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

[Hoa Thai Duy]

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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