[asterisk-users] 1.4 SIP Jitter Buffer

[Luc Moreira]

Gregory

We have many Asterisk 1.4.13 in production solid like a rock.

Couples examples:
a) Asterisk 1.4.13 + Unicall + 2 E1 MFCR2 Digium + Legacy PBX
    60+ Extentions /  IVR / 10~30 concorrent calls

b) Asterisk 1.4.11 + 1 E1 ISDN PRI Digium
    50+ Extentions / IVR / 5 Queues / ~2000 call/day

c) Asterisk 1.4.13 + 4 E1 ISDN Digium (working in progress)
    CallCenter / 150 PAs / 15 Queues / expected 8000 calls/day

-- 
Luc

Gregory Boehnlein escreveu:
> Hello,
> 	I'm running into a few situations on lossy network links where a SIP
> jitter buffer w/ some PLC would be helpful. My main TDM gateways are running
> 1.2 (which is solid, stable, reliable and very very very well behaved when
> you know it's limitations), but I'm considering upgrading them before the
> end of the year to 1.4. Two of the main reasons that I would do this are
> Variable Length DTMF and SIP Jitter Buffering. I would be very interested in
> hearing from anyone that is actually running 1.4 in a PRODUCTION
> environment, gatewaying SIP to TDM using Digium cards. To me, production
> means being able to have 3-4 PRI circuits maxed out for 12+ hours a day and
> 7+ call setups / second. Anything less than that is not really going to be
> an accurate comparison to what I have running.
>
> Anyone have any feedback about this combination? 
>