this is a time issue. change your date to older value. everything works again.
paradise dove On 1/25/06, stevanus <[EMAIL PROTECTED]> wrote: > Hi guys, > > I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at > the third days I activated setting jitterbuffer=yes and suddenly there > is no voice when the call is picked up. It's really weird as if asterisk > stops sending rtp packet. I've checked asterisk log and found nothing > suspicious. Just weird :S. > > I tried it in 3 asterisk server and all of them are having the same > symptoms (i.e: no voice). > There is no sound when the call is pickup, no matter the call is from > sip to sip, sip to zap, zap to sip ,sip to zap through iax, nor sip to > sip through iax... > > Is jitterbuffer really the culprit or it's just a coincidence that I > activated the jitterbuffer and my asterisks stopped working? > Is asterisk 1.2.2 not meant for production use? > Has there someone success story implemented asterisk 1.2.2? If there's, > please share me as it can encouraged me to try this beast again :)... > > Currently, I'm rollback to asterisk 1.0.10 to avoid any unprecedented > issue... > > Regards, > > Stevanus > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
