Hi, Ronald Voermans wrote:
I've set up an Asterisk box with a SIP trunk to our PSTN provider. I've configured two incoming phonenumbers. One phonenumber is for voice-calls, the other one for receiving faxes. I want the incoming voice-calls to be coded by the G.729 codec, and the fax-number by G.711. Can I make a codec-negotation based on the called number?
Nope, but maybe you could separate the traffic in to different SIP peers.
If you need more info on this, i can send it to you.
If you want we could figure something out. Just curious: Which PSTN provider are you using ?
Florian _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
