On 6/18/06, Philippe Lindheimer <[EMAIL PROTECTED]
> wrote:
How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages going on.
p
From: "Il Neofita" < [EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
< asterisk-users@lists.digium.com>
Date: Sun, 18 Jun 2006 05:01:20 -0400
Subject: Re: [Asterisk-Users] Canreinvite
This is the dial in extensions
exten => _40001,1,Dial(SIP/40001,30)
exten => _40002,1,Dial(SIP/40002,30)From: "Il Neofita" <[EMAIL PROTECTED] >
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com >
Date: Sun, 18 Jun 2006 05:22:35 -0400
Subject: Re: [Asterisk-Users] Canreinvite
cosa vedo a console
-- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack
-- Called 40002
-- SIP/40002-4753 is ringing
-- SIP/40002-4753 answered SIP/40001-3760
-- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753
srvlinux*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
82.X2.XX3.X3 40002 146b518a4cd 00103/00000 alaw No Tx: ACK
82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK
2 active SIP channels
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