I'll explore the options outlined in the document below, later tonight. However, I've been able to reproduce the problem! It seems that when one of my users, at a particular site, transfers a call to another extension, asterisk bounces.
They're using Polycom 301's and 501's with SIP version 3.1.4.0070. Without having gotten the debug info, yet, is there anything else I can look at? TIA. On Thursday 10 November 2011 11:46:35 am Leif Madsen wrote: > On 11-11-10 01:15 PM, Eric Wieling wrote: > > The Asterisk source tree has a document with instructions on getting a > > backtrace from the segfaults so you can report it on the issue tracker. > > Most up to date documentation is on the Asterisk wiki: > > https://wiki.asterisk.org/wiki/display/AST/Debugging > > -- > _____________________________________________________________________ > -- 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 -- Take care and have fun, Mike Diehl. -- _____________________________________________________________________ -- 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
