Well, it would help if you would state who the ITSP is, in the event this has been looked at before. Besides that, the port 5060 /5080 thing could indicate one of two thing...
1. If the call originates at sipx, it is always sent outbound VIA 5060. If the call originates at the PSTN it is sent VIA 5080. 2. If the ITSP uses different upstream providers and the call does not follow the same IP as the provider gateway, the upstream provider may only provide signalling on port 5060. (flowroute tends to be this way, for example, media comes from an ip other than the provider gateway, etc., they have no control over calls through their system or who they route through). I also recall we have seen this consitently with Vitelity too. A siptrace might be the only way to truly see what's wrong with the failed call though. You can always change the call timer in the gateway settings to extend it, but that's not really a fix. On Thu, Jan 26, 2012 at 6:31 PM, jnolen <[email protected]> wrote: > Greetings, > > A customer is running sipXecs (4.4.0- 2011-10-12EDT17:34:54 > domU-12-31-39-00-0D-21). > > They complained that outbound calls to their ITSP were dropping consistently > at 15 minutes into the call. The call fails exactly when the re-INVITE > keep-alive is sent. > > Restarting sipxbridge appears to solve the problem which, by the way has > happened only twice in 3 months. > > Researching a 'good' and 'failed' call, the Via Header is wrong in the > failed case -- see re-INVITE messages below. In the 'good' call the Via > Header points to the bridge and port 5080 as expected. In the failed call, > the Via points to the ITSP and port 5060. > > In the logs, I see no errors or message sequences that might explain such > behavior. I have requested the user to increase the log level for SIP > Trunking to DEBUG in hopes of capturing more information. > > In the meantime, any suggestions? > > Thanks, > > jim > > --------------------------------------------------- > > bad call: > > 2012-01-25T21:58:20.914000Z:1361207:OUTGOING:INFO:netvoice:ReInviteSender-1342:00000000:sipXbridge:Sent > SIP Message : > ----Remote Host:10.20.30.37---- Port: 5060---- > INVITE > sip:[email protected]:5060;insideBWWEST=BWWESTSIG-j895mp1bqdiba;transport=udp > SIP/2.0 > Via: SIP/2.0/UDP > 10.20.30.37:5060;branch=z9hG4bK71c18bf42eeee0ab9fb3bc6fcdf260b0353439 > CSeq: 2 INVITE > Call-ID: [email protected] > From: \"sipxbridge\" <sip:[email protected]>;tag=2513112800114058419 > To: <sip:[email protected];user=phone>;tag=1819991104-1327527848452 > Remote-Party-ID: > <sip:[email protected];user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber > Max-Forwards: 70 > User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux) > Allow: INVITE,BYE,ACK,CANCEL,OPTIONS > Accept: application/sdp > Content-Type: application/sdp > Contact: <sip:[email protected]:5080;transport=udp> > Session-Expires: 900;refresher=uac > Min-SE: 90 > Content-Length: 257 > > > good call: > > INVITE sip:[email protected];user=phone SIP/2.0 > Call-ID: [email protected] > CSeq: 1 INVITE > From: "sipxbridge" <sip:[email protected]>;tag=7554767118327400281 > To: <sip:[email protected];user=phone> > Via: SIP/2.0/UDP > 172.20.3.10:5080;branch=z9hG4bK62b99785b9d6ebd48036d5080407660c3934 > Max-Forwards: 70 > User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux) > Contact: <sip:[email protected]:5080;transport=udp> > Route: <sip:10.20.30.37:5060;transport=udp;lr> > Session-Expires: 900;refresher=uac > References: > [email protected];rel=chain;sipxecs-tag=request-invite-z9hg4bk-xx-064a4yl62itzore3npyhwq12za > Allow: INVITE,BYE,ACK,CANCEL,OPTIONS > Supported: timer > Content-Type: application/sdp > Content-Length: 257 > > _______________________________________________ > 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: 434.465.6833 ~~~~~~~~~~~~~~~~~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
