[asterisk-users] Asterisk and Linphone

[Jerry Geis]

I think this is what your looking for:

Found RTP audio format 119
Found audio description format speex for ID 119
Capabilities: us - (speex|speex16|speex32|g722|ulaw|alaw|gsm), peer -
audio=(speex32)/video=(nothing)/text=(nothing), combined - (speex32)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.176:7078
[Jul  5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find
a codec translation path: (speex32) -> (speex)
^M^[[Kdevgeis*CLI> ^M^[[0K[Jul  5 09:55:34] WARNING[19832]: channel.c:5751
set_format: Unable to find a codec translation path: (speex) -> (speex32)

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...

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.

What do I do about that ? - thanks

Jerry

On Fri, Jul 5, 2019 at 8:22 AM Jerry Geis <jerry.geis at gmail.com> wrote:

> Hi all - I am using asterisk 13.27.0 with Linphone.
> I turned off all codes on linphone except the one I want to try. For
> example:
> opus and speex (so only one enabled at a time).
> Then did this same on asterisk for the linphone extension.
> disallow=all
> allow=speex
>
> (for example).
>
> Then I place my call and the call fails.   if I enable something like gsm,
> ulaw, alaw the call works fine. Why does the call fail with opus and speex ?
> Thanks,
>
> Jerry
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190705/b606cf14/attachment.html>