[asterisk-users] Simultaneous Calls at a time

[Danny Nicholas]

Sip show peers will show you the ip address.  Then do sip show channels
during the call and the codec will show as format (xxx) such as format
(ulaw)

 

 

  _____  

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 2:00 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Simultaneous Calls at a time

 

 

Xlite

 

Btw, how to find out which codec a call is using when asterisk is dialing
out ?

On Thu, Apr 16, 2009 at 11:05 PM, David @ULC <ucoms2001 at gmail.com> wrote:

 

Which is the latest version of Asterisk ?

 

On Thu, Apr 16, 2009 at 11:04 PM, David @ULC <ucoms2001 at gmail.com> wrote:

busy-level ?
How to use it and whats the purpose ?

 

On Thu, Apr 16, 2009 at 10:43 PM, David @ULC <ucoms2001 at gmail.com> wrote:

 

http://threebit.net/mail-archive/asterisk-users/msg07138.html

 

Remember that if you want to support attended transfers, you need at least
two
simultaneous calls.

 

So, its safe bet to keep call-limit=2.


Advice ?

 

 

On Thu, Apr 16, 2009 at 10:37 PM, David @ULC <ucoms2001 at gmail.com> wrote:

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 <ucoms2001 at gmail.com> 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 <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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