No compatible codecs, not accepting this offer! Since Module updates

FreePBX
21

This is the workaround we are currently using on FreePBX 13 systems as well, thanks for the suggestion.

22

What version of Asterisk is this?

23

-exp-pbx-01 asterisk]# rasterisk
Asterisk 13.7.1, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 13.7.1 currently running on uh-exp-pbx-01 (pid = 14528)
uh-exp-pbx-01*CLI>
Disconnected from Asterisk server

24

BTW, I appreciate your helping with this.

25

Asterisk 13.7.1 is really old. We are up to 13.21.0 now. WITH security fixes.

26

I can confirm that this issue affects systems running Asterisk 11, you can fix by upgrading to 13:
https://wiki.freepbx.org/display/PPS/Changing+Major+Asterisk+Versions+on+the+Fly

Those running older versions of 13 may see similar issues.

27

https://issues.freepbx.org/browse/FREEPBX-17591

28

framework 13.0.195.4
framework 14.0.3.6

1 Like
29

Thank you!

30

Confirmed that the update fixed the issue.

31

Experiencing the same issue today. Did a bunch of module updates an I get the same error in my logs as the OP about “No compatible codecs, not accepting this offer!”

Asterisk 13.19.1
Framework 14.0.3.6

Any thoughts @tm1000 ?

32

Go to sip settings and select some codecs.

33

I have g722 selected in asterisk sip settings.
I have explicitly specified ulaw:20 in my sip trunk which should override the main asterisk sip settings.

It seems the trunk codec preference is being ignored because internal calls work. Just external calls fail.

34

The update this morning fixed the problem from yesterday. I changed the settings back to what it was under Trunk, SIP settings and it works as before.

35

So there is definitely some weird shananigans going on. Whether it’s a trunk or an extension, only the disallow=xxxxx appears in the sip_additional config file. Any allow parameters are not present.

What have you guys changed? The main asterisk sip settings (which I thought was where default settings could be set) now override anything set at the trunk or extension level?

Is this the expected behavior now? Because it seems weird.

36

No. Nothing here changed. It’s working for others as well.

I had to go ask the Asterisk team what “alaw:20” is as I had never seen it before…

https://issues.freepbx.org/browse/FREEPBX-17600

37

Thank you for the quick fix.

Switching to the edge track and doing a module update fixed the colon suffix on the codec for the packetization time.

1 Like
38

Hi,

For anyone searching the forums, after preforming FreePBX module updates we could no longer make any outgoing calls via our SIP trunk and instead got the following in the console:

No audio format found to offer. Cancelling call to XXXXXXXXX

Our SIP provider investigated and could not see any INVITE being sent however all configuration looked correct.

Since updating our Core module (from 13.0.122.30) to 13.0.122.31 and framework (from 13.0.195.2) to 13.0.195.4 the problem is resolved.

Thanks!
Fraser

39

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