Yes, But without going deeper into OpenSer (since this IS a Asterisk list): With OpenSer I'm using RTPPRoxy. I don't think i can manage rtpproxy to bind to multiple addresses. I'll look for that anyway.
Thanks, Regards, Ronald. -----Oorspronkelijk bericht----- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Florian Overkamp Verzonden: donderdag 9 februari 2006 23:38 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Codec negotiation Hi Ronald, Ronald Voermans wrote: > What exactly do you mean by seperating traffic in to differt SIP peers? > > The situation is as follows: > > I have OpenSer connected to our SIP provider/PSTN Provider (the answer > to your question: Enertel). Ah 'kay. > Asterisk registers to OpenSer, which then forwards the call to PSTN. > Asterisk registers two numbers at OpenSer; one phonenumber and one > faxnumber. I also made two entries in sip.conf. However, the host=... > Is the same for both numbers. So incoming calls are always matched to > one > (1) peer/entry in sip.conf. Hence the problem with negotiating the > right codec (g.729 for voice, g.711 for fax). Hrm, yes for inbound the problem is with the host=.. matching. Maybe Olle has a good suggestion on this :-P. However, if you control the OpenSer yourself you could easily bind another IP, or perhaps use OpenSer rules to do the trick ? Asterisk SIP stack doesn't seem suited for this type of traffic separation I guess... 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 _______________________________________________ --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
