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
