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 [email protected] www.oyster-telecom.ru > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:[email protected]] > 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:[email protected]' > '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 > > [email protected] > > www.oyster-telecom.ru > > > > > > > > > > > > _______________________________________________ > > Users mailing list > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Bogdan-Andrei Iancu > OpenSIPS eBootcamp - 2nd of May 2011 > OpenSIPS solutions and "know-how" _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
