[Asterisk-Dev] SIP media negotiation problem

[anthony thomas]

     Hello all,
 
     We are facing a problem trying to pass faxes between two gateways configured this way (we have tested with Quintum and Cisco gws):
 
     GW1 -------> Asterisk (1.2)-------> GW2
 
     Both gateways and asterisk support g729 and g711, we want to use g729 for voice and fallback to g711 when a fax is detected.
 
     The call goes like this:
 
     - GW1 initiates the call using g729 (at this time, no one knows that the call is a fax)
 
     - Once the call is answered, GW2 detects a fax tone, it sends an INVITE to * to change the media to g711, Asterisk relies the INVITE correctly to GW2 and GW2 switches to g711, the problem seems to be that * replays to the INVITE coming from GW2 with an OK with codec g729 as prefered codec, this fools GW2 and chooses g729 again.
 
     The codec that Asterisk uses in the OK seems to be the codec sent by GW1 when the call was established instead of sending the codec that GW2 sent in the (RE)INVITE.
 
     Can anyone comment if this is a bug or simply we are missunderstanding something?
 
     Thank you,
 
 A

		
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