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<DIV dir=ltr align=left><SPAN class=984384810-18042009><FONT face=Arial
color=#0000ff size=2>Asterisk still controls the signalling, but the audio path
should be going through the phones directly.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=984384810-18042009><FONT face=Arial
color=#0000ff size=2>Fire up a tcpdump on the Asterisk server to varify
this.</FONT></SPAN></DIV>
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color=#0000ff size=2></FONT></SPAN> </DIV>
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color=#0000ff size=2></FONT></SPAN> </DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>jonas
kellens<BR><B>Sent:</B> Saturday, April 18, 2009 5:33 AM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [asterisk-users]
Canreinvite=yes // native bridging // 2 sip channels with different
Call-ID<BR></FONT><BR></DIV>
<DIV></DIV>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. </BODY></HTML>