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I have 2 SIP-clients defined in my sip.conf :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>[GXP1200]</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>type=friend</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>context=intern</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>host=dynamic</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>username=GXP1200</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>secret=testpaswoord</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>canreinvite=yes</I></FONT></FONT><BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>[BT201]</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>type=friend</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>context=intern</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>host=dynamic</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>username=BT201</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>secret=testpaswoord</I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>canreinvite=yes</I></FONT></FONT><BR>
<BR>
When I make a call from one to another this is displayed on the CLI :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>-- Executing [210@intern:1] Dial("SIP/GXP1200-093900c8", "SIP/BT201|30") in new stack </I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>-- Called BT201 </I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>-- SIP/BT201-09395070 is ringing </I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>-- SIP/BT201-09395070 answered SIP/GXP1200-093900c8 </I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>-- Native bridging SIP/GXP1200-093900c8 and SIP/BT201-09395070 </I></FONT></FONT><BR>
<BR>
>From voip-info.org I understand that 'canreinvite' means that the SIP-client will re-invite the other client, so that Asterisk is no longer in the path...<BR>
This is indicated on the CLI with 'native bridging'.<BR>
<BR>
Then why are there 2 sip-channels with a different Call-ID ? The output shows that Asterisk is still in between !<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>asterisk*CLI> sip show channels </I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message </I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>192.168.x.x GXP2020 4684b544470 00103/00000 0x4 (ulaw) No Tx: ACK </I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>192.168.x.x BT201 1212e00ffa1 00102/43234 0x4 (ulaw) No Tx: ACK </I></FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#000000"><I>2 active SIP channels </I></FONT></FONT><BR>
<BR>
Is there something that I misunderstand here ??<BR>
<BR>
Thanks for the feedback on this !<BR>
<BR>
Greetingz,<BR>
Jonas.
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