[Asterisk-Users] Canreinvite

[Philippe Lindheimer]

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" <asteriskmail at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
 <asterisk-users at 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" <asteriskmail at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
 <asterisk-users at 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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