[Asterisk-Users] Codec negotiation

[Ronald Voermans]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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
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