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

Bogdan-Andrei Iancu bogdan at voice-system.ro
Sat Jun 20 00:33:12 CEST 2009


Hi James,

So, continuing the previous email....what you do by playing with the 
dialog expire param is forcing (from proxy) side to terminate the 
ongoing calls after 30 secs. As said, your call was not CANCELed, but 
was established - and you force the termination of the call after 30 
secs, that is why it works now :D....but it is not correct.

Regards,
Bogdan

James Wiegand wrote:
> Don't know if this is the right thing to try, but when I set the
> dialog timeout the session clears after a few moments.  Is 30 seconds
> too short for use on general calling patterns?  I am looking to pass
> on the order of 700 simultaneous calls.
>
> ...
> modparam("dialog", "default_timeout", 30)
> ...
>
> -jim
>
> On Fri, Jun 19, 2009 at 9:28 AM, James
> Wiegand<originaljimdandy at gmail.com> wrote:
>   
>> 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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