Please also have a look at the gateway boxes from berofix
(http://wiki.beronet.com/index.php/Main_Page). I am not affiliated but
have used different products from them over last few yeas and all have
survived and are stable.
Documentation is open and free on their wiki. They provide updates. They
On Monday 17 June 2013, vortex wrote:
That seems suitable.
for the cid name i am using asterisk's internal database for the lookup
such as
*database put cidname 222333 xyzwhoevername*
then how do you create the database for the check_ban_db?
However you like! I'd use a MySQL
On 17 June 2013 11:02, Shanavaz E A shanava...@yahoo.com wrote:
Hi,
I have a requirement, which I am not sure whether it can be implemented. I
had done some searches but didnt find an answer to this. Kindly let me know
if some one has an idea to implement this:
I am not aware of an
You should have different sets of agents logged in to different queues and
you should have a monitor to move them from one queue to the other based on
incoming traffic.
l.
2013/6/17 Shanavaz E A shanava...@yahoo.com
Hi,
I have a requirement, which I am not sure whether it can be implemented.
Hello Shanavaz.,
Please find some quick thoughts:
* 2 main queues
* agents logged on one or on both main queues
* before sending a new call to one of the main queues check the number of
waiting callers (QUEUE_WAITING_COUNT function) and divert (for example for
30 sec) the call on a empty members
My experience was good, Nicolas was very helpful and quick
Regards,
Zohair Raza
On Tue, Jun 18, 2013 at 4:26 AM, Carlos Alvarez car...@televolve.comwrote:
No vacation notice, nothing, other than the system auto-replying saying
that the ticket will be closed because we didn't have any action
I think political calling is less restrictive than you think! This is three
years old, but probably mostly applicable.
http://www.ilga.gov/commission/lru/Feb2010FirstRdg.pdf
--Don
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
I have a setup where there are occasional problems with attended transfers. I have already
checked the devices as well as the relevant DTMF modes (SIP INFO and rfc2833). I could not find
any problems here.
The setup is a follows:
The front desk (F) accepts calls from customers (C). In some
Greetings.
I've plugged 3 analog lines on an ethernet cable in an Khomp card to
receive it's incoming calls. Without any configuration, when I call those
numbers the asterisk server automatically answer the call and play the
default music.
The problem is: I need to discern the lines and redirect
Hi,
I am new to Asterisk. I'm using it behind a kamailio sip-router to provide
voicemail boxes to sip-users.
I followed these instruction:
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x
to set everything up, using ARA with a MySQL DB.
After a few
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