Hoa Thai Duy
2006-Jun-22 00:50 UTC
[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@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@W.X.Y.Z * SIP Call Direction: Outgoing Call-ID: 14448d41170ac3a66a41602575476d5f@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@A.B.C.D:5060 Need Destroy: 2 Last Message: Rx: BYE Promiscuous Redir: No Route: sip:112233445566@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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060622/ce2438bf/attachment.htm
Hoa Thai Duy
2006-Jun-25 22:16 UTC
[Asterisk-Users] SIP Channel hangup problem with re-INVITE enabled- ugrent
Does anyone on this list has idea? _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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@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@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@W.X.Y.Z * SIP Call Direction: Outgoing Call-ID: 14448d41170ac3a66a41602575476d5f@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@A.B.C.D:5060 Need Destroy: 2 Last Message: Rx: BYE Promiscuous Redir: No Route: sip:112233445566@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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060625/306059a8/attachment.htm