a-1e58d20a-1-zvE1E7A74E0q
> Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK5C1C8C
> Contact: <sip:18077005 at 10.98.88.38:5060;maddr=10.98.88.38>
> Date: Thu, 10 Jan 2013 19:52:29 GMT
> P-Asserted-Identity: "My Name" <sip:8077005 at 10.98.88.38>
> Content-Length: 0
>
> From: Mark Holloway [mailto:mh at markholloway.com]
> Sent: Thursday, January 10, 2013 1:51 PM
> To: Heim, Dennis
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SIP CUBE Caller ID
>
> The FROM field should have a name in front of the phone number. (Like the bottom
line of text in this capture). Can you send the SIP INVITE from CUCM > CUBE, and
from CUBE to the PBX?
>
> <image001.png>
>
>
>
> On Jan 10, 2013, at 1:26 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:
>
>
> I am doing a sip integration to a pbx. We are not getting the caller id
name/number when a call is placed form cisco to the pbx via the cube. We do see the
caller id name in the debugs, but not on the phone display.
>
> Received:
> SIP/2.0 200 OK
> Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51
> CSeq: 101 INVITE
> From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B
> To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp
> Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7
> Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>
> Content-Type: application/sdp
> Content-Length: 367
> X-Siemens-Call-Type: ST-insecure
> Accept-Language: en;q=0.0
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
> Date: Thu, 10 Jan 2013 17:20:27 GMT
> P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>
>
> What am I missing?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From Dennis.Heim at wwt.com Thu Jan 10 16:58:23 2013


From: Dennis.Heim at wwt.com (Heim, Dennis)
Date: Thu, 10 Jan 2013 15:58:23 -0600
Subject: [cisco-voip] SIP CUBE Caller ID
In-Reply-To: <F3544AB7-071B-4494-B057-69A7796171CB@markholloway.com>
References: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
<3B793D75-EFA9-4601-AE9B-3BAC2F0B7628@markholloway.com>
<0CC57FCAB07CEB4595526952471493D316F247D358@PRODCMS1.wwt.local>
<CAKBLUbC=90Qjg90c6PHeE8+6YmnaK1mZz79sS2sUP5+AoPS06w@mail.gmail.com>
<0CC57FCAB07CEB4595526952471493D316F247D382@PRODCMS1.wwt.local>
<F3544AB7-071B-4494-B057-69A7796171CB@markholloway.com>
Message-ID: <0CC57FCAB07CEB4595526952471493D316F247D3B8@PRODCMS1.wwt.local>

Is it possible with cube to take the PAI and copy it into the from field?

Call setup

INVITE sip:18077014 at CUBE:5060 SIP/2.0


Via: SIP/2.0/TCP CUCM:5060;branch=z9hG4bK1eda32bbab882
From: "Cisco Phone" <sip:5987800 at CUCM>;tag=2004325~37777082-f7ba-4cb2-8467-
e286cea21f2e-88710674
To: <sip:18077014 at CUBE>
Date: Thu, 10 Jan 2013 21:47:24 GMT
Call-ID: 50330380-ef136ec-1a038-2b3dd20a at CUCM
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 1345520512-0000065536-0000033495-0725471754
Session-Expires: 1800
P-Asserted-Identity: "Cisco Phone" <sip:5987800 at CUCM>
Remote-Party-ID: "Cisco Phone" <sip:5987800 at
CUCM>;party=calling;screen=yes;privacy=off
Contact: <sip:5987800 at CUCM:5060;transport=tcp>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 214

When ringing

Received:
SIP/2.0 180 Ringing
Call-ID: 1A53E43D-5AA611E2-A21DF143-739C4300 at CUBE
CSeq: 101 INVITE
From: sip:5987800 at CUCM;tag=809784F8-1B9F
To: <sip:18077014 at SiemensOSV>;tag=SEC11-a58620a-1e58d20a-1-4197U1P1AG4o
Via: SIP/2.0/UDP CUBE:5060;branch=z9hG4bK7B1AD5
Contact: <sip:18077014 at SiemensOSV:5060;maddr=SiemensOSV>
Content-Type: application/sdp
Content-Length: 320
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
Allow-Events: CCNR
Date: Thu, 10 Jan 2013 21:47:24 GMT
P-Asserted-Identity: "Siemens User" <sip:18077014 at SiemensOSV>
From: Mark Holloway [mailto:mh at markholloway.com]
Sent: Thursday, January 10, 2013 4:23 PM
To: Heim, Dennis
Cc: Kenneth Hayes; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP CUBE Caller ID

What I was looking for, but didn't see in your TXT file, is the SIP INVITE from
CUCM includes the calling party name in the FROM field to CUBE, and CUBE includes
the same FROM name to Siemens. Assuming that is true, Siemens appears to be
disregarding the name portion of the FROM field by not only failing to display it,
but completely removing it in 1XX responses. It would be nice if it processed PAI
but I think the name is being stripped and then Siemens just assumes there is no
name for the call, as if PAI didn't exist.

On Jan 10, 2013, at 3:56 PM, "Heim, Dennis" <Dennis.Heim at


wwt.com<mailto:Dennis.Heim at wwt.com>> wrote:

This is outbound from CUCM. I dial the number it connects and the display just
shows the dialed number, without the name. I did not have a display name on my
phone, when I fixed that, the caller id name is visible to the siemens pbx.

The problem still is that I dial a number and it does not put in the name

The Cisco-Siemens direction, we see Cisco set the From with the name (From: "Cisco
Phone" sip:5987800 at ip. We also see the P-Asserted-Identity set to "Cisco Phone"
sip:5987800 at ip.

On the invite coming back from the siemens we see it set the From, only contain the
number. However, the P-Asserted-Identity, contains the caller id name.

From: Heim, Dennis


Sent: Thursday, January 10, 2013 3:46 PM
To: 'Kenneth Hayes'; Voice_VT
Subject: RE: [cisco-voip] SIP CUBE Caller ID
This is outbound from CUCM. I dial the number it connects and the display just
shows the dialed number, with the name. I did not have a display name on my phone,
when I fixed that, the caller id name is visible to the siemens pbx.

The problem still is that I dial a number and it does not put in the name.

From: Kenneth Hayes [mailto:kennethwhayes at gmail.com]<mailto:


[mailto:kennethwhayes at gmail.com]>
Sent: Thursday, January 10, 2013 3:29 PM
To: Heim, Dennis
Subject: Re: [cisco-voip] SIP CUBE Caller ID

This is inbound correct?


On Thu, Jan 10, 2013 at 2:55 PM, Heim, Dennis <Dennis.Heim at
wwt.com<mailto:Dennis.Heim at wwt.com>> wrote:
It is showing in the PAI, but not the from field.

Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51<mailto:C2451EA-


5A9611E2-92D3F143-739C4300 at 10.98.254.51>
CSeq: 101 INVITE
Timestamp: 1357847526
Content-Length: 0

Jan 10 19:52:06.991: //184752/427582000000/SIP/Info/sipSPICheckResponseExt: INVITE


response with no RSEQ - disable IS_REL1XX
Jan 10 19:52:06.991: //184752/427582000000/SIP/State/sipSPIChangeState: 0x2472CB8 :
State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING,
SUBSTATE_PROCEEDING_PROCEEDING)
Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg
enqueued for SPI with IP addr: [10.98.88.38]:5060, local_address:[10.98.254.51]
Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
ccsip_spi_get_msg_type returned: 2 for event 1
Jan 10 19:52:07.119: //-
1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
Checking Invite Dialog
Jan 10 19:52:07.119: //184752/427582000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51<mailto:C2451EA-
5A9611E2-92D3F143-739C4300 at 10.98.254.51>
CSeq: 101 INVITE
From: sip:XXXX7800 at 10.210.61.43<mailto:sip%3AXXXX7800 at
10.210.61.43>;tag=802E4C74-179B
To: <sip:18077005 at 10.98.88.38<mailto:sip%3A18077005 at 10.98.88.38>>;tag=SEC11-
a58620a-1e58d20a-1-zvE1E7A74E0q
Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK5C1C8C
Contact: <sip:18077005 at 10.98.88.38:5060;maddr=10.98.88.38>
Date: Thu, 10 Jan 2013 19:52:29 GMT
P-Asserted-Identity: "My Name" <sip:8077005 at 10.98.88.38<mailto:sip%3A8077005 at
10.98.88.38>>
Content-Length: 0

From: Mark Holloway [mailto:mh at markholloway.com<mailto:mh at markholloway.com>]


Sent: Thursday, January 10, 2013 1:51 PM
To: Heim, Dennis
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] SIP CUBE Caller ID

The FROM field should have a name in front of the phone number. (Like the bottom
line of text in this capture). Can you send the SIP INVITE from CUCM > CUBE, and
from CUBE to the PBX?

<image001.png>

On Jan 10, 2013, at 1:26 PM, "Heim, Dennis" <Dennis.Heim at


wwt.com<mailto:Dennis.Heim at wwt.com>> wrote:

I am doing a sip integration to a pbx. We are not getting the caller id name/number
when a call is placed form cisco to the pbx via the cube. We do see the caller id
name in the debugs, but not on the phone display.

Received:
SIP/2.0 200 OK
Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51<mailto:CD2B18C9-
5A8011E2-80D5F143-739C4300 at 10.98.254.51>
CSeq: 101 INVITE
From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B
To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp
Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7
Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>
Content-Type: application/sdp
Content-Length: 367
X-Siemens-Call-Type: ST-insecure
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
Date: Thu, 10 Jan 2013 17:20:27 GMT
P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>

What am I missing?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From mh at markholloway.com Thu Jan 10 17:15:24 2013


From: mh at markholloway.com (Mark Holloway)
Date: Thu, 10 Jan 2013 17:15:24 -0500
Subject: [cisco-voip] SIP CUBE Caller ID
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F247D3B8@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
<3B793D75-EFA9-4601-AE9B-3BAC2F0B7628@markholloway.com>
<0CC57FCAB07CEB4595526952471493D316F247D358@PRODCMS1.wwt.local>
<CAKBLUbC=90Qjg90c6PHeE8+6YmnaK1mZz79sS2sUP5+AoPS06w@mail.gmail.com>
<0CC57FCAB07CEB4595526952471493D316F247D382@PRODCMS1.wwt.local>
<F3544AB7-071B-4494-B057-69A7796171CB@markholloway.com>
<0CC57FCAB07CEB4595526952471493D316F247D3B8@PRODCMS1.wwt.local>
Message-ID: <F9FC424D-CEC8-4036-8823-D04AC325F629@markholloway.com>

Maybe with SIP Profiles, but I think the real issue is Siemens isn't honoring the
From name, RPID, or PAI. Even if you add it back in after a 1XX response it may
not solve your problem.

On Jan 10, 2013, at 4:58 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:

> Is it possible with cube to take the PAI and copy it into the from field?
>
> Call setup
>
> INVITE sip:18077014 at CUBE:5060 SIP/2.0
> Via: SIP/2.0/TCP CUCM:5060;branch=z9hG4bK1eda32bbab882
> From: "Cisco Phone" <sip:5987800 at CUCM>;tag=2004325~37777082-f7ba-4cb2-8467-
e286cea21f2e-88710674
> To: <sip:18077014 at CUBE>
> Date: Thu, 10 Jan 2013 21:47:24 GMT
> Call-ID: 50330380-ef136ec-1a038-2b3dd20a at CUCM
> Supported: timer,resource-priority,replaces
> Min-SE: 1800
> User-Agent: Cisco-CUCM8.6
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY
> CSeq: 101 INVITE
> Expires: 180
> Allow-Events: presence, kpml
> Supported: X-cisco-srtp-fallback
> Supported: Geolocation
> Cisco-Guid: 1345520512-0000065536-0000033495-0725471754
> Session-Expires: 1800
> P-Asserted-Identity: "Cisco Phone" <sip:5987800 at CUCM>
> Remote-Party-ID: "Cisco Phone" <sip:5987800 at
CUCM>;party=calling;screen=yes;privacy=off
> Contact: <sip:5987800 at CUCM:5060;transport=tcp>
> Max-Forwards: 69
> Content-Type: application/sdp
> Content-Length: 214
>
> When ringing
>
> Received:
> SIP/2.0 180 Ringing
> Call-ID: 1A53E43D-5AA611E2-A21DF143-739C4300 at CUBE
> CSeq: 101 INVITE
> From: sip:5987800 at CUCM;tag=809784F8-1B9F
> To: <sip:18077014 at SiemensOSV>;tag=SEC11-a58620a-1e58d20a-1-4197U1P1AG4o
> Via: SIP/2.0/UDP CUBE:5060;branch=z9hG4bK7B1AD5
> Contact: <sip:18077014 at SiemensOSV:5060;maddr=SiemensOSV>
> Content-Type: application/sdp
> Content-Length: 320
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
> Allow-Events: CCNR
> Date: Thu, 10 Jan 2013 21:47:24 GMT
> P-Asserted-Identity: "Siemens User" <sip:18077014 at SiemensOSV>
> From: Mark Holloway [mailto:mh at markholloway.com]
> Sent: Thursday, January 10, 2013 4:23 PM
> To: Heim, Dennis
> Cc: Kenneth Hayes; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SIP CUBE Caller ID
>
> What I was looking for, but didn't see in your TXT file, is the SIP INVITE from
CUCM includes the calling party name in the FROM field to CUBE, and CUBE includes
the same FROM name to Siemens. Assuming that is true, Siemens appears to be
disregarding the name portion of the FROM field by not only failing to display it,
but completely removing it in 1XX responses. It would be nice if it processed PAI
but I think the name is being stripped and then Siemens just assumes there is no
name for the call, as if PAI didn't exist.
>
> On Jan 10, 2013, at 3:56 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:
>
>
> This is outbound from CUCM. I dial the number it connects and the display just
shows the dialed number, without the name. I did not have a display name on my
phone, when I fixed that, the caller id name is visible to the siemens pbx.
>
> The problem still is that I dial a number and it does not put in the name
>
> The Cisco-Siemens direction, we see Cisco set the From with the name (From: ?
Cisco Phone? sip:5987800 at ip. We also see the P-Asserted-Identity set to ?Cisco
Phone? sip:5987800 at ip.
>
> On the invite coming back from the siemens we see it set the From, only contain
the number. However, the P-Asserted-Identity, contains the caller id name.
>
> From: Heim, Dennis
> Sent: Thursday, January 10, 2013 3:46 PM
> To: 'Kenneth Hayes'; Voice_VT
> Subject: RE: [cisco-voip] SIP CUBE Caller ID
>
> This is outbound from CUCM. I dial the number it connects and the display just
shows the dialed number, with the name. I did not have a display name on my phone,
when I fixed that, the caller id name is visible to the siemens pbx.
>
> The problem still is that I dial a number and it does not put in the name.
>
> From: Kenneth Hayes [mailto:kennethwhayes at gmail.com]
> Sent: Thursday, January 10, 2013 3:29 PM
> To: Heim, Dennis
> Subject: Re: [cisco-voip] SIP CUBE Caller ID
>
> This is inbound correct?
>
> On Thu, Jan 10, 2013 at 2:55 PM, Heim, Dennis <Dennis.Heim at wwt.com> wrote:
> It is showing in the PAI, but not the from field.
>
> Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51
> CSeq: 101 INVITE
> Timestamp: 1357847526
> Content-Length: 0
>
>
> Jan 10 19:52:06.991: //184752/427582000000/SIP/Info/sipSPICheckResponseExt:
INVITE response with no RSEQ - disable IS_REL1XX
> Jan 10 19:52:06.991: //184752/427582000000/SIP/State/sipSPIChangeState: 0x2472CB8
: State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING,
SUBSTATE_PROCEEDING_PROCEEDING)
> Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg
enqueued for SPI with IP addr: [10.98.88.38]:5060, local_address:[10.98.254.51]
> Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
ccsip_spi_get_msg_type returned: 2 for event 1
> Jan 10 19:52:07.119: //-
1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
> Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
Checking Invite Dialog
> Jan 10 19:52:07.119: //184752/427582000000/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 180 Ringing
> Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51
> CSeq: 101 INVITE
> From: sip:XXXX7800 at 10.210.61.43;tag=802E4C74-179B
> To: <sip:18077005 at 10.98.88.38>;tag=SEC11-a58620a-1e58d20a-1-zvE1E7A74E0q
> Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK5C1C8C
> Contact: <sip:18077005 at 10.98.88.38:5060;maddr=10.98.88.38>
> Date: Thu, 10 Jan 2013 19:52:29 GMT
> P-Asserted-Identity: "My Name" <sip:8077005 at 10.98.88.38>
> Content-Length: 0
>
> From: Mark Holloway [mailto:mh at markholloway.com]
> Sent: Thursday, January 10, 2013 1:51 PM
> To: Heim, Dennis
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SIP CUBE Caller ID
>
> The FROM field should have a name in front of the phone number. (Like the bottom
line of text in this capture). Can you send the SIP INVITE from CUCM > CUBE, and
from CUBE to the PBX?
>
> <image001.png>
>
>
>
> On Jan 10, 2013, at 1:26 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:
>
>
> I am doing a sip integration to a pbx. We are not getting the caller id
name/number when a call is placed form cisco to the pbx via the cube. We do see the
caller id name in the debugs, but not on the phone display.
>
> Received:
> SIP/2.0 200 OK
> Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51
> CSeq: 101 INVITE
> From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B
> To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp
> Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7
> Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>
> Content-Type: application/sdp
> Content-Length: 367
> X-Siemens-Call-Type: ST-insecure
> Accept-Language: en;q=0.0
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
> Date: Thu, 10 Jan 2013 17:20:27 GMT
> P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>
>
> What am I missing?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

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From tman701 at gmail.com Thu Jan 10 17:19:26 2013


From: tman701 at gmail.com (Joel Perez)
Date: Thu, 10 Jan 2013 17:19:26 -0500
Subject: [cisco-voip] SIP CUBE Caller ID
In-Reply-To: <0CC57FCAB07CEB4595526952471493D316F247D3B8@PRODCMS1.wwt.local>
References: <0CC57FCAB07CEB4595526952471493D316F247D315@PRODCMS1.wwt.local>
<3B793D75-EFA9-4601-AE9B-3BAC2F0B7628@markholloway.com>
<0CC57FCAB07CEB4595526952471493D316F247D358@PRODCMS1.wwt.local>
<CAKBLUbC=90Qjg90c6PHeE8+6YmnaK1mZz79sS2sUP5+AoPS06w@mail.gmail.com>
<0CC57FCAB07CEB4595526952471493D316F247D382@PRODCMS1.wwt.local>
<F3544AB7-071B-4494-B057-69A7796171CB@markholloway.com>
<0CC57FCAB07CEB4595526952471493D316F247D3B8@PRODCMS1.wwt.local>
Message-ID: <CABdWoUHTFgdqDFO9jPxKJTUKAOj_L-EUg1e7bZXWZQUtnxLCUg@mail.gmail.com>

You should be able to but it depends on the version of IOS.

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_exampl
e09186a0080982499.shtml

I havent tried it with CUBE, only with ACME but it should be the same or
similar.

Joel P.

On Thu, Jan 10, 2013 at 4:58 PM, Heim, Dennis <Dennis.Heim at wwt.com> wrote:
> Is it possible with cube to take the PAI and copy it into the from field?*
> ***
>
> ** **
>
> *Call setup*
>
> * *
>
> INVITE sip:18077014 at CUBE:5060 SIP/2.0****
>
> Via: SIP/2.0/TCP CUCM:5060;branch=z9hG4bK1eda32bbab882****
>
> From: "Cisco Phone" <sip:5987800 at CUCM
> >;tag=2004325~37777082-f7ba-4cb2-8467-e286cea21f2e-88710674****
>
> To: <sip:18077014 at CUBE>****
>
> Date: Thu, 10 Jan 2013 21:47:24 GMT****
>
> Call-ID: 50330380-ef136ec-1a038-2b3dd20a at CUCM****
>
> Supported: timer,resource-priority,replaces****
>
> Min-SE: 1800****
>
> User-Agent: Cisco-CUCM8.6****
>
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY****
>
> CSeq: 101 INVITE****
>
> Expires: 180****
>
> Allow-Events: presence, kpml****
>
> Supported: X-cisco-srtp-fallback****
>
> Supported: Geolocation****
>
> Cisco-Guid: 1345520512-0000065536-0000033495-0725471754****
>
> Session-Expires: 1800****
>
> P-Asserted-Identity: "Cisco Phone" <sip:5987800 at CUCM>****
>
> Remote-Party-ID: "Cisco Phone" <sip:5987800 at CUCM
> >;party=calling;screen=yes;privacy=off****
>
> Contact: <sip:5987800 at CUCM:5060;transport=tcp>****
>
> Max-Forwards: 69****
>
> Content-Type: application/sdp****
>
> Content-Length: 214****
>
> ** **
>
> *When ringing*
>
> ** **
>
> Received: ****
>
> SIP/2.0 180 Ringing****
>
> Call-ID: 1A53E43D-5AA611E2-A21DF143-739C4300 at CUBE****
>
> CSeq: 101 INVITE****
>
> From: sip:5987800 at CUCM;tag=809784F8-1B9F****
>
> To: <sip:18077014 at SiemensOSV>;tag=SEC11-a58620a-1e58d20a-1-4197U1P1AG4o***
> *
>
> Via: SIP/2.0/UDP CUBE:5060;branch=z9hG4bK7B1AD5****
>
> Contact: <sip:18077014 at SiemensOSV:5060;maddr=SiemensOSV>****
>
> Content-Type: application/sdp****
>
> Content-Length: 320****
>
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO****
>
> Allow-Events: CCNR****
>
> Date: Thu, 10 Jan 2013 21:47:24 GMT****
>
> P-Asserted-Identity: "Siemens User" <sip:18077014 at SiemensOSV>****
>
> *From:* Mark Holloway [mailto:mh at markholloway.com]
> *Sent:* Thursday, January 10, 2013 4:23 PM
> *To:* Heim, Dennis
> *Cc:* Kenneth Hayes; cisco-voip at puck.nether.net
>
> *Subject:* Re: [cisco-voip] SIP CUBE Caller ID****
>
> ** **
>
> What I was looking for, but didn't see in your TXT file, is the SIP INVITE
> from CUCM includes the calling party name in the FROM field to CUBE, and
> CUBE includes the same FROM name to Siemens. Assuming that is true, Siemens
> appears to be disregarding the name portion of the FROM field by not only
> failing to display it, but completely removing it in 1XX responses. It
> would be nice if it processed PAI but I think the name is being stripped
> and then Siemens just assumes there is no name for the call, as if PAI
> didn't exist.****
>
> ** **
>
> On Jan 10, 2013, at 3:56 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:**
> **
>
>
>
> ****
>
> This is outbound from CUCM. I dial the number it connects and the display
> just shows the dialed number, without the name. I did not have a display
> name on my phone, when I fixed that, the caller id name is visible to the
> siemens pbx.****
>
> ****
>
> The problem still is that I dial a number and it does not put in the name*
> ***
>
> ****
>
> The Cisco-Siemens direction, we see Cisco set the From with the name
> (From: ?Cisco Phone? sip:5987800 at ip. We also see the P-Asserted-Identity
> set to ?Cisco Phone? sip:5987800 at ip.****
>
> ****
>
> On the invite coming back from the siemens we see it set the From, only
> contain the number. However, the P-Asserted-Identity, contains the caller
> id name.****
>
> ****
>
> *From:* Heim, Dennis
> *Sent:* Thursday, January 10, 2013 3:46 PM
> *To:* 'Kenneth Hayes'; Voice_VT
> *Subject:* RE: [cisco-voip] SIP CUBE Caller ID****
>
> ****
>
> This is outbound from CUCM. I dial the number it connects and the display
> just shows the dialed number, with the name. I did not have a display name
> on my phone, when I fixed that, the caller id name is visible to the
> siemens pbx.****
>
> ****
>
> The problem still is that I dial a number and it does not put in the name.
> ****
>
> ****
>
> *From:* Kenneth Hayes [mailto:kennethwhayes at gmail.com]
> *Sent:* Thursday, January 10, 2013 3:29 PM
> *To:* Heim, Dennis
> *Subject:* Re: [cisco-voip] SIP CUBE Caller ID****
>
> ****
>
> This is inbound correct?****
>
> On Thu, Jan 10, 2013 at 2:55 PM, Heim, Dennis <Dennis.Heim at wwt.com> wrote:
> ****
>
> It is showing in the PAI, but not the from field.****
>
> ****
>
> Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51****
>
> CSeq: 101 INVITE****
>
> Timestamp: 1357847526****
>
> Content-Length: 0****
>
> ****
>
> ****
>
> Jan 10 19:52:06.991:
> //184752/427582000000/SIP/Info/sipSPICheckResponseExt: INVITE response with
> no RSEQ - disable IS_REL1XX****
>
> Jan 10 19:52:06.991: //184752/427582000000/SIP/State/sipSPIChangeState:
> 0x2472CB8 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to
> (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)****
>
> Jan 10 19:52:07.119: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
> Msg enqueued for SPI with IP addr: [10.98.88.38]:5060,
> local_address:[10.98.254.51]****
>
> Jan 10 19:52:07.119:
> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
> ccsip_spi_get_msg_type returned: 2 for event 1****
>
> Jan 10 19:52:07.119:
> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
> ****
>
> Jan 10 19:52:07.119:
> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite
> Dialog****
>
> Jan 10 19:52:07.119: //184752/427582000000/SIP/Msg/ccsipDisplayMsg:****
>
> Received:****
>
> SIP/2.0 180 Ringing****
>
> Call-ID: C2451EA-5A9611E2-92D3F143-739C4300 at 10.98.254.51****
>
> CSeq: 101 INVITE****
>
> From: sip:XXXX7800 at 10.210.61.43;tag=802E4C74-179B****
>
> To: <sip:18077005 at 10.98.88.38>;tag=SEC11-a58620a-1e58d20a-1-zvE1E7A74E0q**
> **
>
> Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK5C1C8C****
>
> Contact: <sip:18077005 at 10.98.88.38:5060;maddr=10.98.88.38>****
>
> Date: Thu, 10 Jan 2013 19:52:29 GMT****
>
> P-Asserted-Identity: "My Name" <sip:8077005 at 10.98.88.38>****
>
> Content-Length: 0****
>
> ****
>
> *From:* Mark Holloway [mailto:mh at markholloway.com]
> *Sent:* Thursday, January 10, 2013 1:51 PM
> *To:* Heim, Dennis
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] SIP CUBE Caller ID****
>
> ****
>
> The FROM field should have a name in front of the phone number. (Like the
> bottom line of text in this capture). Can you send the SIP INVITE from
> CUCM > CUBE, and from CUBE to the PBX? ****
>
> ****
>
> <image001.png>****
>
> ****
>
> ****
>
> ****
>
> On Jan 10, 2013, at 1:26 PM, "Heim, Dennis" <Dennis.Heim at wwt.com> wrote:**
> **
>
> ****
>
> I am doing a sip integration to a pbx. We are not getting the caller id
> name/number when a call is placed form cisco to the pbx via the cube. We do
> see the caller id name in the debugs, but not on the phone display.****
>
> ****
>
> Received:****
>
> SIP/2.0 200 OK****
>
> Call-ID: CD2B18C9-5A8011E2-80D5F143-739C4300 at 10.98.254.51****
>
> CSeq: 101 INVITE****
>
> From: <sip:XXXXXXXXXX at 10.98.254.51>;tag=7FA30F9C-154B****
>
> To: <sip:18077014 at osv1.XXX.com>;tag=SEC21-a58620a-1e58d20a-2-1e9kU4nFr1Kp*
> ***
>
> Via: SIP/2.0/UDP 10.98.254.51:5060;branch=z9hG4bK456B7****
>
> Contact: <sip:18077014 at 10.210.88.39:5060;maddr=10.210.88.39>****
>
> Content-Type: application/sdp****
>
> Content-Length: 367****
>
> X-Siemens-Call-Type: ST-insecure****
>
> Accept-Language: en;q=0.0****
>
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO****
>
> Date: Thu, 10 Jan 2013 17:20:27 GMT****
>
> P-Asserted-Identity: "My Name" <sip:8077014 at 10.210.88.39>****
>
> ****
>
> What am I missing?****
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ****
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ****
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From mizzou0 at yahoo.com Thu Jan 10 21:08:02 2013


From: mizzou0 at yahoo.com (Me)
Date: Thu, 10 Jan 2013 18:08:02 -0800
Subject: [cisco-voip] Simulate SIP error code 408 and 603 from CUBE
Message-ID: <CAGBqMBjocfDmaMijThnXFWXCtF=_qSbxFuH+EYVyNtpcopztcA@mail.gmail.com>

Is there a way to simulate SIP error code 408 (Not Available) and 603
(Reject) from a Cisco CUBE? Such as making a call to a number router
returns the SIP errors.

Regards,
Dave
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From mh at markholloway.com Thu Jan 10 22:30:41 2013


From: mh at markholloway.com (Mark Holloway)
Date: Thu, 10 Jan 2013 22:30:41 -0500
Subject: [cisco-voip] Simulate SIP error code 408 and 603 from CUBE
In-Reply-To: <CAGBqMBjocfDmaMijThnXFWXCtF=_qSbxFuH+EYVyNtpcopztcA@mail.gmail.com>
References: <CAGBqMBjocfDmaMijThnXFWXCtF=_qSbxFuH+EYVyNtpcopztcA@mail.gmail.com>
Message-ID: <05DC3998-0E6A-4C0F-9C9D-3630BFD9234E@markholloway.com>

You need a SIP device to originate the call to CUBE and something on the "other
side" of CUBE to generate the 603 message. For SIP 408, this usually occurs when a
SIP INVITE Timeout occurs. You could configure CUBE to forward to some null device
to the INVITE will timeout and CUBE will send 408 back to the originating party.

SIPp and X-Lite work well for this sort of thing.

On Jan 10, 2013, at 9:08 PM, Me <mizzou0 at yahoo.com> wrote:

> Is there a way to simulate SIP error code 408 (Not Available) and 603 (Reject)
from a Cisco CUBE? Such as making a call to a number router returns the SIP
errors.
>
> Regards,
> Dave
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From ahmed_elnagar at hotmail.com Fri Jan 11 09:10:37 2013


From: ahmed_elnagar at hotmail.com (Ahmed Elnagar)
Date: Fri, 11 Jan 2013 16:10:37 +0200
Subject: [cisco-voip] Cisco Unified IP Phone Local Kernel System Call Input
Validation Vulnerability
Message-ID: <BAY149-ds36C305B9071A909BD158C87290@phx.gbl>

I think someone was asking about the below a couple of days

Regards,

Ahmed Elnagar | Unified Communication Team Leader | CCIE #24697, Voice

Description: Description: Description: MS Green


From: CiscoNotificationService at cisco.com
[mailto:CiscoNotificationService at cisco.com]
Sent: Thursday, January 10, 2013 3:33 PM
To: Ahmed Elnagar
Subject: Cisco Notification Alert -UC-Products-01/10/2013 13:32 GMT

Cisco Notification Service Alert:


____________________________________________________________________________
____

Security Advisories & Responses for All Voice and Unified Communications

Title

Cisco Unified IP Phone Local Kernel System Call Input Validation


Vulnerability
<http://www.cisco.com/en/US/products/csa/cisco-sa-20130109-uipphone.html>

Description

Cisco Unified IP Phones 7900 Series versions 9.3(1)SR1 and prior contain an
arbitrary code execution vulnerability that could allow a local attacker to
execute code or modify arbitrary memory with elevated privileges. This
vulnerability is due to a failure to properly validate input passed to
kernel system calls from applications running in userspace. An attacker
could exploit this issue by gaining local access to the device using
physical access or authenticated access using SSH and executing an
attacker-controlled binary that is designed to exploit the issue. Such an
attack would originate from an unprivileged context. Ang Cui initially
reported the issue to the Cisco Product Security Incident Response Team
(PSIRT). On November 6, 2012, the Cisco PSIRT disclosed this issue in Cisco
bug ID CSCuc83860 (registered customers only) Release Note Enclosure.
Subsequently, Mr. Cui has spoken at several public conferences and has
performed public demonstrations of a device being compromised and used as a
listening device. Mitigations are available to help reduce the attack
surface of affected devices. See the &quo;Details&quo; section of this
security advisory and the accompanying Cisco Applied Mitigation Bulletin
(AMB) for additional information. This advisory is available at the
following link:
http://tools.cisco.com/security/center/content/CiscoSecurityAdvisory/cisco-s
a-20130109-uipphone

Date

09-JAN-2013
For more information; you can visit Cisco Security Advisories
<http://www.cisco.com/en/US/products/products_security_advisories_listing.ht
ml> & Responses index.
____________________________________________________________________________
____

To unsubscribe this notification click here


<http://www.cisco.com/cisco/support/notifications/addedit.html?notiId=186498
>

Help us improve this facility. To give feedback click here


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From dzhars at gmail.com Fri Jan 11 13:32:53 2013


From: dzhars at gmail.com (David Zhars)
Date: Fri, 11 Jan 2013 13:32:53 -0500
Subject: [cisco-voip] FWD one Ext to Another
Message-ID: <CADe=jTGzpoYcj4FjwqfMWs+5x9mw1Xtg7BSShnCA=WeAyhYYjQ@mail.gmail.com>

Old user had ext 1212 (and this is a DID, so people can call directly from
the outside).
New user has ext 1702.

What I want is:

Internally: User dials 1212, phone rings at 1702.


Internally: Reception takes a call, transfers it with TRANS **1212 TRANS,
call goes to 1702 voicemail.

Externally: Someone calls 555-1212 and the call lands internally at 1702.

Some of this I know how to do, I am not sure about the transfer to
voicemail of the old extension and have it land at the new ext VM.

Appreciate any help!

Dave

UCM 8.0, Unity 8.0


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From tman701 at gmail.com Fri Jan 11 14:24:36 2013


From: tman701 at gmail.com (Joel Perez)
Date: Fri, 11 Jan 2013 14:24:36 -0500
Subject: [cisco-voip] FWD one Ext to Another
In-Reply-To: <CADe=jTGzpoYcj4FjwqfMWs+5x9mw1Xtg7BSShnCA=WeAyhYYjQ@mail.gmail.com>
References: <CADe=jTGzpoYcj4FjwqfMWs+5x9mw1Xtg7BSShnCA=WeAyhYYjQ@mail.gmail.com>
Message-ID: <CABdWoUFirq_PnXgGdThGwgQe+7wVoXNfYqaNier4mBe-DiZL3g@mail.gmail.com>

Hi Dave,

If i'm understanding what you want to do correctly then you can just do a
CFWA on old ext 1212 to 1702. That way any call that comes into 1212 will
be sent to the new one of 1702.
You can also do a translation pattern from 1212 to 1702.
As far as Unity you can keep the VMB for 1212 and just add 1702 as an
alternate or vice-versa.

