LG LIP-6812 & LIP-6830

FAQs

LG LIP-6812 & LIP-6830

Frequently Asked Questions
The following FAQs answer most of the common key questions about the LIP-6812 and
LIP-6830 phones.

General
Q. I am unable to access the Web Manager after entering the correct URL
string in my browser (http://xxx.xxx.xxx.xxx:8000).
A. Check the Display Name parameter configuration on the Web Manager.
Access to a device’s Web Manager may fail if the Display Name parameter is
configured with a value that includes a space.

Example: Test Test

Q. Can calls be placed from one LIP Phone to a different hardware product? If
so, how?
A. The LIP Phones can all communicate with other configured LIP Phones, as
well as with PSTN phones. For example, if you are located within the United
States and would like to call another US phone number, simply dial "1" + the
area code + the number.

Q. Will my LIP Phone work with ISDN, DSL or cable?

A. Yes, it will operate with ISDN, DSL or cable. Please note that the amount of
available bandwidth on your network may determine what codec setting must
be used on your LIP Phone.
Q. I am using a DSL or Cable-based connection to the Internet. Do I need
additional accessories to start making calls?
A. If you are using a PPPoE connection to access the Internet (such as cable
modem or DSL), you will need to connect the LIP Phone to a router, switch,
or hub, which then connects to the cable or DSL modem.

Q. When I call the United States, the call quality is very good, but I experience
a delay when calling other countries. Why?
A. This is most likely occurring in the local phone system and is independent of
the product.
Sometimes call delay issues can be caused by limitations of voice traffic
travel to the terminating carrier at the destination’s end. This is not
something that can be controlled.

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Functionality
Q. What bandwidth is required to use the LIP Phone?
A. Your bandwidth requirement will depend on the type of codec your LIP Phone
is set up to use.
For calls set up with the G.711 (PCMU) codec, you will need a minimum of
160Kbps of bandwidth on a single call.
If you are using G.729 codec, you will need a minimum of 53Kbps on a single
call.
If you are using the G.723 codec, you will need a minimum of 38Kbps per
single call.

Q. I cannot access my Voicemail properly when calling my Voicemailbox

using a Speed Dial programmable key. Why?
A. Outbound calls to phone numbers that are answered by an automated menu
(an Interactive Voice Response, or IVR) may become affected if the dialing is
performed through a programmed Speed dial key. It is recommended that
Speed dial keys do not get programmed with these types of phone numbers.

Q. I cannot log into my Voicemailbox after calling it using the MSG button on
the LIP-6830 phone, or any programmed Speed Dial button. Why?
A. Verify that you have programmed the MSG button or programmable Speed
Dial button to properly dial out to your Voicemail. You can program the
button by entering your Voicemail access number in the “VMS Address” field
in the VoIP Configuration page of the Web Manager.

Outbound calls to phone numbers that are answered by an automated menu

(an Interactive Voice Response, or IVR) may become affected if the dialing is
performed through a programmed Speed dial key or the MSG button. It is
recommended that Speed dial keys do not get programmed with these types
of phone numbers.

Q. I have more than one line configured on my LIP-6830 phone. How can I set
the MSG button on my LIP-6830 phone to access Voicemail for my other
lines. Why?
A. The MSG button on the LIP-6830 phone will only dial out through Line 1.
Voicemail cannot be accessed for accounts other than what is configured on
Line 1 when using this button.

[Note: If you want to have an access shortcut to Voicemail for your other
configured lines, you may designate one of the programmable keys on your
phone as a Speed Dial key.]

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Setup/Connectivity
Q. I deleted all of the entries in my Routing Table and now I can’t place any
calls. Why?
A. Some entries within the Routing table contain settings that are necessary to
place outbound calls. If these values have been deleted, you must reload
your default configuration in order to restore those entries.

[Note: Please take note that restoring defaults will remove all other
configuration from your device, including proxy address and account
information. It is strongly recommended that all configuration parameters are
noted prior to loading the default configuration.]

Q. I set DHCP to ON, but my LIP-6812/LIP-6830 cannot obtain correct network

settings. Why?
A. Check the following:

• Make sure your network supports DHCP.

• Make sure your RJ-45 LAN cable is securely connected to both the LIP-
6812/LIP-6830 and to the LAN, hub, or router.
• Also, make sure that all network ports, routers, or hubs are live.

Q. My analog phone does not have a period. How do I enter the periods for
my IP address and netmask?
A. Use the star key (*) in place of the periods.

Q. When I enter a static IP address into the LIP Phone and restart the unit, the
static IP address that I entered is not saved. What should I check?
A. Make sure that the Network Selection radio button on the LAN Configuration
page of the Web Manager is set to “DHCP”.

