On Thursday 10 November 2011 2:09:09 pm Danny Nicholas wrote: > You might want to see if it is > 1. a phone or an asterisk transfer - phone transfer hits button on phone > and does attended/blind transfer that way; asterisk transfer initiated > with *1 or #2 (whatever value is specified in features.conf)
It's a phone-based transfer. > 2. attended or blind transfer The attended transfers seem to give them the most problems. They say that if they hit the transfer softkey while the destination extension is still ringing, it tends to work well. Mike. > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Mike Diehl > Sent: Thursday, November 10, 2011 3:05 PM > To: [email protected] > Subject: Re: [asterisk-users] Frequent Asterisk Restarts > > 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
