[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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