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IAC / SV9500 to CUCM Integration Technical Information Bulletin
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Document ID: TIB-IAC / SV9500 to CUCM Integration-1.0 NEC Australia Pty. Ltd. © 2015
IAC / SV9500 to CUCM Integration Technical Information Bulletin
Overview
Product Category
General Information
Document ID: TIB-IAC / SV9500 to CUCM Integration-1.0 NEC Australia Pty. Ltd. © 2015
IAC / SV9500 to CUCM Integration Technical Information Bulletin
Table of contents
Revision history..............................................................................................................................................2
Overview........................................................................................................................................................3
1 Introduction.............................................................................................................................................6
6. Conclusion...............................................................................................................................................21
Document ID: TIB-IAC / SV9500 to CUCM Integration-1.0 NEC Australia Pty. Ltd. © 2015
IAC / SV9500 to CUCM Integration Technical Information Bulletin
Document ID: TIB-IAC / SV9500 to CUCM Integration-1.0 NEC Australia Pty. Ltd. © 2015
IAC / SV9500 to CUCM Integration Technical Information Bulletin
1 Introduction
Contained in this report is a step-by-step process for configuring both SV9500 and Cisco
devices, tested / supported features and the service conditions of integration via SIP
Trunks.
Basic Call From the SV9500 – to the CUCM, using Off-hook Pass
dialling.
1
Where the SV9500 is From CUCM to the SV9500, using Pre-dialling. Pass
originator
Basic Call
Call from Cisco to NEC when SIP trunks are busy Pass
4 Calls from Cisco to NEC with NEC Call Forward – Pass Note: - Display on the NEC Phone (Forwarded
Cont. Busy to another NEC Phone Station) does not show ‘FDB’
Calls from NEC to Cisco with Cisco Call Forward Pass Note: - Display on the NEC Phone (Forwarded
– Nans to NEC Phone Station) does not show ‘FDN’
Call from NEC to Cisco Pass CLI is displayed when calls are made from CUCM to
NEC and vice versa.
5 Calling Line
Identification – Call from Cisco to NEC Pass
Presentation (CLIP)
Call from NEC to Cisco Pass Calls established and ensured that opposite party
number was displayed on a Digital Multiline.
6 Calling Party Number
Display Call from Cisco to NEC Pass
Call from NEC to Cisco Pass Calling name is display from the PABX when calls are
made CUCM to NEC and vice versa.
7 Calling Name Display
Call from Cisco to NEC Pass
8 Follow Me Calls from Cisco to NEC Pass Calls made from the CUCM to NEC phones correctly
followed the Call Forward – I’m here feature.
9 Call History Is the call shown in the call history on the phone Pass All calls from CUCM to NEC and NEC to CUCM were
for calls from Cisco to NEC and NEC to Cisco? shown correctly in the Call History.
10 Enquiry / Shuttle Transfer in conversation to another handset. Pass Calls made from CUCM to SV9500, SV9500 transfers
the call to CUCM. CUCM stations connected but
SPTS shows 2 x SIP Trunks busy in the SV9500.
11 Shuttle between two parties Pass Once call is completed, trunks are cleared.
18 Call Park X
Test Call Park – System
20 DID Termination
- Calls from Cisco to IVR on NEC X Does not operate correctly, every time CUCM station
A is set for CFWD – All to Voicemail, the caller will
receive their own Voice Mail greeting and not the
- Calls from Cisco to UM on NEC Fail mailbox of the CUCM Station A.
21 Hold / Retrieve Hold Whilst in Conversation, place the call on hold and Pass
retrieve the caller
22
Day / Night Change – Station Pass
Trunk Answer for Any Station (TAS) X Not a feature of SIP Trunks.
Station Hunting Station Hunting – Circular Pass Circular Hunt programmed in SV9500.
23
Station Hunting – Terminal Pass
24 Station Messaging Ensure that outgoing and incoming calls are Pass
Detail Recording generating SMDR Records.
(SMDR)
CUCM to SV9500
25 Station Speed Dial Ensure that outgoing calls can be made using Pass
Station Speed Dial
SV9500 to CUCM
26 Station Speed Dial Ensure that outgoing calls can be made using X Only Station Speed Dial programmed on CUCM.
Station Speed Dial
CUCM to SV9500
27 Trunk to Trunk Receive an incoming call and transfer the call Pass
Connection outside.
