Well, first, I'd say more information is needed.... I see you're using a Local channel construct, is it pointing to a valid context and extension? Is there any more debugging information you can provide? It seems there's something missing here, if I was debugging the issue to find a solution, I'd be digging up a lot more info, but we're not local to the problem, we have to rely on you the poster.
On Thu, Mar 24, 2011 at 12:21 PM, Ernie Dunbar <[email protected]>wrote: > 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 >
-- _____________________________________________________________________ -- 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
