[Asterisk-Users] Codec negotiation

[Florian Overkamp]

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