CUCM to SV9500
28 Uniform Call Distribution Created a UCD Group and ensured calls are Pass UCD Group programmed in SV9500
(UCD) distributed correctly
CUCM to SV9500
1 Attendant Camp-on with Calls from the CUCM to the LDN of the IAC, Pass
Tone indication transferred and camped onto a Busy Extension
within the SV9500
2 Attendant Controlled Calls from the CUCM to the IAC then the IAC sets Pass
Conference up a conference call to parties within the SV9500.
Call from the CUCM to the IAC console then Pass Calls not answered on the SV9500 will result with the
transferred to a station within the SV9500. The IAC being recalled.
IAC should not be able to re-enter the
conversation until after the pre-determined recall
3 Attendant Lock Out timer
Call from the SV9500 to the IAC console then X Calls not answered on the CUCM do not result with
transferred to a station within the CUCM. The IAC the IAC being recalled.
should not be able to re-enter the conversation
until after the pre-determined recall timer
5 Attendant Override Call from the CUCM to a Station within the Pass
SV9500. Verify that the IAC console can enter the
busy connection and join / monitor the
conversation.
Call from a station in the SV9500 to the IAC X Calls not answered on the CUCM do not result with
Console then transferred to an idle station in the the IAC being recalled.
CUCM
7 Audible Indication Calls to the IAC Console from the CUCM Pass
Control
8 Attendant Night Transfer Calls from the CUCM to a station within the X Only calls direct to ‘9’ from CUCM will follow Night
SV9500, then transferred to the IAC set in Night Destination.
Mode
Calls from the CUCM to the IAC Console X Couldn’t get the feature to work.
transferred to a station in the SV9500 with an
account code entered.
9 Account Code –
Attendant Calls from the SV9500 to the IAC Console X Couldn’t get the feature to work.
transferred to a station in the CUCM with an
account code entered
10 Automatic Change of Calls from the CUCM to the IAC N/A This feature is provided for Incoming calls to the
Night Service Attendant Console [A-3] from a Central Office Trunk
or Ring-Down lines.
12 Attendant Overflow Calls to the IAC Console not answered within the N/A Note: - For Australia, refer to [A-55] Automatic
predetermined time to a station within the SV9500 Change of Night Service
13 Blind Transfer Calls from the CUCM to a station in the SV9500 Pass
blind transferred to the IAC Console
16 Call Waiting Place Consecutive calls from the CUCM to the Pass
IAC
17 Call Pick-up Direct Calls from the CUCM to a station in the SV9500 N/A This feature cannot be accessed from the IAC
picked up from the IAC Console Console.
18 Calling number Display Calls from the CUCM to the IAC Console Pass
– IAC
20 Call Forwarding Calls from the CUCM to an unassigned station Pass Note: - There is No indication on the IAC that the call
Intercept / within the SV9500 system. is an intercepted. Call presented as a direct call from
Announcement the CUCM
21 Day / Night Class of Calls from the CUCM to the SV9500 ? Not tested
Service
22 Incoming Call Calls from the CUCM to the IAC Console Pass
Identification
23 Listed Directory Number Calls from the CUCM to the IAC Console Pass
24 LDN Night Connection / Calls from the CUCM to the LDN of the IAC Pass
Outside Console
26 Night Connection Calls from the CUCM to the IAC Console Pass
Fixed / Flexible
28 Restriction from Calls to the IAC console from station within the ? Not tested
Outgoing Calls SV9500 with restriction class prohibiting outgoing
calls to be transferred to the CUCM.
30 Trunk Answer from any Calls from the CUCM to the SV9500 Pass
Station – Attendant
31 Time Display Calls from the CUCM to the IAC Console Pass
Link Reconstitution between the SV9500 system and CUCM is not supported.
Incoming SIP Trunk calls transferred back onto SIP Trunks will hold up 2 x SIP Trunks for
the duration of the call. I.E.: - An incoming call from the CUCM that is transferred back to
the CUCM will have 2 x SIP Trunk connections.
6. Conclusion
The results of the interoperability tests via direct SIP Trunking clearly indicate that NEC
can seize upon the opportunity of integrating SV9500 platforms and Cisco Call Manager
without carrier’s involvements.
The emphasis of testing was the IAC Console / SV9500 features available to CUCM via SIP
Trunking. These positive results show that NEC should direct its attention to the
opportunity within the WA Health Sector.
Voice traffic must be considered when looking at SIP Trunk integration as Link
Reconstitution between the SV9500 platform and CUCM is not supported and creates a
trombone effect when transferring calls.
