On 09/06/2010 10:28 AM, Olivier wrote: > Hi, > > With a 1.4.35 or 1.6.1.19, I'm facing this behaviour : > > - extension 7002 is a SIP hard phone currently configured to forward > incoming calls to extension 7003, when a call is unanswered within a 10s > time frame > > - when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20) > statement and no one answers, then : > - after 10s, Asterisk receives "SIP 302 Moved temporarily" message and > enters its dialplan to call 7003, as required, > - 10s later (or 20s from the very start), call from 7001 to 7003 is cut > and the next statement after Dial(SIP/7002,20) is run. > > The behaviour I would ideally implement is : > - whenever a "SIP 302 Moved temporarily" message is received, timer > associated to the original call (the one from 7001 to 7002) is reset to > another 20s period
That would change the logic in app_dial quite drastically, as it would have to remember separate timeouts for each of the originally-dialed destinations in case they get forwarded elsewhere. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: [email protected] Check us out at www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
