[OpenSIPS-Users] failure route with $rU == null
Anton Zagorskiy
a.zagorskiy at oyster-telecom.ru
Wed Apr 6 18:05:33 CEST 2011
Hi Bogdan,
Such situation (when in the failure route $rU is NULL) isn't good - I've
checked log using Call-ID, and:
1. I already received BYE on that session (and stored CDR)
2. I'm using call forward on request timeout and use $rU to determine
forward sip account of dialed user. How can I do this if I don't know
request user?
There is no any data with given Call-ID in log between BYE and failure
request. Failure route was totally unexpected...
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagorskiy at oyster-telecom.ru
www.oyster-telecom.ru
> -----Original Message-----
> From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> Sent: Wednesday, April 06, 2011 7:51 PM
> To: OpenSIPS users mailling list
> Cc: Anton Zagorskiy
> Subject: Re: [OpenSIPS-Users] failure route with $rU == null
>
> Hi Anton,
>
> In sequential requests you typically have in RURI the contact URI from the
> original INVITE + 200 OK. And username in contact URIs in most of the case
> irrelevant - important is the IP, port and proto.
>
> Regards,
> Bogdan
>
> On 04/06/2011 06:23 PM, Anton Zagorskiy wrote:
> > Hi.
> >
> > In which cases in the failure route will be passed a request with $rU
> > == null?
> > In a log I see just:
> >
> > DBG:tm:utimer_routine: timer routine:4,tl=0x803ad3218 next=0x0,
> > timeout=256300000
> > DBG:tm:timer_routine: timer routine:2,tl=0x803ae43c0 next=0x0,
> > timeout=260
> > DBG:tm:wait_handler: removing 0x803ae4340 from table
> > DBG:tm:delete_cell: delete transaction 0x803ae4340
> > DBG:dialog:next_state_dlg: dialog 0x803ad68a0 changed from state 5 to
> > state 5, due event 1
> > DBG:dialog:unref_dlg: unref dlg 0x803ad68a0 with 1 -> 2
> > DBG:tm:wait_handler: done
> > DBG:tm:timer_routine: timer routine:1,tl=0x803ad7028 next=0x0,
> > timeout=262
> > DBG:tm:final_response_handler: stop retr. and send CANCEL
> > (0x803ad6dd8)
> > DBG:tm:t_should_relay_response: T_code=100, new_code=408
> > DBG:tm:t_pick_branch: picked branch 0, code 408 (prio=800)
> > DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=408,
> > flags=2
> > INFO:core:buf_init: initializing...
> > *** +++ failure_route[1] has started
> > *** failure_route[1]:mi = 1; R-URI:
'sip:87.249.51.227:5090;transport=udp'
> > '<null>' @ '87.249.51.227'; From-Uri: 'sip:6010666 at mydomain.com'
> '6010666'
> > @ 'mydomain.com'
> > DBG:avpops:ops_delete_avp: 0 avps were removed
> > DBG:core:comp_scriptvar: str 20 : 0
> > DBG:tm:t_check_status: checked status is<408>
> >
> >
> >
> >
> >
> >
> > WBR, Anton Zagorskiy
> > VoIP Developer, Oyster Telecom
> > Phone.: +7 812 601-0666
> > Fax: +7 812 601-0593
> > a.zagorskiy at oyster-telecom.ru
> > www.oyster-telecom.ru
> >
> >
> >
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS eBootcamp - 2nd of May 2011
> OpenSIPS solutions and "know-how"
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