See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532
There isn't even any code for SIP yet. However the iax integration works
wonders for a link with just a bit of packet loss and jitter. Voice
conversations are nice and crisp and without the pops associated with lost
packets or growth of the jitter buffer.
Is there a reason why this isn't in HEAD?
roy
_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
