Did you try having two sip.conf entries for your gateway? Forcing one
with G729 and the other with ulaw? You would obviously need to change
your dialplan accordingly and have each phone configured so that it
would take the proper extension. I have not tried this, it is just
really an idea...
Manuel Wenger wrote:
"If" I understood your initial objective correctly (and I may not have),
the user's phones are negotiating the codec to be used for each rtp session.
Asterisk parameters can be used to dictate rtp sessions between the sip
phone and asterisk, but that won't influence the next step in which the sip
phone negotiates a new rtp session directly with the gateway.
The gateway and the phone will negotiate a common codec based on
whatever logic those two devices have been programmed with by their
respective manufacturers; asterisk isn't involved.
So, it sounds like the issue is understanding the codec selection logic
that has been programmed into the gateway and the phone.
I think you're getting my point, at least I think so (I'm getting more and more
confused myself about this...)
The problem is that the phone negotiates a codec with asterisk when placing the call (remember I have all reinvite's set to "no", so the gateway and the phone won't talk directly to each other!). This negotiation actually works correctly, because I force the phone's codec using "disallow=all; allow=g729" in the SIP phone's peer configuration.
The negotiation which doesn't work the way I want is the asterisk-to-gateway part. The gateway
can talk either ULAW or G729, whatever I tell it, if I "force" it using the
disallow/allow method in sip.conf. The problem is that I need asterisk to talk to the gateway
sometimes with ULAW, sometimes with G729, depending on the SIP phone who placed the call in the
first place.
What I need is some sort of command which says "OK, now Dial(... @gateway), but force
G729" (which works *if* I tell asterisk that the gateway supports G729 *only* in sip.conf,
but we want it to support both codecs, right?). Apparently I can only force the codec on
incoming channels, not on outgoing channels. Is this really an asterisk limitation?
-Manuel
___________________________________________________
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
taridium.communications
dominique kull, partner
the old lodge, london sw6 6ee uk
t: +44 207 731 1562
f: +44 207 900 6564
v: fwd 268167
w: http://taridium.com
e: [EMAIL PROTECTED]
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
