Unable to find a codec translation path: (slin) -> (g729)

Hi, I just started to learn about Asterisk and I am having problems with my first try. I was success installing Asterisk 18.4.0 on an AWC EC2 with Ubuntu 20.04.2 LTS (GNU/Linux 5.4.0-1047-aws x86_64). I also was able to register to my SIP Trunk and place incoming calls from the PSTN to some softphones, The call stablishes and the audio goes through but for each call on the CLI console I get a warning saying “channel.c:5674 set_format: Unable to find a codec translation path: (slin) → (g729)” I don’t know if that is some thing I should care about.

On the general section of the sip.conf file I have the following lines:


I use them because the SIP Trunk provider told me that I have to use G729

The softphones are:

Linphone Android 4.4.3
Zoioer5 5.4.12 for windows 64bits

Please any help will be appreciated

G729 is not an included codec for translation.

Without a G.729 codec, you are not going to be able to play tones, use MixMonitor, have conferences, listen in on calls, etc. All you can do is pass media, unchanged and uninspected, from source to destination.

I believe that G.729 was subject to patents, which have now lapsed, but the preferred Asterisk G.729 codec module is still subject to licensing fees, so not included in the open source code.


Does the .lv one have a valid licence now? The last time I looked at the open source one, it purported to be licensed under the GPL, but contained third party code with a no commercial use restriction. The second point voids the GPL.

Thank very much for your help. I found this page (SUPPORT - ALG: HOW TO: Install the Free G729 Codec for Asterisk (ViciDial and GoAutodial) ) talking about a free g729 codec. I think it is the .lv codec david551 is talking about but I could not find the license information. Is this codec really free and will perform find or does it have some limitations?

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