I have phones registered to a FS box, and an * box. There is a sip trunk between the two boxes. A phone on my * (54321) calls a FS phone (12345); if I hang up the * phone while it's still ringing, this is what I get on the sip trace on FS: ... 2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/12345@11.2.22.45! recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950: ------------------------------------------------------------------------ CANCEL sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345@b-pbx-sip-3.abc.xyz.net>SIP/2.0 Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport From: "Steve" <sip:54321@11.2.22.45 <sip%3A54321@11.2.22.45>
;tag=as25193d44 To: <sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345@b-pbx-sip-3.abc.xyz.net>
Call-ID: 0c0614d866a62841546cbf3340224682@11.2.22.45 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 328 bytes to udp/[11.2.22.45]:5060 at 19:05:44.313572: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport=5060 From: "Steve" <sip:54321@11.2.22.45 <sip%3A54321@11.2.22.45>
;tag=as25193d44 To: <sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345@b-pbx-sip-3.abc.xyz.net> ;tag=c5Z8Q1e93p7KD Call-ID: 0c0614d866a62841546cbf3340224682@11.2.22.45 CSeq: 103 CANCEL Content-Length: 0
-------------------------------------------------------- The effect is that the FS keeps on ringing - it doesn't detect the hangup. When I call from a FS phone (1000) to another FS phone (12345), and I hang up the calling phone while it's still ringing, this is what I get on the sip trace: ... send 425 bytes to udp/[11.2.56.106]:63054 at 19:15:29.737163: ------------------------------------------------------------------------ CANCEL sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3 SIP/2.0 Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a Max-Forwards: 69 From: "Extension 1000" <sip:1000@11.2.22.46 <sip%3A1000@11.2.22.46>
;tag=meK8yUgpgU2Zc To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3> Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0
------------------------------------------------------------------------ recv 427 bytes from udp/[11.2.56.106]:63054 at 19:15:29.838863: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a Contact: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3> To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3>;tag=db12c87a From: "Extension 1000"<sip:1000@11.2.22.46 <sip%3A1000@11.2.22.46>
;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0
------------------------------------------------------------------------ recv 376 bytes from udp/[11.2.56.106]:63054 at 19:15:29.839334: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3>;tag=db12c87a From: "Extension 1000"<sip:1000@11.2.22.46 <sip%3A1000@11.2.22.46>
;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0
... It works just fine. Any ideas? I'm not sure where to go with this. Thanks.
It looks like interop issue with dialog matching between asterisk and freeswitch. Which version of asterisk is it? Which version of FreeSWITCH? You may want to provide a trace of the whole call starting with the invite. FS is having trouble identifying what call asterisk wants to cancel. 2009/3/19 Steven Ward <steve.d.ward@gmail.com>
I have phones registered to a FS box, and an * box. There is a sip trunk between the two boxes.
A phone on my * (54321) calls a FS phone (12345); if I hang up the * phone while it's still ringing, this is what I get on the sip trace on FS:
...
2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/12345@11.2.22.45! recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950: ------------------------------------------------------------------------ CANCEL sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345@b-pbx-sip-3.abc.xyz.net>SIP/2.0 Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport From: "Steve" <sip:54321@11.2.22.45 <sip%3A54321@11.2.22.45>
;tag=as25193d44 To: <sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345@b-pbx-sip-3.abc.xyz.net>
Call-ID: 0c0614d866a62841546cbf3340224682@11.2.22.45 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0
------------------------------------------------------------------------ send 328 bytes to udp/[11.2.22.45]:5060 at 19:05:44.313572: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport=5060 From: "Steve" <sip:54321@11.2.22.45 <sip%3A54321@11.2.22.45>
;tag=as25193d44 To: <sip:12345@b-pbx-sip-3.abc.xyz.net<sip%3A12345@b-pbx-sip-3.abc.xyz.net> ;tag=c5Z8Q1e93p7KD Call-ID: 0c0614d866a62841546cbf3340224682@11.2.22.45 CSeq: 103 CANCEL Content-Length: 0
--------------------------------------------------------
The effect is that the FS keeps on ringing - it doesn't detect the hangup.
When I call from a FS phone (1000) to another FS phone (12345), and I hang up the calling phone while it's still ringing, this is what I get on the sip trace:
...
send 425 bytes to udp/[11.2.56.106]:63054 at 19:15:29.737163: ------------------------------------------------------------------------ CANCEL sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3 SIP/2.0 Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a Max-Forwards: 69 From: "Extension 1000" <sip:1000@11.2.22.46 <sip%3A1000@11.2.22.46>
;tag=meK8yUgpgU2Zc To: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3> Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0
------------------------------------------------------------------------ recv 427 bytes from udp/[11.2.56.106]:63054 at 19:15:29.838863: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a Contact: <sip:12345@11.2.56.106:63054;rinstance=64e968d7a1317bc3> To: <sip:12345@11.2.56.106:63054 ;rinstance=64e968d7a1317bc3>;tag=db12c87a From: "Extension 1000"<sip:1000@11.2.22.46 <sip%3A1000@11.2.22.46>
;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0
------------------------------------------------------------------------ recv 376 bytes from udp/[11.2.56.106]:63054 at 19:15:29.839334: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a To: <sip:12345@11.2.56.106:63054 ;rinstance=64e968d7a1317bc3>;tag=db12c87a From: "Extension 1000"<sip:1000@11.2.22.46 <sip%3A1000@11.2.22.46>
;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0
...
It works just fine. Any ideas? I'm not sure where to go with this. Thanks.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (2)
-
Anthony Minessale
-
Steven Ward