[asterisk-dev] SIP codec selection. Can I select a codec and reinvite?
abalashov at evaristesys.com
Mon Nov 19 15:12:26 CST 2007
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
> My peers are setup like this.
> 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
> Or is there any way to set the codec and do an manual re-invite?
> Doug Gillespie
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