Hi, When *ANY* SIP client (softphone, hardphone, ATA) registers to an Asterisk server on my LAN and the extension dials out through a remote SIP provider, the audio is fine for "a while". It then degrades and starts to be "cracky"/jittery. The extension can call once and again and it will always be bad. The only way to somehow "fix" the audio problem is to unregister the local SIP extension/hardphone/softphone and register it back to the same Asterisk server.
I repeated the test several times and it seems to be reproducible. It apparently has nothing to do with my SIP provider or my DSL connection or router. It doesn't even seem to be a network problem on my side. Curiously though, it only happens if dialing out through the SIP provider... I thought maybe the Asterisk server's system clock could be an issue but it doesn't seem to be skewing off too quickly. Also, this problem started showing up 2 weeks ago. Before that, we've been making a lot of calls through the provider without a glitch. Nothing has changed as far as hardware and software is concerned. What could I try? How can I debug this? Why is re-registering the SIP extension making a difference? Any clues? Asterisk 1.4 Thanks, Vieri -- _____________________________________________________________________ -- 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