[2015-05-18 09:50:32]
[SP-4058:02.00.00.00]
0 = MAC Data
1 = DRS Data
9 = Exit
Ether[1]
set signaling_port_no :
---------------------------------------------------
set registration_port_no :
---------------------------------------------------
set pre_negotiation_port_no :
---------------------------------------------------
set ipx_rtp_port_no :
---------------------------------------------------
set sip_rtp_port_no :
---------------------------------------------------
set h245_base_port_no :
---------------------------------------------------
set drs_qos :
PRECEDENCE :5
DELAY :0
THROUGHPUT :0
RELIABILITY :0
COST :0
---------------------------------------------------
set rtp_qos :
PRECEDENCE :5
DELAY :0
THROUGHPUT :0
RELIABILITY :0
COST :0
set sip_qos :
PRECEDENCE :5
DELAY :0
THROUGHPUT :0
RELIABILITY :0
COST :0
---------------------------------------------------
set domain :
Domain Name :0
---------------------------------------------------
set dnsaddress :
---------------------------------------------------
set hc_alarm :
---------------------------------------------------
set hc_timer :
-----------------------------------------
set musictype :
---------------------------------------------------
set call_hold :
---------------------------------------------------
set name_display :
---------------------------------------------------
set ip_version :
IP Version : IPv4
---------------------------------------------------
set country_code :
---------------------------------------------------
set softdsp :
---------------------------------------------------
Ether[1]
IP Address : 192.168.0.110
set sip_server :
Domain Name :0
---------------------------------------------------
set sip_register :
---------------------------------------------------
set keynumber :
Keynumber :0
---------------------------------------------------
set auth :
User ID :0
Password :0
---------------------------------------------------
set privacy_pattern :
---------------------------------------------------
set session_timer :
Session-timer : enable
Session-expires : 180sec
Refresher : UAC
---------------------------------------------------
set prack :
set cdn_pattern :
---------------------------------------------------
set sip_port_no :
--------------------------------------------------
set 184toprivacy :
---------------------------------------------------
set slipresp :
---------------------------------------------------
set self_sip_domain :
------------------------------------------------
set dtmf_mode :
---------------------------------------------------
set 183rbt :
---------------------------------------------------
set rtp_pathon :
---------------------------------------------------
set another_keynumber :
Another Keynumber 1 :0
Another Keynumber 2 :0
Another Keynumber 3 :0
---------------------------------------------------
set multi_regist :
---------------------------------------------------
set sip_tel_service :
---------------------------------------------------
set reg_interval :
---------------------------------------------------
set update :
---------------------------------------------------
set auth_header :
---------------------------------------------------
set cause_table :
---------------------------------------------------
set response_table :
---------------------------------------------------
set cpn_pattern :
---------------------------------------------------
set cc_convert :
---------------------------------------------------
set check_number :
---------------------------------------------------
set out_of_area_code :
---------------------------------------------------
set sdp_style :
---------------------------------------------------
set call_id_relay :
---------------------------------------------------
set sip_accfilter :
---------------------------------------------------
set sip_transport :
---------------------------------------------------
set dns_option :
---------------------------------------------------
set codec_change :
---------------------------------------------------
set no_media_code :
---------------------------------------------------
set ppi_to_pai :
---------------------------------------------------
Ether[1]
Ether[2]
Numbering Plan
IP Configuration
MGPRI 192.168.0.107
Assignment Procedure
Index64,
Index187=00
Index 810 Bit2 = 0/1 [Internal Holding Tone (IP terminals)/External Holding Tone
Note 4: This system data is required to enable Codec Change function and G.711 Fax
renegotiation in addition to a configuration command setting with “set codec_change” in
the MG-SIP side. For more information, refer to the configuration command for MG-
SIP128/96 and MG-SIP16 in Peripheral Equipment Description (IP Devices).
Note: When a change has been made, it is necessary to initialize the Internal PHI-BRI
module with the ADPM command.
LOC-ID = 0
Select the Type of Signalling Packet Service: Specify the Quality of Service (IP
PRECEDENCE/DIFFSERV).
Note: One of the following is required to allow the changes to Setting Type of Signalling
Packet Service data for already-operating equipment (such as PH and IP station) to take
effect.
• Initialize the Internal PHI-BRI (Initialize the Internal PHI-BRI module with the
ADPM command) or system
Example
Note: When you change the default port number with this command, you need to
initialize the Internal PHI-BRI module with the ADPM command.
Menu = Initialize
Example
PH LENS: Specify an even-numbered Group Number. The LENS can be allocated on either
a Virtual PIR or a Physical PIR.
Note: When the total number of channels using PHI-BRI exceeds 256, increase the value
of REGMAX.
Example
STEP 7: ARTD/ARTDN - Specify the route class data on Virtual Speech Channel as follows.
Note: To change the data for existing route (RT), initialize or reboot the trunk device such
as MG-SIP.
