[Asterisk-Users] Codecs problem

[William Lloyd]

I've found that happens when one version of asterisk is 1.2 and the  
other end is running 1.0.9 and you are connecting over IAX2.

If you bridge the two servers with SIP it will be fine.

-bill

On 9-Nov-05, at 11:52 AM, Olivier Taylor wrote:

> That's a call to pstn
>
> Callee and caller have 9729 but asterisk (astlinux and soekris)  
> tell me that
> there is no match and give me an error :(
>
> Any idea?
>
> Kind regards,
>
> Olivier
>
>
> 9 headers, 11 lines
> Found RTP audio format 18
> Found RTP audio format 101
> Peer audio RTP is at port 82.146.123.246:38098
> Found description format G729
> Found description format telephone-event
> Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer -
> audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
> Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723),  
> combined - 0x1
> (g723)
> Nov  9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format:  
> Unable to
> find a path from g729 to gsm
> Nov  9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format:  
> Unable to
> find a path from ilbc to g729
>
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