[asterisk-users] Simultaneous Calls at a time

[David @ULC]

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 [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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