Joel P

On Fri, Jan 11, 2013 at 1:32 PM, David Zhars <dzhars at gmail.com> wrote:

> Old user had ext 1212 (and this is a DID, so people can call directly from
> the outside).
> New user has ext 1702.
>
> What I want is:
>
> Internally: User dials 1212, phone rings at 1702.
> Internally: Reception takes a call, transfers it with TRANS **1212 TRANS,
> call goes to 1702 voicemail.
>
> Externally: Someone calls 555-1212 and the call lands internally at 1702.
>
> Some of this I know how to do, I am not sure about the transfer to
> voicemail of the old extension and have it land at the new ext VM.
>
> Appreciate any help!
>
> Dave
>
> UCM 8.0, Unity 8.0
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From rratliff at cisco.com Fri Jan 11 14:28:15 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 11 Jan 2013 14:28:15 -0500
Subject: [cisco-voip] FWD one Ext to Another
In-Reply-To: <CADe=jTGzpoYcj4FjwqfMWs+5x9mw1Xtg7BSShnCA=WeAyhYYjQ@mail.gmail.com>
References: <CADe=jTGzpoYcj4FjwqfMWs+5x9mw1Xtg7BSShnCA=WeAyhYYjQ@mail.gmail.com>
Message-ID: <CC5F2DCE-8E8D-46B8-AAB5-FAA32CAB83A4@cisco.com>

Unity call routing rule to send calls from 1212 to 1702, should be pretty straight
forward.

-Ryan

On Jan 11, 2013, at 1:32 PM, David Zhars <dzhars at gmail.com> wrote:

Old user had ext 1212 (and this is a DID, so people can call directly from the
outside).
New user has ext 1702.

What I want is:

Internally: User dials 1212, phone rings at 1702.


Internally: Reception takes a call, transfers it with TRANS **1212 TRANS, call goes
to 1702 voicemail.

Externally: Someone calls 555-1212 and the call lands internally at 1702.

Some of this I know how to do, I am not sure about the transfer to voicemail of the
old extension and have it land at the new ext VM.

Appreciate any help!

Dave

UCM 8.0, Unity 8.0

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From avholloway+cisco-voip at gmail.com Fri Jan 11 15:11:56 2013


From: avholloway+cisco-voip at gmail.com (Anthony Holloway)
Date: Fri, 11 Jan 2013 14:11:56 -0600
Subject: [cisco-voip] CUCM MTP and g729
Message-ID: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>

Hi All,

I have a wireshark capture off of my CUCM 8.6(2) which shows that it is


receiving a g729 audio stream from my VG224.

Long story short, according to the CUCM SRND, the CUCM MTP can only
terminate g711, and yet, attached is a screenshot of the wireshark capture
which clearly shows it terminating g729.

What piece of this puzzle am I missing? Also, the CUCM traces read like
the MTP is being invoked on that CUCM. It's due to the lack of the fm
package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a fun
time resolving that, as you can probably imagine.

Thanks and Happy Friday!


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From jsteinberg at gmail.com Fri Jan 11 15:43:07 2013


From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 11 Jan 2013 15:43:07 -0500
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
Message-ID: <CACCAghZi4gbL6M-iEJcNfR+BS=oMBczN=2jJ7DBWzkmuEqpq7g@mail.gmail.com>

interesting. I always wondered why the software MTP couldn't terminate


G729 for things like DTMF mismatch. It isn't like it is transcoding audio,
it is simply handling different DTMF protocols.

is it possible Wireshark is just identifying it incorrectly? could you try


to convert it to audio and see if you hear it.

On Fri, Jan 11, 2013 at 3:11 PM, Anthony Holloway <


avholloway+cisco-voip at gmail.com> wrote:

> Hi All,
>
> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
> receiving a g729 audio stream from my VG224.
>
> Long story short, according to the CUCM SRND, the CUCM MTP can only
> terminate g711, and yet, attached is a screenshot of the wireshark capture
> which clearly shows it terminating g729.
>
> What piece of this puzzle am I missing? Also, the CUCM traces read like
> the MTP is being invoked on that CUCM. It's due to the lack of the fm
> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a fun
> time resolving that, as you can probably imagine.
>
> Thanks and Happy Friday!
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From wsisk at cisco.com Fri Jan 11 15:48:00 2013


From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 11 Jan 2013 15:48:00 -0500
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
Message-ID: <3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>

Interesting observations.

I am not aware of any changes around CM's software MTP only doing G.711.

The packet capture shows RTP coming into(?) to the MTP. I do not see any sign of
anything egressing the MTP.

ccm has internal logic that attempts to connect RTP streams even if codec
negotiation fails. This is controlled by a service parameter. You may be seeing an
artifact of this behavior where no codec was common but the streams attempted to
setup anyway. Streaming codecs to the MTP that it does not support typically
results in garble or silence on the egress leg.

/wes

On Jan 11, 2013, at 3:11 PM, Anthony Holloway wrote:

Hi All,

I have a wireshark capture off of my CUCM 8.6(2) which shows that it is receiving a
g729 audio stream from my VG224.

Long story short, according to the CUCM SRND, the CUCM MTP can only terminate g711,
and yet, attached is a screenshot of the wireshark capture which clearly shows it
terminating g729.

What piece of this puzzle am I missing? Also, the CUCM traces read like the MTP is
being invoked on that CUCM. It's due to the lack of the fm package on my VG224 and
a mismatch in DTMF to the PSTN (SIP). I had a fun time resolving that, as you can
probably imagine.

Thanks and Happy Friday!

<cucm-mtp-g729-redacted.png><cucm-mtp-redacted.png><cucm-sub02b-
redacted.png>_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From avholloway+cisco-voip at gmail.com Fri Jan 11 16:02:15 2013


From: avholloway+cisco-voip at gmail.com (Anthony Holloway)
Date: Fri, 11 Jan 2013 15:02:15 -0600
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <CACCAghZi4gbL6M-iEJcNfR+BS=oMBczN=2jJ7DBWzkmuEqpq7g@mail.gmail.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<CACCAghZi4gbL6M-iEJcNfR+BS=oMBczN=2jJ7DBWzkmuEqpq7g@mail.gmail.com>
Message-ID: <CACRCJOiYAgisxj-UR_1T-SHmL-yYTJvJhe=kFQ1A8CS6iCp6eg@mail.gmail.com>

When I do a "show call active voice brief" on the VG224, it shows g729 and
the IP Address of the CUCM. That confirms it in another way. Thanks for
responding.

On Fri, Jan 11, 2013 at 2:43 PM, Justin Steinberg <jsteinberg at gmail.com>wrote:

> interesting. I always wondered why the software MTP couldn't terminate
> G729 for things like DTMF mismatch. It isn't like it is transcoding audio,
> it is simply handling different DTMF protocols.
>
> is it possible Wireshark is just identifying it incorrectly? could you
> try to convert it to audio and see if you hear it.
>
> On Fri, Jan 11, 2013 at 3:11 PM, Anthony Holloway <
> avholloway+cisco-voip at gmail.com> wrote:
>
>> Hi All,
>>
>> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
>> receiving a g729 audio stream from my VG224.
>>
>> Long story short, according to the CUCM SRND, the CUCM MTP can only
>> terminate g711, and yet, attached is a screenshot of the wireshark capture
>> which clearly shows it terminating g729.
>>
>> What piece of this puzzle am I missing? Also, the CUCM traces read like
>> the MTP is being invoked on that CUCM. It's due to the lack of the fm
>> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a fun
>> time resolving that, as you can probably imagine.
>>
>> Thanks and Happy Friday!
>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
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From avholloway+cisco-voip at gmail.com Fri Jan 11 16:08:54 2013


From: avholloway+cisco-voip at gmail.com (Anthony Holloway)
Date: Fri, 11 Jan 2013 15:08:54 -0600
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>
Message-ID: <CACRCJOguqqYfJrY38mSG47qgmnkz=RYReZdLA2rTdpPYpuSz0A@mail.gmail.com>

Hey Wes,

The packet capture was done on the CUCM itself via CLI command: "utils
network capture". Also, I filtered the capture to traffic only coming from
the VG224, which is why you do not see any other streams. It was, however,
going to our SBC. So the call flow was: Analog Phone > VG224 > CUCM (MTP)
> SBC > PSTN.

The negotiated CODEC was in fact g729, and both sides support it. The MGCP
SDP shows g729 and the SBC sends back g729 in the SIP SDP. The only thing
that is different in caps is DTMF. MGCP was trying 100 while SBC wanted to
do 101.

As for the garble: I wasn't experiencing any voice quality issues that I
could hear, but I was experiencing double DTMF going out to the PSTN. Not
sure if an artifact of the MTP, or simply a misonconfiguration on the
VG224's MGCP package. Like I said it's the fm package I was missing that
ultimately fixed the issue. The MTP is no longer used, and the double DTMF
is gone. I didn't find very much info on what the fm packages does, only
that it fixes DTMF and Faxing issues when communicating with a SIP device.

Thanks for the late Friday afternoon reply Wes.

On Fri, Jan 11, 2013 at 2:48 PM, Wes Sisk <wsisk at cisco.com> wrote:

> Interesting observations.


>
> I am not aware of any changes around CM's software MTP only doing G.711.
>
> The packet capture shows RTP coming into(?) to the MTP. I do not see any
> sign of anything egressing the MTP.
>
> ccm has internal logic that attempts to connect RTP streams even if codec
> negotiation fails. This is controlled by a service parameter. You may be
> seeing an artifact of this behavior where no codec was common but the
> streams attempted to setup anyway. Streaming codecs to the MTP that it
> does not support typically results in garble or silence on the egress leg.
>
> /wes
>
>
> On Jan 11, 2013, at 3:11 PM, Anthony Holloway wrote:
>
> Hi All,
>
> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
> receiving a g729 audio stream from my VG224.
>
> Long story short, according to the CUCM SRND, the CUCM MTP can only
> terminate g711, and yet, attached is a screenshot of the wireshark capture
> which clearly shows it terminating g729.
>
> What piece of this puzzle am I missing? Also, the CUCM traces read like
> the MTP is being invoked on that CUCM. It's due to the lack of the fm
> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a fun
> time resolving that, as you can probably imagine.
>
> Thanks and Happy Friday!
>
>
>
> <cucm-mtp-g729-redacted.png><cucm-mtp-redacted.png><cucm-sub02b-
redacted.png>_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From peter.slow at gmail.com Fri Jan 11 16:13:17 2013


From: peter.slow at gmail.com (Peter Slow)
Date: Fri, 11 Jan 2013 16:13:17 -0500
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <CACRCJOiYAgisxj-UR_1T-SHmL-yYTJvJhe=kFQ1A8CS6iCp6eg@mail.gmail.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<CACCAghZi4gbL6M-iEJcNfR+BS=oMBczN=2jJ7DBWzkmuEqpq7g@mail.gmail.com>
<CACRCJOiYAgisxj-UR_1T-SHmL-yYTJvJhe=kFQ1A8CS6iCp6eg@mail.gmail.com>
Message-ID: <CAMa5Jw7ZznC4s8ExtRuy3ZTi_S8MrnB0gAuQorCw0FxG+ZFN0w@mail.gmail.com>

Wes,
I once filed a defect against ipvmsapp for, if i remember
correctly, ORCAcking ("successfully") a skinny ORC for 729. I can't
remember the exact heaadline, and I think it may actually have been
against the conference bridge portion of the code. maybe there's
similar behavior in the MTP portion of the code? See anything that
looks similar to what I'm describing? It'd be interesting to find it
again and see exactly what it was.

-Pete

On Fri, Jan 11, 2013 at 4:02 PM, Anthony Holloway


<avholloway+cisco-voip at gmail.com> wrote:
> When I do a "show call active voice brief" on the VG224, it shows g729 and
> the IP Address of the CUCM. That confirms it in another way. Thanks for
> responding.
>
>
> On Fri, Jan 11, 2013 at 2:43 PM, Justin Steinberg <jsteinberg at gmail.com>
> wrote:
>>
>> interesting. I always wondered why the software MTP couldn't terminate
>> G729 for things like DTMF mismatch. It isn't like it is transcoding audio,
>> it is simply handling different DTMF protocols.
>>
>> is it possible Wireshark is just identifying it incorrectly? could you
>> try to convert it to audio and see if you hear it.
>>
>> On Fri, Jan 11, 2013 at 3:11 PM, Anthony Holloway
>> <avholloway+cisco-voip at gmail.com> wrote:
>>>
>>> Hi All,
>>>
>>> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
>>> receiving a g729 audio stream from my VG224.
>>>
>>> Long story short, according to the CUCM SRND, the CUCM MTP can only
>>> terminate g711, and yet, attached is a screenshot of the wireshark capture
>>> which clearly shows it terminating g729.
>>>
>>> What piece of this puzzle am I missing? Also, the CUCM traces read like
>>> the MTP is being invoked on that CUCM. It's due to the lack of the fm
>>> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a fun
>>> time resolving that, as you can probably imagine.
>>>
>>> Thanks and Happy Friday!
>>>
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From peter.slow at gmail.com Fri Jan 11 16:24:35 2013


From: peter.slow at gmail.com (Peter Slow)
Date: Fri, 11 Jan 2013 16:24:35 -0500
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <CACRCJOguqqYfJrY38mSG47qgmnkz=RYReZdLA2rTdpPYpuSz0A@mail.gmail.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>
<CACRCJOguqqYfJrY38mSG47qgmnkz=RYReZdLA2rTdpPYpuSz0A@mail.gmail.com>
Message-ID: <CAMa5Jw7OZ1-cQOo3KVdx56A9FdTDmOqVK9rzxy5+CHr8v8dQ9Q@mail.gmail.com>

PS> ..Also, are you SURE this isn't just a device hearing g.729 hold
music while you've got or had the Duplex Streaming service parameter
enabled? ...'Cause that would totally explain this packet capture. Do
you have the skinny signalling to go with it showing what it was
specifically set up for use as?

Also, I beleive MGCP Endpoints have an initial "state" when they begin
call setup. i think not using g.729 actually entails a switch from 729
to another codec, and perhaps a small delay is causing some packets to
be transmitted using g.729? maybe? that's a complete stretch but who
knows =)

I don't think you're really goign to get an answer unless you can
recreate the issue and we can see traces. you'll also want a packet
capture of the registration of the media device, so if you can make
the MTP register to a different callmanager than what it's running on,
using its CUCM group, we could take a look at what capabilities it was
registering with and if it says it supports 729 now =) ..you'll wnat
to look at the skinny registration of the MTP, ANN ooh, ANN
announcements are in 7.29 also, i think? you coudl be doing duplex
audio while one of those is playing =)....

anyway, can you reproduce it or verify or deny any of those guesses?

Very Interesting,
-Pete

On Fri, Jan 11, 2013 at 4:08 PM, Anthony Holloway


<avholloway+cisco-voip at gmail.com> wrote:
>
> Hey Wes,
>
> The packet capture was done on the CUCM itself via CLI command: "utils
> network capture". Also, I filtered the capture to traffic only coming from
> the VG224, which is why you do not see any other streams. It was, however,
> going to our SBC. So the call flow was: Analog Phone > VG224 > CUCM (MTP) >
> SBC > PSTN.
>
> The negotiated CODEC was in fact g729, and both sides support it. The MGCP
> SDP shows g729 and the SBC sends back g729 in the SIP SDP. The only thing
> that is different in caps is DTMF. MGCP was trying 100 while SBC wanted to
> do 101.
>
> As for the garble: I wasn't experiencing any voice quality issues that I
> could hear, but I was experiencing double DTMF going out to the PSTN. Not
> sure if an artifact of the MTP, or simply a misonconfiguration on the
> VG224's MGCP package. Like I said it's the fm package I was missing that
> ultimately fixed the issue. The MTP is no longer used, and the double DTMF
> is gone. I didn't find very much info on what the fm packages does, only
> that it fixes DTMF and Faxing issues when communicating with a SIP device.
>
> Thanks for the late Friday afternoon reply Wes.
>
> On Fri, Jan 11, 2013 at 2:48 PM, Wes Sisk <wsisk at cisco.com> wrote:
>>
>> Interesting observations.
>>
>> I am not aware of any changes around CM's software MTP only doing G.711.
>>
>> The packet capture shows RTP coming into(?) to the MTP. I do not see any
>> sign of anything egressing the MTP.
>>
>> ccm has internal logic that attempts to connect RTP streams even if codec
>> negotiation fails. This is controlled by a service parameter. You may be
>> seeing an artifact of this behavior where no codec was common but the
>> streams attempted to setup anyway. Streaming codecs to the MTP that it does
>> not support typically results in garble or silence on the egress leg.
>>
>> /wes
>>
>>
>> On Jan 11, 2013, at 3:11 PM, Anthony Holloway wrote:
>>
>> Hi All,
>>
>> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
>> receiving a g729 audio stream from my VG224.
>>
>> Long story short, according to the CUCM SRND, the CUCM MTP can only
>> terminate g711, and yet, attached is a screenshot of the wireshark capture
>> which clearly shows it terminating g729.
>>
>> What piece of this puzzle am I missing? Also, the CUCM traces read like
>> the MTP is being invoked on that CUCM. It's due to the lack of the fm
>> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a fun
>> time resolving that, as you can probably imagine.
>>
>> Thanks and Happy Friday!
>>
>>
>>
>> <cucm-mtp-g729-redacted.png><cucm-mtp-redacted.png><cucm-sub02b-
redacted.png>_______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From avholloway+cisco-voip at gmail.com Fri Jan 11 16:36:13 2013


From: avholloway+cisco-voip at gmail.com (Anthony Holloway)
Date: Fri, 11 Jan 2013 15:36:13 -0600
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <CAMa5Jw7OZ1-cQOo3KVdx56A9FdTDmOqVK9rzxy5+CHr8v8dQ9Q@mail.gmail.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>
<CACRCJOguqqYfJrY38mSG47qgmnkz=RYReZdLA2rTdpPYpuSz0A@mail.gmail.com>
<CAMa5Jw7OZ1-cQOo3KVdx56A9FdTDmOqVK9rzxy5+CHr8v8dQ9Q@mail.gmail.com>
Message-ID: <CACRCJOiRpHqNMHLNYgLEvTtKUunie7Y82HKvFfUW1zCbU-6c5g@mail.gmail.com>

Thanks Pete. I'll see if I can answer or reply to each of your questions
or points.

*"are you SURE this isn't just a device hearing g.729 hold music while
you've got or had the Duplex Streaming service parameter enabled?"*
No, this is a call from an analog device to a PSTN device, and the call is
well established and in progress with two way audio.

*"Do you have the skinny signalling to go with it showing what it was
specifically set up for use as?"*
I'm not sure I understand where the skinny signaling comes in. The VG224
is MGCP, the SBC is a SIP trunk, and the MTP is local to the CUCM. Could
you help me understand? I do have traces off the CUCM if that answers your
question.

*"Also, I beleive MGCP Endpoints have an initial "state" when they begin
call setup. i think not using g.729 actually entails a switch from 729 to
another codec, and perhaps a small delay is causing some packets to be
transmitted using g.729? maybe? that's a complete stretch but who knows =)"*
Again, this is an established call. I get the call setup, verify two way
audio, and then take the capture.

*"I don't think you're really goign to get an answer unless you can
recreate the issue"*
I can. I have the VG224 in my cubicle. Check out the adapter I'm using!
Photo attached. =)

*"and we can see traces."*


I can't upload traces to the list, but if this goes to a TAC case, I will
certainly give them up at that time.

*"you'll also want a packet capture of the registration of the media device"
*
This is a good idea. I'll give it a try and see what it reports.

*"you coudl be doing duplex audio while one of those is playing"*


I'm not sure what that means, or how that would even work, but you sound
excited. =)

*"anyway, can you reproduce it"*


Yes. I can reproduce it at will.

Thanks for asking the questions and commenting.

On Fri, Jan 11, 2013 at 3:24 PM, Peter Slow <peter.slow at gmail.com> wrote:

> PS> ..Also, are you SURE this isn't just a device hearing g.729 hold
> music while you've got or had the Duplex Streaming service parameter
> enabled? ...'Cause that would totally explain this packet capture. Do
> you have the skinny signalling to go with it showing what it was
> specifically set up for use as?
>
> Also, I beleive MGCP Endpoints have an initial "state" when they begin
> call setup. i think not using g.729 actually entails a switch from 729
> to another codec, and perhaps a small delay is causing some packets to
> be transmitted using g.729? maybe? that's a complete stretch but who
> knows =)
>
> I don't think you're really goign to get an answer unless you can
> recreate the issue and we can see traces. you'll also want a packet
> capture of the registration of the media device, so if you can make
> the MTP register to a different callmanager than what it's running on,
> using its CUCM group, we could take a look at what capabilities it was
> registering with and if it says it supports 729 now =) ..you'll wnat
> to look at the skinny registration of the MTP, ANN ooh, ANN
> announcements are in 7.29 also, i think? you coudl be doing duplex
> audio while one of those is playing =)....
>
> anyway, can you reproduce it or verify or deny any of those guesses?
>
> Very Interesting,
> -Pete
>
>
>
> On Fri, Jan 11, 2013 at 4:08 PM, Anthony Holloway
> <avholloway+cisco-voip at gmail.com> wrote:
> >
> > Hey Wes,
> >
> > The packet capture was done on the CUCM itself via CLI command: "utils
> > network capture". Also, I filtered the capture to traffic only coming
> from
> > the VG224, which is why you do not see any other streams. It was,
> however,
> > going to our SBC. So the call flow was: Analog Phone > VG224 > CUCM
> (MTP) >
> > SBC > PSTN.
> >
> > The negotiated CODEC was in fact g729, and both sides support it. The
> MGCP
> > SDP shows g729 and the SBC sends back g729 in the SIP SDP. The only
> thing
> > that is different in caps is DTMF. MGCP was trying 100 while SBC wanted
> to
> > do 101.
> >
> > As for the garble: I wasn't experiencing any voice quality issues that I
> > could hear, but I was experiencing double DTMF going out to the PSTN.
> Not
> > sure if an artifact of the MTP, or simply a misonconfiguration on the
> > VG224's MGCP package. Like I said it's the fm package I was missing that
> > ultimately fixed the issue. The MTP is no longer used, and the double
> DTMF
> > is gone. I didn't find very much info on what the fm packages does, only
> > that it fixes DTMF and Faxing issues when communicating with a SIP
> device.
> >
> > Thanks for the late Friday afternoon reply Wes.
> >
> > On Fri, Jan 11, 2013 at 2:48 PM, Wes Sisk <wsisk at cisco.com> wrote:
> >>
> >> Interesting observations.
> >>
> >> I am not aware of any changes around CM's software MTP only doing G.711.
> >>
> >> The packet capture shows RTP coming into(?) to the MTP. I do not see any
> >> sign of anything egressing the MTP.
> >>
> >> ccm has internal logic that attempts to connect RTP streams even if
> codec
> >> negotiation fails. This is controlled by a service parameter. You may be
> >> seeing an artifact of this behavior where no codec was common but the
> >> streams attempted to setup anyway. Streaming codecs to the MTP that it
> does
> >> not support typically results in garble or silence on the egress leg.
> >>
> >> /wes
> >>
> >>
> >> On Jan 11, 2013, at 3:11 PM, Anthony Holloway wrote:
> >>
> >> Hi All,
> >>
> >> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
> >> receiving a g729 audio stream from my VG224.
> >>
> >> Long story short, according to the CUCM SRND, the CUCM MTP can only
> >> terminate g711, and yet, attached is a screenshot of the wireshark
> capture
> >> which clearly shows it terminating g729.
> >>
> >> What piece of this puzzle am I missing? Also, the CUCM traces read like
> >> the MTP is being invoked on that CUCM. It's due to the lack of the fm
> >> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a
> fun
> >> time resolving that, as you can probably imagine.
> >>
> >> Thanks and Happy Friday!
> >>
> >>
> >>
> >>
> <cucm-mtp-g729-redacted.png><cucm-mtp-redacted.png><cucm-sub02b-
redacted.png>_______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
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From peter.slow at gmail.com Fri Jan 11 16:48:37 2013


From: peter.slow at gmail.com (Peter Slow)
Date: Fri, 11 Jan 2013 16:48:37 -0500
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <CACRCJOiRpHqNMHLNYgLEvTtKUunie7Y82HKvFfUW1zCbU-6c5g@mail.gmail.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>
<CACRCJOguqqYfJrY38mSG47qgmnkz=RYReZdLA2rTdpPYpuSz0A@mail.gmail.com>
<CAMa5Jw7OZ1-cQOo3KVdx56A9FdTDmOqVK9rzxy5+CHr8v8dQ9Q@mail.gmail.com>
<CACRCJOiRpHqNMHLNYgLEvTtKUunie7Y82HKvFfUW1zCbU-6c5g@mail.gmail.com>
Message-ID: <CAMa5Jw6ZJqrba8Hp9OgxcZtsmpXv6W13Pm6-QV99=WsKD3Zs4Q@mail.gmail.com>

the skinny signalling is between the MTP and the CUCM - thats how it
knows what RTP streams to send and expect and which ones to tie with
which. you can tell a CUCM MTP on server A to register with CUCM on
server B - this forces that MTPS skinny signalling over the wire so
you can get a packet capture of it. use the CM group on the device
pool of the MTP, ANN conference or MoH device to do this - it will
work for all of those types of endpoints. seeing the skinny signalling
in your packet catpure woudl be a very easy way of proving
specifically what service that RTP is destined for.

easier than looking at traces anyway =)

...this is also how you'd get a packet capture of the registration of


those devices. that would allow you to see which ones were reporting
729 support.

there HAVE been a couple instances here and there where CUCM
mistakenly signals one device to use a codec that the other device
doesnt support, but thats typically a very very complex call flow and
probably not what's happening here. There is liekly a more simple
explanation =)

-Pete

On Fri, Jan 11, 2013 at 4:36 PM, Anthony Holloway


<avholloway+cisco-voip at gmail.com> wrote:
> Thanks Pete. I'll see if I can answer or reply to each of your questions or
> points.
>
>
> "are you SURE this isn't just a device hearing g.729 hold music while you've
> got or had the Duplex Streaming service parameter enabled?"
> No, this is a call from an analog device to a PSTN device, and the call is
> well established and in progress with two way audio.
>
>
> "Do you have the skinny signalling to go with it showing what it was
> specifically set up for use as?"
> I'm not sure I understand where the skinny signaling comes in. The VG224 is
> MGCP, the SBC is a SIP trunk, and the MTP is local to the CUCM. Could you
> help me understand? I do have traces off the CUCM if that answers your
> question.
>
>
> "Also, I beleive MGCP Endpoints have an initial "state" when they begin call
> setup. i think not using g.729 actually entails a switch from 729 to another
> codec, and perhaps a small delay is causing some packets to be transmitted
> using g.729? maybe? that's a complete stretch but who knows =)"
> Again, this is an established call. I get the call setup, verify two way
> audio, and then take the capture.
>
>
> "I don't think you're really goign to get an answer unless you can recreate
> the issue"
> I can. I have the VG224 in my cubicle. Check out the adapter I'm using!
> Photo attached. =)
>
>
> "and we can see traces."
> I can't upload traces to the list, but if this goes to a TAC case, I will
> certainly give them up at that time.
>
>
> "you'll also want a packet capture of the registration of the media device"
> This is a good idea. I'll give it a try and see what it reports.
>
>
> "you coudl be doing duplex audio while one of those is playing"
> I'm not sure what that means, or how that would even work, but you sound
> excited. =)
>
>
> "anyway, can you reproduce it"
> Yes. I can reproduce it at will.
>
> Thanks for asking the questions and commenting.
>
>
> On Fri, Jan 11, 2013 at 3:24 PM, Peter Slow <peter.slow at gmail.com> wrote:
>>
>> PS> ..Also, are you SURE this isn't just a device hearing g.729 hold
>> music while you've got or had the Duplex Streaming service parameter
>> enabled? ...'Cause that would totally explain this packet capture. Do
>> you have the skinny signalling to go with it showing what it was
>> specifically set up for use as?
>>
>> Also, I beleive MGCP Endpoints have an initial "state" when they begin
>> call setup. i think not using g.729 actually entails a switch from 729
>> to another codec, and perhaps a small delay is causing some packets to
>> be transmitted using g.729? maybe? that's a complete stretch but who
>> knows =)
>>
>> I don't think you're really goign to get an answer unless you can
>> recreate the issue and we can see traces. you'll also want a packet
>> capture of the registration of the media device, so if you can make
>> the MTP register to a different callmanager than what it's running on,
>> using its CUCM group, we could take a look at what capabilities it was
>> registering with and if it says it supports 729 now =) ..you'll wnat
>> to look at the skinny registration of the MTP, ANN ooh, ANN
>> announcements are in 7.29 also, i think? you coudl be doing duplex
>> audio while one of those is playing =)....
>>
>> anyway, can you reproduce it or verify or deny any of those guesses?
>>
>> Very Interesting,
>> -Pete
>>
>>
>>
>> On Fri, Jan 11, 2013 at 4:08 PM, Anthony Holloway
>> <avholloway+cisco-voip at gmail.com> wrote:
>> >
>> > Hey Wes,
>> >
>> > The packet capture was done on the CUCM itself via CLI command: "utils
>> > network capture". Also, I filtered the capture to traffic only coming
>> > from
>> > the VG224, which is why you do not see any other streams. It was,
>> > however,
>> > going to our SBC. So the call flow was: Analog Phone > VG224 > CUCM
>> > (MTP) >
>> > SBC > PSTN.
>> >
>> > The negotiated CODEC was in fact g729, and both sides support it. The
>> > MGCP
>> > SDP shows g729 and the SBC sends back g729 in the SIP SDP. The only
>> > thing
>> > that is different in caps is DTMF. MGCP was trying 100 while SBC wanted
>> > to
>> > do 101.
>> >
>> > As for the garble: I wasn't experiencing any voice quality issues that I
>> > could hear, but I was experiencing double DTMF going out to the PSTN.
>> > Not
>> > sure if an artifact of the MTP, or simply a misonconfiguration on the
>> > VG224's MGCP package. Like I said it's the fm package I was missing
>> > that
>> > ultimately fixed the issue. The MTP is no longer used, and the double
>> > DTMF
>> > is gone. I didn't find very much info on what the fm packages does, only
>> > that it fixes DTMF and Faxing issues when communicating with a SIP
>> > device.
>> >
>> > Thanks for the late Friday afternoon reply Wes.
>> >
>> > On Fri, Jan 11, 2013 at 2:48 PM, Wes Sisk <wsisk at cisco.com> wrote:
>> >>
>> >> Interesting observations.
>> >>
>> >> I am not aware of any changes around CM's software MTP only doing
>> >> G.711.
>> >>
>> >> The packet capture shows RTP coming into(?) to the MTP. I do not see
>> >> any
>> >> sign of anything egressing the MTP.
>> >>
>> >> ccm has internal logic that attempts to connect RTP streams even if
>> >> codec
>> >> negotiation fails. This is controlled by a service parameter. You may
>> >> be
>> >> seeing an artifact of this behavior where no codec was common but the
>> >> streams attempted to setup anyway. Streaming codecs to the MTP that it
>> >> does
>> >> not support typically results in garble or silence on the egress leg.
>> >>
>> >> /wes
>> >>
>> >>
>> >> On Jan 11, 2013, at 3:11 PM, Anthony Holloway wrote:
>> >>
>> >> Hi All,
>> >>
>> >> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
>> >> receiving a g729 audio stream from my VG224.
>> >>
>> >> Long story short, according to the CUCM SRND, the CUCM MTP can only
>> >> terminate g711, and yet, attached is a screenshot of the wireshark
>> >> capture
>> >> which clearly shows it terminating g729.
>> >>
>> >> What piece of this puzzle am I missing? Also, the CUCM traces read
>> >> like
>> >> the MTP is being invoked on that CUCM. It's due to the lack of the fm
>> >> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a
>> >> fun
>> >> time resolving that, as you can probably imagine.
>> >>
>> >> Thanks and Happy Friday!
>> >>
>> >>
>> >>
>> >>
>> >> <cucm-mtp-g729-redacted.png><cucm-mtp-redacted.png><cucm-sub02b-
redacted.png>_______________________________________________
>> >> cisco-voip mailing list
>> >> cisco-voip at puck.nether.net
>> >> https://puck.nether.net/mailman/listinfo/cisco-voip
>> >>
>> >
>> >
>> > _______________________________________________
>> > cisco-voip mailing list
>> > cisco-voip at puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >
>
>

From rratliff at cisco.com Fri Jan 11 17:02:35 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 11 Jan 2013 17:02:35 -0500
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <CACRCJOiRpHqNMHLNYgLEvTtKUunie7Y82HKvFfUW1zCbU-6c5g@mail.gmail.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>
<CACRCJOguqqYfJrY38mSG47qgmnkz=RYReZdLA2rTdpPYpuSz0A@mail.gmail.com>
<CAMa5Jw7OZ1-cQOo3KVdx56A9FdTDmOqVK9rzxy5+CHr8v8dQ9Q@mail.gmail.com>
<CACRCJOiRpHqNMHLNYgLEvTtKUunie7Y82HKvFfUW1zCbU-6c5g@mail.gmail.com>
Message-ID: <A287F450-2809-4BE6-A3D1-9584CC120CE1@cisco.com>

Check the capabilities the MTP advertises to CUCM when it registers. At some point
(8.5 maybe?) IP Voice Media Streaming App began supporting audio passthrough, which
would explain what you are seeing.

-Ryan

On Jan 11, 2013, at 4:36 PM, Anthony Holloway <avholloway+cisco-voip at gmail.com>


wrote:
Thanks Pete. I'll see if I can answer or reply to each of your questions or
points.

"are you SURE this isn't just a device hearing g.729 hold music while you've got or
had the Duplex Streaming service parameter enabled?"
No, this is a call from an analog device to a PSTN device, and the call is well
established and in progress with two way audio.

"Do you have the skinny signalling to go with it showing what it was specifically
set up for use as?"
I'm not sure I understand where the skinny signaling comes in. The VG224 is MGCP,
the SBC is a SIP trunk, and the MTP is local to the CUCM. Could you help me
understand? I do have traces off the CUCM if that answers your question.

"Also, I beleive MGCP Endpoints have an initial "state" when they begin call setup.
i think not using g.729 actually entails a switch from 729 to another codec, and
perhaps a small delay is causing some packets to be transmitted using g.729? maybe?
that's a complete stretch but who knows =)"
Again, this is an established call. I get the call setup, verify two way audio,
and then take the capture.

"I don't think you're really goign to get an answer unless you can recreate the
issue"
I can. I have the VG224 in my cubicle. Check out the adapter I'm using! Photo
attached. =)

"and we can see traces."


I can't upload traces to the list, but if this goes to a TAC case, I will certainly
give them up at that time.

"you'll also want a packet capture of the registration of the media device"
This is a good idea. I'll give it a try and see what it reports.

"you coudl be doing duplex audio while one of those is playing"


I'm not sure what that means, or how that would even work, but you sound excited.
=)

"anyway, can you reproduce it"


Yes. I can reproduce it at will.

Thanks for asking the questions and commenting.

On Fri, Jan 11, 2013 at 3:24 PM, Peter Slow <peter.slow at gmail.com> wrote:
PS> ..Also, are you SURE this isn't just a device hearing g.729 hold
music while you've got or had the Duplex Streaming service parameter
enabled? ...'Cause that would totally explain this packet capture. Do
you have the skinny signalling to go with it showing what it was
specifically set up for use as?

Also, I beleive MGCP Endpoints have an initial "state" when they begin
call setup. i think not using g.729 actually entails a switch from 729
to another codec, and perhaps a small delay is causing some packets to
be transmitted using g.729? maybe? that's a complete stretch but who
knows =)

I don't think you're really goign to get an answer unless you can
recreate the issue and we can see traces. you'll also want a packet
capture of the registration of the media device, so if you can make
the MTP register to a different callmanager than what it's running on,
using its CUCM group, we could take a look at what capabilities it was
registering with and if it says it supports 729 now =) ..you'll wnat
to look at the skinny registration of the MTP, ANN ooh, ANN
announcements are in 7.29 also, i think? you coudl be doing duplex
audio while one of those is playing =)....

anyway, can you reproduce it or verify or deny any of those guesses?

