Mark, well, your ping results look pretty damn good. I get good quality with up to 20 ms jitter and 80 ms pings. If on a very bad day jitter on my line is 30+ ms, calls to PSTN sound "under the water" but calls to other SIP devices through Broadvoice sound fine. I guess the jitter buffer/handling is better in SIP adapters than what Broadvoice uses to terminate to PSTN.
Anyway, I don't see why you get bad quality (it's not that I don't believe you) so can't help you more. You're probably sick and tired and ready to try someone else, which I understand if you've been dealing with this for over a year. If not, you could try changing the RTP packet size in asterisk (rtp.c) to either 10, 30 or 40 ms and see if this makes a difference... --Luki _______________________________________________ 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
