[OpenSIPS-Users] Load balancer sending 403 when caller hangs uo

James Wiegand originaljimdandy at gmail.com
Fri Jun 19 16:28:24 CEST 2009


Hi Bogdan,

Here's the dialog from a test call.
The remote client is Eyebeam on a PC connected to Asterisk.  I made a
call and hung up before answering.  The call has been terminated for
some time.  I can do an lb_reload to clear out the hung lb session.

opensipsctl fifo lb_list
Destination:: sip:XXX.XXX.XXX.6 id=1
        Resource:: pstn max=0 load=0
Destination:: sip:XXX.XXX.XXX.7 id=2
        Resource:: pstn max=0 load=0
Destination:: sip:XXX.XXX.XXX.8 id=3
        Resource:: pstn max=1 load=1
Destination:: sip:XXX.XXX.XXX.9 id=4
        Resource:: pstn max=0 load=0

opensipsctl fifo dlg_list
dialog::  hash=3498:265315739
        state:: 3
        user_flags:: 0
        timestart:: 1245419911
        timeout:: 99843
        callid:: 30cd5dba1a90fbe7023054f8293fc520 at YYY.YYY.YYY.12
        from_uri:: sip:8705082000 at YYY.YYY.YYY.12
        from_tag:: as14720305
        caller_contact:: sip:8705082000 at YYY.YYY.YYY.12
        caller_cseq:: 102
        caller_route_set::
        caller_bind_addr:: udp:XXX.XXX.XXX.24:5060
        to_uri:: sip:8706569978 at XXX.XXX.XXX.24
        to_tag:: as4042950a
        callee_contact:: sip:8706569978 at XXX.XXX.XXX.8
        callee_cseq:: 102
        callee_route_set::
        callee_bind_addr:: udp:XXX.XXX.XXX.24:5060
dialog::  hash=3895:1205860066
        state:: 3
        user_flags:: 0
        timestart:: 1245419947
        timeout:: 99879
        callid:: 768a3fbb026fec2038c9334c05e12298 at YYY.YYY.YYY.12
        from_uri:: sip:8705082000 at YYY.YYY.YYY.12
        from_tag:: as5a726731
        caller_contact:: sip:8705082000 at YYY.YYY.YYY.12
        caller_cseq:: 102
        caller_route_set::
        caller_bind_addr:: udp:XXX.XXX.XXX.24:5060
        to_uri:: sip:8706569978 at XXX.XXX.XXX.24
        to_tag:: as3ac79c83
        callee_contact:: sip:8706569978 at XXX.XXX.XXX.8
        callee_cseq:: 102
        callee_route_set::
        callee_bind_addr:: udp:XXX.XXX.XXX.24:5060


TCP SIP trace, not from the same call, but with the same result:

09:08:37.758213 IP (tos 0x0, ttl  45, id 34347, offset 0, flags
[none], proto: UDP (17), length: 855) YYY.YYY.YYY.12.sip >
XXX.XXX.XXX.24.sip: SIP, length: 827
        INVITE sip:8706569978 at XXX.XXX.XXX.24 SIP/2.0
        Via: SIP/2.0/UDP YYY.YYY.YYY.12:5060;branch=z9hG4bK1fe6bb8d;rport
        From: "device" <sip:8705082000 at YYY.YYY.YYY.12>;tag=as0fb4ac11
        To: <sip:8706569978 at XXX.XXX.XXX.24>
        Contact: <sip:8705082000 at YYY.YYY.YYY.12>
        Call-ID: 6d25870c2c9d32c90c6e4498079dcd84 at YYY.YYY.YYY.12
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX
        Max-Forwards: 70
        Date: Fri, 19 Jun 2009 14:00:50 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
        Supported: replaces
        Content-Type: application/sdp
        Content-Length: 284

        v=0
        o=root 3848 3848 IN IP4 YYY.YYY.YYY.12
        s=session
        c=IN IP4 YYY.YYY.YYY.12
        t=0 0
        m=audio 6962 RTP/AVP 0 3 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:3 GSM/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=silenceSupp:off - - - -
        a=ptime:20
        a=sendrecv