Very Interesting,
-Pete

On Fri, Jan 11, 2013 at 4:08 PM, Anthony Holloway


<avholloway+cisco-voip at gmail.com> wrote:
>
> Hey Wes,
>
> The packet capture was done on the CUCM itself via CLI command: "utils
> network capture". Also, I filtered the capture to traffic only coming from
> the VG224, which is why you do not see any other streams. It was, however,
> going to our SBC. So the call flow was: Analog Phone > VG224 > CUCM (MTP) >
> SBC > PSTN.
>
> The negotiated CODEC was in fact g729, and both sides support it. The MGCP
> SDP shows g729 and the SBC sends back g729 in the SIP SDP. The only thing
> that is different in caps is DTMF. MGCP was trying 100 while SBC wanted to
> do 101.
>
> As for the garble: I wasn't experiencing any voice quality issues that I
> could hear, but I was experiencing double DTMF going out to the PSTN. Not
> sure if an artifact of the MTP, or simply a misonconfiguration on the
> VG224's MGCP package. Like I said it's the fm package I was missing that
> ultimately fixed the issue. The MTP is no longer used, and the double DTMF
> is gone. I didn't find very much info on what the fm packages does, only
> that it fixes DTMF and Faxing issues when communicating with a SIP device.
>
> Thanks for the late Friday afternoon reply Wes.
>
> On Fri, Jan 11, 2013 at 2:48 PM, Wes Sisk <wsisk at cisco.com> wrote:
>>
>> Interesting observations.
>>
>> I am not aware of any changes around CM's software MTP only doing G.711.
>>
>> The packet capture shows RTP coming into(?) to the MTP. I do not see any
>> sign of anything egressing the MTP.
>>
>> ccm has internal logic that attempts to connect RTP streams even if codec
>> negotiation fails. This is controlled by a service parameter. You may be
>> seeing an artifact of this behavior where no codec was common but the
>> streams attempted to setup anyway. Streaming codecs to the MTP that it does
>> not support typically results in garble or silence on the egress leg.
>>
>> /wes
>>
>>
>> On Jan 11, 2013, at 3:11 PM, Anthony Holloway wrote:
>>
>> Hi All,
>>
>> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
>> receiving a g729 audio stream from my VG224.
>>
>> Long story short, according to the CUCM SRND, the CUCM MTP can only
>> terminate g711, and yet, attached is a screenshot of the wireshark capture
>> which clearly shows it terminating g729.
>>
>> What piece of this puzzle am I missing? Also, the CUCM traces read like
>> the MTP is being invoked on that CUCM. It's due to the lack of the fm
>> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a fun
>> time resolving that, as you can probably imagine.
>>
>> Thanks and Happy Friday!
>>
>>
>>
>> <cucm-mtp-g729-redacted.png><cucm-mtp-redacted.png><cucm-sub02b-
redacted.png>_______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From dzhars at gmail.com Fri Jan 11 19:49:08 2013


From: dzhars at gmail.com (David Zhars)
Date: Fri, 11 Jan 2013 19:49:08 -0500
Subject: [cisco-voip] FWD one Ext to Another
In-Reply-To: <253075E8-E4F6-4CA9-B9D8-248D96CC345C@cisco.com>
References: <CADe=jTGzpoYcj4FjwqfMWs+5x9mw1Xtg7BSShnCA=WeAyhYYjQ@mail.gmail.com>
<CC5F2DCE-8E8D-46B8-AAB5-FAA32CAB83A4@cisco.com>
<CADe=jTHy5Po+7wYUFF1bRo7d1xpOxh-HvwuWyP7bpfCGN3nwFQ@mail.gmail.com>
<253075E8-E4F6-4CA9-B9D8-248D96CC345C@cisco.com>
Message-ID: <CADe=jTF5qu4XvN-NRvYKa_OHQNXcXDuWeqD4Z5-9E9b_TE_0Mg@mail.gmail.com>

So my confusion has to extend from not having a solid understanding of an


"alternate extension" in Unity. I assumed (there's that word) that
applying 1212 as an alternate ext to 1702, would mean the user would STILL
have to login SEPARATELY to both VM boxes to get messages. Are you saying
that if I add 1212 as an alternate, he needs only check messages for ext
1702, and anything that went to 1212 would be in the 1702 box?? Cause that
would be way easy!

On Fri, Jan 11, 2013 at 4:21 PM, Ryan Ratliff <rratliff at cisco.com> wrote:

> A translation pattern is one option, the other is a DN for 1212 (not
> assigned to a phone) with CFA set to 1702. You'll want to test your usual
> call flows so that the fact that every call to 1702 will be a forwarded
> calls. For example if you have any calling party selections on outbound
> gateways set to 'first redirecting number' you may end up seeing 1212
> instead of 1702 as the calling party number. Calls that end up in
> voicemail will also be sent to the mailbox of 1212.
>
> The routing rule in Unity would basically be: Any redirected call with an
> original called party number of 1212, send it to standard greeting for
> 1702. I bet you could also just ad 1212 as an alternate extension for
> 1702 and you'll be set.
>
> -Ryan
>
> On Jan 11, 2013, at 2:51 PM, David Zhars <dzhars at gmail.com> wrote:
>
> OK, here's my dilemma. 1212 doesn't exist as far as UCM is concerned. I
> suppose I could recreate it as a CTI Route point??
>
> Ryan, not sure what you mean about Unity sending calls from 1212 to 1702.
>
> Sounds like a translation pattern in UCM may be the way to go, delete the
> 1212 mailbox in Unity, so if someone does try and transfer a call to that
> VM it would error out.
>
> While I want this to be easy for the end users, I also want it to be easy
> for me!
>
>
> On Fri, Jan 11, 2013 at 2:28 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>
>> Unity call routing rule to send calls from 1212 to 1702, should be pretty
>> straight forward.
>> -Ryan
>>
>> On Jan 11, 2013, at 1:32 PM, David Zhars <dzhars at gmail.com> wrote:
>>
>> Old user had ext 1212 (and this is a DID, so people can call directly
>> from the outside).
>> New user has ext 1702.
>>
>> What I want is:
>>
>> Internally: User dials 1212, phone rings at 1702.
>> Internally: Reception takes a call, transfers it with TRANS **1212 TRANS,
>> call goes to 1702 voicemail.
>>
>> Externally: Someone calls 555-1212 and the call lands internally at 1702.
>>
>> Some of this I know how to do, I am not sure about the transfer to
>> voicemail of the old extension and have it land at the new ext VM.
>>
>> Appreciate any help!
>>
>> Dave
>>
>> UCM 8.0, Unity 8.0
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
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voip/attachments/20130111/23ae8f45/attachment.html>

From george.hendrix at l-3com.com Sat Jan 12 07:45:58 2013


From: george.hendrix at l-3com.com (george.hendrix at l-3com.com)
Date: Sat, 12 Jan 2013 12:45:58 +0000
Subject: [cisco-voip] CUC Cluster replication issues
Message-ID: <255F57BB43F894468DA6BD4F2F5DD1CE6490DEF1@RSTN-S-MBX01.net.its.l-
3com.com>

Hey Guys,

Noticed we were having Unity Connection cluster issues when I logged. I went
ahead and restarted the Pub first and then the Sub as this was the fix that TAC
gave me last time we had a similar issue. However, it still having issues. Please
see output below from each server.

Publisher:
admin:show cuc cluster status

Server Name Member ID Server State Internal State


Reason
--------------------- --------- ------------ ----------------------------
------
rstn-2fs-cuc-pub01 0 Primary Pri Active
Normal
annjct-2720-cuc-sub01 1 Primary Sec Act Primary Disconnected
Normal

Database replication is not active

admin:

Subscriber:
admin:show cuc cluster status

Server Name Member ID Server State Internal State Reason


--------------------- --------- ---------------------- -------------- ------
rstn-2fs-cuc-pub01 0 Split Brain Resolution Pri SBR Normal
annjct-2720-cuc-sub01 1 Secondary Sec Active Normal
SERVER ID STATE STATUS QUEUE CONNECTION CHANGED
-----------------------------------------------------------------------
g_ciscounity_pub 100 Active Dropped 568373 Jan 12 07:38:42
g_ciscounity_sub1 101 Active Local 0

admin:

Any ideas of what to do next?

Bill

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From jason.aarons at dimensiondata.com Sat Jan 12 12:45:23 2013


From: jason.aarons at dimensiondata.com (Jason Aarons (AM))
Date: Sat, 12 Jan 2013 12:45:23 -0500
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <A287F450-2809-4BE6-A3D1-9584CC120CE1@cisco.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>
<CACRCJOguqqYfJrY38mSG47qgmnkz=RYReZdLA2rTdpPYpuSz0A@mail.gmail.com>
<CAMa5Jw7OZ1-cQOo3KVdx56A9FdTDmOqVK9rzxy5+CHr8v8dQ9Q@mail.gmail.com>
<CACRCJOiRpHqNMHLNYgLEvTtKUunie7Y82HKvFfUW1zCbU-6c5g@mail.gmail.com>
<A287F450-2809-4BE6-A3D1-9584CC120CE1@cisco.com>
Message-ID:
<4E38DB0A1959B04C8C83EDCF069B53ED0D2C718859@USISPCLEXDB01.na.didata.local>

I got a customer running 8.5.1SU2 and it's not doing IP Voice Media Streaming App
MTP with Pass-Thru. Just had a big TAC case with T38 and took awhile for the TAC
lead to come to that conclusion. A IOS Software MTP was needed to fix it.

I've looked previously and haven't found any details about fix/improvement (eg
what's new) around IP Voice Media Streaming App in newer versions.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Friday, January 11, 2013 5:03 PM
To: Anthony Holloway
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] CUCM MTP and g729

Check the capabilities the MTP advertises to CUCM when it registers. At some point
(8.5 maybe?) IP Voice Media Streaming App began supporting audio passthrough, which
would explain what you are seeing.

-Ryan

On Jan 11, 2013, at 4:36 PM, Anthony Holloway <avholloway+cisco-voip at


gmail.com<mailto:avholloway+cisco-voip at gmail.com>> wrote:

Thanks Pete. I'll see if I can answer or reply to each of your questions or
points.
"are you SURE this isn't just a device hearing g.729 hold music while you've got or
had the Duplex Streaming service parameter enabled?"
No, this is a call from an analog device to a PSTN device, and the call is well
established and in progress with two way audio.

"Do you have the skinny signalling to go with it showing what it was specifically
set up for use as?"
I'm not sure I understand where the skinny signaling comes in. The VG224 is MGCP,
the SBC is a SIP trunk, and the MTP is local to the CUCM. Could you help me
understand? I do have traces off the CUCM if that answers your question.

"Also, I beleive MGCP Endpoints have an initial "state" when they begin call setup.
i think not using g.729 actually entails a switch from 729 to another codec, and
perhaps a small delay is causing some packets to be transmitted using g.729? maybe?
that's a complete stretch but who knows =)"
Again, this is an established call. I get the call setup, verify two way audio,
and then take the capture.

"I don't think you're really goign to get an answer unless you can recreate the
issue"
I can. I have the VG224 in my cubicle. Check out the adapter I'm using! Photo
attached. =)

"and we can see traces."


I can't upload traces to the list, but if this goes to a TAC case, I will certainly
give them up at that time.

"you'll also want a packet capture of the registration of the media device"
This is a good idea. I'll give it a try and see what it reports.

"you coudl be doing duplex audio while one of those is playing"


I'm not sure what that means, or how that would even work, but you sound excited.
=)

"anyway, can you reproduce it"


Yes. I can reproduce it at will.
Thanks for asking the questions and commenting.

On Fri, Jan 11, 2013 at 3:24 PM, Peter Slow <peter.slow at


gmail.com<mailto:peter.slow at gmail.com>> wrote:
PS> ..Also, are you SURE this isn't just a device hearing g.729 hold
music while you've got or had the Duplex Streaming service parameter
enabled? ...'Cause that would totally explain this packet capture. Do
you have the skinny signalling to go with it showing what it was
specifically set up for use as?

Also, I beleive MGCP Endpoints have an initial "state" when they begin
call setup. i think not using g.729 actually entails a switch from 729
to another codec, and perhaps a small delay is causing some packets to
be transmitted using g.729? maybe? that's a complete stretch but who
knows =)

I don't think you're really goign to get an answer unless you can
recreate the issue and we can see traces. you'll also want a packet
capture of the registration of the media device, so if you can make
the MTP register to a different callmanager than what it's running on,
using its CUCM group, we could take a look at what capabilities it was
registering with and if it says it supports 729 now =) ..you'll wnat
to look at the skinny registration of the MTP, ANN ooh, ANN
announcements are in 7.29 also, i think? you coudl be doing duplex
audio while one of those is playing =)....

anyway, can you reproduce it or verify or deny any of those guesses?

Very Interesting,
-Pete

On Fri, Jan 11, 2013 at 4:08 PM, Anthony Holloway


<avholloway+cisco-voip at gmail.com<mailto:avholloway%2Bcisco-voip at gmail.com>>
wrote:
>
> Hey Wes,
>
> The packet capture was done on the CUCM itself via CLI command: "utils
> network capture". Also, I filtered the capture to traffic only coming from
> the VG224, which is why you do not see any other streams. It was, however,
> going to our SBC. So the call flow was: Analog Phone > VG224 > CUCM (MTP) >
> SBC > PSTN.
>
> The negotiated CODEC was in fact g729, and both sides support it. The MGCP
> SDP shows g729 and the SBC sends back g729 in the SIP SDP. The only thing
> that is different in caps is DTMF. MGCP was trying 100 while SBC wanted to
> do 101.
>
> As for the garble: I wasn't experiencing any voice quality issues that I
> could hear, but I was experiencing double DTMF going out to the PSTN. Not
> sure if an artifact of the MTP, or simply a misonconfiguration on the
> VG224's MGCP package. Like I said it's the fm package I was missing that
> ultimately fixed the issue. The MTP is no longer used, and the double DTMF
> is gone. I didn't find very much info on what the fm packages does, only
> that it fixes DTMF and Faxing issues when communicating with a SIP device.
>
> Thanks for the late Friday afternoon reply Wes.
>
> On Fri, Jan 11, 2013 at 2:48 PM, Wes Sisk <wsisk at cisco.com<mailto:wsisk at
cisco.com>> wrote:
>>
>> Interesting observations.
>>
>> I am not aware of any changes around CM's software MTP only doing G.711.
>>
>> The packet capture shows RTP coming into(?) to the MTP. I do not see any
>> sign of anything egressing the MTP.
>>
>> ccm has internal logic that attempts to connect RTP streams even if codec
>> negotiation fails. This is controlled by a service parameter. You may be
>> seeing an artifact of this behavior where no codec was common but the
>> streams attempted to setup anyway. Streaming codecs to the MTP that it does
>> not support typically results in garble or silence on the egress leg.
>>
>> /wes
>>
>>
>> On Jan 11, 2013, at 3:11 PM, Anthony Holloway wrote:
>>
>> Hi All,
>>
>> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
>> receiving a g729 audio stream from my VG224.
>>
>> Long story short, according to the CUCM SRND, the CUCM MTP can only
>> terminate g711, and yet, attached is a screenshot of the wireshark capture
>> which clearly shows it terminating g729.
>>
>> What piece of this puzzle am I missing? Also, the CUCM traces read like
>> the MTP is being invoked on that CUCM. It's due to the lack of the fm
>> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a fun
>> time resolving that, as you can probably imagine.
>>
>> Thanks and Happy Friday!
>>
>>
>>
>> <cucm-mtp-g729-redacted.png><cucm-mtp-redacted.png><cucm-sub02b-
redacted.png>_______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

itevomcid
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From george.hendrix at l-3com.com Sat Jan 12 15:10:27 2013


From: george.hendrix at l-3com.com (george.hendrix at l-3com.com)
Date: Sat, 12 Jan 2013 20:10:27 +0000
Subject: [cisco-voip] CUC Cluster replication issues
In-Reply-To: <549601cdf0f1$35256480$9f702d80$@gmail.com>
References: <255F57BB43F894468DA6BD4F2F5DD1CE6490DEF1@RSTN-S-MBX01.net.its.l-
3com.com>
<549601cdf0f1$35256480$9f702d80$@gmail.com>
Message-ID: <255F57BB43F894468DA6BD4F2F5DD1CE6490DF81@RSTN-S-MBX01.net.its.l-
3com.com>

Both are synch'd

PUB:
admin:utils ntp status
ntpd (pid 18803) is running...

remote refid st t when poll reach delay offset jitter


==============================================================================
*141.199.251.254 166.20.167.29 3 u - 1024 377 0.495 10.195 23.315
+141.199.251.253 166.20.167.29 3 u 114 1024 377 0.562 -11.931 23.057

synchronised to NTP server (141.199.251.254) at stratum 4


time correct to within 135 ms
polling server every 1024 s

Current time in UTC is : Sat Jan 12 20:06:23 UTC 2013


Current time in America/New_York is : Sat Jan 12 15:06:23 EST 2013

SUB:
admin:utils ntp status
ntpd (pid 18919) is running...

remote refid st t when poll reach delay offset jitter


==============================================================================
*172.20.208.7 141.199.251.254 4 u 577 1024 377 2.833 -1.293 0.037

synchronised to NTP server (172.20.208.7) at stratum 5


time correct to within 133 ms
polling server every 1024 s

Current time in UTC is : Sat Jan 12 20:07:17 UTC 2013


Current time in America/New_York is : Sat Jan 12 15:07:17 EST 2013

Bill

From: mike.lydick at gmail.com [mailto:mike.lydick at gmail.com]


Sent: Saturday, January 12, 2013 1:18 PM
To: Hendrix, George (Bill) @ NSS - STRATIS
Subject: RE: [cisco-voip] CUC Cluster replication issues

Check your NTP, high stratum or unsynchronized or insane NTP clocks can cause Unity
Connection Database replication to fail. Fix the ntp status and resolves the issue

Utils ntp status


Looking for ntp to have a peer and the state is synch'd

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
george.hendrix at l-3com.com<mailto:george.hendrix at l-3com.com>
Sent: Saturday, January 12, 2013 7:46 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] CUC Cluster replication issues

Hey Guys,

Noticed we were having Unity Connection cluster issues when I logged. I went
ahead and restarted the Pub first and then the Sub as this was the fix that TAC
gave me last time we had a similar issue. However, it still having issues. Please
see output below from each server.

Publisher:
admin:show cuc cluster status

Server Name Member ID Server State Internal State


Reason
--------------------- --------- ------------ ----------------------------
------
rstn-2fs-cuc-pub01 0 Primary Pri Active
Normal
annjct-2720-cuc-sub01 1 Primary Sec Act Primary Disconnected
Normal

Database replication is not active

admin:

Subscriber:
admin:show cuc cluster status

Server Name Member ID Server State Internal State Reason


--------------------- --------- ---------------------- -------------- ------
rstn-2fs-cuc-pub01 0 Split Brain Resolution Pri SBR Normal
annjct-2720-cuc-sub01 1 Secondary Sec Active Normal

SERVER ID STATE STATUS QUEUE CONNECTION CHANGED


-----------------------------------------------------------------------
g_ciscounity_pub 100 Active Dropped 568373 Jan 12 07:38:42
g_ciscounity_sub1 101 Active Local 0

admin:

Any ideas of what to do next?

Bill

-------------- next part --------------


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From mike.lydick at gmail.com Sat Jan 12 15:49:34 2013


From: mike.lydick at gmail.com (Mike Lydick)
Date: Sat, 12 Jan 2013 15:49:34 -0500
Subject: [cisco-voip] CUC Cluster replication issues
In-Reply-To: <255F57BB43F894468DA6BD4F2F5DD1CE6490DF81@RSTN-S-MBX01.net.its.l-
3com.com>
References: <255F57BB43F894468DA6BD4F2F5DD1CE6490DEF1@RSTN-S-MBX01.net.its.l-
3com.com>
<549601cdf0f1$35256480$9f702d80$@gmail.com>
<255F57BB43F894468DA6BD4F2F5DD1CE6490DF81@RSTN-S-MBX01.net.its.l-3com.com>
Message-ID: <227f901cdf106$54bcf5a0$fe36e0e0$@gmail.com>

Are the NTP server windows based NTP? I have had many issues using windows
NTP because the clocks slip. So the effect will be that NTP will slip in/out
of sync.
From: george.hendrix at l-3com.com [mailto:george.hendrix at l-3com.com]
Sent: Saturday, January 12, 2013 3:10 PM
To: mike.lydick at gmail.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CUC Cluster replication issues

Both are synch'd

PUB:
admin:utils ntp status

ntpd (pid 18803) is running...

remote refid st t when poll reach delay offset


jitter

============================================================================
==

*141.199.251.254 166.20.167.29 3 u - 1024 377 0.495 10.195


23.315

+141.199.251.253 166.20.167.29 3 u 114 1024 377 0.562 -11.931


23.057

synchronised to NTP server (141.199.251.254) at stratum 4

time correct to within 135 ms

polling server every 1024 s

Current time in UTC is : Sat Jan 12 20:06:23 UTC 2013

Current time in America/New_York is : Sat Jan 12 15:06:23 EST 2013

SUB:

admin:utils ntp status

ntpd (pid 18919) is running...


remote refid st t when poll reach delay offset
jitter

============================================================================
==

*172.20.208.7 141.199.251.254 4 u 577 1024 377 2.833 -1.293


0.037

synchronised to NTP server (172.20.208.7) at stratum 5

time correct to within 133 ms

polling server every 1024 s

Current time in UTC is : Sat Jan 12 20:07:17 UTC 2013

Current time in America/New_York is : Sat Jan 12 15:07:17 EST 2013

Bill

From: mike.lydick at gmail.com [mailto:mike.lydick at gmail.com]


Sent: Saturday, January 12, 2013 1:18 PM
To: Hendrix, George (Bill) @ NSS - STRATIS
Subject: RE: [cisco-voip] CUC Cluster replication issues

Check your NTP, high stratum or unsynchronized or insane NTP clocks can
cause Unity Connection Database replication to fail. Fix the ntp status and
resolves the issue

Utils ntp status

Looking for ntp to have a peer and the state is synch'd

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
george.hendrix at l-3com.com
Sent: Saturday, January 12, 2013 7:46 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CUC Cluster replication issues

Hey Guys,

Noticed we were having Unity Connection cluster issues when I logged. I


went ahead and restarted the Pub first and then the Sub as this was the fix
that TAC gave me last time we had a similar issue. However, it still having
issues. Please see output below from each server.

Publisher:

admin:show cuc cluster status

Server Name Member ID Server State Internal State


Reason

--------------------- --------- ------------ ----------------------------


------

rstn-2fs-cuc-pub01 0 Primary Pri Active


Normal

annjct-2720-cuc-sub01 1 Primary Sec Act Primary Disconnected


Normal

Database replication is not active

admin:

Subscriber:

admin:show cuc cluster status

Server Name Member ID Server State Internal State


Reason

--------------------- --------- ---------------------- --------------


------

rstn-2fs-cuc-pub01 0 Split Brain Resolution Pri SBR


Normal

annjct-2720-cuc-sub01 1 Secondary Sec Active


Normal

SERVER ID STATE STATUS QUEUE CONNECTION CHANGED

-----------------------------------------------------------------------

g_ciscounity_pub 100 Active Dropped 568373 Jan 12 07:38:42

g_ciscounity_sub1 101 Active Local 0

admin:

Any ideas of what to do next?

Bill

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From peter.slow at gmail.com Sat Jan 12 16:43:36 2013


From: peter.slow at gmail.com (Peter Slow)
Date: Sat, 12 Jan 2013 16:43:36 -0500
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To:
<4E38DB0A1959B04C8C83EDCF069B53ED0D2C718859@USISPCLEXDB01.na.didata.local>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>
<CACRCJOguqqYfJrY38mSG47qgmnkz=RYReZdLA2rTdpPYpuSz0A@mail.gmail.com>
<CAMa5Jw7OZ1-cQOo3KVdx56A9FdTDmOqVK9rzxy5+CHr8v8dQ9Q@mail.gmail.com>
<CACRCJOiRpHqNMHLNYgLEvTtKUunie7Y82HKvFfUW1zCbU-6c5g@mail.gmail.com>
<A287F450-2809-4BE6-A3D1-9584CC120CE1@cisco.com>
<4E38DB0A1959B04C8C83EDCF069B53ED0D2C718859@USISPCLEXDB01.na.didata.local>
Message-ID: <188F72A9-D5A4-448A-A2CE-059A5942833A@gmail.com>

Ryan, that's awesome, didn't know that.

Anthony, it looks like the pass through codec Ryan is talking about is registered
as codec number 258. Look for that in the capabilitiesresponse message.

Sent from my iPad


On Jan 12, 2013, at 12:45 PM, "Jason Aarons (AM)" <jason.aarons at
dimensiondata.com> wrote:

> I got a customer running 8.5.1SU2 and it?s not doing IP Voice Media Streaming App
MTP with Pass-Thru. Just had a big TAC case with T38 and took awhile for the TAC
lead to come to that conclusion. A IOS Software MTP was needed to fix it.
>
> I?ve looked previously and haven?t found any details about fix/improvement (eg
what?s new) around IP Voice Media Streaming App in newer versions.
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at
puck.nether.net] On Behalf Of Ryan Ratliff
> Sent: Friday, January 11, 2013 5:03 PM
> To: Anthony Holloway
> Cc: Cisco VoIP Group
> Subject: Re: [cisco-voip] CUCM MTP and g729
>
>
>
> Check the capabilities the MTP advertises to CUCM when it registers. At some
point (8.5 maybe?) IP Voice Media Streaming App began supporting audio passthrough,
which would explain what you are seeing.
>
> -Ryan
>
> On Jan 11, 2013, at 4:36 PM, Anthony Holloway <avholloway+cisco-voip at
gmail.com> wrote:
>
> Thanks Pete. I'll see if I can answer or reply to each of your questions or
points.
>
> "are you SURE this isn't just a device hearing g.729 hold music while you've got
or had the Duplex Streaming service parameter enabled?"
> No, this is a call from an analog device to a PSTN device, and the call is well
established and in progress with two way audio.
>
> "Do you have the skinny signalling to go with it showing what it was specifically
set up for use as?"
> I'm not sure I understand where the skinny signaling comes in. The VG224 is
MGCP, the SBC is a SIP trunk, and the MTP is local to the CUCM. Could you help me
understand? I do have traces off the CUCM if that answers your question.
>
> "Also, I beleive MGCP Endpoints have an initial "state" when they begin call
setup. i think not using g.729 actually entails a switch from 729 to another codec,
and perhaps a small delay is causing some packets to be transmitted using g.729?
maybe? that's a complete stretch but who knows =)"
> Again, this is an established call. I get the call setup, verify two way audio,
and then take the capture.
>
> "I don't think you're really goign to get an answer unless you can recreate the
issue"
> I can. I have the VG224 in my cubicle. Check out the adapter I'm using! Photo
attached. =)
>
> "and we can see traces."
> I can't upload traces to the list, but if this goes to a TAC case, I will
certainly give them up at that time.
>
> "you'll also want a packet capture of the registration of the media device"
> This is a good idea. I'll give it a try and see what it reports.
>
> "you coudl be doing duplex audio while one of those is playing"
> I'm not sure what that means, or how that would even work, but you sound excited.
=)
>
> "anyway, can you reproduce it"
> Yes. I can reproduce it at will.
>
> Thanks for asking the questions and commenting.
>
>
> On Fri, Jan 11, 2013 at 3:24 PM, Peter Slow <peter.slow at gmail.com> wrote:
> PS> ..Also, are you SURE this isn't just a device hearing g.729 hold
> music while you've got or had the Duplex Streaming service parameter
> enabled? ...'Cause that would totally explain this packet capture. Do
> you have the skinny signalling to go with it showing what it was
> specifically set up for use as?
>
> Also, I beleive MGCP Endpoints have an initial "state" when they begin
> call setup. i think not using g.729 actually entails a switch from 729
> to another codec, and perhaps a small delay is causing some packets to
> be transmitted using g.729? maybe? that's a complete stretch but who
> knows =)
>
> I don't think you're really goign to get an answer unless you can
> recreate the issue and we can see traces. you'll also want a packet
> capture of the registration of the media device, so if you can make
> the MTP register to a different callmanager than what it's running on,
> using its CUCM group, we could take a look at what capabilities it was
> registering with and if it says it supports 729 now =) ..you'll wnat
> to look at the skinny registration of the MTP, ANN ooh, ANN
> announcements are in 7.29 also, i think? you coudl be doing duplex
> audio while one of those is playing =)....
>
> anyway, can you reproduce it or verify or deny any of those guesses?
>
> Very Interesting,
> -Pete
>
>
>
> On Fri, Jan 11, 2013 at 4:08 PM, Anthony Holloway
> <avholloway+cisco-voip at gmail.com> wrote:
> >
> > Hey Wes,
> >
> > The packet capture was done on the CUCM itself via CLI command: "utils
> > network capture". Also, I filtered the capture to traffic only coming from
> > the VG224, which is why you do not see any other streams. It was, however,
> > going to our SBC. So the call flow was: Analog Phone > VG224 > CUCM (MTP) >
> > SBC > PSTN.
> >
> > The negotiated CODEC was in fact g729, and both sides support it. The MGCP
> > SDP shows g729 and the SBC sends back g729 in the SIP SDP. The only thing
> > that is different in caps is DTMF. MGCP was trying 100 while SBC wanted to
> > do 101.
> >
> > As for the garble: I wasn't experiencing any voice quality issues that I
> > could hear, but I was experiencing double DTMF going out to the PSTN. Not
> > sure if an artifact of the MTP, or simply a misonconfiguration on the
> > VG224's MGCP package. Like I said it's the fm package I was missing that
> > ultimately fixed the issue. The MTP is no longer used, and the double DTMF
> > is gone. I didn't find very much info on what the fm packages does, only
> > that it fixes DTMF and Faxing issues when communicating with a SIP device.
> >
> > Thanks for the late Friday afternoon reply Wes.
> >
> > On Fri, Jan 11, 2013 at 2:48 PM, Wes Sisk <wsisk at cisco.com> wrote:
> >>
> >> Interesting observations.
> >>
> >> I am not aware of any changes around CM's software MTP only doing G.711.
> >>
> >> The packet capture shows RTP coming into(?) to the MTP. I do not see any
> >> sign of anything egressing the MTP.
> >>
> >> ccm has internal logic that attempts to connect RTP streams even if codec
> >> negotiation fails. This is controlled by a service parameter. You may be
> >> seeing an artifact of this behavior where no codec was common but the
> >> streams attempted to setup anyway. Streaming codecs to the MTP that it does
> >> not support typically results in garble or silence on the egress leg.
> >>
> >> /wes
> >>
> >>
> >> On Jan 11, 2013, at 3:11 PM, Anthony Holloway wrote:
> >>
> >> Hi All,
> >>
> >> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
> >> receiving a g729 audio stream from my VG224.
> >>
> >> Long story short, according to the CUCM SRND, the CUCM MTP can only
> >> terminate g711, and yet, attached is a screenshot of the wireshark capture
> >> which clearly shows it terminating g729.
> >>
> >> What piece of this puzzle am I missing? Also, the CUCM traces read like
> >> the MTP is being invoked on that CUCM. It's due to the lack of the fm
> >> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a fun
> >> time resolving that, as you can probably imagine.
> >>
> >> Thanks and Happy Friday!
> >>
> >>
> >>
> >> <cucm-mtp-g729-redacted.png><cucm-mtp-redacted.png><cucm-sub02b-
redacted.png>_______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> itevomcid
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From erickbee at gmail.com Sun Jan 13 00:22:27 2013


From: erickbee at gmail.com (Erick B.)
Date: Sat, 12 Jan 2013 23:22:27 -0600
Subject: [cisco-voip] FWD one Ext to Another
In-Reply-To: <CADe=jTF5qu4XvN-NRvYKa_OHQNXcXDuWeqD4Z5-9E9b_TE_0Mg@mail.gmail.com>
References: <CADe=jTGzpoYcj4FjwqfMWs+5x9mw1Xtg7BSShnCA=WeAyhYYjQ@mail.gmail.com>
<CC5F2DCE-8E8D-46B8-AAB5-FAA32CAB83A4@cisco.com>
<CADe=jTHy5Po+7wYUFF1bRo7d1xpOxh-HvwuWyP7bpfCGN3nwFQ@mail.gmail.com>
<253075E8-E4F6-4CA9-B9D8-248D96CC345C@cisco.com>
<CADe=jTF5qu4XvN-NRvYKa_OHQNXcXDuWeqD4Z5-9E9b_TE_0Mg@mail.gmail.com>
Message-ID: <CAHSnBQyyhz7b=LHzHtan_a0HfCiyt73GOvZgWnHnmxnq8Cg0zg@mail.gmail.com>

Yes, in unity add 1212 as alternate extension to the 1702 users mailbox and
any calls for 1702 or 1212 will go to that users mailbox. The ** transfer
method will still work with this to (assuming the ** pattern sends call
those calls to voicemail directly on your setup)

I would use a translation pattern on CUCM to route calls from 1212 to 1702
unless you need to keep 1212 on a phone/etc for some reason.

On Fri, Jan 11, 2013 at 6:49 PM, David Zhars <dzhars at gmail.com> wrote:

> So my confusion has to extend from not having a solid understanding of an


> "alternate extension" in Unity. I assumed (there's that word) that
> applying 1212 as an alternate ext to 1702, would mean the user would STILL
> have to login SEPARATELY to both VM boxes to get messages. Are you saying
> that if I add 1212 as an alternate, he needs only check messages for ext
> 1702, and anything that went to 1212 would be in the 1702 box?? Cause that
> would be way easy!
>
>
>
> On Fri, Jan 11, 2013 at 4:21 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>
>> A translation pattern is one option, the other is a DN for 1212 (not
>> assigned to a phone) with CFA set to 1702. You'll want to test your usual
>> call flows so that the fact that every call to 1702 will be a forwarded
>> calls. For example if you have any calling party selections on outbound
>> gateways set to 'first redirecting number' you may end up seeing 1212
>> instead of 1702 as the calling party number. Calls that end up in
>> voicemail will also be sent to the mailbox of 1212.
>>
>> The routing rule in Unity would basically be: Any redirected call with an
>> original called party number of 1212, send it to standard greeting for
>> 1702. I bet you could also just ad 1212 as an alternate extension for
>> 1702 and you'll be set.
>>
>> -Ryan
>>
>> On Jan 11, 2013, at 2:51 PM, David Zhars <dzhars at gmail.com> wrote:
>>
>> OK, here's my dilemma. 1212 doesn't exist as far as UCM is concerned. I
>> suppose I could recreate it as a CTI Route point??
>>
>> Ryan, not sure what you mean about Unity sending calls from 1212 to
>> 1702.
>>
>> Sounds like a translation pattern in UCM may be the way to go, delete the
>> 1212 mailbox in Unity, so if someone does try and transfer a call to that
>> VM it would error out.
>>
>> While I want this to be easy for the end users, I also want it to be easy
>> for me!
>>
>>
>> On Fri, Jan 11, 2013 at 2:28 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>>
>>> Unity call routing rule to send calls from 1212 to 1702, should be
>>> pretty straight forward.
>>> -Ryan
>>>
>>> On Jan 11, 2013, at 1:32 PM, David Zhars <dzhars at gmail.com> wrote:
>>>
>>> Old user had ext 1212 (and this is a DID, so people can call directly
>>> from the outside).
>>> New user has ext 1702.
>>>
>>> What I want is:
>>>
>>> Internally: User dials 1212, phone rings at 1702.
>>> Internally: Reception takes a call, transfers it with TRANS **1212
>>> TRANS, call goes to 1702 voicemail.
>>>
>>> Externally: Someone calls 555-1212 and the call lands internally at 1702.
>>>
>>> Some of this I know how to do, I am not sure about the transfer to
>>> voicemail of the old extension and have it land at the new ext VM.
>>>
>>> Appreciate any help!
>>>
>>> Dave
>>>
>>> UCM 8.0, Unity 8.0
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From MLoraditch at heliontechnologies.com Sun Jan 13 08:58:28 2013


From: MLoraditch at heliontechnologies.com (Matthew Loraditch)
Date: Sun, 13 Jan 2013 13:58:28 +0000
Subject: [cisco-voip] 9.1 Upgrade Times
Message-ID: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>

On a Unity Connection system and CUCM both 8.6.2aSU1 started at 11:45PM and 12:12AM
Last night, both are STILL running at 8:45 AM this morning. The system I am doing
this test on has about 60 Phones/Users/VM. These are the publishers of each
install, but I have never had an upgrade take this long, ever.

