[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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