[asterisk-dev] SIP codec selection. Can I select a codec and reinvite?

[Alex Balashov]

Try "disallow" instead of "deny."

On Mon, 19 Nov 2007, asterisk wrote:

> I am having an issue with codec selections. I am using only 2 codecs
> ULAW & G729.  On some peers I want the proffered codec to be G729 other
> ULAW.    I have a SIP trunk form a carrier that supports both G729 and
> ULAW. I would like asterisk to use the [proffered codec and not
> transcode.
>
> My peers are setup like this.
>
> [siptrunk]
> Deny=all
> Allow=ulaw
> Allow=g729
>
> [UPeer]
> Deny=all
> Allow=ulaw
>
> [Gpeer]
> Deny=all
> Allow=G729
>
>
> On outbound calls every thing works fine. The phone picks the codec and
> asterisk passes it on to the sip trunk.   On inbound calls  (from the
> SIP trunk) the Sip trunks proffered codec is selected.
>
> Is there any way (in asterisk) that I can select the codec per call?
>
> i.e.  Call comes in. before answering the call look up the destination
> peers codec, (function SIPPEER?) and set the codec to use and then
> answer?
>
> Or is there any way to set the codec and do an manual re-invite?
>
> Thanks
> Doug Gillespie
>
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--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : +1-678-954-0670
Direct : +1-678-954-0671