I will try your suggestion and I will let you know. Thank you <br><br><div><span class="gmail_quote">On 6/18/06, <b class="gmail_sendername">Philippe Lindheimer</b> <<a href="mailto:p_lindheimer@yahoo.com">p_lindheimer@yahoo.com
</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div>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.
<br><br>p<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"><br>From: "Il Neofita" <<a href="mailto:asteriskmail@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
asteriskmail@gmail.com</a>><br>To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br> <<a href="mailto:asterisk-users@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
asterisk-users@lists.digium.com</a>><br>Date: Sun, 18 Jun 2006 05:01:20 -0400<br>Subject: Re: [Asterisk-Users] Canreinvite</blockquote></div><div><span class="q"><br><br> This is the dial in extensions<br>exten => _40001,1,Dial(SIP/40001,30)
<br>exten =>
_40002,1,Dial(SIP/40002,30) <br><br></span></div><div> From: "Il Neofita" <<a href="mailto:asteriskmail@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asteriskmail@gmail.com
</a>><br>To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br> <<a href="mailto:asterisk-users@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users@lists.digium.com
</a>><br>Date: Sun, 18 Jun 2006 05:22:35 -0400<br>Subject: Re: [Asterisk-Users] Canreinvite</div><div><span class="e" id="q_10be804c51ac3401_3"><br><br> cosa vedo a console<br><br> -- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack
<br> -- Called 40002<br> -- SIP/40002-4753 is ringing<br> -- SIP/40002-4753 answered SIP/40001-3760 <br> -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753<br>srvlinux*CLI> sip show channels<br>
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message<br>82.X2.XX3.X3
40002 146b518a4cd 00103/00000 alaw No Tx: ACK <br>82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK<br>2 active SIP channels<br><br></span></div><div></div><div><span class="ad">
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