09:08:37.759853 IP (tos 0x10, ttl  64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 345) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip:
SIP, length: 317
        SIP/2.0 100 Giving a try
        Via: SIP/2.0/UDP YYY.YYY.YYY.12:5060;branch=z9hG4bK1fe6bb8d;rport=5060
        From: "device" <sip:8705082000 at YYY.YYY.YYY.12>;tag=as0fb4ac11
        To: <sip:8706569978 at XXX.XXX.XXX.24>
        Call-ID: 6d25870c2c9d32c90c6e4498079dcd84 at YYY.YYY.YYY.12
        CSeq: 102 INVITE
        Server: VistaVox SIP Service
        Content-Length: 0


09:08:40.113592 IP (tos 0x10, ttl  64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 874) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip:
SIP, length: 846
        SIP/2.0 183 Session Progress
        Via: SIP/2.0/UDP
YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060
        Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c>
        From: "device" <sip:8705082000 at YYY.YYY.YYY.12>;tag=as0fb4ac11
        To: <sip:8706569978 at XXX.XXX.XXX.24>;tag=as2661bdde
        Call-ID: 6d25870c2c9d32c90c6e4498079dcd84 at YYY.YYY.YYY.12
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
        Supported: replaces
        Contact: <sip:8706569978 at XXX.XXX.XXX.8>
        Content-Type: application/sdp
        Content-Length: 262

        v=0
        o=root 18239 18239 IN IP4 XXX.XXX.XXX.8
        s=session
        c=IN IP4 XXX.XXX.XXX.8
        t=0 0
        m=audio 16734 RTP/AVP 0 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=silenceSupp:off - - - -
        a=ptime:20
        a=sendrecv

09:08:55.476673 IP (tos 0x0, ttl  45, id 34348, offset 0, flags
[none], proto: UDP (17), length: 372) YYY.YYY.YYY.12.sip >
XXX.XXX.XXX.24.sip: SIP, length: 344
        CANCEL sip:8706569978 at XXX.XXX.XXX.24 SIP/2.0
        Via: SIP/2.0/UDP YYY.YYY.YYY.12:5060;branch=z9hG4bK1fe6bb8d;rport
        From: "device" <sip:8705082000 at YYY.YYY.YYY.12>;tag=as0fb4ac11
        To: <sip:8706569978 at XXX.XXX.XXX.24>
        Call-ID: 6d25870c2c9d32c90c6e4498079dcd84 at YYY.YYY.YYY.12
        CSeq: 102 CANCEL
        User-Agent: Asterisk PBX
        Max-Forwards: 70
        Content-Length: 0


09:08:55.477405 IP (tos 0x10, ttl  64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 393) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip:
SIP, length: 365
        SIP/2.0 405 Method Not Allowed
        Via: SIP/2.0/UDP YYY.YYY.YYY.12:5060;branch=z9hG4bK1fe6bb8d;rport=5060
        From: "device" <sip:8705082000 at YYY.YYY.YYY.12>;tag=as0fb4ac11
        To: <sip:8706569978 at XXX.XXX.XXX.24>;tag=9508a3e09327a949e746abbd3d262852.51a3
        Call-ID: 6d25870c2c9d32c90c6e4498079dcd84 at YYY.YYY.YYY.12
        CSeq: 102 CANCEL
        Server: VistaVox SIP Service
        Content-Length: 0


09:09:04.124970 IP (tos 0x10, ttl  64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 860) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip:
SIP, length: 832
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP
YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060
        Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c>
        From: "device" <sip:8705082000 at YYY.YYY.YYY.12>;tag=as0fb4ac11
        To: <sip:8706569978 at XXX.XXX.XXX.24>;tag=as2661bdde
        Call-ID: 6d25870c2c9d32c90c6e4498079dcd84 at YYY.YYY.YYY.12
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
        Supported: replaces
        Contact: <sip:8706569978 at XXX.XXX.XXX.8>
        Content-Type: application/sdp
        Content-Length: 262

        v=0
        o=root 18239 18240 IN IP4 XXX.XXX.XXX.8
        s=session
        c=IN IP4 XXX.XXX.XXX.8
        t=0 0
        m=audio 16734 RTP/AVP 0 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=silenceSupp:off - - - -
        a=ptime:20
        a=sendrecv

