-----Original Message----- From: WORLEY, DALE R (DALE) [mailto:[email protected]] Sent: Thursday, March 18, 2010 11:19 AM To: Todd Hodgen; [email protected] Subject: RE: [sipx-users] Call failures via UDP
_______________________________________ From: [email protected] [[email protected]] On Behalf Of Todd Hodgen [[email protected]] 17/03 12:10:10.231 DEBUG1 UA [75:2xxx273807] <-- INVITE SDP 6835264022320204725 1 10.110.2.10:30002 PCMU/G729/DTMF sendrecv 17/03 12:10:10.231 DEBUG1 UA [75:2xxx273807] --> 180 Ringing 17/03 12:10:18.106 DEBUG1 UA [75:2xxx273807] --> OK SDP 384 384 2xx.xxx.131.152:26154 PCMU/DTMF 17/03 12:10:18.293 DEBUG1 UA [75:2xxx273807] --> INVITE SDP 384 385 10.110.2.101:2234 PCMU/DTMF A call is offered (INVITE), VOP answers (OK), then connects it to the slave phone (INVITE). At this time the answer from sipXecs is "408 Request Timeout" only 3 seconds after the INVITE. ____________________________________________ Have you got a sipXecs trace of the the failed call? Dale --------------------------- Attached is a link to a merged xml file from a failed call. Here is the scenario. The customer is running on sipXecs 4.0.4. All calls come into an alias that is assigned an extension - 398. I won't get into all the details, but 398 forwards these calls to extension 300 during working hours. Ext 300 is a Voice Operator Panel soft client. It is programmed to be "Tethered" to extension 301. When 300 gets a call, it forks the call to 301, which is used to answer it, while 300 provides call control, presence, etc. This works flawlessly for hundreds of calls each day, with the exception of calls from one of their customers. This customer that calls them is on an older digital key system, fed by analog trunks, that are provisioned on an IAD fed by channelized T1. Per this company, they are having no reports of issues with other calls them make. When the user from the digital key system calls the sipXecs number, the console rings, when they pick up the tethered call it drops. The ITSP in the middle is Broadvox. They did a trace and state they see a disconnect from the digital key system end. I suspect that is not where the issue is. The debug from the console reported the data that I previously provided - TCP calls work, UDP don't. The link to the xml file is http://misiusystems.com/temp/cust_merged.xml Any ideas would be appreciated. Regards, Todd _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
