I see nothing wrong with either invite, except the fact that there is a cancel between sipxbridge and the provider on the failed call.
Time: 2010-07-29T05:33:09.656000Z Frame: 27 /tmp/trace.aWD29XQ8/_.sipxbridge.trace.xml:68 Source: pfcpbx01-sipXbridge Dest: 67.158.102.9:5060 CANCEL sip:[email protected] <sip%[email protected]>;user=phone SIP/2.0 Call-ID: [email protected] To: <sip:[email protected] <sip%[email protected]> ;user=phone> CSeq: 1 CANCEL From: "sipxbridge" <sip:[email protected]<sip%[email protected]> >;tag=7639194180084180466 Via: SIP/2.0/UDP 173.210.70.22:5080 ;branch=z9hG4bK9385ab1cd6314357907220b0785be7d9363535 Max-Forwards: 70 Route: <sip:67.158.102.9:5060;transport=udp;lr> User-Agent: sipXecs/4.2.1 sipXecs/sipxbridge (Linux) Content-Length: 0 I see 91NPANXXNNNN on each invite going out. If these are in a siptrunk do they even need to send a "9"? I see when it gets to the provider its only 10 digits. I think I'd ask the provider what they. Who is the provider? On Thu, Jul 29, 2010 at 3:03 PM, Matt White <[email protected]>wrote: > Attached is a good call trace. > > The only thing I see in this good trace is the 200 OK comes back in very > quickly..in about 1ms. In the bad trace the phone goes at least 2 ms before > sending the "cancel". 2ms is hardly a long time to wait. > > -M > > > >>> On 7/29/2010 at 02:20 PM, in message < > [email protected]>, "Matt White" < > [email protected]> wrote: > > Ok, been working on this all day. > > > I've attached a sipx-trace file for you to view. Here is the > background. > > > A Polycom 450 (3.1.3 firmware and 3.2.3 firmware both tested) sends a > call to a 1800 number and works > > Same phone calls a local number and the call fails. > > > But this is not a dial plan. I've only configured the long distance dial > plan and dial each with the full 10 digits. If I simply change the dial > plan to use a different providers sip trunk the calls work. > > > The trouble provider is convinced its not thier fault. The weird thing > is the siptrace for a completed tollfree call and failed regular call look > exactly the same. > > > In fact, Tthe sip trace shows it is the handset that is terminating the > call. > > > Can anybody see what the polycom would choke on here? I'll send another > email with a trace from the same phone but to a completed toll free number. > > > -Matt > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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