I'm now not sure I'll even be able to finish in my window since I haven't even
touched the subscribers yet.

Any clues as to why this is taking so long? I used to disable iothrottle during an
upgrade but that command is deprecated now.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

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From MLoraditch at heliontechnologies.com Sun Jan 13 15:06:25 2013


From: MLoraditch at heliontechnologies.com (Matthew Loraditch)
Date: Sun, 13 Jan 2013 20:06:25 +0000
Subject: [cisco-voip] 9.1 Upgrade Times
In-Reply-To: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>
References: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>
Message-ID: <C75AF2AD9308C246AFBDDB994E3E2983110B2E69@PHANES.helion.local>

Well my CUCM publisher took almost 13 hours, my sub took 1 hour and change... the
unity connection pub is still running, I think we are on hour 15 of that...

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Matthew Loraditch
Sent: Sunday, January 13, 2013 8:58 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 9.1 Upgrade Times

On a Unity Connection system and CUCM both 8.6.2aSU1 started at 11:45PM and 12:12AM
Last night, both are STILL running at 8:45 AM this morning. The system I am doing
this test on has about 60 Phones/Users/VM. These are the publishers of each
install, but I have never had an upgrade take this long, ever.

I'm now not sure I'll even be able to finish in my window since I haven't even
touched the subscribers yet.

Any clues as to why this is taking so long? I used to disable iothrottle during an
upgrade but that command is deprecated now.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

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From rratliff at cisco.com Mon Jan 14 11:06:07 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 14 Jan 2013 11:06:07 -0500
Subject: [cisco-voip] FWD one Ext to Another
In-Reply-To: <CAHSnBQyyhz7b=LHzHtan_a0HfCiyt73GOvZgWnHnmxnq8Cg0zg@mail.gmail.com>
References: <CADe=jTGzpoYcj4FjwqfMWs+5x9mw1Xtg7BSShnCA=WeAyhYYjQ@mail.gmail.com>
<CC5F2DCE-8E8D-46B8-AAB5-FAA32CAB83A4@cisco.com>
<CADe=jTHy5Po+7wYUFF1bRo7d1xpOxh-HvwuWyP7bpfCGN3nwFQ@mail.gmail.com>
<253075E8-E4F6-4CA9-B9D8-248D96CC345C@cisco.com>
<CADe=jTF5qu4XvN-NRvYKa_OHQNXcXDuWeqD4Z5-9E9b_TE_0Mg@mail.gmail.com>
<CAHSnBQyyhz7b=LHzHtan_a0HfCiyt73GOvZgWnHnmxnq8Cg0zg@mail.gmail.com>
Message-ID: <3F1A3BD3-9CA1-4600-8C60-4510D2DD8959@cisco.com>

> I assumed (there's that word) that applying 1212 as an alternate ext to 1702,
would mean the user would STILL have to login SEPARATELY to both VM boxes to get
messages.

The use of an alternate extension is for the case when there is only one mailbox in
Unity, with two extensions that could be calling into it (forwarded or not).

-Ryan

On Jan 13, 2013, at 12:22 AM, Erick B. <erickbee at gmail.com> wrote:

Yes, in unity add 1212 as alternate extension to the 1702 users mailbox and any
calls for 1702 or 1212 will go to that users mailbox. The ** transfer method will
still work with this to (assuming the ** pattern sends call those calls to
voicemail directly on your setup)

I would use a translation pattern on CUCM to route calls from 1212 to 1702 unless
you need to keep 1212 on a phone/etc for some reason.

On Fri, Jan 11, 2013 at 6:49 PM, David Zhars <dzhars at gmail.com> wrote:
So my confusion has to extend from not having a solid understanding of an
"alternate extension" in Unity. I assumed (there's that word) that applying 1212
as an alternate ext to 1702, would mean the user would STILL have to login
SEPARATELY to both VM boxes to get messages. Are you saying that if I add 1212 as
an alternate, he needs only check messages for ext 1702, and anything that went to
1212 would be in the 1702 box?? Cause that would be way easy!

On Fri, Jan 11, 2013 at 4:21 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
A translation pattern is one option, the other is a DN for 1212 (not assigned to a
phone) with CFA set to 1702. You'll want to test your usual call flows so that the
fact that every call to 1702 will be a forwarded calls. For example if you have
any calling party selections on outbound gateways set to 'first redirecting number'
you may end up seeing 1212 instead of 1702 as the calling party number. Calls
that end up in voicemail will also be sent to the mailbox of 1212.

The routing rule in Unity would basically be: Any redirected call with an original
called party number of 1212, send it to standard greeting for 1702. I bet you
could also just ad 1212 as an alternate extension for 1702 and you'll be set.

-Ryan

On Jan 11, 2013, at 2:51 PM, David Zhars <dzhars at gmail.com> wrote:

OK, here's my dilemma. 1212 doesn't exist as far as UCM is concerned. I suppose I
could recreate it as a CTI Route point??

Ryan, not sure what you mean about Unity sending calls from 1212 to 1702.

Sounds like a translation pattern in UCM may be the way to go, delete the 1212
mailbox in Unity, so if someone does try and transfer a call to that VM it would
error out.

While I want this to be easy for the end users, I also want it to be easy for me!

On Fri, Jan 11, 2013 at 2:28 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
Unity call routing rule to send calls from 1212 to 1702, should be pretty straight
forward.
-Ryan

On Jan 11, 2013, at 1:32 PM, David Zhars <dzhars at gmail.com> wrote:

Old user had ext 1212 (and this is a DID, so people can call directly from the
outside).
New user has ext 1702.

What I want is:

Internally: User dials 1212, phone rings at 1702.


Internally: Reception takes a call, transfers it with TRANS **1212 TRANS, call goes
to 1702 voicemail.

Externally: Someone calls 555-1212 and the call lands internally at 1702.

Some of this I know how to do, I am not sure about the transfer to voicemail of the
old extension and have it land at the new ext VM.

Appreciate any help!

Dave

UCM 8.0, Unity 8.0

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From MLoraditch at heliontechnologies.com Mon Jan 14 11:26:39 2013


From: MLoraditch at heliontechnologies.com (Matthew Loraditch)
Date: Mon, 14 Jan 2013 16:26:39 +0000
Subject: [cisco-voip] 9.1 Upgrade Times
In-Reply-To: <C75AF2AD9308C246AFBDDB994E3E2983110B2E69@PHANES.helion.local>
References: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B2E69@PHANES.helion.local>
Message-ID: <C75AF2AD9308C246AFBDDB994E3E2983110B5E52@PHANES.helion.local>

So I figured I'd loop around and let everyone know how things ended up.
Unity Connection never finished the first attempt and wouldn't cancel either. I had
to force a shutdown. After it came back up I tried again... 90 minutes boom.
Suffice it to say. I wish had done that sooner.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: Matthew Loraditch


Sent: Sunday, January 13, 2013 3:06 PM
To: Matthew Loraditch; cisco-voip at puck.nether.net
Subject: RE: 9.1 Upgrade Times

Well my CUCM publisher took almost 13 hours, my sub took 1 hour and change... the
unity connection pub is still running, I think we are on hour 15 of that...

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
Matthew Loraditch
Sent: Sunday, January 13, 2013 8:58 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] 9.1 Upgrade Times

On a Unity Connection system and CUCM both 8.6.2aSU1 started at 11:45PM and 12:12AM
Last night, both are STILL running at 8:45 AM this morning. The system I am doing
this test on has about 60 Phones/Users/VM. These are the publishers of each
install, but I have never had an upgrade take this long, ever.

I'm now not sure I'll even be able to finish in my window since I haven't even
touched the subscribers yet.

Any clues as to why this is taking so long? I used to disable iothrottle during an
upgrade but that command is deprecated now.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

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From petrybr at gmail.com Mon Jan 14 11:42:30 2013


From: petrybr at gmail.com (=?ISO-8859-1?Q?Jos=E9_Paulo_de_Oliveira_Petry?=)
Date: Mon, 14 Jan 2013 14:42:30 -0200
Subject: [cisco-voip] License migration: 8.6 -> 9.0
Message-ID: <CAMX+=-X2NE+DguYETCZ4TOJoC=Hbj27=TxCDwnh7tfBb=FDnRA@mail.gmail.com>

Hello,

I have a 8.6 CUCM running with 1700 DLUs and did a upgrade to 9.0.

To migrate licenses (DLU to CUWL) im using this doc:


http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/9x/migratn.html#wp1067112

After i add the CUCM instance, im trying to use the Migration Utility, but
im receiving this error:

"There are no Unified CM product instances with pre-9.0 licenses available


for upgrade"

Any suggestion?

Regards,

Jos? Paulo de Oliveira Petry


petrybr at gmail.com
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From tfrazee at gmail.com Mon Jan 14 11:45:00 2013


From: tfrazee at gmail.com (Tim Frazee)
Date: Mon, 14 Jan 2013 10:45:00 -0600
Subject: [cisco-voip] 9.1 Upgrade Times
In-Reply-To: <C75AF2AD9308C246AFBDDB994E3E2983110B5E52@PHANES.helion.local>
References: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B2E69@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B5E52@PHANES.helion.local>
Message-ID: <CABzsfHxm-nH3DW-mYvFc8TqsWVjV7dHm+1DonOdR=g-q+ecwhg@mail.gmail.com>

so would you suggest a clean reboot before the upgrade of CUC to at least
9.1 ?

On Mon, Jan 14, 2013 at 10:26 AM, Matthew Loraditch <


MLoraditch at heliontechnologies.com> wrote:

> So I figured I?d loop around and let everyone know how things ended up.**
> **
>
> Unity Connection never finished the first attempt and wouldn?t cancel
> either. I had to force a shutdown. After it came back up I tried again? 90
> minutes boom. Suffice it to say. I wish had done that sooner. ****
>
> ** **
>
> ** **
>
> Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA
>
> 1965 Greenspring Drive
> Timonium, MD 21093
>
> voice. 410.252.8830
> fax. 410.252.9284
>
> Twitter <http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296>
> | Website <http://www.heliontechnologies.com/> | Email Support<support at
heliontechnologies.com?subject=Technical%20Support%20Request>
> ****
>
> ** **
>
> ** **
>
> *From:* Matthew Loraditch
> *Sent:* Sunday, January 13, 2013 3:06 PM
> *To:* Matthew Loraditch; cisco-voip at puck.nether.net
> *Subject:* RE: 9.1 Upgrade Times****
>
> ** **
>
> Well my CUCM publisher took almost 13 hours, my sub took 1 hour and
> change? the unity connection pub is still running, I think we are on hour
> 15 of that?****
>
> ** **
>
> ** **
>
> Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA
>
> 1965 Greenspring Drive
> Timonium, MD 21093
>
> voice. 410.252.8830
> fax. 410.252.9284
>
> Twitter <http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296>
> | Website <http://www.heliontechnologies.com/> | Email Support<support at
heliontechnologies.com?subject=Technical%20Support%20Request>
> ****
>
> ** **
>
> ** **
>
> *From:* cisco-voip-bounces at puck.nether.net [
> mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at
puck.nether.net>]
> *On Behalf Of *Matthew Loraditch
> *Sent:* Sunday, January 13, 2013 8:58 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] 9.1 Upgrade Times****
>
> ** **
>
> On a Unity Connection system and CUCM both 8.6.2aSU1 started at 11:45PM
> and 12:12AM Last night, both are STILL running at 8:45 AM this morning. The
> system I am doing this test on has about 60 Phones/Users/VM. These are the
> publishers of each install, but I have never had an upgrade take this long,
> ever.****
>
> ** **
>
> I?m now not sure I?ll even be able to finish in my window since I haven?t
> even touched the subscribers yet.****
>
> ** **
>
> Any clues as to why this is taking so long? I used to disable iothrottle
> during an upgrade but that command is deprecated now.****
>
> ** **
>
> ** **
>
> Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA
>
> 1965 Greenspring Drive
> Timonium, MD 21093
>
> voice. 410.252.8830
> fax. 410.252.9284
>
> Twitter <http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296>
> | Website <http://www.heliontechnologies.com/> | Email Support<support at
heliontechnologies.com?subject=Technical%20Support%20Request>
> ****
>
> ** **
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From VanMarenNP at ldschurch.org Mon Jan 14 11:45:26 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Mon, 14 Jan 2013 16:45:26 +0000
Subject: [cisco-voip] License migration: 8.6 -> 9.0
In-Reply-To: <CAMX+=-X2NE+DguYETCZ4TOJoC=Hbj27=TxCDwnh7tfBb=FDnRA@mail.gmail.com>
References: <CAMX+=-X2NE+DguYETCZ4TOJoC=Hbj27=TxCDwnh7tfBb=FDnRA@mail.gmail.com>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E71580@W12112.ldschurch.org>

Run from 9.0 and do 9.1. You'll have to redo all of the licenses when you go to
9.1 anyways.

All of the 9.0 upgrades that I did the ELM work flawlessly once the ESXi host that
it was on had a proper NIC teaming configuration.

-Nate

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Jos? Paulo de Oliveira Petry
Sent: Monday, January 14, 2013 9:43 AM
To: Cisco VoIPoE List
Subject: [cisco-voip] License migration: 8.6 -> 9.0

Hello,

I have a 8.6 CUCM running with 1700 DLUs and did a upgrade to 9.0.

To migrate licenses (DLU to CUWL) im using this doc:


http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/9x/migratn.html#wp1067112

After i add the CUCM instance, im trying to use the Migration Utility, but im
receiving this error:

"There are no Unified CM product instances with pre-9.0 licenses available for
upgrade"

Any suggestion?
Regards,

Jos? Paulo de Oliveira Petry


petrybr at gmail.com<mailto:petrybr at gmail.com>

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
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From Leslie.Meade at lvs1.com Mon Jan 14 11:56:35 2013


From: Leslie.Meade at lvs1.com (Leslie Meade)
Date: Mon, 14 Jan 2013 16:56:35 +0000
Subject: [cisco-voip] UCCX cluster
Message-ID:
<F64719604B4E6F41BDBB2AF38E7609F433DF4D9E@LVSCGYEX03.longviewsystems.com>

I am adding a new uccx server into a cluster. I have been told that once this is
done, there will be an outage while the Data base is replicated to the HA.
Is this true ? I am trying to find doc on this

Cheers

Leslie

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From MLoraditch at heliontechnologies.com Mon Jan 14 12:08:18 2013


From: MLoraditch at heliontechnologies.com (Matthew Loraditch)
Date: Mon, 14 Jan 2013 17:08:18 +0000
Subject: [cisco-voip] 9.1 Upgrade Times
In-Reply-To: <CABzsfHxm-nH3DW-mYvFc8TqsWVjV7dHm+1DonOdR=g-q+ecwhg@mail.gmail.com>
References: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B2E69@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B5E52@PHANES.helion.local>
<CABzsfHxm-nH3DW-mYvFc8TqsWVjV7dHm+1DonOdR=g-q+ecwhg@mail.gmail.com>
Message-ID: <C75AF2AD9308C246AFBDDB994E3E2983110B618A@PHANES.helion.local>

To answer you and Nate


The server had been up for Several Months and yes it looks like it wouldn't hurt to
do a reboot.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: Tim Frazee [mailto:tfrazee at gmail.com]


Sent: Monday, January 14, 2013 11:45 AM
To: Matthew Loraditch
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 9.1 Upgrade Times

so would you suggest a clean reboot before the upgrade of CUC to at least 9.1 ?
On Mon, Jan 14, 2013 at 10:26 AM, Matthew Loraditch <MLoraditch at
heliontechnologies.com<mailto:MLoraditch at heliontechnologies.com>> wrote:
So I figured I'd loop around and let everyone know how things ended up.
Unity Connection never finished the first attempt and wouldn't cancel either. I had
to force a shutdown. After it came back up I tried again... 90 minutes boom.
Suffice it to say. I wish had done that sooner.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830<tel:410.252.8830>
fax. 410.252.9284<tel:410.252.9284>

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: Matthew Loraditch


Sent: Sunday, January 13, 2013 3:06 PM
To: Matthew Loraditch; cisco-voip at puck.nether.net<mailto:cisco-voip at
puck.nether.net>
Subject: RE: 9.1 Upgrade Times

Well my CUCM publisher took almost 13 hours, my sub took 1 hour and change... the
unity connection pub is still running, I think we are on hour 15 of that...

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830<tel:410.252.8830>
fax. 410.252.9284<tel:410.252.9284>

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
Matthew Loraditch
Sent: Sunday, January 13, 2013 8:58 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] 9.1 Upgrade Times

On a Unity Connection system and CUCM both 8.6.2aSU1 started at 11:45PM and 12:12AM
Last night, both are STILL running at 8:45 AM this morning. The system I am doing
this test on has about 60 Phones/Users/VM. These are the publishers of each
install, but I have never had an upgrade take this long, ever.

I'm now not sure I'll even be able to finish in my window since I haven't even
touched the subscribers yet.

Any clues as to why this is taking so long? I used to disable iothrottle during an
upgrade but that command is deprecated now.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830<tel:410.252.8830>
fax. 410.252.9284<tel:410.252.9284>

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

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From avholloway+cisco-voip at gmail.com Mon Jan 14 12:08:22 2013


From: avholloway+cisco-voip at gmail.com (Anthony Holloway)
Date: Mon, 14 Jan 2013 11:08:22 -0600
Subject: [cisco-voip] UCCX cluster
In-Reply-To:
<F64719604B4E6F41BDBB2AF38E7609F433DF4D9E@LVSCGYEX03.longviewsystems.com>
References:
<F64719604B4E6F41BDBB2AF38E7609F433DF4D9E@LVSCGYEX03.longviewsystems.com>
Message-ID: <CACRCJOgW0jKdKiYJrjb8E8nGPAEN7VZwn2k9PL2rBNDL2=igOA@mail.gmail.com>
It's in the Admin Guide
[link<http://www.cisco.com/en/US/partner/products/sw/custcosw/ps1846/products_insta
llation_and_configuration_guides_list.html>],
and it's a note, that calls *may* be dropped. I don't have any concrete
data to share with you, as I have only ever built HA from the get-go.

[image: Inline image 1]

On Mon, Jan 14, 2013 at 10:56 AM, Leslie Meade <Leslie.Meade at lvs1.com>wrote:

> I am adding a new uccx server into a cluster. I have been told that once
> this is done, there will be an outage while the Data base is replicated to
> the HA.****
>
> Is this true ? I am trying to find doc on this****
>
> ** **
>
> ** **
>
> Cheers****
>
> ** **
>
> ** **
>
> Leslie****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From VanMarenNP at ldschurch.org Mon Jan 14 11:33:25 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Mon, 14 Jan 2013 16:33:25 +0000
Subject: [cisco-voip] 9.1 Upgrade Times
In-Reply-To: <C75AF2AD9308C246AFBDDB994E3E2983110B5E52@PHANES.helion.local>
References: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B2E69@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B5E52@PHANES.helion.local>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E71556@W12112.ldschurch.org>
How long had the server been up before the you started the first upgrade?

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Matthew Loraditch
Sent: Monday, January 14, 2013 9:27 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 9.1 Upgrade Times

So I figured I'd loop around and let everyone know how things ended up.
Unity Connection never finished the first attempt and wouldn't cancel either. I had
to force a shutdown. After it came back up I tried again... 90 minutes boom.
Suffice it to say. I wish had done that sooner.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: Matthew Loraditch


Sent: Sunday, January 13, 2013 3:06 PM
To: Matthew Loraditch; cisco-voip at puck.nether.net<mailto:cisco-voip at
puck.nether.net>
Subject: RE: 9.1 Upgrade Times

Well my CUCM publisher took almost 13 hours, my sub took 1 hour and change... the
unity connection pub is still running, I think we are on hour 15 of that...

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
Matthew Loraditch
Sent: Sunday, January 13, 2013 8:58 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] 9.1 Upgrade Times

On a Unity Connection system and CUCM both 8.6.2aSU1 started at 11:45PM and 12:12AM
Last night, both are STILL running at 8:45 AM this morning. The system I am doing
this test on has about 60 Phones/Users/VM. These are the publishers of each
install, but I have never had an upgrade take this long, ever.

I'm now not sure I'll even be able to finish in my window since I haven't even
touched the subscribers yet.

Any clues as to why this is taking so long? I used to disable iothrottle during an
upgrade but that command is deprecated now.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

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From jbuchanan at presidio.com Mon Jan 14 13:03:39 2013


From: jbuchanan at presidio.com (Buchanan, James)
Date: Mon, 14 Jan 2013 18:03:39 +0000
Subject: [cisco-voip] UCCX cluster
In-Reply-To: <CACRCJOgW0jKdKiYJrjb8E8nGPAEN7VZwn2k9PL2rBNDL2=igOA@mail.gmail.com>
References:
<F64719604B4E6F41BDBB2AF38E7609F433DF4D9E@LVSCGYEX03.longviewsystems.com>
<CACRCJOgW0jKdKiYJrjb8E8nGPAEN7VZwn2k9PL2rBNDL2=igOA@mail.gmail.com>
Message-ID: <12D6A6A157B44348974E5ED93738628C25369137@HQEXCHMBX03.Presidio.Corp>

What will happen is that the CCX Engine will need to restart (at least). There may
also be issues while the port group is being rebuilt with the added ports. This
will create an outage.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Anthony Holloway
Sent: Monday, January 14, 2013 7:08 PM
To: Leslie Meade
Cc: cisco-voip (cisco-voip at puck.nether.net)
Subject: Re: [cisco-voip] UCCX cluster

It's in the Admin Guide


[link<http://www.cisco.com/en/US/partner/products/sw/custcosw/ps1846/products_insta
llation_and_configuration_guides_list.html>], and it's a note, that calls *may* be
dropped. I don't have any concrete data to share with you, as I have only ever
built HA from the get-go.

[Inline image 1]

On Mon, Jan 14, 2013 at 10:56 AM, Leslie Meade <Leslie.Meade at


lvs1.com<mailto:Leslie.Meade at lvs1.com>> wrote:
I am adding a new uccx server into a cluster. I have been told that once this is
done, there will be an outage while the Data base is replicated to the HA.
Is this true ? I am trying to find doc on this

Cheers

Leslie

James Buchanan | Sr. Network Engineer


Presidio | www.presidio.com<http://www.presidio.com>
12 Cadillac Drive Suite 130, Brentwood, TN 37027
D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 | jbuchanan at
presidio.com<mailto:jbuchanan at presidio.com>

[Be Secure In The Knowledge]<http://www.presidio.com>

Follow us:

[Follow Presidio on Twitter]<http://www.twitter.com/presidio>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip

This message w/attachments (message) is intended solely for the use of the intended
recipient(s) and may contain information that is privileged, confidential or
proprietary. If you are not an intended recipient, please notify the sender, and
then please delete and destroy all copies and attachments. Please be advised that
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From tanner.ezell at gmail.com Mon Jan 14 13:11:17 2013


From: tanner.ezell at gmail.com (Tanner Ezell)
Date: Mon, 14 Jan 2013 11:11:17 -0700
Subject: [cisco-voip] UCCX cluster
In-Reply-To: <12D6A6A157B44348974E5ED93738628C25369137@HQEXCHMBX03.Presidio.Corp>
References:
<F64719604B4E6F41BDBB2AF38E7609F433DF4D9E@LVSCGYEX03.longviewsystems.com>
<CACRCJOgW0jKdKiYJrjb8E8nGPAEN7VZwn2k9PL2rBNDL2=igOA@mail.gmail.com>
<12D6A6A157B44348974E5ED93738628C25369137@HQEXCHMBX03.Presidio.Corp>
Message-ID: <CADsErBYOzGxYPQF2Aiyqmf+ZN=FqG9w=b9McfXVyyUP=BfK18w@mail.gmail.com>

Note that the call control groups will not expand automatically, you'll
need to create the second set of ports on the CCG (at least that is the
behavior on SU4)

On Mon, Jan 14, 2013 at 11:03 AM, Buchanan, James <jbuchanan at presidio.com>wrote:

> What will happen is that the CCX Engine will need to restart (at
> least). There may also be issues while the port group is being rebuilt with
> the added ports. This will create an outage.****
>
> ** **
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Anthony Holloway
> *Sent:* Monday, January 14, 2013 7:08 PM
> *To:* Leslie Meade
> *Cc:* cisco-voip (cisco-voip at puck.nether.net)
> *Subject:* Re: [cisco-voip] UCCX cluster****
>
> ** **
>
> It's in the Admin Guide
[link<http://www.cisco.com/en/US/partner/products/sw/custcosw/ps1846/products_insta
llation_and_configuration_guides_list.html>],
> and it's a note, that calls *may* be dropped. I don't have any concrete
> data to share with you, as I have only ever built HA from the get-go.
>
> [image: Inline image 1]****
>
> ** **
>
> On Mon, Jan 14, 2013 at 10:56 AM, Leslie Meade <Leslie.Meade at lvs1.com>
> wrote:****
>
> I am adding a new uccx server into a cluster. I have been told that once
> this is done, there will be an outage while the Data base is replicated to
> the HA.****
>
> Is this true ? I am trying to find doc on this****
>
> ****
>
> ****
>
> Cheers****
>
> ****
>
> ****
>
> Leslie****
>
> ****
>
>
>
> James Buchanan | Sr. Network Engineer
> Presidio | www.presidio.com
> 12 Cadillac Drive Suite 130, Brentwood, TN 37027
> D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 |
> jbuchanan at presidio.com
>
>
>
> [image: Be Secure In The Knowledge] <http://www.presidio.com>
>
>
> Follow us:
>
> [image: Follow Presidio on Twitter] <http://www.twitter.com/presidio>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ** **
>
> *This message w/attachments (message) is intended solely for the use of
> the intended recipient(s) and may contain information that is privileged,
> confidential or proprietary. If you are not an intended recipient, please
> notify the sender, and then please delete and destroy all copies and
> attachments. Please be advised that any review or dissemination of, or
> the taking of any action in reliance on, the information contained in or
> attached to this message is prohibited.*****
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From jmad at cityofevanston.org Mon Jan 14 12:35:02 2013


From: jmad at cityofevanston.org (Madziarczyk, Jonathan)
Date: Mon, 14 Jan 2013 11:35:02 -0600
Subject: [cisco-voip] Upgrading from 6x to 9x?
Message-ID:
<4604610D6AAC164AA12B9E03AB79028B0B7B9E82@EXCHANGE3.local.cityofevanston.org>

So I know Cisco did a full presentation at Live last year on this


(including moving from physical to virtual), but they didn't think to
record it, so all I have is the powerpoint, which is sadly lacking in
information. Has anyone seen a webex or similar presentation that
actually goes through the process and mentions all the caveats?

JonM

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From dane.newman at gmail.com Mon Jan 14 15:40:33 2013


From: dane.newman at gmail.com (Dane Newman)
Date: Mon, 14 Jan 2013 15:40:33 -0500
Subject: [cisco-voip] Mobility Issue
Message-ID: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>

Hello

I have an issue when users are connected to a call and hit the mobility
soft key button on 9971 phones when a call is active, the phone system
rings on the mobile number configured in the system. When they pick up the
the mobile number it just plays what sounds like hold music on both ends of
the call (I believe this music is coming from cucm but I haven't heard it
before) instead of providing 2 way voice.

In another senario with what I believe is the same issue. If a user picks
up on there cell phone first (using single number reach) opposed to the
deskphone the call is connected with 2 way voice and no issues exist. If
the user then hangs up his cell phone with the intent to take the call on
his deskphone the calling party starts hearing the hold music. Once the
user picks up the call on his deskphone he hears nothing but the calling
party is still hearing the hold music. It is not working as intended where
2 way voice happens once the user hangs up his mobile phone and picks up on
his deskphone 2 way voice should happen.

My topology is as follows..

PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE

Calls are sent back out the SIP trunk to the ITSP when using mobile
connect/snr.

Does anyone have any ideas how I can make 2 way voice happen instead of the
hold music when the calls are picked up?
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From dane.newman at gmail.com Mon Jan 14 15:47:04 2013


From: dane.newman at gmail.com (Dane Newman)
Date: Mon, 14 Jan 2013 15:47:04 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
Message-ID: <BEF7E91B-C188-4968-A4FF-DF445665CE58@gmail.com>

This is what the music sounds like

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Sent from my mobile device

On Jan 14, 2013, at 3:40 PM, Dane Newman <dane.newman at gmail.com> wrote:

> Hello
>
> I have an issue when users are connected to a call and hit the mobility soft key
button on 9971 phones when a call is active, the phone system rings on the mobile
number configured in the system. When they pick up the the mobile number it just
plays what sounds like hold music on both ends of the call (I believe this music is
coming from cucm but I haven't heard it before) instead of providing 2 way voice.
>
> In another senario with what I believe is the same issue. If a user picks up on
there cell phone first (using single number reach) opposed to the deskphone the
call is connected with 2 way voice and no issues exist. If the user then hangs up
his cell phone with the intent to take the call on his deskphone the calling party
starts hearing the hold music. Once the user picks up the call on his deskphone he
hears nothing but the calling party is still hearing the hold music. It is not
working as intended where 2 way voice happens once the user hangs up his mobile
phone and picks up on his deskphone 2 way voice should happen.
>
> My topology is as follows..
>
>
> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>
> Calls are sent back out the SIP trunk to the ITSP when using mobile connect/snr.
>
> Does anyone have any ideas how I can make 2 way voice happen instead of the hold
music when the calls are picked up?

From jbalbuena at desca.com Mon Jan 14 14:34:39 2013


From: jbalbuena at desca.com (Jose Balbuena)
Date: Mon, 14 Jan 2013 19:34:39 +0000
Subject: [cisco-voip] ERROR UPDATING LOCALE - CUCM 8.0
Message-ID: <6EDB87B1248B1B47AAD57EE99E053B48825A66DC@desca-mia-e10n2.desca.com>

Hello,

Someone can help me to figure out why some IP Phones 7911 & 7942 are not loading
correctly the file that permit the phone is in Spanish. For what I understand of
the next log extracted of a Phone (7942, Firmware: SCCP42.9-0-3S), is that the file
is well received but finally the phone does not execute correctly.

Any help, please!

3270: ERR 17:33:25.187583 JVM: tftpClient Spanish_Colombia/mk-sccp.jar


/usr/ram/L10N1063478883 550001 1
3271: NOT 17:33:25.191035 tftpClient: tftp request rcv'd from /usr/tmp/tftp,
srcFile = Spanish_Colombia/mk-sccp.jar, dstFile = /usr/ram/L10N1063478883 max size
= 550001
3272: NOT 17:33:25.193072 CDP-D: catchipcfg:getdhcpinfo IP:a224f23 domain:
chngVal:1
3273: NOT 17:33:25.193903 CDP-D: cdpGetPortCfg SPANTOPC CFG:11
3274: NOT 17:33:25.205494 tftpClient: auth server - tftpList[0] =
::ffff:10.33.168.250
3275: NOT 17:33:25.206047 tftpClient: look up server - 0
3276: NOT 17:33:25.207805 SECD: lookupCTL: TFTP SRVR secure
3277: NOT 17:33:25.210974 tftpClient: secVal = 0x9
3278: NOT 17:33:25.211628 tftpClient: ::ffff:10.33.168.250 is a secure server
3279: NOT 17:33:25.212120 tftpClient: retval = SRVR_SECURE
3280: NOT 17:33:25.212597 tftpClient: Secure file requested
3281: NOT 17:33:25.213060 tftpClient: authenticated file approved - add .sgn --
Spanish_Colombia/mk-sccp.jar.sgn
3282: ERR 17:33:25.240988 RTSOLD: opvvlan:594->594 advvlan:4095->4095 vvlanState=1
3283: NOT 17:33:25.249036 TFTP: [26]:Requesting Spanish_Colombia/mk-sccp.jar.sgn
from 10.33.168.250 with size limit of 550001

Best & Regards.

Rodrigo B.
CCNA VOICE
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whom they are addressed and it may contain information that is privileged or
confidential. If you have received this communication by mistake, please notify us
immediately by e-mail or telephone.The storage, recording, use or disclosure of
this e-mail and its attachments by anyone other than the intended recipient is
strictly prohibited. This message has been verified using antivirus software;
however, the sender is not responsible for any damage to hardware or software
resulting from the presence of any virus.

Este mensaje y cualquier anexo son exclusivamente para la persona a quien van
dirigidos y pueden contener informaci?n privilegiada o confidencial. Si usted ha
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o divulgaci?n con cualquier prop?sito. Este mensaje ha sido verificado con software
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From kennethwhayes at gmail.com Mon Jan 14 16:11:52 2013


From: kennethwhayes at gmail.com (Kenneth Hayes)
Date: Mon, 14 Jan 2013 16:11:52 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
Message-ID: <-4286606911770185612@unknownmsgid>

What version of code are you running on the CUBE?

Sent from my iPhone

On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:

> Hello
>
> I have an issue when users are connected to a call and hit the mobility soft key
button on 9971 phones when a call is active, the phone system rings on the mobile
number configured in the system. When they pick up the the mobile number it just
plays what sounds like hold music on both ends of the call (I believe this music is
coming from cucm but I haven't heard it before) instead of providing 2 way voice.
>
> In another senario with what I believe is the same issue. If a user picks up on
there cell phone first (using single number reach) opposed to the deskphone the
call is connected with 2 way voice and no issues exist. If the user then hangs up
his cell phone with the intent to take the call on his deskphone the calling party
starts hearing the hold music. Once the user picks up the call on his deskphone he
hears nothing but the calling party is still hearing the hold music. It is not
working as intended where 2 way voice happens once the user hangs up his mobile
phone and picks up on his deskphone 2 way voice should happen.
>
> My topology is as follows..
>
>
> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>
> Calls are sent back out the SIP trunk to the ITSP when using mobile connect/snr.
>
> Does anyone have any ideas how I can make 2 way voice happen instead of the hold
music when the calls are picked up?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From dane.newman at gmail.com Mon Jan 14 16:18:16 2013


From: dane.newman at gmail.com (Dane Newman)
Date: Mon, 14 Jan 2013 16:18:16 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <-4286606911770185612@unknownmsgid>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
Message-ID: <07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>

Using keyboard-interactive authentication.


