Hi Boris, as it also appears to just be a ast bridge interaction issue, not just a SIP issue, I doubt this will help.
Cheers, Ben. On Thu, 2005-11-03 at 18:01 +1100, Boris Bakchiev wrote: > Perhaps the async patch will help you with this. > Its will use independent timing for in/out channels. > http://bugs.digium.com/view.php?id=5374 > > > >Asterisk is just the "slave" of the incoming SIP frames. So as it > >receives them it sends them out to the outgoing channel. > > >Now the sirrix is externally synced to the ISDN line so can only send > out > >audio at the rate clocked by the line. > > >So the excess samples accumulated in a buffer in the sirrix kernel > driver. > >Worse, when the buffer filled, the channel driver actually waited for > the > >write - which then caused trouble for other things as the thread got > >blocked. > > _______________________________________________ > Asterisk-Dev mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > _______________________________________________ Asterisk-Dev mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
