So... no solution to this problem? > It does depend on how you set up the call forwarding on asterisk and > sometimes when the ATA sends the forwarding call to the Voip provider > server it has nothing to do with it which causes a problem. if you > disable call forwarding remotely see if that works also. its a tricky > situation. > > > > > On Wed, 2011-03-23 at 16:18 -0700, Ernie Dunbar wrote: >> I have a Linksys 2102 ATA here that does call forwarding internally with >> the *72 code, however our Asterisk 1.6.2.17 server doesn't forward the >> call properly. This is what shows up in the console when an incoming >> call >> is made while the ATA is call-forwarded: >> >> -- Called Username >> -- Got SIP response 302 "Moved Temporarily" back from XX.XXX.XX.XXX >> -- Now forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' (thanks >> to SIP/Username-00000045) >> -- Forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' prevented. >> == Everyone is busy/congested at this time (1:1/0/0) >> >> The SIP configuration allows call forwarding (cancallforward=yes), so >> I'm >> at a loss as to what is preventing the forwarding. It's not like >> Asterisk >> is very specific about that. >> >> >> -- >> _____________________________________________________________________ >> -- 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 >
-- _____________________________________________________________________ -- 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