09:09:05.123714 IP (tos 0x10, ttl  64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 860) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip:
SIP, length: 832
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP
YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060
        Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c>
        From: "device" <sip:8705082000 at YYY.YYY.YYY.12>;tag=as0fb4ac11
        To: <sip:8706569978 at XXX.XXX.XXX.24>;tag=as2661bdde
        Call-ID: 6d25870c2c9d32c90c6e4498079dcd84 at YYY.YYY.YYY.12
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
        Supported: replaces
        Contact: <sip:8706569978 at XXX.XXX.XXX.8>
        Content-Type: application/sdp
        Content-Length: 262

        v=0
        o=root 18239 18240 IN IP4 XXX.XXX.XXX.8
        s=session
        c=IN IP4 XXX.XXX.XXX.8
        t=0 0
        m=audio 16734 RTP/AVP 0 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=silenceSupp:off - - - -
        a=ptime:20
        a=sendrecv

09:09:06.123020 IP (tos 0x10, ttl  64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 860) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip:
SIP, length: 832
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP
YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060
        Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c>
        From: "device" <sip:8705082000 at YYY.YYY.YYY.12>;tag=as0fb4ac11
        To: <sip:8706569978 at XXX.XXX.XXX.24>;tag=as2661bdde
        Call-ID: 6d25870c2c9d32c90c6e4498079dcd84 at YYY.YYY.YYY.12
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
        Supported: replaces
        Contact: <sip:8706569978 at XXX.XXX.XXX.8>
        Content-Type: application/sdp
        Content-Length: 262

        v=0
        o=root 18239 18240 IN IP4 XXX.XXX.XXX.8
        s=session
        c=IN IP4 XXX.XXX.XXX.8
        t=0 0
        m=audio 16734 RTP/AVP 0 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=silenceSupp:off - - - -
        a=ptime:20
        a=sendrecv

09:09:08.123528 IP (tos 0x10, ttl  64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 860) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip:
SIP, length: 832
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP
YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060
        Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c>
        From: "device" <sip:8705082000 at YYY.YYY.YYY.12>;tag=as0fb4ac11
        To: <sip:8706569978 at XXX.XXX.XXX.24>;tag=as2661bdde
        Call-ID: 6d25870c2c9d32c90c6e4498079dcd84 at YYY.YYY.YYY.12
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
        Supported: replaces
        Contact: <sip:8706569978 at XXX.XXX.XXX.8>
        Content-Type: application/sdp
        Content-Length: 262

        v=0
        o=root 18239 18240 IN IP4 XXX.XXX.XXX.8
        s=session
        c=IN IP4 XXX.XXX.XXX.8
        t=0 0
        m=audio 16734 RTP/AVP 0 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=silenceSupp:off - - - -
        a=ptime:20
        a=sendrecv


On Fri, Jun 19, 2009 at 2:18 AM, Bogdan-Andrei
Iancu<bogdan at voice-system.ro> wrote:
> Hi James,
>
> Could you please check if the "dialog" module sees the call as ended? Use
> "opensipsctl fifo dlg_list"
>  (http://www.opensips.org/html/docs/modules/1.5.x/dialog.html#id272726) and
> paste the output here.
>
> Also, do you have a full SIP trace of the call (ngrep) ?
>
> Regards,
> Bogdan
>
>
>
> James Wiegand wrote:
>>
>> Hi all,
>>
>> I am using OpenSIPS 1.5.1 and the lb module.  Following the example I
>> see this chunk of code execute when the caller hangs up as the dial
>> progresses (but before the other side answers):
>>
>>        # from now on we have only the initial requests
>>        if (!is_method("INVITE")) {
>>                send_reply("405","Method Not Allowed");
>>                exit;
>>        }
>>
>> This leaves a session hanging in the load balancer:
>>
>> Destination:: sip:XXX.XXX.XXX.XXX id=3
>>        Resource:: pstn max=1 load=1
>>
>> I'm seeing CANCEL come in from the caller and it looks like
>> !t_check_trans() is not picking this up?  How do I catch this case?
>>
>> Thanks for the help,
>>
>> -jim
>>
>>
>>
>
>



-- 
-- 
Jim Wiegand
-----------
Home:  originaljimdandy at gmail.com
AIM:     originaljimdandy



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