[Asterisk-Users] Codec negotiation

[Ronald Voermans]

Florian,

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).

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). 


Met vriendelijke groet,
-----------------------------------------------
R.L.L.M. Voermans
Access & Hosting
E-mail: r.voermans at global-e.nl
Tel.: +31 (0)161 - 88.88.88
Fax: +31 (0)161 - 88.88.99
Global-e
Raadhuisstraat 32
5126 CJ GILZE
http://www.global-e.nl
-----------------------------------------------


-----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 18:27
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] Codec negotiation

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
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