Thomas Winter wrote:
> On Sunday 09 November 2008 20:14, Eric "ManxPower" Wieling wrote:
>> The best (and maybe only way) is to set your client and your service
>> provider to only do G.723.
>
> Really, thats not the way it should work.
>
> How I can find out the codec of an incomming call?
>
> Is there any way to use ${SIP_CODEC} to try to change to G.723 and then check
> success?
> If OK use provider with only allowed G.723 and if not use provider with
> allowed alaw and ulaw?
I didn't say that is how it should work. I said that is how it does
work. No, you cannot change the codec of the incoming call in the
dialplan. SIP_CODEC only sets the codec for the outgoing leg of the call.
Remember, Asterisk cannot transcode to/from G.723
--
Consulting and design services for LAN, WAN, voice and data. Based near
Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs
echo canceling systems. Also see http://www.fnords.org/skillslist.html
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users