PAETEC uses a Broadsoft platform. "I think" their preferred refresh timer might be 10 minutes.
On Fri, Jan 27, 2012 at 4:29 AM, Tony Graziano <[email protected]> wrote: > 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! -- ~~~~~~~~~~~~~~~~~~ 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/
