Busy-level is a nice idea, but is a 1.6.X feature and the question relates to 1.2.X. As for call-limit=2, that's my blind advice since I don't have a similar environment to verify the answer with.
_____ From: [email protected] [mailto:[email protected]] On Behalf Of Geraint Lee Sent: Thursday, April 16, 2009 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simultaneous Calls at a time i haven't understood any of this thread... but i'm going to throw "busy-level in sip.conf" in to the mix... i have no idea if this is a useful contribution... but i felt i should contribute something :) 2009/4/16 David @ULC <[email protected]> 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 Wink <http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm 9ydW0vaW1hZ2VzL3NtaWxlcy9pY29uX3dpbmsuZ2lm&b=2> } Also, this thing happens even when we have just 5 agents on a single server. Sad <http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm 9ydW0vaW1hZ2VzL3NtaWxlcy9pY29uX3NhZC5naWY%3D&b=2> Our version is Asterisk 1.2.27 Any Solutions ? _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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