At 23:33 12/21/2009, Doug wrote: >At 00:46 12/21/2009, Alex Balashov wrote: > >A packet capture would be needed to illuminate the source of the problem. > >Thanks, Alex for your suggestion.
>I just don't see where the extension responds to >the INVITE. What would prevent that? Problem solved: Each peer in sip.conf needs: qualify=yes > >By the way, I have a bunch of phones behind this >same router that work just fine on our old v1.2 >system. > > > > > > > > >On 12/21/2009 01:39 AM, Doug wrote: > > > >> I've turned on NAT everywhere I can think, but > >> even though I hear ringing on the calling > >> phone (different system) the called phone does > >> not ring. > >> > >> Has anyone bumped into this lately? > > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
