Let's say this is your sip.conf entry for an agent
[104] type=peer context=phones host=dynamic fromuser=104 secret=mickey canreinvite=yes directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register => 104:[email protected]/104 defaultip=192.168.23.114 mailbox=104 disallow=all allow=ulaw,alaw The call-limit=3 allows this phone to do 3 conversations simultaneously; I think the value has to be at least 2 for bridging, but the default is some int4 number like 4096 (someone in a higher pay grade feel free to provide the correct value). You are really wanting the agent to carry on one conversation and something is allowing your calls to "bleed over" into one another. _____ From: [email protected] [mailto:[email protected]] On Behalf Of David @ULC Sent: Thursday, April 16, 2009 11:52 AM To: [email protected] Subject: Re: [asterisk-users] Simultaneous Calls at a time "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 ?
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