Password:

Cisco3825#
Cisco3825#sh ver
Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1
(4)M5, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2012 by Cisco Systems, Inc.
Compiled Tue 04-Sep-12 17:25 by prod_rel_team

ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)

Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes


System returned to ROM by power-on
System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
Last reload type: Normal Reload

This product contains cryptographic features and is subject to United


States and local country laws governing import, export, transfer and
use. Delivery of Cisco cryptographic products does not imply
third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for
compliance with U.S. and local country laws. By using this product you
agree to comply with applicable laws and regulations. If you are unable
to comply with U.S. and local laws, return this product immediately.

A summary of U.S. laws governing Cisco cryptographic products may be found at:
http://www.cisco.com/wwl/export/crypto/tool/stqrg.html

If you require further assistance please contact us by sending email to


export at cisco.com.

Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.


Processor board ID FTX1237A1T0
2 Gigabit Ethernet interfaces
2 Channelized T1/PRI ports
1 Virtual Private Network (VPN) Module
DRAM configuration is 64 bits wide with parity enabled.
479K bytes of NVRAM.
500472K bytes of ATA System CompactFlash (Read/Write)

License Info:

License UDI:

-------------------------------------------------
Device# PID SN

Sent from my mobile device

On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
> What version of code are you running on the CUBE?
>
> Sent from my iPhone
>
> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
>> Hello
>>
>> I have an issue when users are connected to a call and hit the mobility soft
key button on 9971 phones when a call is active, the phone system rings on the
mobile number configured in the system. When they pick up the the mobile number it
just plays what sounds like hold music on both ends of the call (I believe this
music is coming from cucm but I haven't heard it before) instead of providing 2 way
voice.
>>
>> In another senario with what I believe is the same issue. If a user picks up on
there cell phone first (using single number reach) opposed to the deskphone the
call is connected with 2 way voice and no issues exist. If the user then hangs up
his cell phone with the intent to take the call on his deskphone the calling party
starts hearing the hold music. Once the user picks up the call on his deskphone he
hears nothing but the calling party is still hearing the hold music. It is not
working as intended where 2 way voice happens once the user hangs up his mobile
phone and picks up on his deskphone 2 way voice should happen.
>>
>> My topology is as follows..
>>
>>
>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>
>> Calls are sent back out the SIP trunk to the ITSP when using mobile connect/snr.
>>
>> Does anyone have any ideas how I can make 2 way voice happen instead of the hold
music when the calls are picked up?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip

From rratliff at cisco.com Mon Jan 14 17:25:41 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 14 Jan 2013 17:25:41 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
Message-ID: <C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>

Do you get similar behavior if you just hold and resume the call outside SNR
features?

-Ryan

On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:

Using keyboard-interactive authentication.


Password:
Cisco3825#
Cisco3825#sh ver
Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1
(4)M5, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2012 by Cisco Systems, Inc.
Compiled Tue 04-Sep-12 17:25 by prod_rel_team

ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)

Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes


System returned to ROM by power-on
System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
Last reload type: Normal Reload

This product contains cryptographic features and is subject to United


States and local country laws governing import, export, transfer and
use. Delivery of Cisco cryptographic products does not imply
third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for
compliance with U.S. and local country laws. By using this product you
agree to comply with applicable laws and regulations. If you are unable
to comply with U.S. and local laws, return this product immediately.

A summary of U.S. laws governing Cisco cryptographic products may be found at:
http://www.cisco.com/wwl/export/crypto/tool/stqrg.html

If you require further assistance please contact us by sending email to


export at cisco.com.

Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.


Processor board ID FTX1237A1T0
2 Gigabit Ethernet interfaces
2 Channelized T1/PRI ports
1 Virtual Private Network (VPN) Module
DRAM configuration is 64 bits wide with parity enabled.
479K bytes of NVRAM.
500472K bytes of ATA System CompactFlash (Read/Write)

License Info:

License UDI:

-------------------------------------------------
Device# PID SN

Sent from my mobile device

On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:

> What version of code are you running on the CUBE?


>
> Sent from my iPhone
>
> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
>> Hello
>>
>> I have an issue when users are connected to a call and hit the mobility soft
key button on 9971 phones when a call is active, the phone system rings on the
mobile number configured in the system. When they pick up the the mobile number it
just plays what sounds like hold music on both ends of the call (I believe this
music is coming from cucm but I haven't heard it before) instead of providing 2 way
voice.
>>
>> In another senario with what I believe is the same issue. If a user picks up on
there cell phone first (using single number reach) opposed to the deskphone the
call is connected with 2 way voice and no issues exist. If the user then hangs up
his cell phone with the intent to take the call on his deskphone the calling party
starts hearing the hold music. Once the user picks up the call on his deskphone he
hears nothing but the calling party is still hearing the hold music. It is not
working as intended where 2 way voice happens once the user hangs up his mobile
phone and picks up on his deskphone 2 way voice should happen.
>>
>> My topology is as follows..
>>
>>
>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>
>> Calls are sent back out the SIP trunk to the ITSP when using mobile connect/snr.
>>
>> Does anyone have any ideas how I can make 2 way voice happen instead of the hold
music when the calls are picked up?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-------------- next part --------------


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From kennethwhayes at gmail.com Mon Jan 14 17:36:35 2013


From: kennethwhayes at gmail.com (Kenneth Hayes)
Date: Mon, 14 Jan 2013 17:36:35 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
Message-ID: <1397723124845575534@unknownmsgid>

What codec are you using?

Sent from my iPhone

On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:

> Using keyboard-interactive authentication.


> Password:
>
> Cisco3825#
> Cisco3825#sh ver
> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1
> (4)M5, RELEASE SOFTWARE (fc1)
> Technical Support: http://www.cisco.com/techsupport
> Copyright (c) 1986-2012 by Cisco Systems, Inc.
> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>
> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>
> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
> System returned to ROM by power-on
> System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
> Last reload type: Normal Reload
>
>
> This product contains cryptographic features and is subject to United
> States and local country laws governing import, export, transfer and
> use. Delivery of Cisco cryptographic products does not imply
> third-party authority to import, export, distribute or use encryption.
> Importers, exporters, distributors and users are responsible for
> compliance with U.S. and local country laws. By using this product you
> agree to comply with applicable laws and regulations. If you are unable
> to comply with U.S. and local laws, return this product immediately.
>
> A summary of U.S. laws governing Cisco cryptographic products may be found at:
> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>
> If you require further assistance please contact us by sending email to
> export at cisco.com.
>
> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
> Processor board ID FTX1237A1T0
> 2 Gigabit Ethernet interfaces
> 2 Channelized T1/PRI ports
> 1 Virtual Private Network (VPN) Module
> DRAM configuration is 64 bits wide with parity enabled.
> 479K bytes of NVRAM.
> 500472K bytes of ATA System CompactFlash (Read/Write)
>
>
> License Info:
>
> License UDI:
>
> -------------------------------------------------
> Device# PID SN
>
> Sent from my mobile device
>
> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
>
>> What version of code are you running on the CUBE?
>>
>> Sent from my iPhone
>>
>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>>> Hello
>>>
>>> I have an issue when users are connected to a call and hit the mobility soft
key button on 9971 phones when a call is active, the phone system rings on the
mobile number configured in the system. When they pick up the the mobile number it
just plays what sounds like hold music on both ends of the call (I believe this
music is coming from cucm but I haven't heard it before) instead of providing 2 way
voice.
>>>
>>> In another senario with what I believe is the same issue. If a user picks up on
there cell phone first (using single number reach) opposed to the deskphone the
call is connected with 2 way voice and no issues exist. If the user then hangs up
his cell phone with the intent to take the call on his deskphone the calling party
starts hearing the hold music. Once the user picks up the call on his deskphone he
hears nothing but the calling party is still hearing the hold music. It is not
working as intended where 2 way voice happens once the user hangs up his mobile
phone and picks up on his deskphone 2 way voice should happen.
>>>
>>> My topology is as follows..
>>>
>>>
>>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>>
>>> Calls are sent back out the SIP trunk to the ITSP when using mobile
connect/snr.
>>>
>>> Does anyone have any ideas how I can make 2 way voice happen instead of the
hold music when the calls are picked up?
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip

From dane.newman at gmail.com Mon Jan 14 18:06:25 2013


From: dane.newman at gmail.com (Dane Newman)
Date: Mon, 14 Jan 2013 18:06:25 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
Message-ID: <CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>

Ryan (sorry I forgot to reply to all)

Thanks for the Reply


Oddly enough we are.
This probably has something to do with MOH in general?

Internally when I user puts another user on hold everything works. No MOH
plays and they can hold and unhold the call just fine.
I tested calling from an external number. Once I put the external caller on
hold the MOH played but I was unable to resume the call. When I hit resume
on the deskphone the MOH still played to the external caller and there was
no sound on the deskphone.
On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com> wrote:

> Do you get similar behavior if you just hold and resume the call outside
> SNR features?
>
> -Ryan
>
> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> Using keyboard-interactive authentication.
>
> Password:
>
>
> Cisco3825#
>
> Cisco3825#sh ver
>
> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M),
> Version 15.1
> (4)M5, RELEASE SOFTWARE (fc1)
>
> Technical Support: http://www.cisco.com/techsupport
> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>
> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>
>
> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>
>
> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>
> System returned to ROM by power-on
>
> System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>
> Last reload type: Normal Reload
>
>
>
> This product contains cryptographic features and is subject to United
>
> States and local country laws governing import, export, transfer and
>
> use. Delivery of Cisco cryptographic products does not imply
>
> third-party authority to import, export, distribute or use encryption.
>
> Importers, exporters, distributors and users are responsible for
>
> compliance with U.S. and local country laws. By using this product you
>
> agree to comply with applicable laws and regulations. If you are unable
>
> to comply with U.S. and local laws, return this product immediately.
>
>
> A summary of U.S. laws governing Cisco cryptographic products may be found
> at:
> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>
> If you require further assistance please contact us by sending email to
>
> export at cisco.com.
>
>
> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>
> Processor board ID FTX1237A1T0
>
> 2 Gigabit Ethernet interfaces
>
> 2 Channelized T1/PRI ports
>
> 1 Virtual Private Network (VPN) Module
>
> DRAM configuration is 64 bits wide with parity enabled.
>
> 479K bytes of NVRAM.
>
> 500472K bytes of ATA System CompactFlash (Read/Write)
>
>
>
> License Info:
>
>
> License UDI:
>
>
> -------------------------------------------------
>
> Device# PID SN
>
> Sent from my mobile device
>
> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com>
> wrote:
>
> What version of code are you running on the CUBE?
>
> Sent from my iPhone
>
> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> Hello
>
> I have an issue when users are connected to a call and hit the mobility
> soft key button on 9971 phones when a call is active, the phone system
> rings on the mobile number configured in the system. When they pick up the
> the mobile number it just plays what sounds like hold music on both ends of
> the call (I believe this music is coming from cucm but I haven't heard it
> before) instead of providing 2 way voice.
>
> In another senario with what I believe is the same issue. If a user picks
> up on there cell phone first (using single number reach) opposed to the
> deskphone the call is connected with 2 way voice and no issues exist. If
> the user then hangs up his cell phone with the intent to take the call on
> his deskphone the calling party starts hearing the hold music. Once the
> user picks up the call on his deskphone he hears nothing but the calling
> party is still hearing the hold music. It is not working as intended where
> 2 way voice happens once the user hangs up his mobile phone and picks up on
> his deskphone 2 way voice should happen.
>
> My topology is as follows..
>
>
> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>
> Calls are sent back out the SIP trunk to the ITSP when using mobile
> connect/snr.
>
> Does anyone have any ideas how I can make 2 way voice happen instead of
> the hold music when the calls are picked up?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
-------------- next part --------------
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voip/attachments/20130114/0e759021/attachment.html>

From kennethwhayes at gmail.com Mon Jan 14 18:09:07 2013


From: kennethwhayes at gmail.com (Kenneth Hayes)
Date: Mon, 14 Jan 2013 18:09:07 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
Message-ID: <-5635080451137355987@unknownmsgid>

Have you tried different audio codecs?

Sent from my iPhone

On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:

Ryan (sorry I forgot to reply to all)

Thanks for the Reply


Oddly enough we are.
This probably has something to do with MOH in general?
Internally when I user puts another user on hold everything works. No MOH
plays and they can hold and unhold the call just fine.
I tested calling from an external number. Once I put the external caller
on hold the MOH played but I was unable to resume the call. When I hit
resume on the deskphone the MOH still played to the external caller and
there was no sound on the deskphone.

On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com> wrote:

> Do you get similar behavior if you just hold and resume the call outside
> SNR features?
>
> -Ryan
>
> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> Using keyboard-interactive authentication.
>
> Password:
>
>
> Cisco3825#
>
> Cisco3825#sh ver
>
> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M),
> Version 15.1
> (4)M5, RELEASE SOFTWARE (fc1)
>
> Technical Support: http://www.cisco.com/techsupport
> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>
> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>
>
> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>
>
> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>
> System returned to ROM by power-on
>
> System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>
> Last reload type: Normal Reload
>
>
>
> This product contains cryptographic features and is subject to United
>
> States and local country laws governing import, export, transfer and
>
> use. Delivery of Cisco cryptographic products does not imply
>
> third-party authority to import, export, distribute or use encryption.
>
> Importers, exporters, distributors and users are responsible for
>
> compliance with U.S. and local country laws. By using this product you
>
> agree to comply with applicable laws and regulations. If you are unable
>
> to comply with U.S. and local laws, return this product immediately.
>
>
> A summary of U.S. laws governing Cisco cryptographic products may be found
> at:
> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>
> If you require further assistance please contact us by sending email to
>
> export at cisco.com.
>
>
> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>
> Processor board ID FTX1237A1T0
>
> 2 Gigabit Ethernet interfaces
>
> 2 Channelized T1/PRI ports
>
> 1 Virtual Private Network (VPN) Module
>
> DRAM configuration is 64 bits wide with parity enabled.
>
> 479K bytes of NVRAM.
>
> 500472K bytes of ATA System CompactFlash (Read/Write)
>
>
>
> License Info:
>
>
> License UDI:
>
>
> -------------------------------------------------
>
> Device# PID SN
>
> Sent from my mobile device
>
> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com>
> wrote:
>
> What version of code are you running on the CUBE?
>
> Sent from my iPhone
>
> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> Hello
>
> I have an issue when users are connected to a call and hit the mobility
> soft key button on 9971 phones when a call is active, the phone system
> rings on the mobile number configured in the system. When they pick up the
> the mobile number it just plays what sounds like hold music on both ends of
> the call (I believe this music is coming from cucm but I haven't heard it
> before) instead of providing 2 way voice.
>
> In another senario with what I believe is the same issue. If a user picks
> up on there cell phone first (using single number reach) opposed to the
> deskphone the call is connected with 2 way voice and no issues exist. If
> the user then hangs up his cell phone with the intent to take the call on
> his deskphone the calling party starts hearing the hold music. Once the
> user picks up the call on his deskphone he hears nothing but the calling
> party is still hearing the hold music. It is not working as intended where
> 2 way voice happens once the user hangs up his mobile phone and picks up on
> his deskphone 2 way voice should happen.
>
> My topology is as follows..
>
>
> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>
> Calls are sent back out the SIP trunk to the ITSP when using mobile
> connect/snr.
>
> Does anyone have any ideas how I can make 2 way voice happen instead of
> the hold music when the calls are picked up?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
-------------- next part --------------
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From dane.newman at gmail.com Mon Jan 14 18:12:55 2013


From: dane.newman at gmail.com (Dane Newman)
Date: Mon, 14 Jan 2013 18:12:55 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <-5635080451137355987@unknownmsgid>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
Message-ID: <CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>

My ITSP will only support 711ulaw for me currently I believe. They hard
coded it with me when I was initially setting it up.

Do you think this could be a codec issue? How would I go about identifying
if it is?

Dane

On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <kennethwhayes at gmail.com>wrote:

> Have you tried different audio codecs?


>
> Sent from my iPhone
>
> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> Ryan (sorry I forgot to reply to all)
>
> Thanks for the Reply
> Oddly enough we are.
> This probably has something to do with MOH in general?
>
> Internally when I user puts another user on hold everything works. No MOH
> plays and they can hold and unhold the call just fine.
> I tested calling from an external number. Once I put the external caller
> on hold the MOH played but I was unable to resume the call. When I hit
> resume on the deskphone the MOH still played to the external caller and
> there was no sound on the deskphone.
>
> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>
>> Do you get similar behavior if you just hold and resume the call outside
>> SNR features?
>>
>> -Ryan
>>
>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Using keyboard-interactive authentication.
>>
>> Password:
>>
>>
>> Cisco3825#
>>
>> Cisco3825#sh ver
>>
>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M),
>> Version 15.1
>> (4)M5, RELEASE SOFTWARE (fc1)
>>
>> Technical Support: http://www.cisco.com/techsupport
>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>>
>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>
>>
>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>
>>
>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>>
>> System returned to ROM by power-on
>>
>> System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>>
>> Last reload type: Normal Reload
>>
>>
>>
>> This product contains cryptographic features and is subject to United
>>
>> States and local country laws governing import, export, transfer and
>>
>> use. Delivery of Cisco cryptographic products does not imply
>>
>> third-party authority to import, export, distribute or use encryption.
>>
>> Importers, exporters, distributors and users are responsible for
>>
>> compliance with U.S. and local country laws. By using this product you
>>
>> agree to comply with applicable laws and regulations. If you are unable
>>
>> to comply with U.S. and local laws, return this product immediately.
>>
>>
>> A summary of U.S. laws governing Cisco cryptographic products may be
>> found at:
>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>
>> If you require further assistance please contact us by sending email to
>>
>> export at cisco.com.
>>
>>
>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>>
>> Processor board ID FTX1237A1T0
>>
>> 2 Gigabit Ethernet interfaces
>>
>> 2 Channelized T1/PRI ports
>>
>> 1 Virtual Private Network (VPN) Module
>>
>> DRAM configuration is 64 bits wide with parity enabled.
>>
>> 479K bytes of NVRAM.
>>
>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>
>>
>>
>> License Info:
>>
>>
>> License UDI:
>>
>>
>> -------------------------------------------------
>>
>> Device# PID SN
>>
>> Sent from my mobile device
>>
>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com>
>> wrote:
>>
>> What version of code are you running on the CUBE?
>>
>> Sent from my iPhone
>>
>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Hello
>>
>> I have an issue when users are connected to a call and hit the mobility
>> soft key button on 9971 phones when a call is active, the phone system
>> rings on the mobile number configured in the system. When they pick up the
>> the mobile number it just plays what sounds like hold music on both ends of
>> the call (I believe this music is coming from cucm but I haven't heard it
>> before) instead of providing 2 way voice.
>>
>> In another senario with what I believe is the same issue. If a user picks
>> up on there cell phone first (using single number reach) opposed to the
>> deskphone the call is connected with 2 way voice and no issues exist. If
>> the user then hangs up his cell phone with the intent to take the call on
>> his deskphone the calling party starts hearing the hold music. Once the
>> user picks up the call on his deskphone he hears nothing but the calling
>> party is still hearing the hold music. It is not working as intended where
>> 2 way voice happens once the user hangs up his mobile phone and picks up on
>> his deskphone 2 way voice should happen.
>>
>> My topology is as follows..
>>
>>
>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>
>> Calls are sent back out the SIP trunk to the ITSP when using mobile
>> connect/snr.
>>
>> Does anyone have any ideas how I can make 2 way voice happen instead of
>> the hold music when the calls are picked up?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>
-------------- next part --------------
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From kennethwhayes at gmail.com Mon Jan 14 18:20:11 2013


From: kennethwhayes at gmail.com (Kenneth Hayes)
Date: Mon, 14 Jan 2013 18:20:11 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
<CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
Message-ID: <5095685926238469350@unknownmsgid>

Well have you tried debugging the codec to see if it has any errors!

Sent from my iPhone

On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com> wrote:

My ITSP will only support 711ulaw for me currently I believe. They hard
coded it with me when I was initially setting it up.

Do you think this could be a codec issue? How would I go about identifying
if it is?

Dane

On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <kennethwhayes at gmail.com>wrote:

> Have you tried different audio codecs?


>
> Sent from my iPhone
>
> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> Ryan (sorry I forgot to reply to all)
>
> Thanks for the Reply
> Oddly enough we are.
> This probably has something to do with MOH in general?
>
> Internally when I user puts another user on hold everything works. No MOH
> plays and they can hold and unhold the call just fine.
> I tested calling from an external number. Once I put the external caller
> on hold the MOH played but I was unable to resume the call. When I hit
> resume on the deskphone the MOH still played to the external caller and
> there was no sound on the deskphone.
>
> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>
>> Do you get similar behavior if you just hold and resume the call outside
>> SNR features?
>>
>> -Ryan
>>
>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Using keyboard-interactive authentication.
>>
>> Password:
>>
>>
>> Cisco3825#
>>
>> Cisco3825#sh ver
>>
>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M),
>> Version 15.1
>> (4)M5, RELEASE SOFTWARE (fc1)
>>
>> Technical Support: http://www.cisco.com/techsupport
>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>>
>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>
>>
>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>
>>
>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>>
>> System returned to ROM by power-on
>>
>> System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>>
>> Last reload type: Normal Reload
>>
>>
>>
>> This product contains cryptographic features and is subject to United
>>
>> States and local country laws governing import, export, transfer and
>>
>> use. Delivery of Cisco cryptographic products does not imply
>>
>> third-party authority to import, export, distribute or use encryption.
>>
>> Importers, exporters, distributors and users are responsible for
>>
>> compliance with U.S. and local country laws. By using this product you
>>
>> agree to comply with applicable laws and regulations. If you are unable
>>
>> to comply with U.S. and local laws, return this product immediately.
>>
>>
>> A summary of U.S. laws governing Cisco cryptographic products may be
>> found at:
>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>
>> If you require further assistance please contact us by sending email to
>>
>> export at cisco.com.
>>
>>
>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>>
>> Processor board ID FTX1237A1T0
>>
>> 2 Gigabit Ethernet interfaces
>>
>> 2 Channelized T1/PRI ports
>>
>> 1 Virtual Private Network (VPN) Module
>>
>> DRAM configuration is 64 bits wide with parity enabled.
>>
>> 479K bytes of NVRAM.
>>
>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>
>>
>>
>> License Info:
>>
>>
>> License UDI:
>>
>>
>> -------------------------------------------------
>>
>> Device# PID SN
>>
>> Sent from my mobile device
>>
>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com>
>> wrote:
>>
>> What version of code are you running on the CUBE?
>>
>> Sent from my iPhone
>>
>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Hello
>>
>> I have an issue when users are connected to a call and hit the mobility
>> soft key button on 9971 phones when a call is active, the phone system
>> rings on the mobile number configured in the system. When they pick up the
>> the mobile number it just plays what sounds like hold music on both ends of
>> the call (I believe this music is coming from cucm but I haven't heard it
>> before) instead of providing 2 way voice.
>>
>> In another senario with what I believe is the same issue. If a user picks
>> up on there cell phone first (using single number reach) opposed to the
>> deskphone the call is connected with 2 way voice and no issues exist. If
>> the user then hangs up his cell phone with the intent to take the call on
>> his deskphone the calling party starts hearing the hold music. Once the
>> user picks up the call on his deskphone he hears nothing but the calling
>> party is still hearing the hold music. It is not working as intended where
>> 2 way voice happens once the user hangs up his mobile phone and picks up on
>> his deskphone 2 way voice should happen.
>>
>> My topology is as follows..
>>
>>
>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>
>> Calls are sent back out the SIP trunk to the ITSP when using mobile
>> connect/snr.
>>
>> Does anyone have any ideas how I can make 2 way voice happen instead of
>> the hold music when the calls are picked up?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>
-------------- next part --------------
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voip/attachments/20130114/3b4877c3/attachment.html>

From kennethwhayes at gmail.com Mon Jan 14 18:20:43 2013


From: kennethwhayes at gmail.com (Kenneth Hayes)
Date: Mon, 14 Jan 2013 18:20:43 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
<CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
Message-ID: <1680820783548761203@unknownmsgid>

Have you also restarted the Cisco IP Media Services?

Sent from my iPhone

On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com> wrote:

My ITSP will only support 711ulaw for me currently I believe. They hard
coded it with me when I was initially setting it up.

Do you think this could be a codec issue? How would I go about identifying
if it is?

Dane

On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <kennethwhayes at gmail.com>wrote:

> Have you tried different audio codecs?


>
> Sent from my iPhone
>
> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> Ryan (sorry I forgot to reply to all)
>
> Thanks for the Reply
> Oddly enough we are.
> This probably has something to do with MOH in general?
>
> Internally when I user puts another user on hold everything works. No MOH
> plays and they can hold and unhold the call just fine.
> I tested calling from an external number. Once I put the external caller
> on hold the MOH played but I was unable to resume the call. When I hit
> resume on the deskphone the MOH still played to the external caller and
> there was no sound on the deskphone.
>
> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>
>> Do you get similar behavior if you just hold and resume the call outside
>> SNR features?
>>
>> -Ryan
>>
>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Using keyboard-interactive authentication.
>>
>> Password:
>>
>>
>> Cisco3825#
>>
>> Cisco3825#sh ver
>>
>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M),
>> Version 15.1
>> (4)M5, RELEASE SOFTWARE (fc1)
>>
>> Technical Support: http://www.cisco.com/techsupport
>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>>
>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>
>>
>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>
>>
>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>>
>> System returned to ROM by power-on
>>
>> System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>>
>> Last reload type: Normal Reload
>>
>>
>>
>> This product contains cryptographic features and is subject to United
>>
>> States and local country laws governing import, export, transfer and
>>
>> use. Delivery of Cisco cryptographic products does not imply
>>
>> third-party authority to import, export, distribute or use encryption.
>>
>> Importers, exporters, distributors and users are responsible for
>>
>> compliance with U.S. and local country laws. By using this product you
>>
>> agree to comply with applicable laws and regulations. If you are unable
>>
>> to comply with U.S. and local laws, return this product immediately.
>>
>>
>> A summary of U.S. laws governing Cisco cryptographic products may be
>> found at:
>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>
>> If you require further assistance please contact us by sending email to
>>
>> export at cisco.com.
>>
>>
>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>>
>> Processor board ID FTX1237A1T0
>>
>> 2 Gigabit Ethernet interfaces
>>
>> 2 Channelized T1/PRI ports
>>
>> 1 Virtual Private Network (VPN) Module
>>
>> DRAM configuration is 64 bits wide with parity enabled.
>>
>> 479K bytes of NVRAM.
>>
>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>
>>
>>
>> License Info:
>>
>>
>> License UDI:
>>
>>
>> -------------------------------------------------
>>
>> Device# PID SN
>>
>> Sent from my mobile device
>>
>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com>
>> wrote:
>>
>> What version of code are you running on the CUBE?
>>
>> Sent from my iPhone
>>
>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Hello
>>
>> I have an issue when users are connected to a call and hit the mobility
>> soft key button on 9971 phones when a call is active, the phone system
>> rings on the mobile number configured in the system. When they pick up the
>> the mobile number it just plays what sounds like hold music on both ends of
>> the call (I believe this music is coming from cucm but I haven't heard it
>> before) instead of providing 2 way voice.
>>
>> In another senario with what I believe is the same issue. If a user picks
>> up on there cell phone first (using single number reach) opposed to the
>> deskphone the call is connected with 2 way voice and no issues exist. If
>> the user then hangs up his cell phone with the intent to take the call on
>> his deskphone the calling party starts hearing the hold music. Once the
>> user picks up the call on his deskphone he hears nothing but the calling
>> party is still hearing the hold music. It is not working as intended where
>> 2 way voice happens once the user hangs up his mobile phone and picks up on
>> his deskphone 2 way voice should happen.
>>
>> My topology is as follows..
>>
>>
>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>
>> Calls are sent back out the SIP trunk to the ITSP when using mobile
>> connect/snr.
>>
>> Does anyone have any ideas how I can make 2 way voice happen instead of
>> the hold music when the calls are picked up?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>
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From dane.newman at gmail.com Mon Jan 14 18:40:30 2013


From: dane.newman at gmail.com (Dane Newman)
Date: Mon, 14 Jan 2013 18:40:30 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <1680820783548761203@unknownmsgid>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
<CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
<1680820783548761203@unknownmsgid>
Message-ID: <CAL-DCK0NHxrOgk4RM=CX_absw=Kc2+aqYLOzR-9Oc3_CGEvsEQ@mail.gmail.com>

*Hello Kenneth*
**
*I have restarted both CUCM servers so this should have restarted the
services when the utils system restart happened*
**

*on my router I see I am using g711 from the debug *


**
*I ran a debug voip ccapi inout *
**
*I connected a call calling from an external number to a DiD inside of my
system. Once the call was connected I put the call on hold and the
following debug came out..the music on hold played for the external caller*

*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:


Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.783:
//12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742, Xmit Function=0x64204BAC
*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12742
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12741
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=96, Call Id=12742
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.839:
//12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741, Xmit Function=0x64204BAC
*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12741
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12742
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
Cisco3825#
Cisco3825#
Cisco3825#

*I then after that took off the hold and the following debug came out. The
call on the PSDN side still played the hold music while there was no voice
on the deskphone side.*

*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:


Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.783:
//12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742, Xmit Function=0x64204BAC
*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12742
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12741
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=96, Call Id=12742
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.839:
//12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741, Xmit Function=0x64204BAC
*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12741
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12742
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
Cisco3825#
Cisco3825#
Cisco3825#

On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <kennethwhayes at gmail.com>wrote:

> Have you also restarted the Cisco IP Media Services?


>
> Sent from my iPhone
>
> On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> My ITSP will only support 711ulaw for me currently I believe. They hard
> coded it with me when I was initially setting it up.
>
> Do you think this could be a codec issue? How would I go about
> identifying if it is?
>
> Dane
>
> On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <kennethwhayes at gmail.com>wrote:
>
>> Have you tried different audio codecs?
>>
>> Sent from my iPhone
>>
>> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Ryan (sorry I forgot to reply to all)
>>
>> Thanks for the Reply
>> Oddly enough we are.
>> This probably has something to do with MOH in general?
>>
>> Internally when I user puts another user on hold everything works. No MOH
>> plays and they can hold and unhold the call just fine.
>> I tested calling from an external number. Once I put the external
>> caller on hold the MOH played but I was unable to resume the call. When I
>> hit resume on the deskphone the MOH still played to the external caller and
>> there was no sound on the deskphone.
>>
>> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>>
>>> Do you get similar behavior if you just hold and resume the call outside
>>> SNR features?
>>>
>>> -Ryan
>>>
>>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>> Using keyboard-interactive authentication.
>>>
>>> Password:
>>>
>>>
>>> Cisco3825#
>>>
>>> Cisco3825#sh ver
>>>
>>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M),
>>> Version 15.1
>>> (4)M5, RELEASE SOFTWARE (fc1)
>>>
>>> Technical Support: http://www.cisco.com/techsupport
>>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>>>
>>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>>
>>>
>>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>>
>>>
>>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>>>
>>> System returned to ROM by power-on
>>>
>>> System image file is
>>> "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>>> Last reload type: Normal Reload
>>>
>>>
>>>
>>> This product contains cryptographic features and is subject to United
>>>
>>> States and local country laws governing import, export, transfer and
>>>
>>> use. Delivery of Cisco cryptographic products does not imply
>>>
>>> third-party authority to import, export, distribute or use encryption.
>>>
>>> Importers, exporters, distributors and users are responsible for
>>>
>>> compliance with U.S. and local country laws. By using this product you
>>>
>>> agree to comply with applicable laws and regulations. If you are unable
>>>
>>> to comply with U.S. and local laws, return this product immediately.
>>>
>>>
>>> A summary of U.S. laws governing Cisco cryptographic products may be
>>> found at:
>>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>>
>>> If you require further assistance please contact us by sending email to
>>>
>>> export at cisco.com.
>>>
>>>
>>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>>>
>>> Processor board ID FTX1237A1T0
>>>
>>> 2 Gigabit Ethernet interfaces
>>>
>>> 2 Channelized T1/PRI ports
>>>
>>> 1 Virtual Private Network (VPN) Module
>>>
>>> DRAM configuration is 64 bits wide with parity enabled.
>>>
>>> 479K bytes of NVRAM.
>>>
>>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>>
>>>
>>>
>>> License Info:
>>>
>>>
>>> License UDI:
>>>
>>>
>>> -------------------------------------------------
>>>
>>> Device# PID SN
>>>
>>> Sent from my mobile device
>>>
>>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com>
>>> wrote:
>>>
>>> What version of code are you running on the CUBE?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>> Hello
>>>
>>> I have an issue when users are connected to a call and hit the mobility
>>> soft key button on 9971 phones when a call is active, the phone system
>>> rings on the mobile number configured in the system. When they pick up the
>>> the mobile number it just plays what sounds like hold music on both ends of
>>> the call (I believe this music is coming from cucm but I haven't heard it
>>> before) instead of providing 2 way voice.
>>>
>>> In another senario with what I believe is the same issue. If a user
>>> picks up on there cell phone first (using single number reach) opposed to
>>> the deskphone the call is connected with 2 way voice and no issues exist.
>>> If the user then hangs up his cell phone with the intent to take the call
>>> on his deskphone the calling party starts hearing the hold music. Once the
>>> user picks up the call on his deskphone he hears nothing but the calling
>>> party is still hearing the hold music. It is not working as intended where
>>> 2 way voice happens once the user hangs up his mobile phone and picks up on
>>> his deskphone 2 way voice should happen.
>>>
>>> My topology is as follows..
>>>
>>>
>>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>>
>>> Calls are sent back out the SIP trunk to the ITSP when using mobile
>>> connect/snr.
>>>
>>> Does anyone have any ideas how I can make 2 way voice happen instead of
>>> the hold music when the calls are picked up?
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>>
>>
>
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From svoll.voip at gmail.com Mon Jan 14 18:45:41 2013


From: svoll.voip at gmail.com (Scott Voll)
Date: Mon, 14 Jan 2013 15:45:41 -0800
Subject: [cisco-voip] 2960s-48fps-l flex stack
Message-ID: <CAHgd+3_H1PcGKB06TWvwZOkDMFm7736Tfw2Od0Pb8tPSB3Mr7g@mail.gmail.com>

I have a 2960s-48fps-l and when I inserted the flex stack module I get:

%PLATFORM-6-FLEXSTACK_UNSUPPORTED_MODULE: Unsupported FlexStack module


inserted in Switch 1. C2960S-F-STACK
Is this not supported? I'm running 15.0.2se1. How do I get it talking to
the other switches?

