I checked the bug reports and all I could find was similar issues with the Asterisk 1.6 which (according to the reports) have been resolved. I couldnt find anyone talking about 1.4 so I created a new issue and someone closed the case and added this note:-
> This does not appear to be a bug, but rather a support issue. Please use the > asterisk-users mailing list for such issues. > The problem looks like your device has not re-registered after your 'sip > reload' which means it does not exist in memory, and thus causes Asterisk to > not know where to send the call. Your device needs to re-register after a > 'sip reload' in order for Asterisk to know where to send the call. I really think that "sip reload" shouldn't purge all the realtime peer registrations. It should treat the realtime peers the same way as the hardcoded peers. As i've said, the hardcoded peers don't lose registration when I issue a SIP RELOAD. Asterisk should be flexible enough to allow modification of the sip.conf file without losing all the realtime registrations. Does anyone have a comment on the subject? Am I expecting too much? I'm open to feedback. Thanks Dan -- _____________________________________________________________________ -- 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
