Hello All,
I have a request from a prospectieve client to deploy a PBX capacity
that can do up to 3000+ lines within a geographic region similar to a
campus. The client wants analog lines for extensions and maybe VoIP
for some backhaul traffic while the other traffic would be carrid via
E1 channel
Could you explain further?
On Thu, Jul 24, 2008 at 4:13 AM, Gregory Malsack <[EMAIL PROTECTED]> wrote:
> I resolved this problem. The key was to get the right combination of
> self/callee and peer/caller. Read the instructions regarding the application
> map very closely. My problem was that
Why not get a TDMoE multiplexer, check out http://spidermux.com/
On Sat, Mar 8, 2008 at 4:03 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, Mar 08, 2008 at 07:41:50PM +0500, Faraz Khan wrote:
> >
> > > Just think of a different alternative: If you consider the cost of a
> > > 24/32/48
hi,
I had a similar problem with FreePBX last year, follow this mail
thread, it might help solve ur problem
http://lists.digium.com/pipermail/asterisk-users/2006-October/169568.html
On 12/13/07, Paul Hales <[EMAIL PROTECTED]> wrote:
>
> Using the 'read' function you should be able to do somethin
Hi All,
I need help with CDR issues but first let me describe the problem.
My office has 2 Asterisk PBX the first pbx is termed as the gateway
PBX (because it carries the TE card and thus Telco E1s) since all
calls are routed via this PBX. the second pbx is know as the office
pbx. this pbx contro
Hello Don,
thanks for the helpful pointers, i'll push my quotes on these and
hopefully they will be accepeted.
The only drawback on this is the fact that i would have to use an ATA
to complete the loop. This will rais the unit cost of the deployment.
I was thinking of usin SOEKRIS installed with
Hi All,
Does anyone know of a working SNMPC implementation for asterisk 1.2
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All,
I recently got a request to deploy VOIP helplines with a bank's ATM
site locations. The purpose of this solution was to be able to resolve
cash dispensing errors as quickly as possible.
My problem is that i do not have any idea about outdoor type VOIP
equipments. Does anyone have ideas???
__
It is true what Eric and Steve have said, you do need a licensed GSM
frequency to operate and sell GSM services (even for rural areas).
however, this link might be of interest to you
http://rfdesign.com/mag/radio_field_trials_allsoftware/
On 1/10/07, Eric ManxPower Wieling <[EMAIL PROTECTED]
Hi,
in addition to my previous post about the OSP support on Asterisk,
does anyone know if there existst OSP peering VOIP hosts who are
willing to connect to simple users like me using OSP protocol
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hello everyone,
i have been researching into transnexus (http://www.transnexus.com/)
OSP (open settlement protocol) server. i am really interested in its
routing flextbility and call clearing capabilities. Has anyone
implemented OSP with Asterisk or Cisco voice devices. I would like to
have produc
i had the same problem. the GUI stopped responding to configuration changes.
On 11/28/06, James Willing <[EMAIL PROTECTED]> wrote:
"Geoff Karl" <[EMAIL PROTECTED]> wrote:
> I just downloaded and installed the AsteriskNow appliance
> (http://www.asterisknow.org) . This looks like it has lots
hi,
i have tried to use asterisk now but it seems to me that it doesnt retain
configurations. have you experinced the same thing ??
On 11/26/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:
Hello,
Anyone saw asterisknow, ?
Regards
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Hi,
Has anyone installed Asterisk on FreeBSD? i need help/steps on this task
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i also used to have this problem, for instance we use the pin code
functionality of FreePBX and whenever i add or modify a pin number, it is
not effected or changed in the config files. i dont know what causes this
error but i have noticed that restarting FreePBX or re-installing the
application s
) {return '1';} } return '0';
}
you have to install the Asterisk::AGI module for thi script to work. i used the authentication macros from AMP (FreePBX) to pug in this agi.
