Strange problem with transcoding

Asterisk Asterisk Support
1

Colleagues, I have a strange situation - the codec is present in Asterisk, but transcoding does not turn on. What could be the reason?

There is a certain remote SIP client that should work only with the g729 codec. The others use the alaw and ulaw codecs.
When inviting the specified client, the audio stream cannot be negotiated.

messages:[May  9 00:58:59] WARNING[100800][C-000000c0] channel.c: Unable to find a codec translation path: (g729) -> (alaw)
messages:[May  9 00:58:59] WARNING[100800][C-000000c0] channel.c: Unable to find a codec translation path: (alaw) -> (g729)

At the same time, if I understand correctly, the codec is present in the system.

pbx*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
	It does not indicate anything about your configuration.
      ID TYPE  NAME         FORMAT           DESCRIPTION
------------------------------------------------------------------------------------------------
...
      18 audio g729         g729             (G.729A)
...
pbx*CLI>

What am I doing wrong and how can I fix the situation?

I would be grateful for advice,
Ogogon.

2

So the console output doesn’t tell you if the codec is present.

The normal codec used for G.729 is commercial, and requires licences to be bought. There is now an unoptimised open source one, as the patent has expired.

There really is no valid reason to use G.729, nowadays. Most people prefer higher audio quality to its bandwidth saving, and there are better low bit rate codecs which have always been opoen source.

3

This codec is used because this SIP client is connected via a disgustingly bad channel and there is no hope for improvement yet. Solid fluttering of artifacts.