TIA

Scott
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From joshua.morgan at gmail.com Mon Jan 14 19:00:12 2013


From: joshua.morgan at gmail.com (Joshua Morgan)
Date: Tue, 15 Jan 2013 11:00:12 +1100
Subject: [cisco-voip] 2960s-48fps-l flex stack
In-Reply-To: <CAHgd+3_H1PcGKB06TWvwZOkDMFm7736Tfw2Od0Pb8tPSB3Mr7g@mail.gmail.com>
References: <CAHgd+3_H1PcGKB06TWvwZOkDMFm7736Tfw2Od0Pb8tPSB3Mr7g@mail.gmail.com>
Message-ID: <CAP-90TnrJqcQcdSj4TgqS-YV2bSmf39OGPQiwgVgB60mgaRVhA@mail.gmail.com>

That module may only be for 2960SF series switches (FastEthernet version).

On Tue, Jan 15, 2013 at 10:45 AM, Scott Voll <svoll.voip at gmail.com> wrote:

> I have a 2960s-48fps-l and when I inserted the flex stack module I get:
>
> %PLATFORM-6-FLEXSTACK_UNSUPPORTED_MODULE: Unsupported FlexStack module
> inserted in Switch 1. C2960S-F-STACK
>
> Is this not supported? I'm running 15.0.2se1. How do I get it talking to
> the other switches?
>
> TIA
>
> Scott
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From mikeeo at msn.com Mon Jan 14 21:01:21 2013


From: mikeeo at msn.com (Mike )
Date: Mon, 14 Jan 2013 21:01:21 -0500
Subject: [cisco-voip] Design question CTI or DN?
Message-ID: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>

I have about 1500 voice mail only users that I have in Unity connections and
I was wondering that the best way to get their DIDs into unity cx. I was
thinking I can create CTI ports send them all to voicemail. The problem is
the customer owns the whole block 0000-9999 and the DIDs are all over the
place or I'd create a wild card hunt pilot.
Any ideas?

Thanks,

Mike

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From VanMarenNP at ldschurch.org Mon Jan 14 22:19:28 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Tue, 15 Jan 2013 03:19:28 +0000
Subject: [cisco-voip] Design question CTI or DN?
In-Reply-To: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>
References: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E7561D@W12112.ldschurch.org>

What needs to happen to an assigned DIDs? If it's ok for them to go to a call


handler, then just have a wildcard for the whole block that sends anything un
assigned to Unity, then it will match their mailbox if it can or they can go to a
callhandler.

Or build DNs for each number, not a ton of fun but can be made easier with AXL or
BAT, the real trouble is keeping them up to date.

-Nate

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Mike
Sent: Monday, January 14, 2013 7:01 PM
To: 'cisco voip'
Subject: [cisco-voip] Design question CTI or DN?

I have about 1500 voice mail only users that I have in Unity connections and I was
wondering that the best way to get their DIDs into unity cx. I was thinking I can
create CTI ports send them all to voicemail. The problem is the customer owns the
whole block 0000-9999 and the DIDs are all over the place or I'd create a wild card
hunt pilot.

Any ideas?

Thanks,
Mike

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

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From mikeeo at msn.com Mon Jan 14 22:24:38 2013


From: mikeeo at msn.com (Mike )
Date: Mon, 14 Jan 2013 22:24:38 -0500
Subject: [cisco-voip] Design question CTI or DN?
In-Reply-To: <2F143E71016CA34C924BF4C33AEF211056E7561D@W12112.ldschurch.org>
References: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E7561D@W12112.ldschurch.org>
Message-ID: <BLU0-SMTP35978042C1DFCB75E7430F3C52D0@phx.gbl>

I was thinking of doing a wildcard 1XXX, 2XXX, 3XXX etc , but that might get
ugly. Its weird they don't want to use SNR they just want the call to go to
voicemail. I don't think you can BAT in DNs only CTI ports right?

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nate VanMaren
Sent: Monday, January 14, 2013 10:19 PM
To: Mike ; 'cisco voip'
Subject: Re: [cisco-voip] Design question CTI or DN?

What needs to happen to an assigned DIDs? If it's ok for them to go to a


call handler, then just have a wildcard for the whole block that sends
anything un assigned to Unity, then it will match their mailbox if it can or
they can go to a callhandler.

Or build DNs for each number, not a ton of fun but can be made easier with
AXL or BAT, the real trouble is keeping them up to date.

-Nate

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike
Sent: Monday, January 14, 2013 7:01 PM
To: 'cisco voip'
Subject: [cisco-voip] Design question CTI or DN?

I have about 1500 voice mail only users that I have in Unity connections and
I was wondering that the best way to get their DIDs into unity cx. I was
thinking I can create CTI ports send them all to voicemail. The problem is
the customer owns the whole block 0000-9999 and the DIDs are all over the
place or I'd create a wild card hunt pilot.
Any ideas?

Thanks,

Mike

NOTICE: This email message is for the sole use of the intended recipient(s)
and may contain confidential and privileged information. Any unauthorized
review, use, disclosure or distribution is prohibited. If you are not the
intended recipient, please contact the sender by reply email and destroy all
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From VanMarenNP at ldschurch.org Mon Jan 14 22:26:34 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Tue, 15 Jan 2013 03:26:34 +0000
Subject: [cisco-voip] Design question CTI or DN?
In-Reply-To: <BLU0-SMTP35978042C1DFCB75E7430F3C52D0@phx.gbl>
References: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E7561D@W12112.ldschurch.org>
<BLU0-SMTP35978042C1DFCB75E7430F3C52D0@phx.gbl>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E75651@W12112.ldschurch.org>

You don't need DNs on a device anymore for them to be active. They just need to be
marked as "active"

From: Mike [mailto:mikeeo at msn.com]


Sent: Monday, January 14, 2013 8:25 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

I was thinking of doing a wildcard 1XXX, 2XXX, 3XXX etc , but that might get ugly.
Its weird they don't want to use SNR they just want the call to go to voicemail. I
don't think you can BAT in DNs only CTI ports right?

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nate
VanMaren
Sent: Monday, January 14, 2013 10:19 PM
To: Mike ; 'cisco voip'
Subject: Re: [cisco-voip] Design question CTI or DN?

What needs to happen to an assigned DIDs? If it's ok for them to go to a call


handler, then just have a wildcard for the whole block that sends anything un
assigned to Unity, then it will match their mailbox if it can or they can go to a
callhandler.

Or build DNs for each number, not a ton of fun but can be made easier with AXL or
BAT, the real trouble is keeping them up to date.

-Nate

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike
Sent: Monday, January 14, 2013 7:01 PM
To: 'cisco voip'
Subject: [cisco-voip] Design question CTI or DN?

I have about 1500 voice mail only users that I have in Unity connections and I was
wondering that the best way to get their DIDs into unity cx. I was thinking I can
create CTI ports send them all to voicemail. The problem is the customer owns the
whole block 0000-9999 and the DIDs are all over the place or I'd create a wild card
hunt pilot.

Any ideas?

Thanks,
Mike

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
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From jonvoip at gmail.com Mon Jan 14 22:28:54 2013


From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 14 Jan 2013 21:28:54 -0600
Subject: [cisco-voip] Can't download List.xml via TFTP... file not found...
Message-ID: <CAPLPVZip4H=A2v8u16F71TnfMS32CZgKe-4WWuJMRhMpcK7JYw@mail.gmail.com>

Trying to deploy images to 7970/71, 7975 and 7965s...

Cisco 7970/71s work fine, 7975s and 65s fail... with a File Not Found,
confirmed repeatedly that the file is there; logs show file being there,
but no signed:

1209: NOT 19:25:02.412509 tftpClient: tftp request rcv'd from


/usr/tmp/tftp, srcFile = Desktops/320x212x16/Large.png, dstFile =
/flash0/F-1580749268 max size = 550001
1210: NOT 19:25:02.430896 tftpClient: auth server - tftpList[0] =
::ffff:172.20.1.100
1211: NOT 19:25:02.431523 tftpClient: look up server - 0
1212: NOT 19:25:02.433489 SECD: lookupCTL: TFTP SRVR secure
1213: NOT 19:25:02.436291 tftpClient: secVal = 0x9
1214: NOT 19:25:02.437012 tftpClient: ::ffff:172.20.1.100 is a secure server
1215: NOT 19:25:02.437538 tftpClient: retval = SRVR_SECURE
1216: NOT 19:25:02.438074 tftpClient: Secure file requested
1217: NOT 19:25:02.438619 tftpClient: authenticated file approved -
add .sgn -- Desktops/320x212x16/HLarge.png.sgn
1218: NOT 19:25:02.470339 TFTP: [18]:Requesting
Desktops/320x212x16/HTALarge.png.sgn from 172.20.1.100 with size limit
of 550001
1219: NOT 19:25:02.585485 TFTP: [18]:Error --> File not found
1220: NOT 19:25:02.593573 SYSMSG: pid 18 (/sbin/tftpd) Normal Exit, status = 2
1221: INF 19:25:02.593616 runtime = 0.150 secs

Looks like it finds the file, wants a signed one, appears to convert
it, then fails to find it.

How do I generate the secure file?

Jonathan
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From mikeeo at msn.com Mon Jan 14 22:30:15 2013


From: mikeeo at msn.com (Mike )
Date: Mon, 14 Jan 2013 22:30:15 -0500
Subject: [cisco-voip] Design question CTI or DN?
In-Reply-To: <2F143E71016CA34C924BF4C33AEF211056E75651@W12112.ldschurch.org>
References: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E7561D@W12112.ldschurch.org>
<BLU0-SMTP35978042C1DFCB75E7430F3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E75651@W12112.ldschurch.org>
Message-ID: <BLU0-SMTP35668CEDEABCD8D477E09FEC52D0@phx.gbl>

True, but I never seen a way to BAT in DNs.

From: Nate VanMaren [mailto:VanMarenNP at ldschurch.org]


Sent: Monday, January 14, 2013 10:27 PM
To: Mike ; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

You don't need DNs on a device anymore for them to be active. They just
need to be marked as "active"
From: Mike [mailto:mikeeo at msn.com]
Sent: Monday, January 14, 2013 8:25 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

I was thinking of doing a wildcard 1XXX, 2XXX, 3XXX etc , but that might get
ugly. Its weird they don't want to use SNR they just want the call to go to
voicemail. I don't think you can BAT in DNs only CTI ports right?

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nate VanMaren
Sent: Monday, January 14, 2013 10:19 PM
To: Mike ; 'cisco voip'
Subject: Re: [cisco-voip] Design question CTI or DN?

What needs to happen to an assigned DIDs? If it's ok for them to go to a


call handler, then just have a wildcard for the whole block that sends
anything un assigned to Unity, then it will match their mailbox if it can or
they can go to a callhandler.

Or build DNs for each number, not a ton of fun but can be made easier with
AXL or BAT, the real trouble is keeping them up to date.

-Nate

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike
Sent: Monday, January 14, 2013 7:01 PM
To: 'cisco voip'
Subject: [cisco-voip] Design question CTI or DN?

I have about 1500 voice mail only users that I have in Unity connections and
I was wondering that the best way to get their DIDs into unity cx. I was
thinking I can create CTI ports send them all to voicemail. The problem is
the customer owns the whole block 0000-9999 and the DIDs are all over the
place or I'd create a wild card hunt pilot.

Any ideas?

Thanks,
Mike

NOTICE: This email message is for the sole use of the intended recipient(s)
and may contain confidential and privileged information. Any unauthorized
review, use, disclosure or distribution is prohibited. If you are not the
intended recipient, please contact the sender by reply email and destroy all
copies of the original message.

NOTICE: This email message is for the sole use of the intended recipient(s)
and may contain confidential and privileged information. Any unauthorized
review, use, disclosure or distribution is prohibited. If you are not the
intended recipient, please contact the sender by reply email and destroy all
copies of the original message.

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From VanMarenNP at ldschurch.org Mon Jan 14 22:35:17 2013


From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Tue, 15 Jan 2013 03:35:17 +0000
Subject: [cisco-voip] Design question CTI or DN?
In-Reply-To: <BLU0-SMTP35668CEDEABCD8D477E09FEC52D0@phx.gbl>
References: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E7561D@W12112.ldschurch.org>
<BLU0-SMTP35978042C1DFCB75E7430F3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E75651@W12112.ldschurch.org>
<BLU0-SMTP35668CEDEABCD8D477E09FEC52D0@phx.gbl>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E756D6@W12112.ldschurch.org>

Ah, my new best friend is import/export.

It's really just a CSV that does anything.

I haven't done it with DNs, but I just imported all of the permutations of phone
button templates in a few seconds.

Export something, look at the CSV in the TAR, edit it to what you want, and load it
back in.

Thanks,
-Nate

From: Mike [mailto:mikeeo at msn.com]


Sent: Monday, January 14, 2013 8:30 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

True, but I never seen a way to BAT in DNs.


From: Nate VanMaren [mailto:VanMarenNP at ldschurch.org]
Sent: Monday, January 14, 2013 10:27 PM
To: Mike ; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

You don't need DNs on a device anymore for them to be active. They just need to be
marked as "active"

From: Mike [mailto:mikeeo at msn.com]


Sent: Monday, January 14, 2013 8:25 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

I was thinking of doing a wildcard 1XXX, 2XXX, 3XXX etc , but that might get ugly.
Its weird they don't want to use SNR they just want the call to go to voicemail. I
don't think you can BAT in DNs only CTI ports right?

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nate
VanMaren
Sent: Monday, January 14, 2013 10:19 PM
To: Mike ; 'cisco voip'
Subject: Re: [cisco-voip] Design question CTI or DN?

What needs to happen to an assigned DIDs? If it's ok for them to go to a call


handler, then just have a wildcard for the whole block that sends anything un
assigned to Unity, then it will match their mailbox if it can or they can go to a
callhandler.

Or build DNs for each number, not a ton of fun but can be made easier with AXL or
BAT, the real trouble is keeping them up to date.

-Nate

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike
Sent: Monday, January 14, 2013 7:01 PM
To: 'cisco voip'
Subject: [cisco-voip] Design question CTI or DN?

I have about 1500 voice mail only users that I have in Unity connections and I was
wondering that the best way to get their DIDs into unity cx. I was thinking I can
create CTI ports send them all to voicemail. The problem is the customer owns the
whole block 0000-9999 and the DIDs are all over the place or I'd create a wild card
hunt pilot.

Any ideas?

Thanks,
Mike

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.
NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

NOTICE: This email message is for the sole use of the intended recipient(s) and
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disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

-------------- next part --------------


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From mikeeo at msn.com Mon Jan 14 22:38:50 2013


From: mikeeo at msn.com (Mike )
Date: Mon, 14 Jan 2013 22:38:50 -0500
Subject: [cisco-voip] Design question CTI or DN?
In-Reply-To: <2F143E71016CA34C924BF4C33AEF211056E756D6@W12112.ldschurch.org>
References: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E7561D@W12112.ldschurch.org>
<BLU0-SMTP35978042C1DFCB75E7430F3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E75651@W12112.ldschurch.org>
<BLU0-SMTP35668CEDEABCD8D477E09FEC52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E756D6@W12112.ldschurch.org>
Message-ID: <BLU0-SMTP366CE9668B959DD8AD94713C52D0@phx.gbl>

I'll give it a shot thanks!

From: Nate VanMaren [mailto:VanMarenNP at ldschurch.org]


Sent: Monday, January 14, 2013 10:35 PM
To: Mike ; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

Ah, my new best friend is import/export.

It's really just a CSV that does anything.

I haven't done it with DNs, but I just imported all of the permutations of
phone button templates in a few seconds.

Export something, look at the CSV in the TAR, edit it to what you want, and
load it back in.
Thanks,

-Nate

From: Mike [mailto:mikeeo at msn.com]


Sent: Monday, January 14, 2013 8:30 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

True, but I never seen a way to BAT in DNs.

From: Nate VanMaren [mailto:VanMarenNP at ldschurch.org]


Sent: Monday, January 14, 2013 10:27 PM
To: Mike ; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

You don't need DNs on a device anymore for them to be active. They just
need to be marked as "active"

From: Mike [mailto:mikeeo at msn.com]


Sent: Monday, January 14, 2013 8:25 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

I was thinking of doing a wildcard 1XXX, 2XXX, 3XXX etc , but that might get
ugly. Its weird they don't want to use SNR they just want the call to go to
voicemail. I don't think you can BAT in DNs only CTI ports right?

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nate VanMaren
Sent: Monday, January 14, 2013 10:19 PM
To: Mike ; 'cisco voip'
Subject: Re: [cisco-voip] Design question CTI or DN?

What needs to happen to an assigned DIDs? If it's ok for them to go to a


call handler, then just have a wildcard for the whole block that sends
anything un assigned to Unity, then it will match their mailbox if it can or
they can go to a callhandler.
Or build DNs for each number, not a ton of fun but can be made easier with
AXL or BAT, the real trouble is keeping them up to date.

-Nate

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike
Sent: Monday, January 14, 2013 7:01 PM
To: 'cisco voip'
Subject: [cisco-voip] Design question CTI or DN?

I have about 1500 voice mail only users that I have in Unity connections and
I was wondering that the best way to get their DIDs into unity cx. I was
thinking I can create CTI ports send them all to voicemail. The problem is
the customer owns the whole block 0000-9999 and the DIDs are all over the
place or I'd create a wild card hunt pilot.

Any ideas?

Thanks,

Mike

NOTICE: This email message is for the sole use of the intended recipient(s)
and may contain confidential and privileged information. Any unauthorized
review, use, disclosure or distribution is prohibited. If you are not the
intended recipient, please contact the sender by reply email and destroy all
copies of the original message.

NOTICE: This email message is for the sole use of the intended recipient(s)
and may contain confidential and privileged information. Any unauthorized
review, use, disclosure or distribution is prohibited. If you are not the
intended recipient, please contact the sender by reply email and destroy all
copies of the original message.

NOTICE: This email message is for the sole use of the intended recipient(s)
and may contain confidential and privileged information. Any unauthorized
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-------------- next part --------------


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From John.VanLaecke at ghd.com Tue Jan 15 00:36:26 2013


From: John.VanLaecke at ghd.com (John Van Laecke)
Date: Tue, 15 Jan 2013 05:36:26 +0000
Subject: [cisco-voip] Design question CTI or DN?
In-Reply-To: <BLU0-SMTP366CE9668B959DD8AD94713C52D0@phx.gbl>
References: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E7561D@W12112.ldschurch.org>
<BLU0-SMTP35978042C1DFCB75E7430F3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E75651@W12112.ldschurch.org>
<BLU0-SMTP35668CEDEABCD8D477E09FEC52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E756D6@W12112.ldschurch.org>
<BLU0-SMTP366CE9668B959DD8AD94713C52D0@phx.gbl>
Message-ID: <CF2E16827204E249BB155D2C1F5C1EAB1ABAB413@GLB-EXMBX-
001.ghdnet.internal>

In call manager you can make bulk dn's.

Goto
call routing
Directory number
Add new
And you can add a range in and create the call forwards.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Mike
Sent: Tuesday, 15 January 2013 1:39 PM
To: 'Nate VanMaren'; 'cisco voip'
Subject: Re: [cisco-voip] Design question CTI or DN?

I'll give it a shot thanks!

From: Nate VanMaren [mailto:VanMarenNP at ldschurch.org]


Sent: Monday, January 14, 2013 10:35 PM
To: Mike ; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

Ah, my new best friend is import/export.

It's really just a CSV that does anything.

I haven't done it with DNs, but I just imported all of the permutations of phone
button templates in a few seconds.

Export something, look at the CSV in the TAR, edit it to what you want, and load it
back in.

Thanks,
-Nate

From: Mike [mailto:mikeeo at msn.com]


Sent: Monday, January 14, 2013 8:30 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

True, but I never seen a way to BAT in DNs.

From: Nate VanMaren [mailto:VanMarenNP at ldschurch.org]


Sent: Monday, January 14, 2013 10:27 PM
To: Mike ; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

You don't need DNs on a device anymore for them to be active. They just need to be
marked as "active"

From: Mike [mailto:mikeeo at msn.com]


Sent: Monday, January 14, 2013 8:25 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

I was thinking of doing a wildcard 1XXX, 2XXX, 3XXX etc , but that might get ugly.
Its weird they don't want to use SNR they just want the call to go to voicemail. I
don't think you can BAT in DNs only CTI ports right?

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nate
VanMaren
Sent: Monday, January 14, 2013 10:19 PM
To: Mike ; 'cisco voip'
Subject: Re: [cisco-voip] Design question CTI or DN?

What needs to happen to an assigned DIDs? If it's ok for them to go to a call


handler, then just have a wildcard for the whole block that sends anything un
assigned to Unity, then it will match their mailbox if it can or they can go to a
callhandler.

Or build DNs for each number, not a ton of fun but can be made easier with AXL or
BAT, the real trouble is keeping them up to date.

-Nate

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike
Sent: Monday, January 14, 2013 7:01 PM
To: 'cisco voip'
Subject: [cisco-voip] Design question CTI or DN?

I have about 1500 voice mail only users that I have in Unity connections and I was
wondering that the best way to get their DIDs into unity cx. I was thinking I can
create CTI ports send them all to voicemail. The problem is the customer owns the
whole block 0000-9999 and the DIDs are all over the place or I'd create a wild card
hunt pilot.
Any ideas?

Thanks,
Mike

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

_____________________
This e-mail has been scanned for viruses by MessageLabs.

_____________________
This email and all attachments are confidential. For further important information
about emails sent to or from GHD or if you have received this email in error,
please refer to http://www.ghd.com/emaildisclaimer.html .
_____________________
This e-mail has been scanned for viruses by MessageLabs.
-------------- next part --------------
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From ahmed_elnagar at hotmail.com Tue Jan 15 01:39:59 2013


From: ahmed_elnagar at hotmail.com (Ahmed Elnagar)
Date: Tue, 15 Jan 2013 08:39:59 +0200
Subject: [cisco-voip] 9.1 Upgrade Times
In-Reply-To: <2F143E71016CA34C924BF4C33AEF211056E71556@W12112.ldschurch.org>
References: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B2E69@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B5E52@PHANES.helion.local>
<2F143E71016CA34C924BF4C33AEF211056E71556@W12112.ldschurch.org>
Message-ID: <BAY149-ds10D2ABFE4E797176881759872D0@phx.gbl>

I usually make a cluster reboot for CUCM or unity connection before I


upgrade.
Regards,

Ahmed Elnagar | Unified Communication Team Leader | CCIE #24697, Voice

Description: Description: Description: MS Green

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nate VanMaren
Sent: Monday, January 14, 2013 6:33 PM
To: Matthew Loraditch; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 9.1 Upgrade Times

How long had the server been up before the you started the first upgrade?

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matthew Loraditch
Sent: Monday, January 14, 2013 9:27 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 9.1 Upgrade Times

So I figured I'd loop around and let everyone know how things ended up.

Unity Connection never finished the first attempt and wouldn't cancel
either. I had to force a shutdown. After it came back up I tried again. 90
minutes boom. Suffice it to say. I wish had done that sooner.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

<http://twitter.com/heliontech> Twitter |
<http://www.facebook.com/#!/pages/Helion/252157915296> Facebook |
<http://www.heliontechnologies.com/> Website |
<mailto:support at heliontechnologies.com?subject=Technical%20Support%20Request
> Email Support

From: Matthew Loraditch


Sent: Sunday, January 13, 2013 3:06 PM
To: Matthew Loraditch; cisco-voip at puck.nether.net
Subject: RE: 9.1 Upgrade Times

Well my CUCM publisher took almost 13 hours, my sub took 1 hour and change.
the unity connection pub is still running, I think we are on hour 15 of
that.

Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

<http://twitter.com/heliontech> Twitter |
<http://www.facebook.com/#!/pages/Helion/252157915296> Facebook |
<http://www.heliontechnologies.com/> Website |
<mailto:support at heliontechnologies.com?subject=Technical%20Support%20Request
> Email Support

From: cisco-voip-bounces at puck.nether.net


[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matthew Loraditch
Sent: Sunday, January 13, 2013 8:58 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 9.1 Upgrade Times

On a Unity Connection system and CUCM both 8.6.2aSU1 started at 11:45PM and
12:12AM Last night, both are STILL running at 8:45 AM this morning. The
system I am doing this test on has about 60 Phones/Users/VM. These are the
publishers of each install, but I have never had an upgrade take this long,
ever.

I'm now not sure I'll even be able to finish in my window since I haven't
even touched the subscribers yet.

Any clues as to why this is taking so long? I used to disable iothrottle


during an upgrade but that command is deprecated now.
Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

<http://twitter.com/heliontech> Twitter |
<http://www.facebook.com/#!/pages/Helion/252157915296> Facebook |
<http://www.heliontechnologies.com/> Website |
<mailto:support at heliontechnologies.com?subject=Technical%20Support%20Request
> Email Support

NOTICE: This email message is for the sole use of the intended recipient(s)
and may contain confidential and privileged information. Any unauthorized
review, use, disclosure or distribution is prohibited. If you are not the
intended recipient, please contact the sender by reply email and destroy all
copies of the original message.

-------------- next part --------------


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From jbuchanan at presidio.com Tue Jan 15 02:24:54 2013


From: jbuchanan at presidio.com (Buchanan, James)
Date: Tue, 15 Jan 2013 07:24:54 +0000
Subject: [cisco-voip] 9.1 Upgrade Times
In-Reply-To: <BAY149-ds10D2ABFE4E797176881759872D0@phx.gbl>
References: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B2E69@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B5E52@PHANES.helion.local>
<2F143E71016CA34C924BF4C33AEF211056E71556@W12112.ldschurch.org>
<BAY149-ds10D2ABFE4E797176881759872D0@phx.gbl>
Message-ID: <12D6A6A157B44348974E5ED93738628C25369417@HQEXCHMBX03.Presidio.Corp>

Cleaning out log files and CDRs might also be a good idea.

James Buchanan | Sr. Network Engineer


Presidio | www.presidio.com<http://www.presidio.com>
12 Cadillac Drive Suite 130, Brentwood, TN 37027
D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 | jbuchanan at
presidio.com<mailto:jbuchanan at presidio.com>

[Be Secure In The Knowledge]<http://www.presidio.com>

Follow us:

[Follow Presidio on Twitter]<http://www.twitter.com/presidio>

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Tuesday, January 15, 2013 8:40 AM
To: 'Nate VanMaren'; 'Matthew Loraditch'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 9.1 Upgrade Times

I usually make a cluster reboot for CUCM or unity connection before I upgrade.

Regards,
Ahmed Elnagar | Unified Communication Team Leader | CCIE #24697, Voice
[Description: Description: Description: MS Green]

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net]<mailto:
[mailto:cisco-voip-bounces at puck.nether.net]> On Behalf Of Nate VanMaren
Sent: Monday, January 14, 2013 6:33 PM
To: Matthew Loraditch; cisco-voip at puck.nether.net<mailto:cisco-voip at
puck.nether.net>
Subject: Re: [cisco-voip] 9.1 Upgrade Times

How long had the server been up before the you started the first upgrade?

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net]<mailto:
[mailto:cisco-voip-bounces at puck.nether.net]> On Behalf Of Matthew Loraditch
Sent: Monday, January 14, 2013 9:27 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] 9.1 Upgrade Times

So I figured I?d loop around and let everyone know how things ended up.
Unity Connection never finished the first attempt and wouldn?t cancel either. I had
to force a shutdown. After it came back up I tried again? 90 minutes boom. Suffice
it to say. I wish had done that sooner.

Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: Matthew Loraditch


Sent: Sunday, January 13, 2013 3:06 PM
To: Matthew Loraditch; cisco-voip at puck.nether.net<mailto:cisco-voip at
puck.nether.net>
Subject: RE: 9.1 Upgrade Times

Well my CUCM publisher took almost 13 hours, my sub took 1 hour and change? the
unity connection pub is still running, I think we are on hour 15 of that?

Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
Matthew Loraditch
Sent: Sunday, January 13, 2013 8:58 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] 9.1 Upgrade Times

On a Unity Connection system and CUCM both 8.6.2aSU1 started at 11:45PM and 12:12AM
Last night, both are STILL running at 8:45 AM this morning. The system I am doing
this test on has about 60 Phones/Users/VM. These are the publishers of each
install, but I have never had an upgrade take this long, ever.

I?m now not sure I?ll even be able to finish in my window since I haven?t even
touched the subscribers yet.

Any clues as to why this is taking so long? I used to disable iothrottle during an
upgrade but that command is deprecated now.

Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email Support<mailto:support at
heliontechnologies.com?subject=Technical%20Support%20Request>
NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

This message w/attachments (message) is intended solely for the use of the intended
recipient(s) and may contain information that is privileged, confidential or
proprietary. If you are not an intended recipient, please notify the sender, and
then please delete and destroy all copies and attachments. Please be advised that
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From A.L.M.Buxey at lboro.ac.uk Tue Jan 15 02:35:12 2013


From: A.L.M.Buxey at lboro.ac.uk (Alan Buxey)
Date: Tue, 15 Jan 2013 07:35:12 +0000
Subject: [cisco-voip] 2960s-48fps-l flex stack
Message-ID: <A5BBBF24-B07B-4BAD-8C37-FF571A201638@lboro.ac.uk>

I've seen this on ios 15 with flexstack modules. Have you tried removing the module
and re-inserting, or a power cycle with the module installed?

alan

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From jbuchanan at presidio.com Tue Jan 15 08:58:54 2013


From: jbuchanan at presidio.com (Buchanan, James)
Date: Tue, 15 Jan 2013 13:58:54 +0000
Subject: [cisco-voip] 2960s-48fps-l flex stack
In-Reply-To: <A5BBBF24-B07B-4BAD-8C37-FF571A201638@lboro.ac.uk>
References: <A5BBBF24-B07B-4BAD-8C37-FF571A201638@lboro.ac.uk>
Message-ID: <12D6A6A157B44348974E5ED93738628C25369491@HQEXCHMBX03.Presidio.Corp>

What is the IOS being run? Is it IP Base? I don?t think anything lower will run the
stacking module.

James Buchanan | Sr. Network Engineer


Presidio | www.presidio.com<http://www.presidio.com>
12 Cadillac Drive Suite 130, Brentwood, TN 37027
D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 | jbuchanan at
presidio.com<mailto:jbuchanan at presidio.com>

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From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Alan Buxey
Sent: Tuesday, January 15, 2013 9:35 AM
To: svoll.voip at gmail.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 2960s-48fps-l flex stack

I've seen this on ios 15 with flexstack modules. Have you tried removing the module
and re-inserting, or a power cycle with the module installed?

alan

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From rratliff at cisco.com Tue Jan 15 09:42:37 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 15 Jan 2013 09:42:37 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <CAL-DCK0NHxrOgk4RM=CX_absw=Kc2+aqYLOzR-9Oc3_CGEvsEQ@mail.gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
<CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
<1680820783548761203@unknownmsgid>
<CAL-DCK0NHxrOgk4RM=CX_absw=Kc2+aqYLOzR-9Oc3_CGEvsEQ@mail.gmail.com>
Message-ID: <F51BAB8A-F89A-41A3-94DA-9B3D899D1C75@cisco.com>

Given you have an ITSP it's most likely the initial hold that's failing, which is
only manifesting when you try to resume it. If you haven't noticed already this
is also very likely causing transfers to fail.

Take a look at the SIP signaling for a call. I believe the most common cause to
this is the ITSP not handling our transition from active->inactive->sendonly-
>active from hold to MOH to resume. The "Duplex Streaming Enabled" parameter is
there just for this type of problem.

-Ryan

On Jan 14, 2013, at 6:40 PM, Dane Newman <dane.newman at gmail.com> wrote:

Hello Kenneth

I have restarted both CUCM servers so this should have restarted the services when
the utils system restart happened

on my router I see I am using g711 from the debug

I ran a debug voip ccapi inout

I connected a call calling from an external number to a DiD inside of my system.


Once the call was connected I put the call on hold and the following debug came
out..the music on hold played for the external caller

*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:


Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742, Xmit Function=0x64204BAC
*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12742
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12741
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=96, Call Id=12742
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741, Xmit Function=0x64204BAC
*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12741
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12742
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
Cisco3825#
Cisco3825#
Cisco3825#

I then after that took off the hold and the following debug came out. The call on
the PSDN side still played the hold music while there was no voice on the deskphone
side.

*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:


Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742, Xmit Function=0x64204BAC
*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12742
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12741
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=96, Call Id=12742
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741, Xmit Function=0x64204BAC
*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12741
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12742
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
Cisco3825#
Cisco3825#
Cisco3825#

On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
Have you also restarted the Cisco IP Media Services?

Sent from my iPhone

On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com> wrote:

> My ITSP will only support 711ulaw for me currently I believe. They hard coded it
with me when I was initially setting it up.
>
> Do you think this could be a codec issue? How would I go about identifying if it
is?
>
> Dane
>
> On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <kennethwhayes at gmail.com>
wrote:
> Have you tried different audio codecs?
>
> Sent from my iPhone
>
> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
>> Ryan (sorry I forgot to reply to all)
>>
>> Thanks for the Reply
>> Oddly enough we are.
>> This probably has something to do with MOH in general?
>>
>> Internally when I user puts another user on hold everything works. No MOH plays
and they can hold and unhold the call just fine.
>> I tested calling from an external number. Once I put the external caller on hold
the MOH played but I was unable to resume the call. When I hit resume on the
deskphone the MOH still played to the external caller and there was no sound on the
deskphone.
>>
>> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>> Do you get similar behavior if you just hold and resume the call outside SNR
features?
>>
>> -Ryan
>>
>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Using keyboard-interactive authentication.
>> Password:
>>
>> Cisco3825#
>> Cisco3825#sh ver
>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1
>> (4)M5, RELEASE SOFTWARE (fc1)
>> Technical Support: http://www.cisco.com/techsupport
>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>
>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>
>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>> System returned to ROM by power-on
>> System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>> Last reload type: Normal Reload
>>
>>
>> This product contains cryptographic features and is subject to United
>> States and local country laws governing import, export, transfer and
>> use. Delivery of Cisco cryptographic products does not imply
>> third-party authority to import, export, distribute or use encryption.
>> Importers, exporters, distributors and users are responsible for
>> compliance with U.S. and local country laws. By using this product you
>> agree to comply with applicable laws and regulations. If you are unable
>> to comply with U.S. and local laws, return this product immediately.
>>
>> A summary of U.S. laws governing Cisco cryptographic products may be found at:
>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>
>> If you require further assistance please contact us by sending email to
>> export at cisco.com.
>>
>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>> Processor board ID FTX1237A1T0
>> 2 Gigabit Ethernet interfaces
>> 2 Channelized T1/PRI ports
>> 1 Virtual Private Network (VPN) Module
>> DRAM configuration is 64 bits wide with parity enabled.
>> 479K bytes of NVRAM.
>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>
>>
>> License Info:
>>
>> License UDI:
>>
>> -------------------------------------------------
>> Device# PID SN
>>
>> Sent from my mobile device
>>
>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
>>
>>> What version of code are you running on the CUBE?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>>> Hello
>>>>
>>>> I have an issue when users are connected to a call and hit the mobility soft
key button on 9971 phones when a call is active, the phone system rings on the
mobile number configured in the system. When they pick up the the mobile number it
just plays what sounds like hold music on both ends of the call (I believe this
music is coming from cucm but I haven't heard it before) instead of providing 2 way
voice.
>>>>
>>>> In another senario with what I believe is the same issue. If a user picks up
on there cell phone first (using single number reach) opposed to the deskphone the
call is connected with 2 way voice and no issues exist. If the user then hangs up
his cell phone with the intent to take the call on his deskphone the calling party
starts hearing the hold music. Once the user picks up the call on his deskphone he
hears nothing but the calling party is still hearing the hold music. It is not
working as intended where 2 way voice happens once the user hangs up his mobile
phone and picks up on his deskphone 2 way voice should happen.
>>>>
>>>> My topology is as follows..
>>>>
>>>>
>>>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>>>
>>>> Calls are sent back out the SIP trunk to the ITSP when using mobile
connect/snr.
>>>>
>>>> Does anyone have any ideas how I can make 2 way voice happen instead of the
hold music when the calls are picked up?
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>

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From rratliff at cisco.com Tue Jan 15 09:51:04 2013
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 15 Jan 2013 09:51:04 -0500
Subject: [cisco-voip] 9.1 Upgrade Times
In-Reply-To: <12D6A6A157B44348974E5ED93738628C25369417@HQEXCHMBX03.Presidio.Corp>
References: <C75AF2AD9308C246AFBDDB994E3E2983110B24E7@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B2E69@PHANES.helion.local>
<C75AF2AD9308C246AFBDDB994E3E2983110B5E52@PHANES.helion.local>
<2F143E71016CA34C924BF4C33AEF211056E71556@W12112.ldschurch.org>
<BAY149-ds10D2ABFE4E797176881759872D0@phx.gbl>
<12D6A6A157B44348974E5ED93738628C25369417@HQEXCHMBX03.Presidio.Corp>
Message-ID: <A7C3FDE4-31FF-40F1-80B3-128AF17039D6@cisco.com>

Also take a look at disk space so you don't run out during the upgrade.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/rel_notes/9_1_1/CUCM_BK_R6F8DBD4
_00_release-notes-for-cucm-91_chapter_011.html#CUCM_RF_C4B0C2D8_00

-Ryan

On Jan 15, 2013, at 2:24 AM, "Buchanan, James" <jbuchanan at presidio.com> wrote:

Cleaning out log files and CDRs might also be a good idea.