Every trunk thatrequiers authentication calls this macros. The macros is as follows..
[macro-pi
HI,
I am trying to write an AGI that will authenticate users befoer the call is allowed to proceed. Befor you ask, i have tried using the authenticate() function but it does not work for me as this function messes up call accounting (authenticat() awayas awnseres the channel, thus causes CDR to bil
other thread
Thanks for your help
On 10/17/06, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:
this Cdr Record if from the Primary PBX
'2006-10-17 07:11:37', 'Admin', 'XXX, 'aa', 'from-internal', 'IAX2/[EMAIL PROTECTED]'
,
Plesae, doesthe 'authenticate' application awnser a channel when requesting for password ?
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#x27;, '', ''
this is the CDR record from the secondsry for the same call
'2006-10-17 13:31:57', '"Admin" ', 'X', 'aa', 'from-internal', 'SIP/401-8f0c', 'IAX2/TRUNK1-2', 'Dial'
Well I am using APM on the two boxes i have modified the srripts extensievely and i am sure that there is no Awnser befor a dial when Dialing through the PBX trunks
On 10/17/06, Steve Davies <[EMAIL PROTECTED]> wrote:
On 10/17/06, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:
>
.
Thanks
On 10/17/06, yusuf <[EMAIL PROTECTED]> wrote:
Dumpolid Exeplish wrote:> Hello,> i have call time irregularites in my asterisk CDR. I a currently using a
> mysqly backent to save CDR records and use this to generate bills at the> end of each month. However, my users are complain
Hello,
i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls (
i.e. calls they make whaich have already
this is nice, how much does the Card Cost?
On 10/13/06, matteo brancaleoni <[EMAIL PROTECTED]> wrote:
Hi again,On Thu, 2006-10-12 at 12:22 +0200, matteo brancaleoni wrote:> Hi all,>> for all those using asterisk + voismart gsm cards,
> we have released a new package that fixes a lot of issue>
How can I download the Sourcecode? went to the site but there wasnt a single .zip or tar file
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Hi Everyone,
Currently in my country, there is no toll free service provider. My company has been thinking of starting such a service (using Asterisk as a soft switch) but really we dont know how to go about this. Can anyone assist us with information/documentations, etc
Thanks
Hi list,
i am currently having problems with CDR accuracy on my asterisk PBX
MY SETUP
=
I have two Asterisk systems, the first one (tagged primary pbx) has E1 lines connected to it and this processes calls on behalf of the secondary pbx. Now the CDR on the primary PBX are very accurate bu
hi list,
i have tried to set the call waiting function using freePBX but it dosent work. i think there is something wrong with the coding. Has anyone experienced this sort of problems?
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Asterisk-U
nstall unixODBC and configure it then use the> cdr_odbc.conf file to specify. Check in your source files or svn
> checkout for a docs folder and the cdt.txt file.>>> On 5/16/06, *Dumpolid Exeplish* < [EMAIL PROTECTED]>
[EMAIL PROTECTED]>> wrote:>
ave it working for CDR logs, I had to install unixODBC and configure it then use the cdr_odbc.conf file to specify. Check in your source files or svn checkout for a docs folder and the
cdt.txt file.On 5/16/06, Dumpolid Exeplish <
[EMAIL PROTECTED]> wrote:
Hi All,How can i use Microsoft SQL
Hi All,How can i use Microsoft SQL server with asterisk, Can the unix ODBC diriver interface MSQL?? and what module would i be using to access ODBC from asterisk??
