[asterisk-users] Codec problems when using G.723
Eric "ManxPower" Wieling
eric at fnords.org
Mon Nov 10 09:52:15 CST 2008
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
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