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
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