[Asterisk-Users] SIP Channel hangup problem with re-INVITE enabled- ugrent

[Hoa Thai Duy]

Does anyone on this list has idea?

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Hoa Thai Duy
Sent: Thursday, June 22, 2006 2:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: asterisk-dev at lists.digium.com
Subject: [Asterisk-Users] SIP Channel hangup problem with re-INVITE enabled-
ugrent



Hi List 

I have UAs  registered with Asterisk and make outbound calls via ITSP1,
everything is fine without re-INVITE. When people call 178, the actual
number 112233445566 at ITSP1 network will be called.

When UA or called telephone (112233445566) hang up, the call and associated
channels are cleared. 

Sip.conf 

[general] 
canreinvite=no 
nat=no                  

[ITSP1] 
type=peer 
host=A.B.C.D 

Extensions.conf 

exten => 178,1,Answer() 
exten => 178,n,Dial(SIP/112233445566 at ITSP1,60)        
exten => 178,n,Hangup() 


However, when I enabled re-INVITE like below, the call still happen, people
can talk with each other. If remote called telephone (112233445566) hang up,
then the call is cleared. But if the Asterisk user (US) Softphone hang up
first, the remote telephone still in talking mode (with no sound, of
course).

Sip.conf 
[ITSP1] 
type=peer 
host=A.B.C.D 
Canreinvite=yes 
Nat=yes 


In this case, when Asterisk user hang up and remote phone still not hang up,
I do show like this 

Show channel verbose 
0 active channels 
0 active calls 


Sip show channels 
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last
Message   
A.B.C.D    112233445566  14448d41170  00103/00104  unkn  No  (d)  Rx: BYE  

CLI> sip show channel 14448d41170ac3a66a41602575476d5f at W.X.Y.Z 
  * SIP Call 
  Direction:              Outgoing 
  Call-ID:                14448d41170ac3a66a41602575476d5f at W.X.Y.Z 
  Our Codec Capability:   256 
  Non-Codec Capability:   1 
  Their Codec Capability:   256 
  Joint Codec Capability:   256 
  Format                  unknown 
  Theoretical Address:    A.B.C.D:5060 
  Received Address:       A.B.C.D:5060 
  NAT Support:            Always 
  Audio IP:               W.X.Y.Z(local) 
  Our Tag:                as5436f254 
  Their Tag:              caba969d04802f1091a1000000000000--558 
  SIP User agent:         Asterisk 
  Username:               112233445566 
  Peername:               112233445566 
  Original uri:           sip:112233445566 at A.B.C.D:5060 
  Need Destroy:           2 
  Last Message:           Rx: BYE 
  Promiscuous Redir:      No 
  Route:                  sip:112233445566 at A.B.C.D:5060;transport=UDP 
  DTMF Mode:              rfc2833 
  SIP Options:            (none) 

In this case, when Asterisk user hang up and remote phone still not hang up,
there's still active SIP channel, which should be cleared when BYE received
from any of peers.

In Asterisk Console, I can see BYE from Asterisk user (UA Softphone) to
Asterisk and OK from Asterisk to UA. But Asterisk DO NOT send BYE to ITSP1,
which is wrong?

Pls. advice 

Brgds 

Hoa 




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