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 <ucoms2...@gmail.com>
> 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 <ucoms2...@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 <ucoms2...@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 <ucoms2...@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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