My SIP config is below : [sip64] type=peer username=fiduci fromuser=fiduci authuser=fiduci secret=pass host=64.33.22.11 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833
Now, I need to add another element as call-limit=1 and this should solve my problem ? If yes. Great. Kindly advice. But will that allow 3 party conference ? On Thu, Apr 16, 2009 at 10:22 PM, David @ULC <[email protected]> wrote: > "call-limit in sip.conf" > > Can you elaborate please and how to set that. > > Lets presume I have 10 agents and dial ratio is 4. > > > On Thu, Apr 16, 2009 at 10:06 PM, David @ULC <[email protected]> wrote: > >> >> Even I thought so thats why I tried with 4 VOIP provider and things didn't >> change. :-( >> >> >> On Thu, Apr 16, 2009 at 8:36 PM, David @ULC <[email protected]> wrote: >> >>> >>> >>> Many time we face an issue where even if an agent is on Call, another >>> call comes in. >>> >>> Sometimes, even if agent hang up the call, call stays back and another >>> come sin and then both customers can hear each other { which i think is VERY >>> dangerous [image: Wink] } >>> >>> Also, this thing happens even when we have just 5 agents on a single >>> server. [image: Sad] >>> >>> Our version is Asterisk 1.2.27 >>> >>> Any Solutions ? >>> >>> >>> >> >
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