Well after doing some more testing it seems the problem is really when going out via SIP to our termination gateway, IAX seems to be working well.
Is there anything SIP specific I should be looking at which might be causing this issue. Regards, -Jed --- Jed Stafford <[EMAIL PROTECTED]> wrote: > This is driving me crazy, when making an outgoing > call > the first 30 seconds is always perfect, then the > party > on the receiving end can always hear me perfectly > but > after 30-60 seconds the audio coming back to me from > them starts to get choppy and drops out. > > I've tried this with multiple devices, from multiple > locations some behind NAT, others not. This is using > the ulaw codec, although i've tried it with alaw as > well. Problem happens via IAX as well as a SIP > channel, both calling PSTN numbers. > > Network performance to the Asterisk server is good, > 15-20ms, performed ping tests for 1-2 hours with > almost no packet loss. > > I'm willing to check or post anything needed here, > but > I need some fresh ideas since i've checked > everything > I can. > > Regards, > > -J > _______________________________________________ > 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 > _______________________________________________ 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
