I need a quick tutorial on codecs please!

Hi all, some background first.

In Australia we are all being forced to use the governments high speed internet service called NBN, if you are establishing new services in a new office for example this is your only option. This caused an issue for me as the new NBN service only supported 4 PSTN lines and I needed 8. So I decided to try switching to a sip based solution and roll out asterisk.

I purchased a trunk from our local provide called Engin. A friend of mine gave me 20 older cisco phones from his office 7961G. I spent a month pulling my hair out with it but managed to finally get it working nicely.

I now have one annoying problem, every so often we try to call someone and the line hangs up after one or two rings. After checking logs I found this which I believe is causing the issue.

[2018-11-04 03:30:47] WARNING[3949][C-00000899]: channel.c:5740 set_format: Unable to find a codec translation path: (g729) -> (ulaw)
[2018-11-04 03:30:47] WARNING[3949][C-00000899]: channel.c:5740 set_format: Unable to find a codec translation path: (ulaw) -> (g729)

I have logged a few topics on this in varying forms and people have suggested just switching off support for g729 or buying licencs and installing the codec and even asking why I need it.

The ansers in order:
tried switching off support on my sip settings and disallow the codec on my extension - -Same issue
I downloaded the codec, Cant install it. See my other thread on this one

As to why I need it, this thread is nswering that question unless someone can tell me how I can circumvent the need in the first place.

I really don’t understand how I can take hundreds a calls a day on the same channel and it works fine however some specific numbers require this codec, why would that be. I would have thought that my sip provider would always talk to me with the same codec. Do you think I can call them and ask them not to use g729, or is it outside of their control? I’m thinking maybe these particular numbers are using sip also and prefer to use g729 and that is just being passed through by my trunk provider.

Appreciate any thoughts.

Make sure that the trunk has only alaw and ulaw enabled, in that order. (For best quality, alaw should also be first in the extension’s codec list.)

If this doesn’t help, enable SIP debug. At the Asterisk command prompt, type
pjsip set logger on
if pjsip trunk, or
sip set debug on
for a chan_sip trunk.

Make a failing call and paste the Asterisk log to pastebin.freepbx.org and post the link here.
If possible, use a destination number that you don’t have to redact, e.g. a business where the calls are answered by an IVR so our test calls won’t disturb anyone.

Have you noticed any pattern to which calls fail, e.g. a specific mobile operator, a specific city or country, etc?

Thanks very much Stewart for your help. although I had disabled the codec in the extension and in asterisk sip settings I hadn’t done it on the actual trunk itself. in fact I had it allowed. Removing has enabled me to call numbers that I previously couldn’t.

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