James Buchanan | Sr. Network Engineer


Presidio | www.presidio.com
12 Cadillac Drive Suite 130, Brentwood, TN 37027
D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 | jbuchanan at presidio.com

Follow us:

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Tuesday, January 15, 2013 8:40 AM
To: 'Nate VanMaren'; 'Matthew Loraditch'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 9.1 Upgrade Times

I usually make a cluster reboot for CUCM or unity connection before I upgrade.

Regards,
Ahmed Elnagar | Unified Communication Team Leader | CCIE #24697, Voice
<image001.jpg>

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Nate VanMaren
Sent: Monday, January 14, 2013 6:33 PM
To: Matthew Loraditch; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 9.1 Upgrade Times

How long had the server been up before the you started the first upgrade?

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Matthew Loraditch
Sent: Monday, January 14, 2013 9:27 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 9.1 Upgrade Times

So I figured I?d loop around and let everyone know how things ended up.
Unity Connection never finished the first attempt and wouldn?t cancel either. I had
to force a shutdown. After it came back up I tried again? 90 minutes boom. Suffice
it to say. I wish had done that sooner.

Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter | Facebook | Website | Email Support

From: Matthew Loraditch


Sent: Sunday, January 13, 2013 3:06 PM
To: Matthew Loraditch; cisco-voip at puck.nether.net
Subject: RE: 9.1 Upgrade Times

Well my CUCM publisher took almost 13 hours, my sub took 1 hour and change? the
unity connection pub is still running, I think we are on hour 15 of that?

Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter | Facebook | Website | Email Support

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at


puck.nether.net] On Behalf Of Matthew Loraditch
Sent: Sunday, January 13, 2013 8:58 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 9.1 Upgrade Times

On a Unity Connection system and CUCM both 8.6.2aSU1 started at 11:45PM and 12:12AM
Last night, both are STILL running at 8:45 AM this morning. The system I am doing
this test on has about 60 Phones/Users/VM. These are the publishers of each
install, but I have never had an upgrade take this long, ever.

I?m now not sure I?ll even be able to finish in my window since I haven?t even
touched the subscribers yet.

Any clues as to why this is taking so long? I used to disable iothrottle during an
upgrade but that command is deprecated now.

Matthew G. Loraditch ? CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive


Timonium, MD 21093

voice. 410.252.8830
fax. 410.252.9284

Twitter | Facebook | Website | Email Support

NOTICE: This email message is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. Any unauthorized review, use,
disclosure or distribution is prohibited. If you are not the intended recipient,
please contact the sender by reply email and destroy all copies of the original
message.

This message w/attachments (message) is intended solely for the use of the intended
recipient(s) and may contain information that is privileged, confidential or
proprietary. If you are not an intended recipient, please notify the sender, and
then please delete and destroy all copies and attachments. Please be advised that
any review or dissemination of, or the taking of any action in reliance on, the
information contained in or attached to this message is prohibited.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From avholloway+cisco-voip at gmail.com Tue Jan 15 11:08:14 2013


From: avholloway+cisco-voip at gmail.com (Anthony Holloway)
Date: Tue, 15 Jan 2013 10:08:14 -0600
Subject: [cisco-voip] CUCM MTP and g729
In-Reply-To: <188F72A9-D5A4-448A-A2CE-059A5942833A@gmail.com>
References: <CACRCJOi5SVz2He4nuzDHX-xS7SFGphgbww-PxqacCTo9sPm__A@mail.gmail.com>
<3440C2FC-FC42-4B2D-B4EB-2F76B76BC6F7@cisco.com>
<CACRCJOguqqYfJrY38mSG47qgmnkz=RYReZdLA2rTdpPYpuSz0A@mail.gmail.com>
<CAMa5Jw7OZ1-cQOo3KVdx56A9FdTDmOqVK9rzxy5+CHr8v8dQ9Q@mail.gmail.com>
<CACRCJOiRpHqNMHLNYgLEvTtKUunie7Y82HKvFfUW1zCbU-6c5g@mail.gmail.com>
<A287F450-2809-4BE6-A3D1-9584CC120CE1@cisco.com>
<4E38DB0A1959B04C8C83EDCF069B53ED0D2C718859@USISPCLEXDB01.na.didata.local>
<188F72A9-D5A4-448A-A2CE-059A5942833A@gmail.com>
Message-ID: <CACRCJOgsQ82pJf7HaP30XPFJ8xK0YjuqQci4vsNQL-wpws_hzg@mail.gmail.com>

Here is the trace from the MTP registering, showing that it supports
pass-through:

10:49:20.397 |StationInit: (0000001) CapabilitiesRes capCount=6 *caps=


258(0)*
...
10:49:20.397
|MediaTerminationPointControl(1)::wait_capabilities_StationCapRes - Device
= SUB02B_MTP - *Pass-Through Supported = 1*|[redacted]

Looks like it's the 258 you and Ryan spoke of. Mystery solved! Thanks to
all who provided input.

On Sat, Jan 12, 2013 at 3:43 PM, Peter Slow <peter.slow at gmail.com> wrote:

> Ryan, that's awesome, didn't know that.


>
> Anthony, it looks like the pass through codec Ryan is talking about is
> registered as codec number 258. Look for that in the capabilitiesresponse
> message.
>
> Sent from my iPad
>
> On Jan 12, 2013, at 12:45 PM, "Jason Aarons (AM)" <
> jason.aarons at dimensiondata.com> wrote:
>
> I got a customer running 8.5.1SU2 and it?s not doing IP Voice Media
> Streaming App MTP with Pass-Thru. Just had a big TAC case with T38 and
> took awhile for the TAC lead to come to that conclusion. A IOS Software
> MTP was needed to fix it.****
>
> ** **
>
> I?ve looked previously and haven?t found any details about fix/improvement
> (eg what?s new) around IP Voice Media Streaming App in newer versions.****
>
> ** **
>
> *From:* cisco-voip-bounces at puck.nether.net [
> mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at
puck.nether.net>]
> *On Behalf Of *Ryan Ratliff
> *Sent:* Friday, January 11, 2013 5:03 PM
> *To:* Anthony Holloway
> *Cc:* Cisco VoIP Group
> *Subject:* Re: [cisco-voip] CUCM MTP and g729****
>
> ** **
>
>
>
> Check the capabilities the MTP advertises to CUCM when it registers. At
> some point (8.5 maybe?) IP Voice Media Streaming App began supporting audio
> passthrough, which would explain what you are seeing.****
>
> ** **
>
> -Ryan ****
>
> ** **
>
> On Jan 11, 2013, at 4:36 PM, Anthony Holloway <
> avholloway+cisco-voip at gmail.com> wrote:****
>
> ** **
>
> Thanks Pete. I'll see if I can answer or reply to each of your questions
> or points.
>
> *"are you SURE this isn't just a device hearing g.729 hold music while
> you've got or had the Duplex Streaming service parameter enabled?"*****
>
> No, this is a call from an analog device to a PSTN device, and the call is
> well established and in progress with two way audio.
>
> *"Do you have the skinny signalling to go with it showing what it was
> specifically set up for use as?"*****
>
> I'm not sure I understand where the skinny signaling comes in. The VG224
> is MGCP, the SBC is a SIP trunk, and the MTP is local to the CUCM. Could
> you help me understand? I do have traces off the CUCM if that answers your
> question.
>
> *"Also, I beleive MGCP Endpoints have an initial "state" when they begin
> call setup. i think not using g.729 actually entails a switch from 729 to
> another codec, and perhaps a small delay is causing some packets to be
> transmitted using g.729? maybe? that's a complete stretch but who knows =)"
> *****
>
> Again, this is an established call. I get the call setup, verify two way
> audio, and then take the capture.
>
> *"I don't think you're really goign to get an answer unless you can
> recreate the issue"*****
>
> I can. I have the VG224 in my cubicle. Check out the adapter I'm using!
> Photo attached. =)
>
> *"and we can see traces."*
> I can't upload traces to the list, but if this goes to a TAC case, I will
> certainly give them up at that time.
>
> *"you'll also want a packet capture of the registration of the media
> device"*****
>
> This is a good idea. I'll give it a try and see what it reports.
>
> *"you coudl be doing duplex audio while one of those is playing"*****
>
> I'm not sure what that means, or how that would even work, but you sound
> excited. =)
>
> *"anyway, can you reproduce it"*****
>
> Yes. I can reproduce it at will.****
>
> Thanks for asking the questions and commenting.****
>
> ** **
>
> On Fri, Jan 11, 2013 at 3:24 PM, Peter Slow <peter.slow at gmail.com> wrote:*
> ***
>
> PS> ..Also, are you SURE this isn't just a device hearing g.729 hold
> music while you've got or had the Duplex Streaming service parameter
> enabled? ...'Cause that would totally explain this packet capture. Do
> you have the skinny signalling to go with it showing what it was
> specifically set up for use as?
>
> Also, I beleive MGCP Endpoints have an initial "state" when they begin
> call setup. i think not using g.729 actually entails a switch from 729
> to another codec, and perhaps a small delay is causing some packets to
> be transmitted using g.729? maybe? that's a complete stretch but who
> knows =)
>
> I don't think you're really goign to get an answer unless you can
> recreate the issue and we can see traces. you'll also want a packet
> capture of the registration of the media device, so if you can make
> the MTP register to a different callmanager than what it's running on,
> using its CUCM group, we could take a look at what capabilities it was
> registering with and if it says it supports 729 now =) ..you'll wnat
> to look at the skinny registration of the MTP, ANN ooh, ANN
> announcements are in 7.29 also, i think? you coudl be doing duplex
> audio while one of those is playing =)....
>
> anyway, can you reproduce it or verify or deny any of those guesses?
>
> Very Interesting,
> -Pete****
>
>
>
>
> On Fri, Jan 11, 2013 at 4:08 PM, Anthony Holloway
> <avholloway+cisco-voip at gmail.com> wrote:
> >
> > Hey Wes,
> >
> > The packet capture was done on the CUCM itself via CLI command: "utils
> > network capture". Also, I filtered the capture to traffic only coming
> from
> > the VG224, which is why you do not see any other streams. It was,
> however,
> > going to our SBC. So the call flow was: Analog Phone > VG224 > CUCM
> (MTP) >
> > SBC > PSTN.
> >
> > The negotiated CODEC was in fact g729, and both sides support it. The
> MGCP
> > SDP shows g729 and the SBC sends back g729 in the SIP SDP. The only
> thing
> > that is different in caps is DTMF. MGCP was trying 100 while SBC wanted
> to
> > do 101.
> >
> > As for the garble: I wasn't experiencing any voice quality issues that I
> > could hear, but I was experiencing double DTMF going out to the PSTN.
> Not
> > sure if an artifact of the MTP, or simply a misonconfiguration on the
> > VG224's MGCP package. Like I said it's the fm package I was missing that
> > ultimately fixed the issue. The MTP is no longer used, and the double
> DTMF
> > is gone. I didn't find very much info on what the fm packages does, only
> > that it fixes DTMF and Faxing issues when communicating with a SIP
> device.
> >
> > Thanks for the late Friday afternoon reply Wes.
> >
> > On Fri, Jan 11, 2013 at 2:48 PM, Wes Sisk <wsisk at cisco.com> wrote:
> >>
> >> Interesting observations.
> >>
> >> I am not aware of any changes around CM's software MTP only doing G.711.
> >>
> >> The packet capture shows RTP coming into(?) to the MTP. I do not see any
> >> sign of anything egressing the MTP.
> >>
> >> ccm has internal logic that attempts to connect RTP streams even if
> codec
> >> negotiation fails. This is controlled by a service parameter. You may be
> >> seeing an artifact of this behavior where no codec was common but the
> >> streams attempted to setup anyway. Streaming codecs to the MTP that it
> does
> >> not support typically results in garble or silence on the egress leg.
> >>
> >> /wes
> >>
> >>
> >> On Jan 11, 2013, at 3:11 PM, Anthony Holloway wrote:
> >>
> >> Hi All,
> >>
> >> I have a wireshark capture off of my CUCM 8.6(2) which shows that it is
> >> receiving a g729 audio stream from my VG224.
> >>
> >> Long story short, according to the CUCM SRND, the CUCM MTP can only
> >> terminate g711, and yet, attached is a screenshot of the wireshark
> capture
> >> which clearly shows it terminating g729.
> >>
> >> What piece of this puzzle am I missing? Also, the CUCM traces read like
> >> the MTP is being invoked on that CUCM. It's due to the lack of the fm
> >> package on my VG224 and a mismatch in DTMF to the PSTN (SIP). I had a
> fun
> >> time resolving that, as you can probably imagine.
> >>
> >> Thanks and Happy Friday!
> >>
> >>
> >>
> >>
> <cucm-mtp-g729-redacted.png><cucm-mtp-redacted.png><cucm-sub02b-
redacted.png>_______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ** **
>
>
>
> itevomcid ****
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From tednugent73 at gmail.com Tue Jan 15 11:19:14 2013


From: tednugent73 at gmail.com (Ted Nugent)
Date: Tue, 15 Jan 2013 11:19:14 -0500
Subject: [cisco-voip] Gathering Licenses from CUCM?
Message-ID: <CAHs2VYsenEWypHOWa7BRbcTwLvxRPGshzMv49_QqOd2xCsr-Ng@mail.gmail.com>

I'm attempting to gather ~50 individual license files from a CUCM 7.1
server to zip them up and send to licensing.... Is there anyway to pull
them off the server without having to copy/paste into the individual
files...?

TIA
T
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From VanMarenNP at ldschurch.org Tue Jan 15 10:23:27 2013
From: VanMarenNP at ldschurch.org (Nate VanMaren)
Date: Tue, 15 Jan 2013 15:23:27 +0000
Subject: [cisco-voip] Design question CTI or DN?
In-Reply-To: <CF2E16827204E249BB155D2C1F5C1EAB1ABAB413@GLB-EXMBX-
001.ghdnet.internal>
References: <BLU0-SMTP21829B96B1F0A32A113A6D3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E7561D@W12112.ldschurch.org>
<BLU0-SMTP35978042C1DFCB75E7430F3C52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E75651@W12112.ldschurch.org>
<BLU0-SMTP35668CEDEABCD8D477E09FEC52D0@phx.gbl>
<2F143E71016CA34C924BF4C33AEF211056E756D6@W12112.ldschurch.org>
<BLU0-SMTP366CE9668B959DD8AD94713C52D0@phx.gbl>
<CF2E16827204E249BB155D2C1F5C1EAB1ABAB413@GLB-EXMBX-001.ghdnet.internal>
Message-ID: <2F143E71016CA34C924BF4C33AEF211056E75ECA@W12112.ldschurch.org>

Yeah that is really good for consecutive ranges, but I don't think that is his case
here.

From: John Van Laecke [mailto:John.VanLaecke at ghd.com]


Sent: Monday, January 14, 2013 10:36 PM
To: Mike ; Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

In call manager you can make bulk dn's.

Goto
call routing
Directory number
Add new
And you can add a range in and create the call forwards.

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike
Sent: Tuesday, 15 January 2013 1:39 PM
To: 'Nate VanMaren'; 'cisco voip'
Subject: Re: [cisco-voip] Design question CTI or DN?

I'll give it a shot thanks!

From: Nate VanMaren [mailto:VanMarenNP at ldschurch.org]


Sent: Monday, January 14, 2013 10:35 PM
To: Mike ; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

Ah, my new best friend is import/export.

It's really just a CSV that does anything.

I haven't done it with DNs, but I just imported all of the permutations of phone
button templates in a few seconds.

Export something, look at the CSV in the TAR, edit it to what you want, and load it
back in.

Thanks,
-Nate

From: Mike [mailto:mikeeo at msn.com]


Sent: Monday, January 14, 2013 8:30 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

True, but I never seen a way to BAT in DNs.

From: Nate VanMaren [mailto:VanMarenNP at ldschurch.org]


Sent: Monday, January 14, 2013 10:27 PM
To: Mike ; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

You don't need DNs on a device anymore for them to be active. They just need to be
marked as "active"

From: Mike [mailto:mikeeo at msn.com]


Sent: Monday, January 14, 2013 8:25 PM
To: Nate VanMaren; 'cisco voip'
Subject: RE: [cisco-voip] Design question CTI or DN?

I was thinking of doing a wildcard 1XXX, 2XXX, 3XXX etc , but that might get ugly.
Its weird they don't want to use SNR they just want the call to go to voicemail. I
don't think you can BAT in DNs only CTI ports right?

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nate
VanMaren
Sent: Monday, January 14, 2013 10:19 PM
To: Mike ; 'cisco voip'
Subject: Re: [cisco-voip] Design question CTI or DN?

What needs to happen to an assigned DIDs? If it's ok for them to go to a call


handler, then just have a wildcard for the whole block that sends anything un
assigned to Unity, then it will match their mailbox if it can or they can go to a
callhandler.

Or build DNs for each number, not a ton of fun but can be made easier with AXL or
BAT, the real trouble is keeping them up to date.

-Nate

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at


puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike
Sent: Monday, January 14, 2013 7:01 PM
To: 'cisco voip'
Subject: [cisco-voip] Design question CTI or DN?

I have about 1500 voice mail only users that I have in Unity connections and I was
wondering that the best way to get their DIDs into unity cx. I was thinking I can
create CTI ports send them all to voicemail. The problem is the customer owns the
whole block 0000-9999 and the DIDs are all over the place or I'd create a wild card
hunt pilot.

Any ideas?

Thanks,
Mike

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From rratliff at cisco.com Tue Jan 15 11:31:25 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 15 Jan 2013 11:31:25 -0500
Subject: [cisco-voip] Gathering Licenses from CUCM?
In-Reply-To: <CAHs2VYsenEWypHOWa7BRbcTwLvxRPGshzMv49_QqOd2xCsr-Ng@mail.gmail.com>
References: <CAHs2VYsenEWypHOWa7BRbcTwLvxRPGshzMv49_QqOd2xCsr-Ng@mail.gmail.com>
Message-ID: <AE3BBFE7-2F2B-4081-9AC0-16D297A1D07B@cisco.com>

admin:file get license *


Please wait while the system is gathering files info ...done.
Sub-directories were not traversed.
Number of files affected: 7
Total size in Bytes: 3450
Total size in Kbytes: 3.3691406
Would you like to proceed [y/n]?
-Ryan

On Jan 15, 2013, at 11:19 AM, Ted Nugent <tednugent73 at gmail.com> wrote:

I'm attempting to gather ~50 individual license files from a CUCM 7.1 server to zip
them up and send to licensing.... Is there anyway to pull them off the server
without having to copy/paste into the individual files...?

TIA
T
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From tednugent73 at gmail.com Tue Jan 15 12:05:58 2013


From: tednugent73 at gmail.com (Ted Nugent)
Date: Tue, 15 Jan 2013 12:05:58 -0500
Subject: [cisco-voip] Gathering Licenses from CUCM?
In-Reply-To: <AE3BBFE7-2F2B-4081-9AC0-16D297A1D07B@cisco.com>
References: <CAHs2VYsenEWypHOWa7BRbcTwLvxRPGshzMv49_QqOd2xCsr-Ng@mail.gmail.com>
<AE3BBFE7-2F2B-4081-9AC0-16D297A1D07B@cisco.com>
Message-ID: <CAHs2VYtsCkxJupLyVXy4zgBaFQwah-4qcyc2Xw8pMEGE+Ts5Gg@mail.gmail.com>

Awesome that did it! Thanks!... is there a similar command of Unity


Connection? I see the command there but I get...

admin:file get license *


Please wait while the system is gathering files info ...done.
No such file or directory can be found.
admin:

On Tue, Jan 15, 2013 at 11:31 AM, Ryan Ratliff <rratliff at cisco.com> wrote:

> file get license *


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From dane.newman at gmail.com Tue Jan 15 12:35:58 2013


From: dane.newman at gmail.com (Dane Newman)
Date: Tue, 15 Jan 2013 12:35:58 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <F51BAB8A-F89A-41A3-94DA-9B3D899D1C75@cisco.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
<CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
<1680820783548761203@unknownmsgid>
<CAL-DCK0NHxrOgk4RM=CX_absw=Kc2+aqYLOzR-9Oc3_CGEvsEQ@mail.gmail.com>
<F51BAB8A-F89A-41A3-94DA-9B3D899D1C75@cisco.com>
Message-ID: <CAL-DCK1fBRyZqK+dZq08go_FaFXvGzuYwHU311vrWuLbe3XL-A@mail.gmail.com>

Thanks Ryan for the input

*On the call when I hold the call the following debug pops out....*

*Jan 15 17:56:05.246:
//13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
passthru hdrs to
container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13938) Group (a= group line) attribute, level 65535 instance 1 not
found.
*Jan 15 17:56:05.274:
//13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
passthru headers to container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not
found.
*Jan 15 17:56:05.286:
//13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
passthru hdrs to
container
*Jan 15 17:56:05.302:
//13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
passthru headers to container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not
found.
SIP: Attribute mid, level 1 instance 1 not found.
*Jan 15 17:56:05.322: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia:
Could not modify QoS params for midcall INVITE

*After I try to unhold the call the following debug comes out....*
**

*Jan 15 17:56:18.874:
//13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
passthru hdrs to
container
*Jan 15 17:56:18.894:
//13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
passthru headers to container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not
found.
SIP: Attribute mid, level 1 instance 1 not found.
*Jan 15 17:56:18.906: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia:
Could not modify QoS params for midcall INVITE
Cisco3825#
Cisco3825#
Cisco3825#

On Tue, Jan 15, 2013 at 9:42 AM, Ryan Ratliff <rratliff at cisco.com> wrote:

> Given you have an ITSP it's most likely the initial hold that's failing,
> which is only manifesting when you try to resume it. If you haven't
> noticed already this is also very likely causing transfers to fail.
>
> Take a look at the SIP signaling for a call. I believe the most common
> cause to this is the ITSP not handling our transition from
> active->inactive->sendonly->active from hold to MOH to resume. The
> "Duplex Streaming Enabled" parameter is there just for this type of problem.
>
> -Ryan
>
> On Jan 14, 2013, at 6:40 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> *Hello Kenneth*
> **
> *I have restarted both CUCM servers so this should have restarted the
> services when the utils system restart happened*
> **
>
> *on my router I see I am using g711 from the debug *
> **
> *I ran a debug voip ccapi inout *
> **
> *I connected a call calling from an external number to a DiD inside of my
> system. Once the call was connected I put the call on hold and the
> following debug came out..the music on hold played for the external caller
> *
>
> *Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
> Stop Tone On Digit=FALSE, Tone=Null,
> Tone Direction=Sum Network, Params=0x0, Call Id=12741
> *Jan 14 23:47:40.783:
> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742, Xmit Function=0x64204BAC
> *Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
> Modem=0x0, Codec Bytes=20, Signal Type=2)
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
> Playout Max=1000(ms), Fax Nom=300(ms))
> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=170, Call Id=12742
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
> Feature Type=50, Interface=0xC05A65AC, Call Id=12742
> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
> Modem=0x0, Codec Bytes=20, Signal Type=2)
> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
> Playout Max=1000(ms), Fax Nom=300(ms))
> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=171, Call Id=12741
> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
> Interface=0xC05A65AC, Call Id=12742
> *Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
> Stop Tone On Digit=FALSE, Tone=Null,
> Tone Direction=Sum Network, Params=0x0, Call Id=12741
> *Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=96, Call Id=12742
> *Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.839:
> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741, Xmit Function=0x64204BAC
> *Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
> Modem=0x0, Codec Bytes=20, Signal Type=2)
> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
> Playout Max=1000(ms), Fax Nom=300(ms))
> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=170, Call Id=12741
> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
> Modem=0x0, Codec Bytes=20, Signal Type=2)
> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
> Playout Max=1000(ms), Fax Nom=300(ms))
> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=171, Call Id=12742
> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
> Interface=0xC05A65AC, Call Id=12742
> Cisco3825#
> Cisco3825#
> Cisco3825#
>
>
> *I then after that took off the hold and the following debug came out.
> The call on the PSDN side still played the hold music while there was no
> voice on the deskphone side.*
>
> *Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
> Stop Tone On Digit=FALSE, Tone=Null,
> Tone Direction=Sum Network, Params=0x0, Call Id=12741
> *Jan 14 23:47:40.783:
> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742, Xmit Function=0x64204BAC
> *Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
> Modem=0x0, Codec Bytes=20, Signal Type=2)
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
> Playout Max=1000(ms), Fax Nom=300(ms))
> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=170, Call Id=12742
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
> Feature Type=50, Interface=0xC05A65AC, Call Id=12742
> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
> Modem=0x0, Codec Bytes=20, Signal Type=2)
> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
> Playout Max=1000(ms), Fax Nom=300(ms))
> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=171, Call Id=12741
> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
> Interface=0xC05A65AC, Call Id=12742
> *Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
> Stop Tone On Digit=FALSE, Tone=Null,
> Tone Direction=Sum Network, Params=0x0, Call Id=12741
> *Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=96, Call Id=12742
> *Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.839:
> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741, Xmit Function=0x64204BAC
> *Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
> Modem=0x0, Codec Bytes=20, Signal Type=2)
> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
> Playout Max=1000(ms), Fax Nom=300(ms))
> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=170, Call Id=12741
> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
> Call Id=12742,
> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
> Modem=0x0, Codec Bytes=20, Signal Type=2)
> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
> Playout Max=1000(ms), Fax Nom=300(ms))
> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
> Call Id=12741,
> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event=171, Call Id=12742
> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
> Event Is Sent To Conferenced SPI(s) Directly
> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
> Interface=0xC05A65AC, Call Id=12742
> Cisco3825#
> Cisco3825#
> Cisco3825#
>
> On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <kennethwhayes at gmail.com>wrote:
>
>> Have you also restarted the Cisco IP Media Services?
>>
>> Sent from my iPhone
>>
>> On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> My ITSP will only support 711ulaw for me currently I believe. They hard
>> coded it with me when I was initially setting it up.
>>
>> Do you think this could be a codec issue? How would I go about
>> identifying if it is?
>>
>> Dane
>>
>> On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <kennethwhayes at
gmail.com>wrote:
>>
>>> Have you tried different audio codecs?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>> Ryan (sorry I forgot to reply to all)
>>>
>>> Thanks for the Reply
>>> Oddly enough we are.
>>> This probably has something to do with MOH in general?
>>>
>>> Internally when I user puts another user on hold everything works. No
>>> MOH plays and they can hold and unhold the call just fine.
>>> I tested calling from an external number. Once I put the external
>>> caller on hold the MOH played but I was unable to resume the call. When I
>>> hit resume on the deskphone the MOH still played to the external caller and
>>> there was no sound on the deskphone.
>>>
>>> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>>
>>>> Do you get similar behavior if you just hold and resume the call
>>>> outside SNR features?
>>>>
>>>> -Ryan
>>>>
>>>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>>
>>>> Using keyboard-interactive authentication.
>>>>
>>>> Password:
>>>>
>>>>
>>>> Cisco3825#
>>>>
>>>> Cisco3825#sh ver
>>>>
>>>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M),
>>>> Version 15.1
>>>> (4)M5, RELEASE SOFTWARE (fc1)
>>>>
>>>> Technical Support: http://www.cisco.com/techsupport
>>>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>>>>
>>>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>>>
>>>>
>>>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>>>
>>>>
>>>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>>>>
>>>> System returned to ROM by power-on
>>>>
>>>> System image file is
>>>> "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>>>> Last reload type: Normal Reload
>>>>
>>>>
>>>>
>>>> This product contains cryptographic features and is subject to United
>>>>
>>>> States and local country laws governing import, export, transfer and
>>>>
>>>> use. Delivery of Cisco cryptographic products does not imply
>>>>
>>>> third-party authority to import, export, distribute or use encryption.
>>>>
>>>> Importers, exporters, distributors and users are responsible for
>>>>
>>>> compliance with U.S. and local country laws. By using this product you
>>>>
>>>> agree to comply with applicable laws and regulations. If you are unable
>>>>
>>>> to comply with U.S. and local laws, return this product immediately.
>>>>
>>>>
>>>> A summary of U.S. laws governing Cisco cryptographic products may be
>>>> found at:
>>>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>>>
>>>> If you require further assistance please contact us by sending email to
>>>>
>>>> export at cisco.com.
>>>>
>>>>
>>>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>>>>
>>>> Processor board ID FTX1237A1T0
>>>>
>>>> 2 Gigabit Ethernet interfaces
>>>>
>>>> 2 Channelized T1/PRI ports
>>>>
>>>> 1 Virtual Private Network (VPN) Module
>>>>
>>>> DRAM configuration is 64 bits wide with parity enabled.
>>>>
>>>> 479K bytes of NVRAM.
>>>>
>>>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>>>
>>>>
>>>>
>>>> License Info:
>>>>
>>>>
>>>> License UDI:
>>>>
>>>>
>>>> -------------------------------------------------
>>>>
>>>> Device# PID SN
>>>>
>>>> Sent from my mobile device
>>>>
>>>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com>
>>>> wrote:
>>>>
>>>> What version of code are you running on the CUBE?
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>>
>>>> Hello
>>>>
>>>> I have an issue when users are connected to a call and hit the
>>>> mobility soft key button on 9971 phones when a call is active, the phone
>>>> system rings on the mobile number configured in the system. When they pick
>>>> up the the mobile number it just plays what sounds like hold music on both
>>>> ends of the call (I believe this music is coming from cucm but I haven't
>>>> heard it before) instead of providing 2 way voice.
>>>>
>>>> In another senario with what I believe is the same issue. If a user
>>>> picks up on there cell phone first (using single number reach) opposed to
>>>> the deskphone the call is connected with 2 way voice and no issues exist.
>>>> If the user then hangs up his cell phone with the intent to take the call
>>>> on his deskphone the calling party starts hearing the hold music. Once the
>>>> user picks up the call on his deskphone he hears nothing but the calling
>>>> party is still hearing the hold music. It is not working as intended where
>>>> 2 way voice happens once the user hangs up his mobile phone and picks up on
>>>> his deskphone 2 way voice should happen.
>>>>
>>>> My topology is as follows..
>>>>
>>>>
>>>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>>>
>>>> Calls are sent back out the SIP trunk to the ITSP when using mobile
>>>> connect/snr.
>>>>
>>>> Does anyone have any ideas how I can make 2 way voice happen instead of
>>>> the hold music when the calls are picked up?
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>>
>>>>
>>>
>>
>
>
-------------- next part --------------
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From rratliff at cisco.com Tue Jan 15 12:42:51 2013
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 15 Jan 2013 12:42:51 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <CAL-DCK1fBRyZqK+dZq08go_FaFXvGzuYwHU311vrWuLbe3XL-A@mail.gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
<CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
<1680820783548761203@unknownmsgid>
<CAL-DCK0NHxrOgk4RM=CX_absw=Kc2+aqYLOzR-9Oc3_CGEvsEQ@mail.gmail.com>
<F51BAB8A-F89A-41A3-94DA-9B3D899D1C75@cisco.com>
<CAL-DCK1fBRyZqK+dZq08go_FaFXvGzuYwHU311vrWuLbe3XL-A@mail.gmail.com>
Message-ID: <AAA51FE5-3A02-44BA-844E-1BE005EA3ABD@cisco.com>

Without sip messages I can't get any clues from that.

-Ryan

On Jan 15, 2013, at 12:35 PM, Dane Newman <dane.newman at gmail.com> wrote:

Thanks Ryan for the input

On the call when I hold the call the following debug pops out....

*Jan 15 17:56:05.246: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer:


Unable to add passthru hdrs to
container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13938) Group (a= group line) attribute, level 65535 instance 1 not found.
*Jan 15 17:56:05.274: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer:
Unable to add
passthru headers to container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.
*Jan 15 17:56:05.286: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer:
Unable to add passthru hdrs to
container
*Jan 15 17:56:05.302: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer:
Unable to add
passthru headers to container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: Attribute mid, level 1 instance 1 not found.
*Jan 15 17:56:05.322: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could
not modify QoS params for midcall INVITE

After I try to unhold the call the following debug comes out....

*Jan 15 17:56:18.874: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer:


Unable to add passthru hdrs to
container
*Jan 15 17:56:18.894: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer:
Unable to add
passthru headers to container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: Attribute mid, level 1 instance 1 not found.
*Jan 15 17:56:18.906: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could
not modify QoS params for midcall INVITE
Cisco3825#
Cisco3825#
Cisco3825#

On Tue, Jan 15, 2013 at 9:42 AM, Ryan Ratliff <rratliff at cisco.com> wrote:
Given you have an ITSP it's most likely the initial hold that's failing, which is
only manifesting when you try to resume it. If you haven't noticed already this
is also very likely causing transfers to fail.

Take a look at the SIP signaling for a call. I believe the most common cause to
this is the ITSP not handling our transition from active->inactive->sendonly-
>active from hold to MOH to resume. The "Duplex Streaming Enabled" parameter is
there just for this type of problem.

-Ryan

On Jan 14, 2013, at 6:40 PM, Dane Newman <dane.newman at gmail.com> wrote:

Hello Kenneth

I have restarted both CUCM servers so this should have restarted the services when
the utils system restart happened

on my router I see I am using g711 from the debug

I ran a debug voip ccapi inout

I connected a call calling from an external number to a DiD inside of my system.


Once the call was connected I put the call on hold and the following debug came
out..the music on hold played for the external caller

*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:


Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742, Xmit Function=0x64204BAC
*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12742
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12741
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=96, Call Id=12742
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741, Xmit Function=0x64204BAC
*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12741
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12742
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
Cisco3825#
Cisco3825#
Cisco3825#

I then after that took off the hold and the following debug came out. The call on
the PSDN side still played the hold music while there was no voice on the deskphone
side.

