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

[asterisk]

That's the ticket!   Thanks!

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Eric
"ManxPower" Wieling
Sent: Monday, November 19, 2007 3:50 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] SIP codec selection. Can I select a codec
and reinvite?

There is this super secret document in the Asterisk source code called 
"channelvariables.txt" (or README.variables if using 1.2.x) that 
contains information on an even more super secret channel variable 
called SIP_CODEC.  You might want to look into it.

Gregory Boehnlein wrote:
>> 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
> 
> BJ Wescke wrote a patch for me last year for 1.2 that allowed us to do
codec
> based routing via a new DialPlan Function called "SIPPrefCodec". I
don't
> know if it will work for you, but I've attached a link to the patch.
> 
> http://www.nacs.net/~damin/sip_codec-1.2.diff
> 
> Basically, that would let you define multiple peers ala:
> 
> [itspG723]
> disallow=all
> allow=g723
> 
> [itspG729]
> disallow=all
> allow=g729
> 
> Using the function, you could then build your Dial string to send it
to the
> specific peer that matched the codec of the inbound call leg.
> 
> Hackish, and not clean, but it worked for us at the time.
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
> 


_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev