[OpenSIPS-Users] Load balancer sending 403 when caller hangs uo
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Tue Jun 23 10:38:09 CEST 2009
Hi James,
It might not be correct - by setting the expires to 600, you will not
allow calls longer than 10 minutes! I think you should rather try to
fix the problem with the CANCEL !
BTW, please get a full openips log (use debug=6) for the entire call and
send it to me - just want to run some checks on what is going on there.
Regards,
Bogdan
James Wiegand wrote:
> Hmm,
>
> How about 600 seconds? The only problem is that I can't use OpenSIPS
> if the sessions get hung - especially when I am getting 20 calls per
> second.
>
> -jim
>
> On Fri, Jun 19, 2009 at 5:33 PM, Bogdan-Andrei
> Iancu<bogdan at voice-system.ro> wrote:
>
>> 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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