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To
ax: 330.882.0455 x25
[EMAIL PROTECTED]
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Dumpolid Exeplish
Sent: Monday, May 15, 2006 6:44 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] queue
help
hi all,
i am having
hi all, i am having problems with my queue. I just found out that my queue is unable to hold simultaneous calls. What i mean is that when a call comes into the queue and it hasent been answered, and a second call come into the queue, the second call is dropped while the first is left ringing. if h
thanks for your help, I really appreciate itOn 4/25/06, Kevin Smith <[EMAIL PROTECTED]> wrote:
Yes there is. QUEUE_MEMBER_LIST()This should return you a list of comman-separated list of the members in
a queue. After that you would need to format it (if needed) so asteriskcan read it back to you. Of
Thanks Kevin,the tip worked like a charm. However, there are newer issues now! Is there any way of knowing which users are looed in? sometimes, customer support users forget to login B4 they shutdown their computers (we use soft phones) and "presistentmembers=yes" is set in
queues.conf so the use
Hi everyoneI am having problems with my call queueWe currently run a customer care call center which has attendants login during the daytime. Customers who call the 'customer care line (a specific number) always get routed to the cutomer care queue (called 124). After hours, staffs of the Network o
e very interested in that!Alex On Mar Abr 18 15:50 , 'Dumpolid Exeplish' sent:
Hi all,
i have been having problems with voice quality. We run asterisk Asterisk CVS-HEAD version 2.4 as a production servre as a call centre/customer support engine. Pressenty, we have about 25 soft phones an
Hi all,
i have been having problems with voice quality. We run asterisk Asterisk CVS-HEAD version 2.4 as a production servre as a call centre/customer support engine. Pressenty, we have about 25 soft phones and 10 hard phones (Perfect Tone SIP phones). when handling internal calls, i usually noti
I am currently using * with MySql server version 3.23.58 by using the sql driver compiled in the asterisk-addon tar file. Whenever i try to update to a later version of MySql, the channel driver brakes and the client is unable to connect to the newer verson of MySql. I have also tried updating th
So how do I enable a High mem Kernel? Do i have to recomplile the kernel to use highmem ??On 3/9/06, Joseph Tanner <
[EMAIL PROTECTED]> wrote:The answer's just below the part you bolded. "Use a HIGHMEM enabled kernel."
Joseph TannerOn 3/8/06, Dumpolid Exeplish <[EMAIL P
Hello,This is not a question directly related to asterisk.I am currently rinning ansterisk on a Debian server and i just upgraded my memory from 1GB to 2GB. However, my linux OS does not recognise the memory upgrade. The BIOS does, but the Debian Linux refuses to use the entier memory, currently,
i am taking overr the administration of an existing production * PBX but i cant seem to find out which version of * this is. When i use the 'show version' coomandat the cli, i get this:Asterisk CVS-HEAD built by [EMAIL PROTECTED]
on a i686 running Linux on 2005-08-10 05:57:59 UTCcan anyone tell me
Hi everyone,
i am face with an asterisk use management interface, at the pressent, i am using AMP (asterisk Management Portal:
http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 ). Does anyone know a better and more documented management interface for * ?
Thanks
Hello all,
this is not really an * question but it is somehow related, i am trying to develop a working proposal for cheap and quick telephony services using Voip running over *. By running a wireless network (over 802.11 a/b/g devices), i plan to be able to reach customers directly with eithe tab
I kind of like the idea of 2n's stargate but when i read the
manual (the one available for download), there were a lot of
complicated issues in configuring the device, (i mean, you have to like
set jumbers on the m/board,etc) and there was a clause that said that
callc could only be routed form the
well, does this gateway support SIP?? and does it generate its own CDR? could you send the devices brocure/tech spec.??
thanks
On 2/19/06, Sam Tam <[EMAIL PROTECTED]> wrote:
Why not get 30 GSM Gateway from us at £60 each and then get an asterisk or some voip gateway like A800 and then link
Hi everyone,
Can anyone give me suggestions on any equipment that can connect from VOIP to a GSM gateway (channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device.
hi everyone,
i am trying to configure an * server to route calls from the PSTN to our internal PBX
This is the IDEA
Currently we have a PANASONIC KT1232 PBX that provides intercom calls and facility for call out lines on it (8 call out lines are pressent on it)
we also currently have 4
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