[Freeswitch-users] not hanging up

Anthony Minessale anthony.minessale at gmail.com
Fri Mar 20 06:48:41 PDT 2009


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 at 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 at 11.2.22.45!
> recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950:
>    ------------------------------------------------------------------------
>    CANCEL sip:12345 at b-pbx-sip-3.abc.xyz.net<sip%3A12345 at 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 at 11.2.22.45 <sip%3A54321 at 11.2.22.45>
> >;tag=as25193d44
>    To: <sip:12345 at b-pbx-sip-3.abc.xyz.net<sip%3A12345 at b-pbx-sip-3.abc.xyz.net>
> >
>    Call-ID: 0c0614d866a62841546cbf3340224682 at 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 at 11.2.22.45 <sip%3A54321 at 11.2.22.45>
> >;tag=as25193d44
>    To: <sip:12345 at b-pbx-sip-3.abc.xyz.net<sip%3A12345 at b-pbx-sip-3.abc.xyz.net>
> >;tag=c5Z8Q1e93p7KD
>    Call-ID: 0c0614d866a62841546cbf3340224682 at 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 at 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 at 11.2.22.46 <sip%3A1000 at 11.2.22.46>
> >;tag=meK8yUgpgU2Zc
>    To: <sip:12345 at 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 at 11.2.56.106:63054;rinstance=64e968d7a1317bc3>
>    To: <sip:12345 at 11.2.56.106:63054
> ;rinstance=64e968d7a1317bc3>;tag=db12c87a
>    From: "Extension 1000"<sip:1000 at 11.2.22.46 <sip%3A1000 at 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 at 11.2.56.106:63054
> ;rinstance=64e968d7a1317bc3>;tag=db12c87a
>    From: "Extension 1000"<sip:1000 at 11.2.22.46 <sip%3A1000 at 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.
>
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>


-- 
Anthony Minessale II

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