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/[EMAIL PROTECTED],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 [EMAIL PROTECTED]
* SIP Call
Direction: Outgoing
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:5060
Need Destroy: 2
Last Message: Rx: BYE
Promiscuous Redir: No
Route: sip:[EMAIL PROTECTED]: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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