to test any VoIP
platform.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S //
On 8/10/2010 11:33 AM, kamrun nahar bina wrote:
Dear Faisal Hanif,
Thanks for your reply.
What is the purpose of using SER ?
What is the purpose of using SIPp -I know little bit about this.
But I know nothing
Hi,
We are using 4-PRI card from http://atcom.cn for our development LAB and
we are satisfied with performance. It is also cheaper then other
products. They also have analog.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S
On 8/10/2010 6:40 PM, Jeremy Betts wrote:
I have always had very
read the value of var ${HANGUPCAUSE} next line to dial command.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S
On 8/10/2010 9:51 PM, bruce bruce wrote:
Hi Everyone
Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to
Bell Canada.
User claims that call hangup without
If Caller party hangups next to dial line wil not be executed but
control will hit to h extension of fame context but if Called party
hangups next to dial ine will be executed.
Faisal Hanif
On 8/11/2010 10:16 AM, Philipp von Klitzing wrote:
read the value of var ${HANGUPCAUSE} next line
Hi,
It is simple to use max_limit perameter in dial command.
Regards,
Faisal Hanif
On 8/9/2010 2:01 PM, Catalin S. wrote:
Hello,
I wish to make a simple system to limit peers at x minutes depending
of buyer voip packet. Can someone help me with some directions?
I intend to make a separate
lback file to sent alert
call.
Signatures fai...@vopium.com
Regards,
Faisal
Hanif
VoIP Manager
n 8/9/2010 11:56 PM, Felipe Figueiredo wrote:
Hi guys,
is t
Hi,
SER is a most powerful SIP router but a SIPp is a VoIP load generation
software. So both are totally different and can not be used interchangably.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S//
On 8/10/2010 10:44 AM, kamrun nahar bina wrote:
Dear all,
What is the difference
We are having good results with
maxexp 120
minexp 90
defexp 100
qualify = yes
qualify = 500
qualifyfreq=5
registerattempts = 0
registertimeout = 10
maxexpiry = 60
minexpiry = 20
defaultexpiry = 600
---///---
Can someone more experienced with these settings to help me to
optimize connections
You need to do it by manager interface
Regards,
Faisal Hanif
On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1
We are not using qualify for the peers which are not on static IP and
registering to server.
Regards,
Faisal Hanif
//
On 7/26/2010 5:06 PM, Catalin S. wrote:
did you also hav qualify and qualifyfreq?
Thank you for reply,
On Mon, Jul 26, 2010 at 1:55 PM, Faisal Haniffai...@vopium.com
there's a variable
capturing who answered.
Zarko
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Faisal
Hanif
*Sent:* Monday, July 26, 2010 12:57 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] URgent
use cacti
Regards,
Faisal Hanif
/Think about the environment before printing this mail /P/ Tænk på
miljøet før du printer denne mail/
On 7/26/2010 5:15 PM, Tony LaMear wrote:
I need graph the utilization of my t1s. Does anyone know of a plug-in,
code, or web interface I can use to help
you can use all asterisk dial-plan functions and application in lua
plus additional complete lua features. so answer is yes.
Regards,
Faisal Hanif
On 7/26/2010 5:34 PM, Bruce McAlister wrote:
Hi All,
I have a quick question with regards the pbx_lua module.
Would the lua dialplan have
You need to create a function is res_odbc for each of required query
and then u can use that function as normal asterisk dialplan function.
Regards,
Faisal Hanif
On 7/26/2010 7:02 PM, Bruce McAlister wrote:
Thanks for the quick response, however, how would I access an odbc dsn
from
You may need to add r as option perameter to dial command.
Regards,
Faisal Hanif
On 7/26/2010 9:39 PM, Chris Ramirez wrote:
The problem we are having with Asterisk is when we initiate a call via
a Zap line and it goes out on a Sip line. When it goes out via Sip we
hear no sound until
using YUM.
Download and Compile latest release of asterisk 1.6.2.
Try to start start asterisk in console mode.
It will crash on LUA and will give a core dump
Did any one got it solved? If yes how?
Regards,
Faisal Hanif
,
Faisal
Hanif
VoIP Manager
m
+45 72 72 00 01
m
+92 32 1405 9996
Vopium A/S
| Office
Do some R D with asterisk function AMD (Answering Machine Detection)
if that can help you.
Regards,
Faisal Hanif
On 7/9/2010 11:24 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
and
asterisk will use configuration returned by that external application
and will treat it same as in static file. Here you again have full power
of programming language in you hand.
Regards,
Faisal Hanif
On 7/7/2010 1:08 PM, Hans Witvliet wrote:
On Wed, 2010-07-07 at 12:12 +0600, ABBAS
Hi,
I am in process of merging all my AGIs+Dialplan to a single LUA
dialplan. It seems much interesting to me spacial LUA tables which allow
me to support a complete object like programming. Yet I did not
completed / tested.
Regards,
*Faisal Hanif*On 7/1/2010 2:37 PM, Gilles wrote
Hi,
If you use curl realtime for registrations you can add useragnet check
in your CGI and also lot of else as well.
Regards,
*Faisal Hanif
*On 6/29/2010 4:48 PM, Tarek Sawah wrote:
well there are two restrictions.. the IP address of the station they
are using it .. and the UserAgent..
one
Simply set it to costume field of cdrs in dialplan and you will have
it a part of native cdr
Regards,
*Faisal Hanif*
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,
Faisal Hanif
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asterisk-users mailing list
Hi,
I am also wonder that same SRV record is working fine on one machine but not
on 2nd while both have same asterisk version.
It may be some missing OS utilities which asterisk using to resolve SRV?
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun
You need to copy or soft link a2billing.conf to /etc/ folder as by default
latest version search for it in /etc/
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent
Try setting insecure=port,invite in sip peer config.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Tuesday, June 15, 2010 9:14 PM
To: Asterisk Users Mailing List
Till now I am not able to find any difference between both machines.
Can you please tell me how I can try to resolve it on OS level using some
utility like dig?
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Both have CentOS 5.2.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Tuesday, June 15, 2010 11:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
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