[asterisk-users] Asterisk and Linphone

Antony Stone Antony.Stone at asterisk.open.source.it
Fri Jul 5 09:28:06 CDT 2019

On Friday 05 July 2019 at 16:03:42, Jerry Geis wrote:

> I think this is what your looking for:

> [Jul  5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find a
> codec translation path: (speex) -> (speex32)

Indeed, it was.

> My linphone side only has speex at 32K enabled.
> My extension definition has:
> disallow=all
> allow=speex
> allow=speex16
> allow=speex32
> allow=g722
> allow=ulaw
> allow=alaw
> allow=gsm
> It looks like its the codec translation ?   So then I enabled speex and
> speex32 on Linphone.... Got past that - I presume it will use speex32 for
> audio...

You can always see which codec is in use by doing a SIPpacket capture and 
looking at the above negotiation exchange to see what got agreed on.

> But then I am trying to place that call in a conference (confbridge) and I
> get this error:
> Unable to find a codec translation path: (slin) -> (speex)
> so I think then it hangs up.

Try "core show translation" on your Asterisk command line and check that the 
table has an entries in both directions for speex (left) to slin (top) and 
slin (left) to speex (top).

The numbers tell you how many microseconds *your* server takes to transcode 1 
second of audio between the two codecs.

You can also try "core show translation paths speex" to get a list of the 
codecs which can and cannot be converted to, with a guide to the method used 
for trancoding that combination where possible.


All matter in the Universe can be placed into one of two categories:

1. Things which need to be fixed.
2. Things which need to be fixed once you've had a few minutes to play with 

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