this is a time issue.
change your date to older value. everything works again.
paradise dove
On 1/25/06, stevanus <step-one-too@bdg.centrin.net.id>
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
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