*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:


Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742, Xmit Function=0x64204BAC
*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12742
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12741
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=96, Call Id=12742
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741, Xmit Function=0x64204BAC
*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12741
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12742
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
Cisco3825#
Cisco3825#
Cisco3825#

On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
Have you also restarted the Cisco IP Media Services?

Sent from my iPhone

On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com> wrote:

> My ITSP will only support 711ulaw for me currently I believe. They hard coded it
with me when I was initially setting it up.
>
> Do you think this could be a codec issue? How would I go about identifying if it
is?
>
> Dane
>
> On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <kennethwhayes at gmail.com>
wrote:
> Have you tried different audio codecs?
>
> Sent from my iPhone
>
> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
>> Ryan (sorry I forgot to reply to all)
>>
>> Thanks for the Reply
>> Oddly enough we are.
>> This probably has something to do with MOH in general?
>>
>> Internally when I user puts another user on hold everything works. No MOH plays
and they can hold and unhold the call just fine.
>> I tested calling from an external number. Once I put the external caller on hold
the MOH played but I was unable to resume the call. When I hit resume on the
deskphone the MOH still played to the external caller and there was no sound on the
deskphone.
>>
>> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>> Do you get similar behavior if you just hold and resume the call outside SNR
features?
>>
>> -Ryan
>>
>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Using keyboard-interactive authentication.
>> Password:
>>
>> Cisco3825#
>> Cisco3825#sh ver
>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1
>> (4)M5, RELEASE SOFTWARE (fc1)
>> Technical Support: http://www.cisco.com/techsupport
>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>
>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>
>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>> System returned to ROM by power-on
>> System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>> Last reload type: Normal Reload
>>
>>
>> This product contains cryptographic features and is subject to United
>> States and local country laws governing import, export, transfer and
>> use. Delivery of Cisco cryptographic products does not imply
>> third-party authority to import, export, distribute or use encryption.
>> Importers, exporters, distributors and users are responsible for
>> compliance with U.S. and local country laws. By using this product you
>> agree to comply with applicable laws and regulations. If you are unable
>> to comply with U.S. and local laws, return this product immediately.
>>
>> A summary of U.S. laws governing Cisco cryptographic products may be found at:
>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>
>> If you require further assistance please contact us by sending email to
>> export at cisco.com.
>>
>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>> Processor board ID FTX1237A1T0
>> 2 Gigabit Ethernet interfaces
>> 2 Channelized T1/PRI ports
>> 1 Virtual Private Network (VPN) Module
>> DRAM configuration is 64 bits wide with parity enabled.
>> 479K bytes of NVRAM.
>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>
>>
>> License Info:
>>
>> License UDI:
>>
>> -------------------------------------------------
>> Device# PID SN
>>
>> Sent from my mobile device
>>
>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
>>
>>> What version of code are you running on the CUBE?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>>> Hello
>>>>
>>>> I have an issue when users are connected to a call and hit the mobility soft
key button on 9971 phones when a call is active, the phone system rings on the
mobile number configured in the system. When they pick up the the mobile number it
just plays what sounds like hold music on both ends of the call (I believe this
music is coming from cucm but I haven't heard it before) instead of providing 2 way
voice.
>>>>
>>>> In another senario with what I believe is the same issue. If a user picks up
on there cell phone first (using single number reach) opposed to the deskphone the
call is connected with 2 way voice and no issues exist. If the user then hangs up
his cell phone with the intent to take the call on his deskphone the calling party
starts hearing the hold music. Once the user picks up the call on his deskphone he
hears nothing but the calling party is still hearing the hold music. It is not
working as intended where 2 way voice happens once the user hangs up his mobile
phone and picks up on his deskphone 2 way voice should happen.
>>>>
>>>> My topology is as follows..
>>>>
>>>>
>>>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>>>
>>>> Calls are sent back out the SIP trunk to the ITSP when using mobile
connect/snr.
>>>>
>>>> Does anyone have any ideas how I can make 2 way voice happen instead of the
hold music when the calls are picked up?
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>

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From dane.newman at gmail.com Tue Jan 15 14:11:05 2013


From: dane.newman at gmail.com (Dane Newman)
Date: Tue, 15 Jan 2013 14:11:05 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <AAA51FE5-3A02-44BA-844E-1BE005EA3ABD@cisco.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
<CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
<1680820783548761203@unknownmsgid>
<CAL-DCK0NHxrOgk4RM=CX_absw=Kc2+aqYLOzR-9Oc3_CGEvsEQ@mail.gmail.com>
<F51BAB8A-F89A-41A3-94DA-9B3D899D1C75@cisco.com>
<CAL-DCK1fBRyZqK+dZq08go_FaFXvGzuYwHU311vrWuLbe3XL-A@mail.gmail.com>
<AAA51FE5-3A02-44BA-844E-1BE005EA3ABD@cisco.com>
Message-ID: <CAL-DCK08ubjtxC2GDE6n+ORDBBP2P=j=D9P2L12=BBrgJrW7ww@mail.gmail.com>

Ryan

What is the proper debug to use to caputre the useful information?

Dane

On Tue, Jan 15, 2013 at 12:42 PM, Ryan Ratliff <rratliff at cisco.com> wrote:

> Without sip messages I can't get any clues from that.
>
> -Ryan
>
> On Jan 15, 2013, at 12:35 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> Thanks Ryan for the input
>
>
> *On the call when I hold the call the following debug pops out....*
>
>
> *Jan 15 17:56:05.246:
> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
> passthru hdrs to
> container
> SIP: Attribute mid, level 1 instance 1 not found.
> SIP: (13938) Group (a= group line) attribute, level 65535 instance 1 not
> found.
> *Jan 15 17:56:05.274:
> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
> passthru headers to container
> SIP: Attribute mid, level 1 instance 1 not found.
> SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not
> found.
> *Jan 15 17:56:05.286:
> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
> passthru hdrs to
> container
> *Jan 15 17:56:05.302:
> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
> passthru headers to container
> SIP: Attribute mid, level 1 instance 1 not found.
> SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not
> found.
> SIP: Attribute mid, level 1 instance 1 not found.
> *Jan 15 17:56:05.322:
> //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS
> params for midcall INVITE
>
> *After I try to unhold the call the following debug comes out....*
> **
>
> *Jan 15 17:56:18.874:
> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
> passthru hdrs to
> container
> *Jan 15 17:56:18.894:
> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
> passthru headers to container
> SIP: Attribute mid, level 1 instance 1 not found.
> SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not
> found.
> SIP: Attribute mid, level 1 instance 1 not found.
> *Jan 15 17:56:18.906:
> //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS
> params for midcall INVITE
> Cisco3825#
> Cisco3825#
> Cisco3825#
>
> On Tue, Jan 15, 2013 at 9:42 AM, Ryan Ratliff <rratliff at cisco.com> wrote:
>
>> Given you have an ITSP it's most likely the initial hold that's failing,
>> which is only manifesting when you try to resume it. If you haven't
>> noticed already this is also very likely causing transfers to fail.
>>
>> Take a look at the SIP signaling for a call. I believe the most common
>> cause to this is the ITSP not handling our transition from
>> active->inactive->sendonly->active from hold to MOH to resume. The
>> "Duplex Streaming Enabled" parameter is there just for this type of problem.
>>
>> -Ryan
>>
>> On Jan 14, 2013, at 6:40 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> *Hello Kenneth*
>> **
>> *I have restarted both CUCM servers so this should have restarted the
>> services when the utils system restart happened*
>> **
>>
>> *on my router I see I am using g711 from the debug *
>> **
>> *I ran a debug voip ccapi inout *
>> **
>> *I connected a call calling from an external number to a DiD inside of
>> my system. Once the call was connected I put the call on hold and the
>> following debug came out..the music on hold played for the external caller
>> *
>>
>> *Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>> Stop Tone On Digit=FALSE, Tone=Null,
>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>> *Jan 14 23:47:40.783:
>> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742, Xmit Function=0x64204BAC
>> *Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>> Playout Max=1000(ms), Fax Nom=300(ms))
>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=170, Call Id=12742
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
>> Feature Type=50, Interface=0xC05A65AC, Call Id=12742
>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>> Playout Max=1000(ms), Fax Nom=300(ms))
>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=171, Call Id=12741
>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>> Interface=0xC05A65AC, Call Id=12742
>> *Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>> Stop Tone On Digit=FALSE, Tone=Null,
>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>> *Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=96, Call Id=12742
>> *Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.839:
>> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741, Xmit Function=0x64204BAC
>> *Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>> Playout Max=1000(ms), Fax Nom=300(ms))
>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=170, Call Id=12741
>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>> Playout Max=1000(ms), Fax Nom=300(ms))
>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=171, Call Id=12742
>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>> Interface=0xC05A65AC, Call Id=12742
>> Cisco3825#
>> Cisco3825#
>> Cisco3825#
>>
>>
>> *I then after that took off the hold and the following debug came out.
>> The call on the PSDN side still played the hold music while there was no
>> voice on the deskphone side.*
>>
>> *Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>> Stop Tone On Digit=FALSE, Tone=Null,
>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>> *Jan 14 23:47:40.783:
>> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742, Xmit Function=0x64204BAC
>> *Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>> Playout Max=1000(ms), Fax Nom=300(ms))
>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=170, Call Id=12742
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
>> Feature Type=50, Interface=0xC05A65AC, Call Id=12742
>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>> Playout Max=1000(ms), Fax Nom=300(ms))
>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=171, Call Id=12741
>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>> Interface=0xC05A65AC, Call Id=12742
>> *Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>> Stop Tone On Digit=FALSE, Tone=Null,
>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>> *Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=96, Call Id=12742
>> *Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.839:
>> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741, Xmit Function=0x64204BAC
>> *Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>> Playout Max=1000(ms), Fax Nom=300(ms))
>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=170, Call Id=12741
>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>> Call Id=12742,
>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>> Playout Max=1000(ms), Fax Nom=300(ms))
>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>> Call Id=12741,
>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event=171, Call Id=12742
>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>> Event Is Sent To Conferenced SPI(s) Directly
>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>> Interface=0xC05A65AC, Call Id=12742
>> Cisco3825#
>> Cisco3825#
>> Cisco3825#
>>
>> On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <kennethwhayes at
gmail.com>wrote:
>>
>>> Have you also restarted the Cisco IP Media Services?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>> My ITSP will only support 711ulaw for me currently I believe. They hard
>>> coded it with me when I was initially setting it up.
>>>
>>> Do you think this could be a codec issue? How would I go about
>>> identifying if it is?
>>>
>>> Dane
>>>
>>> On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <kennethwhayes at
gmail.com>wrote:
>>>
>>>> Have you tried different audio codecs?
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>>
>>>> Ryan (sorry I forgot to reply to all)
>>>>
>>>> Thanks for the Reply
>>>> Oddly enough we are.
>>>> This probably has something to do with MOH in general?
>>>>
>>>> Internally when I user puts another user on hold everything works. No
>>>> MOH plays and they can hold and unhold the call just fine.
>>>> I tested calling from an external number. Once I put the external
>>>> caller on hold the MOH played but I was unable to resume the call. When I
>>>> hit resume on the deskphone the MOH still played to the external caller and
>>>> there was no sound on the deskphone.
>>>>
>>>> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>>>
>>>>> Do you get similar behavior if you just hold and resume the call
>>>>> outside SNR features?
>>>>>
>>>>> -Ryan
>>>>>
>>>>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com>
>>>>> wrote:
>>>>>
>>>>> Using keyboard-interactive authentication.
>>>>>
>>>>> Password:
>>>>>
>>>>>
>>>>> Cisco3825#
>>>>>
>>>>> Cisco3825#sh ver
>>>>>
>>>>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M),
>>>>> Version 15.1
>>>>> (4)M5, RELEASE SOFTWARE (fc1)
>>>>>
>>>>> Technical Support: http://www.cisco.com/techsupport
>>>>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>>>>>
>>>>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>>>>
>>>>>
>>>>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>>>>
>>>>>
>>>>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>>>>>
>>>>> System returned to ROM by power-on
>>>>>
>>>>> System image file is
>>>>> "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>>>>> Last reload type: Normal Reload
>>>>>
>>>>>
>>>>>
>>>>> This product contains cryptographic features and is subject to United
>>>>>
>>>>> States and local country laws governing import, export, transfer and
>>>>>
>>>>> use. Delivery of Cisco cryptographic products does not imply
>>>>>
>>>>> third-party authority to import, export, distribute or use encryption.
>>>>>
>>>>> Importers, exporters, distributors and users are responsible for
>>>>>
>>>>> compliance with U.S. and local country laws. By using this product you
>>>>>
>>>>> agree to comply with applicable laws and regulations. If you are
>>>>> unable
>>>>> to comply with U.S. and local laws, return this product immediately.
>>>>>
>>>>>
>>>>> A summary of U.S. laws governing Cisco cryptographic products may be
>>>>> found at:
>>>>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>>>>
>>>>> If you require further assistance please contact us by sending email
>>>>> to
>>>>> export at cisco.com.
>>>>>
>>>>>
>>>>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>>>>>
>>>>> Processor board ID FTX1237A1T0
>>>>>
>>>>> 2 Gigabit Ethernet interfaces
>>>>>
>>>>> 2 Channelized T1/PRI ports
>>>>>
>>>>> 1 Virtual Private Network (VPN) Module
>>>>>
>>>>> DRAM configuration is 64 bits wide with parity enabled.
>>>>>
>>>>> 479K bytes of NVRAM.
>>>>>
>>>>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>>>>
>>>>>
>>>>>
>>>>> License Info:
>>>>>
>>>>>
>>>>> License UDI:
>>>>>
>>>>>
>>>>> -------------------------------------------------
>>>>>
>>>>> Device# PID SN
>>>>>
>>>>> Sent from my mobile device
>>>>>
>>>>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com>
>>>>> wrote:
>>>>>
>>>>> What version of code are you running on the CUBE?
>>>>>
>>>>> Sent from my iPhone
>>>>>
>>>>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com>
>>>>> wrote:
>>>>>
>>>>> Hello
>>>>>
>>>>> I have an issue when users are connected to a call and hit the
>>>>> mobility soft key button on 9971 phones when a call is active, the phone
>>>>> system rings on the mobile number configured in the system. When they pick
>>>>> up the the mobile number it just plays what sounds like hold music on both
>>>>> ends of the call (I believe this music is coming from cucm but I haven't
>>>>> heard it before) instead of providing 2 way voice.
>>>>>
>>>>> In another senario with what I believe is the same issue. If a user
>>>>> picks up on there cell phone first (using single number reach) opposed to
>>>>> the deskphone the call is connected with 2 way voice and no issues exist.
>>>>> If the user then hangs up his cell phone with the intent to take the call
>>>>> on his deskphone the calling party starts hearing the hold music. Once the
>>>>> user picks up the call on his deskphone he hears nothing but the calling
>>>>> party is still hearing the hold music. It is not working as intended where
>>>>> 2 way voice happens once the user hangs up his mobile phone and picks up on
>>>>> his deskphone 2 way voice should happen.
>>>>>
>>>>> My topology is as follows..
>>>>>
>>>>>
>>>>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>>>>
>>>>> Calls are sent back out the SIP trunk to the ITSP when using mobile
>>>>> connect/snr.
>>>>>
>>>>> Does anyone have any ideas how I can make 2 way voice happen instead
>>>>> of the hold music when the calls are picked up?
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
>>>>>
>>>>
>>>
>>
>>
>
>
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From rratliff at cisco.com Tue Jan 15 14:28:59 2013


From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 15 Jan 2013 14:28:59 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <CAL-DCK08ubjtxC2GDE6n+ORDBBP2P=j=D9P2L12=BBrgJrW7ww@mail.gmail.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
<CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
<1680820783548761203@unknownmsgid>
<CAL-DCK0NHxrOgk4RM=CX_absw=Kc2+aqYLOzR-9Oc3_CGEvsEQ@mail.gmail.com>
<F51BAB8A-F89A-41A3-94DA-9B3D899D1C75@cisco.com>
<CAL-DCK1fBRyZqK+dZq08go_FaFXvGzuYwHU311vrWuLbe3XL-A@mail.gmail.com>
<AAA51FE5-3A02-44BA-844E-1BE005EA3ABD@cisco.com>
<CAL-DCK08ubjtxC2GDE6n+ORDBBP2P=j=D9P2L12=BBrgJrW7ww@mail.gmail.com>
Message-ID: <3F5FC5E5-D7C0-40D8-B0FE-FC15A24FC9F6@cisco.com>

ccsip message is what I'd go with just to see the signaling with no other stuff.
Depending on what that shows and what your gateway is doing to the signals you may
need to expand from there.

-Ryan

On Jan 15, 2013, at 2:11 PM, Dane Newman <dane.newman at gmail.com> wrote:

Ryan

What is the proper debug to use to caputre the useful information?

Dane

On Tue, Jan 15, 2013 at 12:42 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
Without sip messages I can't get any clues from that.

-Ryan

On Jan 15, 2013, at 12:35 PM, Dane Newman <dane.newman at gmail.com> wrote:

Thanks Ryan for the input

On the call when I hold the call the following debug pops out....

*Jan 15 17:56:05.246: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer:


Unable to add passthru hdrs to
container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13938) Group (a= group line) attribute, level 65535 instance 1 not found.
*Jan 15 17:56:05.274: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer:
Unable to add
passthru headers to container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.
*Jan 15 17:56:05.286: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer:
Unable to add passthru hdrs to
container
*Jan 15 17:56:05.302: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer:
Unable to add
passthru headers to container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: Attribute mid, level 1 instance 1 not found.
*Jan 15 17:56:05.322: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could
not modify QoS params for midcall INVITE

After I try to unhold the call the following debug comes out....

*Jan 15 17:56:18.874: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer:


Unable to add passthru hdrs to
container
*Jan 15 17:56:18.894: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer:
Unable to add
passthru headers to container
SIP: Attribute mid, level 1 instance 1 not found.
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: Attribute mid, level 1 instance 1 not found.
*Jan 15 17:56:18.906: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could
not modify QoS params for midcall INVITE
Cisco3825#
Cisco3825#
Cisco3825#

On Tue, Jan 15, 2013 at 9:42 AM, Ryan Ratliff <rratliff at cisco.com> wrote:
Given you have an ITSP it's most likely the initial hold that's failing, which is
only manifesting when you try to resume it. If you haven't noticed already this
is also very likely causing transfers to fail.

Take a look at the SIP signaling for a call. I believe the most common cause to
this is the ITSP not handling our transition from active->inactive->sendonly-
>active from hold to MOH to resume. The "Duplex Streaming Enabled" parameter is
there just for this type of problem.

-Ryan

On Jan 14, 2013, at 6:40 PM, Dane Newman <dane.newman at gmail.com> wrote:

Hello Kenneth

I have restarted both CUCM servers so this should have restarted the services when
the utils system restart happened

on my router I see I am using g711 from the debug

I ran a debug voip ccapi inout

I connected a call calling from an external number to a DiD inside of my system.


Once the call was connected I put the call on hold and the following debug came
out..the music on hold played for the external caller

*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:


Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742, Xmit Function=0x64204BAC
*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12742
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12741
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=96, Call Id=12742
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741, Xmit Function=0x64204BAC
*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12741
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12742
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
Cisco3825#
Cisco3825#
Cisco3825#

I then after that took off the hold and the following debug came out. The call on
the PSDN side still played the hold music while there was no voice on the deskphone
side.

*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:


Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742, Xmit Function=0x64204BAC
*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12742
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12741
*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=12741
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=96, Call Id=12742
*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741, Xmit Function=0x64204BAC
*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event=170, Call Id=12741
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call
Id=12742,
Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call
Id=12741,
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event=171, Call Id=12742
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
Interface=0xC05A65AC, Call Id=12742
Cisco3825#
Cisco3825#
Cisco3825#

On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
Have you also restarted the Cisco IP Media Services?

Sent from my iPhone

On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com> wrote:

> My ITSP will only support 711ulaw for me currently I believe. They hard coded it
with me when I was initially setting it up.
>
> Do you think this could be a codec issue? How would I go about identifying if it
is?
>
> Dane
>
> On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <kennethwhayes at gmail.com>
wrote:
> Have you tried different audio codecs?
>
> Sent from my iPhone
>
> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
>> Ryan (sorry I forgot to reply to all)
>>
>> Thanks for the Reply
>> Oddly enough we are.
>> This probably has something to do with MOH in general?
>>
>> Internally when I user puts another user on hold everything works. No MOH plays
and they can hold and unhold the call just fine.
>> I tested calling from an external number. Once I put the external caller on hold
the MOH played but I was unable to resume the call. When I hit resume on the
deskphone the MOH still played to the external caller and there was no sound on the
deskphone.
>>
>> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>> Do you get similar behavior if you just hold and resume the call outside SNR
features?
>>
>> -Ryan
>>
>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Using keyboard-interactive authentication.
>> Password:
>>
>> Cisco3825#
>> Cisco3825#sh ver
>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1
>> (4)M5, RELEASE SOFTWARE (fc1)
>> Technical Support: http://www.cisco.com/techsupport
>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>
>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>
>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>> System returned to ROM by power-on
>> System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>> Last reload type: Normal Reload
>>
>>
>> This product contains cryptographic features and is subject to United
>> States and local country laws governing import, export, transfer and
>> use. Delivery of Cisco cryptographic products does not imply
>> third-party authority to import, export, distribute or use encryption.
>> Importers, exporters, distributors and users are responsible for
>> compliance with U.S. and local country laws. By using this product you
>> agree to comply with applicable laws and regulations. If you are unable
>> to comply with U.S. and local laws, return this product immediately.
>>
>> A summary of U.S. laws governing Cisco cryptographic products may be found at:
>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>
>> If you require further assistance please contact us by sending email to
>> export at cisco.com.
>>
>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>> Processor board ID FTX1237A1T0
>> 2 Gigabit Ethernet interfaces
>> 2 Channelized T1/PRI ports
>> 1 Virtual Private Network (VPN) Module
>> DRAM configuration is 64 bits wide with parity enabled.
>> 479K bytes of NVRAM.
>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>
>>
>> License Info:
>>
>> License UDI:
>>
>> -------------------------------------------------
>> Device# PID SN
>>
>> Sent from my mobile device
>>
>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
>>
>>> What version of code are you running on the CUBE?
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>>> Hello
>>>>
>>>> I have an issue when users are connected to a call and hit the mobility soft
key button on 9971 phones when a call is active, the phone system rings on the
mobile number configured in the system. When they pick up the the mobile number it
just plays what sounds like hold music on both ends of the call (I believe this
music is coming from cucm but I haven't heard it before) instead of providing 2 way
voice.
>>>>
>>>> In another senario with what I believe is the same issue. If a user picks up
on there cell phone first (using single number reach) opposed to the deskphone the
call is connected with 2 way voice and no issues exist. If the user then hangs up
his cell phone with the intent to take the call on his deskphone the calling party
starts hearing the hold music. Once the user picks up the call on his deskphone he
hears nothing but the calling party is still hearing the hold music. It is not
working as intended where 2 way voice happens once the user hangs up his mobile
phone and picks up on his deskphone 2 way voice should happen.
>>>>
>>>> My topology is as follows..
>>>>
>>>>
>>>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>>>
>>>> Calls are sent back out the SIP trunk to the ITSP when using mobile
connect/snr.
>>>>
>>>> Does anyone have any ideas how I can make 2 way voice happen instead of the
hold music when the calls are picked up?
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>

-------------- next part --------------


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From dane.newman at gmail.com Tue Jan 15 15:12:44 2013


From: dane.newman at gmail.com (Dane Newman)
Date: Tue, 15 Jan 2013 15:12:44 -0500
Subject: [cisco-voip] Mobility Issue
In-Reply-To: <3F5FC5E5-D7C0-40D8-B0FE-FC15A24FC9F6@cisco.com>
References: <CAL-DCK0WQqJiKyFC0d81-4S8zjk5BU+mumsyK7Pk_qm1Vvzfcw@mail.gmail.com>
<-4286606911770185612@unknownmsgid>
<07980D5A-B85E-4C15-9970-3F1697C7BDB0@gmail.com>
<C63D5DCC-4316-4477-8A93-8775E089CEA0@cisco.com>
<CAL-DCK2pOUx1_-EvvABnLTrW-kAqQGgu_JNd0JbJJumCnDGk7w@mail.gmail.com>
<-5635080451137355987@unknownmsgid>
<CAL-DCK1vz8Wmk_3x+w-kVhHY=kM-Ad1GcaLUaPU3u1qBmeMmrw@mail.gmail.com>
<1680820783548761203@unknownmsgid>
<CAL-DCK0NHxrOgk4RM=CX_absw=Kc2+aqYLOzR-9Oc3_CGEvsEQ@mail.gmail.com>
<F51BAB8A-F89A-41A3-94DA-9B3D899D1C75@cisco.com>
<CAL-DCK1fBRyZqK+dZq08go_FaFXvGzuYwHU311vrWuLbe3XL-A@mail.gmail.com>
<AAA51FE5-3A02-44BA-844E-1BE005EA3ABD@cisco.com>
<CAL-DCK08ubjtxC2GDE6n+ORDBBP2P=j=D9P2L12=BBrgJrW7ww@mail.gmail.com>
<3F5FC5E5-D7C0-40D8-B0FE-FC15A24FC9F6@cisco.com>
Message-ID: <CAL-DCK0fuh79KenRe0P-SXGwC2xJj5+X4C-h_fM0Wx=+6m_mHg@mail.gmail.com>

Thanks Ryan

I see I am always getting a 200 ok message after my invites from the debug

*Putting a call on HOLD*

*Jan 15 20:19:28.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22

Date: Tue, 15 Jan 2013 19:57:35 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

Supported: timer,resource-priority,replaces

Min-SE: 1800

User-Agent: Cisco-CUCM8.6

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY

CSeq: 102 INVITE

Max-Forwards: 70

Expires: 180

Allow-Events: presence

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uas

P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>

Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;party=calling;screen=yes;privacy=off

Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video

Content-Type: application/sdp
Content-Length: 240

v=0

o=CiscoSystemsCCM-SIP 7322 3 IN IP4 10.1.80.10

s=SIP Call

c=IN IP4 0.0.0.0

b=TIAS:64000

b=AS:64

t=0 0

m=audio 21476 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=inactive

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

*Jan 15 20:19:28.094: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0

Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0

From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net


>;tag=2E6BC0B0-2268

To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e

Date: Tue, 15 Jan 2013 20:19:28 GMT

Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 3257897472-0000065536-0000000035-0173015306

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER

CSeq: 103 INVITE


Max-Forwards: 70

Timestamp: 1358281168

Contact: <sip:6784563290 at 98.192.104.214:5060>

Expires: 180

Allow-Events: telephone-event

Authorization: Digest username="6784563290",realm="asterisk",uri="


sip:17705439047 at 64.154.41.150:5060
",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5

Session-Expires: 1800;refresher=uas

Content-Type: application/sdp

Content-Length: 289

v=0

o=CiscoSystemsSIP-GW-UserAgent 3168 2739 IN IP4 98.192.104.214

s=SIP Call

c=IN IP4 98.192.104.214

t=0 0

m=audio 19458 RTP/AVP 0 101 19

c=IN IP4 98.192.104.214

a=inactive

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:19 CN/8000

a=ptime:20

*Jan 15 20:19:28.094: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22


Date: Tue, 15 Jan 2013 20:19:28 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 98.192.104.214:5060


;branch=z9hG4bK691F12E0;received=98.192.104.214

From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net


>;tag=2E6BC0B0-2268

To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e

Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214

CSeq: 103 INVITE

Server: Asterisk PBX 1.6.2.13

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Require: timer

Session-Expires: 1800;refresher=uas

Contact: <sip:17705439047 at 64.154.41.150>

Content-Length: 0

*Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 98.192.104.214:5060


;branch=z9hG4bK691F12E0;received=98.192.104.214

From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net


>;tag=2E6BC0B0-2268

To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e

Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214


CSeq: 103 INVITE

Server: Asterisk PBX 1.6.2.13

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Require: timer

Session-Expires: 1800;refresher=uas

Contact: <sip:17705439047 at 64.154.41.150>

Content-Type: application/sdp

Content-Length: 239

v=0

o=root 1685873050 1685873052 IN IP4 64.154.41.150

s=Asterisk PBX 1.6.2.13

c=IN IP4 64.154.41.150

t=0 0

m=audio 13014 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=inactive

*Jan 15 20:19:28.118: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22

Date: Tue, 15 Jan 2013 20:19:28 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

CSeq: 102 INVITE


Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <sip:17705439047 at 10.1.200.1


>;party=called;screen=no;privacy=off

Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Session-Expires: 1800;refresher=uas

Require: timer

Supported: timer

Content-Type: application/sdp

Content-Length: 253

v=0

o=CiscoSystemsSIP-GW-UserAgent 4444 5479 IN IP4 10.1.200.1

s=SIP Call

c=IN IP4 10.1.200.1

t=0 0

m=audio 19514 RTP/AVP 0 101

c=IN IP4 10.1.200.1

a=inactive

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

*Jan 15 20:19:28.118: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0

Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK6920266D


From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>;tag=2E6BC0B0-2268

To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e

Date: Tue, 15 Jan 2013 20:19:28 GMT

Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214

Max-Forwards: 70

CSeq: 103 ACK

Authorization: Digest username="6784563290",realm="asterisk",uri="


sip:17705439047 at 64.154.41.150:5060
",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5

Allow-Events: telephone-event

Content-Length: 0

*Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28b4b1305a0

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22

Date: Tue, 15 Jan 2013 19:57:35 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

Max-Forwards: 70

CSeq: 102 ACK

Allow-Events: presence

Content-Length: 0

*Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22


Date: Tue, 15 Jan 2013 19:57:35 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

Supported: timer,resource-priority,replaces

Min-SE: 1800

User-Agent: Cisco-CUCM8.6

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY

CSeq: 103 INVITE

Max-Forwards: 70

Expires: 180

Allow-Events: presence

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uas

P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>

Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;party=calling;screen=yes;privacy=off

Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video

Content-Length: 0

*Jan 15 20:19:28.126: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0

Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3

From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net


>;tag=2E6BC0B0-2268

To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e

Date: Tue, 15 Jan 2013 20:19:28 GMT

Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER

CSeq: 104 INVITE

Max-Forwards: 70

Timestamp: 1358281168

Contact: <sip:6784563290 at 98.192.104.214:5060>

Expires: 180

Allow-Events: telephone-event

Authorization: Digest username="6784563290",realm="asterisk",uri="


sip:17705439047 at 64.154.41.150:5060
",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5

Session-Expires: 1800;refresher=uas

Content-Length: 0

*Jan 15 20:19:28.126: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22

Date: Tue, 15 Jan 2013 20:19:28 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

CSeq: 103 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 98.192.104.214:5060


;branch=z9hG4bK69211AB3;received=98.192.104.214
From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>;tag=2E6BC0B0-2268

To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e

Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214

CSeq: 104 INVITE

Server: Asterisk PBX 1.6.2.13

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Require: timer

Session-Expires: 1800;refresher=uas

Contact: <sip:17705439047 at 64.154.41.150>

Content-Length: 0

*Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 98.192.104.214:5060


;branch=z9hG4bK69211AB3;received=98.192.104.214

From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net


>;tag=2E6BC0B0-2268

To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e

Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214

CSeq: 104 INVITE

Server: Asterisk PBX 1.6.2.13

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Require: timer

Session-Expires: 1800;refresher=uas

Contact: <sip:17705439047 at 64.154.41.150>

Content-Type: application/sdp

Content-Length: 333

v=0
o=root 1685873050 1685873053 IN IP4 64.154.41.150

s=Asterisk PBX 1.6.2.13

c=IN IP4 64.154.41.150

t=0 0

m=audio 13014 RTP/AVP 3 8 0 18 101

a=rtpmap:3 GSM/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=inactive

*Jan 15 20:19:28.150: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22

Date: Tue, 15 Jan 2013 20:19:28 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

CSeq: 103 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <sip:17705439047 at 10.1.200.1


>;party=called;screen=no;privacy=off

Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>

Supported: replaces
Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Session-Expires: 1800;refresher=uas

Require: timer

Supported: timer

Content-Type: application/sdp

Content-Length: 277

v=0

o=CiscoSystemsSIP-GW-UserAgent 4444 5480 IN IP4 10.1.200.1

s=SIP Call

c=IN IP4 10.1.200.1

t=0 0

m=audio 19514 RTP/AVP 0 101 19

c=IN IP4 10.1.200.1

a=inactive

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20

*Jan 15 20:19:28.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28d3eadaab3

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22

Date: Tue, 15 Jan 2013 19:57:35 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

Max-Forwards: 70
CSeq: 103 ACK

Allow-Events: presence

Content-Type: application/sdp

Content-Length: 209

v=0

o=CiscoSystemsCCM-SIP 7322 4 IN IP4 10.1.80.10

s=SIP Call

c=IN IP4 0.0.0.0

b=TIAS:64000

b=AS:64

t=0 0

m=audio 21476 RTP/AVP 0

a=X-cisco-media:nomedia

a=rtpmap:0 PCMU/8000

a=ptime:20

a=inactive

*Jan 15 20:19:28.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0

Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK692226EA

From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net


>;tag=2E6BC0B0-2268

To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e

Date: Tue, 15 Jan 2013 20:19:28 GMT

Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214

Max-Forwards: 70

CSeq: 104 ACK

Authorization: Digest username="6784563290",realm="asterisk",uri="


sip:17705439047 at 64.154.41.150:5060
",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5

Allow-Events: telephone-event
Content-Type: application/sdp

Content-Length: 251

v=0

o=CiscoSystemsSIP-GW-UserAgent 3168 2740 IN IP4 98.192.104.214

s=SIP Call

c=IN IP4 0.0.0.0

t=0 0

m=audio 19458 RTP/AVP 0 101

c=IN IP4 0.0.0.0

a=inactive

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

*Unholding the call the MOH continues on the previously held caller while
the user hears nothing*

**

*Jan 15 20:19:35.166: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22

Date: Tue, 15 Jan 2013 19:57:42 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

Supported: timer,resource-priority,replaces

Min-SE: 1800

User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY

CSeq: 104 INVITE

Max-Forwards: 70

Expires: 180

Allow-Events: presence

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uas

P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>

Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;party=calling;screen=yes;privacy=off

Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video;audio;video

Content-Length: 0

*Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0

Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672

From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net


>;tag=2E6BC0B0-2268

To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e

Date: Tue, 15 Jan 2013 20:19:35 GMT

Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 3257897472-0000065536-0000000035-0173015306

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER

CSeq: 105 INVITE

Max-Forwards: 70
Timestamp: 1358281175

Contact: <sip:6784563290 at 98.192.104.214:5060>

Expires: 180

Allow-Events: telephone-event

Authorization: Digest username="6784563290",realm="asterisk",uri="


sip:17705439047 at 64.154.41.150:5060
",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5

Session-Expires: 1800;refresher=uas

Content-Length: 0

*Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c

From: "Dane Newman" <sip:6784563290 at 10.1.80.10


>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957

To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22

Date: Tue, 15 Jan 2013 20:19:35 GMT

Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10

CSeq: 104 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 98.192.104.214:5060


;branch=z9hG4bK69232672;received=98.192.104.214

From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net


>;tag=2E6BC0B0-22