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

[Tom Moore]

Asterisk still controls the signalling, but the audio path should be going
through the phones directly.
Fire up a tcpdump on the Asterisk server to varify this.
 
 

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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jonas kellens
Sent: Saturday, April 18, 2009 5:33 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Canreinvite=yes // native bridging // 2 sip
channels with different Call-ID


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. 
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