Hot CISCO Labs: Installing The Free G729 Codec For AsteriskNow

Installing The Free G729 Codec For AsteriskNow

Posted: Saturday, June 1, 2013 by Unknown in Labels: AsteriskNOW / FreePBX., AsteriskNOW.

G729 is a low bandwidth codec, which can operates at a lower bandwidth of 8Kb/s per call instead of 64kb/s when using G711a-law /G711u-law. This describes how to install the G729 free codec on AsteriskNow without the need to license it from Digium Addons.

Installing The Free G729 Codec For AsteriskNow:

1-Infomation you need to gather to download the proper codec .SO binary file.
2-Download the G729 codec source or binary.
3-Rename the binary file.
4-Give it the necessary permissions.
5-Load the module into Asterisk.
6-Verify the module has loaded correctly.
7-Configure your phones to use G729 instead of choosing the G711a-law /G711u-law.
8-Verifiy the codec usage by your phones on your asterisk server.

1-Infomation you need to gather to download the proper codec .SO binary file.

-Processor Architecture: Check the kernel type installed on your system.It's 64 Bit OS.

-Asterisk Version: As you can see here the Asterisk version is V11.2.1.

2-Download the G729 codec source or binary.
-To download the codec binary click here.and choose the codec source / binary that is compatible with your Asterisk version and architecture.
-Go to the modules directory in " /usr/lib64/asterisk/modules " .

-Download the codec file using the " wget " command in the " /usr/lib64/asterisk/modules " directory.

3-Rename the binary file.
-Now copy or move the binary to rename it using the "cp codec_g729-ast110-gcc4-glibc-x86_64-core2.so codec_g729.so "

4-Give it the necessary permissions.
-Use the " chmod +x codec_g729.so " to set the proper permissions.

5-Load the module into Asterisk.

-Log into Asterisk’s CLI using " asterisk -r ".

-Load the module using " module load codec_g729.so " command.

6-Verify the module has loaded correctly.
-Verify the codec has been loaded correctly using the "core show translation" command.
Great! you can see the codec at the end of the list.

7-Configure your phones to use G729 instead of choosing the G711a-law /G711u-law.
-Now,time to configure your phone to use G729.Connect to the FreePBX admin page,choose the extension that you want to configure it to use G729. disallow all codecs and allow only g729 to be used.

8-Verifiy the codec usage by your phones on your Asterisk server.

-Make a call to any phone and use the " sip show channels " and congratulations!

here you can see the 2002 user is using G729 when placing or receiving calls.

Note : NOT all the softphone will support the G729 Codec.To download a softphone that can support G729 you can use SIPLite.Click here and after installing the softphone configure it to use only G729 from the codec settings as follows.

Comments (12)

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0 Vote up Vote down

mahesh · 511 weeks ago

Hi Ali,

I have followed your tutorial from starting (installation AsteriskNow) and grate instruction for the beginners.
I installed Asterisk v. 11.16.0 and trying to integrate trunk with cucm 10.5
g729 codec has been installed. Im able to call from asterisk to cucm fine but cucm to asterisk calls not working.
route pattern configured o both system.

Appreciate your help to resolve this issue.

thanks.

1 reply · active 511 weeks ago

+1 Vote up Vote down

hotciscolabs 18p · 511 weeks ago

Hi Mahesh,
Can u give me more INFO about the SIP trunk configuration on CUCM,I guess the CUCM configuration side is NOT complete.For successful call routing you should associate the route pattern with the SIP Trunk directly or using a hunt list to forward the calls to the Asterisk IP address.Also check my guide for more INFO http://hotciscolabs.blogspot.com/2014/04/dawnload...

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mahesh · 511 weeks ago

Hi Ali,

Thanks for response.
I have created sip trunk with asterisk IP on cucm (use same setting (http://hotciscolabs.blogspot.com/2014/04/dawnload-cisco-asterisk-integration.html)

created route pattern 888X and associate with asterisk sip trunk.

Asterisk extension is 8880

do i need to add incoming route pattern on asterisk(i set type=friend)

host=172.28.28.10
type=friend
canreinvite=no
insecure=port, invite
qualify=yes
dtmfmode=rfc2833
nat=no
disallow=all
allow=g729

thanks.

1 reply · active 511 weeks ago

+2 Vote up Vote down

hotciscolabs 18p · 511 weeks ago

You need to be sure that the call can get out of CUCM to Asterisk,Use the " Dialed Number Analyzer" in CUCM and trace that call from CUCM to extension 8880 and if the call gets routed using the configured SIP trunk try to dial the extension and debug the events on Asterisk to get some logs...this should help u !

0 Vote up Vote down

mahesh · 510 weeks ago

Hi Ali,

It was a simple mistake. I have added cucm pub IP on asterisk but calls coming from CUCM sub IP.
Once i changed it i was able to dial both ways.

Thanks for the help and looking to go forward with asterisk

1 reply · active 510 weeks ago

+1 Vote up Vote down

hotciscolabs 18p · 510 weeks ago

Good work Mahesh ! Kindly stay in touch with me if you need any kind of help :)

0 Vote up Vote down

mahesh · 510 weeks ago

Hi Ali,
How can i configure class of restriction for seem extensions?
I need to allow IDD dial only for specific extension can we implement this on asterisknow?

Thanks.

0 Vote up Vote down

mahesh · 510 weeks ago

for some extensions? *

2 replies · active 510 weeks ago

+2 Vote up Vote down

hotciscolabs 18p · 510 weeks ago

Yes you can implement COR (Class Of Restrictions ) in Asterisk by using different "Contexts" to allow dialing some extensions from only specific extensions.

0 Vote up Vote down

mahesh · 510 weeks ago

Hi Ali,

I used From-internal,outbound-all-routes and from-trunk option. and it worked as default COR.
Is there any place to add new customized context. i need specific extension to dial

1. National and local mobile calls but not IDD
2. National land line only
3. All destination

i used same trunk for all external calls.

Thanks.