> I just upgraded to the latest cvs (3/23/04 @ about 7:00am MST) and > bumped into a little problem... > > Dialing in from the pstn to sip phones (x-lite softphone on winders and > a grandstream handytone-286 ata), I hear the sip phone ring a few times, > but hear nothing from the pstn side 'til the timeout. Then the sip > phone stops ringing and I hear ringback at the pstn side. > > Could this be a latent configuration problem on my end that didn't > manifest until I upgraded, or is it a problem with this morning's cvs? > > (not urgent, of course -- just put cvs from a few weeks ago back, and > all is well...)
I ran into the same thing with Cisco 7960. Looks like the logic in the sip channel has changed recently. Add a ",r" to the end of your Dial statements in extensions.conf and the issue should go away. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
