[asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID

[jonas kellens]

I have 2 SIP-clients defined in my sip.conf :

[GXP1200]
type=friend
context=intern
host=dynamic
username=GXP1200
secret=testpaswoord
canreinvite=yes

[BT201]
type=friend
context=intern
host=dynamic
username=BT201
secret=testpaswoord
canreinvite=yes

When I make a call from one to another this is displayed on the CLI :

-- Executing [210 at intern:1] Dial("SIP/GXP1200-093900c8", "SIP/BT201|30")
in new stack 
-- Called BT201 
-- SIP/BT201-09395070 is ringing 
-- SIP/BT201-09395070 answered SIP/GXP1200-093900c8 
-- Native bridging SIP/GXP1200-093900c8 and SIP/BT201-09395070 

>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...
This is indicated on the CLI with 'native bridging'.

Then why are there 2 sip-channels with a different Call-ID ? The output
shows that Asterisk is still in between !

asterisk*CLI> sip show channels 
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 
192.168.x.x GXP2020 4684b544470 00103/00000 0x4 (ulaw) No Tx: ACK 
192.168.x.x BT201 1212e00ffa1 00102/43234 0x4 (ulaw) No Tx: ACK 
2 active SIP channels 

Is there something that I misunderstand here ??

Thanks for the feedback on this !

Greetingz,
Jonas.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090418/304fea4f/attachment.htm