CDN72: FXJS (Jitter Buffer Size for IP FAX [1-30] 10ms) ex. Enter “15” for 150ms.
CDN74: FXPS (Payload Size for IP FAX [1-4] 10ms) ex. Enter “2” for 20ms.
Note: To change the data for existing route (RT), initialize or reboot the trunk device such
as MG-SIP.
STEP 9: ATRK - Specify trunk data on Virtual Speech Channel and Virtual Signaling
Channel.
STEP 10: ADPC - Register Point Code on Virtual Speech Channel and Virtual Signalling
Channel.
PC: Specify unused Point Code, not Point Code used in CCIS No.7 [1-16383]
STEP 12: ACIC1 - Specify the accommodated location of Virtual Control Channel on Point
Code as follows:
STEP 13: ACIC2 - Specify the accommodated location of Virtual Control Channel on Point
Code as follows:
MB = 0 (Make idle)
FUNCTION = Setting
KIND = SIP-MG
LINE = 0 (fixed)
CH = 0 (fixed)
RETRY = 0
Service Type: Specify QoS settings of Signalling that is used when Internal PHI-BRI
connects MGSIP (128/96/16).
Usable CH Num: Enter the number of channels to be used per MG-SIP (1-128).
Service of Authentication exists: Select the check box if registration is required per
number.
MG Group: Enter MG-SIP Group Number 1-15 to make an alternate route for redundancy
configuration.
STEP 16: AMGVL - Register the voice control data when connecting to MG-SIP128/96/16.
If the settings are not specified by this command, the values assigned by the AIVCL
command will be set automatically.
B RT: Displayed when TYPE = Route Specified is selected. Specify Physical Route Number
(MGSIP128/96/16) in the case of tandem connection between MGs.
Setting Type of Voice Packet Service: Specify QoS settings of Voice Speech
TYPE: Displayed when the check box of “Setting of Payload” is ON. Specify Payload type.
*G.711 and 20ms must be set for public SIP network.
SIZE: Displayed when the check box of “Setting of Payload” is ON. Specify Payload size.
Index31=04
Index78, Bit3: 0/1(Station Number with Name Display on Desktop terminal is out
of service/in service.)
Index79, Bit5: 0/1 (Name Display Enhance Service is out of service/in service.)
1 = Dialed number
INDEX241, Bit1: The number of digits displayed in Name Display service. 0/1 (8-
digit/16-digit)
I NDEX390, Bit0: Caller ID display pattern on the LCD (Top line). 0/1 (Calling
Name/Calling Number has priority.)
STEP 2: ARTD - Assign route class data for the MG-SIP route.
Note: To change the data for existing route (RT), initialize or reboot the trunk device such
as MG-SIP.
Please follow the example as shown below for LCR Programing. In the example below
when SV9500 Station 4xxx dials CUCM Station 7xxx
ASAPL
AFRSL
AADCL
AMNDL
ANPDL
Please take note on the PRI Protocol Type and the PCM type (this is country specific normally)
The Significant Digits field signifies that CUCM will use the last 4 digits from the right to route the incoming
call. The Calling Search Space specific the right to call extension.
Please take note that different systems accept different transport, make sure proper outgoing transport
type is selected, the option available includes TCP, UDP and TLS.
Partitions are a logical grouping of directory numbers and route patterns with similar reachability. The
Communication Managers will be configured with the partitions shown below.
Calling Search Spaces are an ordered list of partitions that are searched when attempting to complete a call
Route patterns are used to direct calls to specific gateway devices or to block calls.
A route list contains a prioritised list of route groups to use for outbound calls. Route patterns are used to
map calls to route lists.
Route groups contain a prioritised list of gateways and trunks used for outbound calls. The route groups to
be configured are shown
controller E1 0/0/1
interface GigabitEthernet0/0
description Management
interface Serial0/0/1:15
no ip address
encapsulation hdlc
no cdp enable
voice-port 0/0/1:15
cptone AU
mgcp
ccm-manager music-on-hold
ccm-manager mgcp
ccm-manager config
Cisco IOS Software, C3900 Software (C3900-UNIVERSALK9-M), Version 15.4(1)T, RELEASE SOFTWARE (fc2)
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
Cisco CISCO3925-CHASSIS (revision 1.0) with C3900-SPE100/K9 with 987136K/61440K bytes of memory.
61 Serial interfaces
2 terminal lines
License Info:
License UDI:
-------------------------------------------------
Device# PID SN
-------------------------------------------------
*1 C3900-SPE100/K9 FOC16154QA9
------------------------------------------------------------------------
------------------------------------------------------------------------
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