Q. I changed the login password on my LIP-6812/LIP-6830 and I forgot the new

password.
A. To recover the default password on your device, you must load the default
configuration.
[Note: Please take note that restoring defaults will remove all other
configuration from your device, including proxy address and account
information.]

If you are also locked out from accessing the LIP Phone’s LCD menu, then

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access the Web Manager page and log in as common. Note that the default
password for both is lip.

Q. When I attempt to access the Web Manager, I am prompted for a user name
and password. However, when I enter them and press the OK button, the
login window reappears.
A. Make sure both the user name and password are correct (note that the user
name and password are case sensitive).

Q. If I already have a LIP Phone and want to add more, what are the
installation issues involved?
A. You may add additional LIP Phones to a network just as you would add
additional computers to a network. Make sure there is enough bandwidth
present and the appropriate IP addresses are available.

Troubleshooting
Q. When I attempt to place a call from my LIP Phone to another phone
number, I get a busy signal. What could be wrong?
A. There could be several possible causes for this problem:

• Make sure the Name, Authentication Name, and Authentication Password

are configured correctly on the LIP Phone’s Web Manager.
• Verify that a proper Proxy Address and Proxy Port has been configured
on your LIP Phone’s Web Manager.

Q. Even though I am able to place calls on my LIP Phone, I am unable to

receive incoming calls. What could be wrong?
A. If you use a minute plan with specified inbound and outbound call
allowances, check that your account is sufficiently funded to allow incoming
calls to your LIP Phone.

Q. When I call the United States, the call quality is very good, but I experience
a delay when calling other countries. Why?
A. This is most likely occurring in the local phone system and is independent of
the product.

Sometimes call delay issues can be caused by limitations of voice traffic

travel to the terminating carrier at the destination’s end. This is not
something that can be controlled.

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Q. When I make a call from one LIP Phone to another number, the other party
cannot hear my voice. What could be the problem?
A. There could be various reasons for this problem:
First, make sure that both units have been upgraded to the latest
firmware.
One of the phones could be behind a firewall that is blocking the RTP
Port used on the call. Check the RTP port settings on the phone behind
firewall and make sure that port is not blocked. The RTP port used by the
phone can also be changed if required.

Q. My audio cuts in and out or is distorted. What can I do to help fix this
problem?
A. The codec settings on your LIP Phone may need to be changed in order to
achieve better audio quality. If you have low bandwidth on your network, set
your Codec Priority 1 (primary) setting to a more compressed codec setting
(either G.723 or G.729) within the VoIP Config page of the MAX Web
Manager. The Codec Priority 2 (secondary) setting can then be set to the
other codec which was not selected in the first option. Codec Priority 3 can
then be configured with G.711 (either PCMU or PCMA) or none.
If this does not help, you may need to readjust the Jitter Buffer Bounds on the
QoS Configuration page of the LIP Web Manager. Please note that the
following default values apply:

Minimum Delay: 30 ms
Normal Delay: 60 ms
Maximum Delay: 120 ms

Please note that while setting a higher overall jitter buffer range may
decrease the risk of audio packet loss, it may increase the chance for audio
delays.

If you are still encountering problems with audio, also note that call quality
may be affected by the available bandwidth of the network in which your
device is connected.

Q. Why is Call Transferring not working when performing the steps in the User
Guide?
A. Call Transfer is not supported on the MaxLine service.

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Q. My LIP Phone has a tendency to freeze during regular use. Why?

A. Check your Network Time Confuguration settings. The LIP Phone may lockup
momentarily if it is configured with certain SNTP Server Address values. For
best results, it is recommended that you configure your LIP Phone with either
of the following time server values:

ntp.nasa.gov
clock.via.net
tick.ucla.edu

Q. Why do I not see the change to my Time Zone setting get applied
immediately?
A. Verify that your LIP Phone is configured with a proper SNTP Server Address
prior to making changes to the Time Zone setting.

For best results, it is recommended that you configure your SNTP Server
Address with any of the time server values listed above in the previous FAQ.

Q. The “Input IP Address” and “Input URL” call modes do not work when they
are used.
A. These call modes are not supported. The only call mode that is supported
from the [Cmod] (Call Mode) menu is the default calling option, “Input dial
number”.

Q. When I try to dial out automatically from an entry listed in my Call Log, I get
a busy signal.
A. Verify that the account you are using for your call has sufficient funds.

Q. When do I know that my device has successfully registered with the

configured SIP proxy server and is ready to make phone calls?
A. Enter a valid Name, Authentication Name, Authentication Password, and
Proxy Address through the VoIP Config page of the LIP Phone’s Web
Manager, and then save these settings and reboot your device.

After rebooting, the device should attempt to register onto the configured SIP
server. If the Registration Status field in the Web Manager reads OK, then
the device has successfully registered with the SIP proxy server and is ready
to make calls.

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