[asterisk-users] Simultaneous Calls at a time

[Danny Nicholas]

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:mouse_inc at mickey.com/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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David @ULC
Sent: Thursday, April 16, 2009 11:52 AM
To: asterisk-users at lists.digium.com
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 <ucoms2001 at gmail.com> 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 <ucoms2001 at gmail.com> 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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