Yes, these are our session-timer settings in sip.conf:
session-timers=originate session-expires=600 session-minse=90 session-refresher=uas Quoting Faisal Hanif <[email protected]>:
Have you tried SIP session timer values in sip.conf -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Ernie Dunbar Sent: Tuesday, June 28, 2011 9:33 PM To: [email protected] Subject: [asterisk-users] Outgoing calls get dropped on high-latency connections. We're a VoIP provider essentially competing with our local incumbent Telco, and a sizeable number of our customers use satellite internet. As a result, these customers never have ping times less than 500ms, and are often exceeding 2500ms. I manually apply a "patch" to the Asterisk source code every time we upgrade Asterisk, described here: http://www.mail-archive.com/[email protected]/msg178034.html This change allows our satellite customers to maintain their SIP connection for more than 5 minutes. But we're currently using Asterisk 1.6.2.17, and this version seems to have one very strange bug on these high latency connections. On outgoing and *only* outgoing calls, the call drops after two or three minutes. Incoming calls do not have this problem, so I don't think it's the SIP connection getting killed due to a slow INVITE response. Has anyone heard of this bug? Or should I submit a new bug report to the Asterisk project? ---------------------------------------------------------------- This message was sent using Lightspeed.ca's Advanced Webmail. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
---------------------------------------------------------------- This message was sent using Lightspeed.ca's Advanced Webmail. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
