> You have hardware echo canceling *outside* of your T1 card? No, on the card.
> The DAHDI layer has some buffering that can help with jitter, but the > default buffers can only handle 80ms of jitter. You can increase this by > setting the 'buffers' option in chan_dahdi.conf; each buffer is 20ms by > default. I'm running 1.6.2 and it appears that this is called jitterbuffers there. Is that right? I've set it to 20 and it did indeed help quite a bit, so I tried 30. > It sounds like the lack of a proper jitter buffer (of adequate size) is > the issue here, since when the audio is directed at endpoints outside of > Asterisk that have them, the audio is as you'd expect it to be. Interestingly, that isn't completely true. If it goes out a SIP trunk to PSTN, it works fine, but when it goes out a SIP trunk to the SV8300 (where the T1 goes), it has the same problem. This was leading me to believe that the problem was on the 8300. -- _____________________________________________________________________ -- 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
