We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway
provider supports both, so that's not a problem, and if I force him (the PSTN gateway)
to allow G729 only, the outgoing call will take place with G729.
The problem is that I want to have my PSTN provider configured to allow ULAW as a
first priority, then G729. I did it like that:
[mypstngate]
type=friend
host=192.168.0.100
port=5060
context=pstn-in
canreinvite=no
disallow=all
allow=ulaw
allow=g729
Then, in the outgoing context for our G729 SIP customers, I've put something like that:
exten => _0NXXXXXXXX,1,setvar(SIP_CODEC=g729)
exten => _0NXXXXXXXX,2,Dial(SIP/0041${EXTEN:[EMAIL PROTECTED],90)
What happens now when placing a call is very interesting. As you can see, Asterisk
wants to change the codec to g729, but on the outgoing call to the PSTN gateway it
remains ULAW. Like this, I'm using up one of my G729 licenses, and Asterisk is doing
the transcoding between G729 and ULAW. That's definitely not what I want. Any ideas
about how to force both channels to G729? By the way, if I use a client which doesn't
support G729, this call doesn't even take place, it hangs up, because Asterisk tries
to force G729 on the client's channel (but not on the PSTN gateway's channel).
In other words, the "setvar(SIP_CODEC=g729)" only forces the codec on the "calling
channel", not on the "called channel". How can I change that?
Another interesting thing, the "show g729" after the call hangs up: I have "-1/-2
encoders/decoders in use". Maybe a bug?
Thanks
-Manuel
*CLI> -- Executing SetVar("SIP/2016-b119", "SIP_CODEC=g729") in new stack
-- Executing Dial("SIP/2016-b119", "SIP/[EMAIL PROTECTED]|90") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/mypstngate-caed is making progress passing it to SIP/2016-b119
-- SIP/mypstngate-caed is ringing
-- SIP/mypstngate-caed answered SIP/2016-b119
Jun 24 09:49:23 NOTICE[1094450096]: chan_sip.c:1314 sip_answer: Changing codec to
'g729' for this call because of ${SIP_CODEC) variable
-- Attempting native bridge of SIP/2016-b119 and SIP/mypstngate-caed
*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format
192.168.0.100 0041911234 1f7d34e3642 00102/00000 00000ms 0000ms ULAW
192.168.0.2 2016 4977-4F41-7 00101/00003 00000ms 0000ms G729A
2 active SIP channel(s)
[... after hangup ...]
== Spawn extension (auth-out, 0911234567, 2) exited non-zero on 'SIP/2016-b119'
-- Executing Hangup("SIP/2016-b119", "") in new stack
== Spawn extension (auth-out, h, 1) exited non-zero on 'SIP/2016-b119'
> cdr_odbc: Query Successful!
*CLI> show g729
-1/-2 encoders/decoders of 30 licensed channels are currently in use
*CLI>
___________________________________________________
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
