Displaying 20 results from an estimated 26 matches for "ghayad".
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ghayyad
2007 Aug 27
4
Prepaid Billing: A2Billing, AstBill, ASTCC
...t;
I need to use an prepaid billing system with Asterisk,
and I do not know which one is more stable and
integrated with Asterisk?
A2Billing or AstBill or ASTCC?
Also, from where I can download it and ready about its
configuration?
Regards
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 009659849460
____________________________________________________________________________________
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2007 Jul 13
3
Macro: s-NOANSWER, _s-.
...when it will go for the line:
exten => s-BUSY,2,Goto(incoming,s,1)
And when it will go for the line:
exten => _s-.,1,Goto(s-NOANSWER,1)
Also, what does it mean _s-. ? It indicated for which
dialing number?
Any help?
Regards
----------
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
____________________________________________________________________________________
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2007 Aug 23
3
Asterisk Prompt
...PROMPT UNIX
environment variable"
In the following link:
http://www.voip-info.org/wiki/index.php
page=Asterisk+CLI+prompt
The question is: what is the ASTERISK_PROMPT UNIX
environment variable and where I can access it to
change it? Also where I can find information about it?
Regards
Bilal Ghayad
____________________________________________________________________________________
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2007 May 01
10
Digital Phones
Hi List;
Asterisk does not have any kind of cards that can work
with it to be used with Digital Phones (digital phones
differ than analoge phone and differ than IP Phones).
Anyone can advise about this as I did not find this on
Diguim
Regards
Bilal Ghayad
__________________________________________________
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2007 Jul 01
1
How can we block the calls for specific code
Hi List;
What is the command and where I can write it to block
specific code from calls (then no one will be able to
place call for any number start by that code)?
---------------
Regards
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: + (965) 9849460
Yahoo ID: bilmar_gh at yahoo.com
MSN ID: bghayad at hotmail.com
____________________________________________________________________________________
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2007 Jul 23
2
Upgrade and keep the configuration
Hi List;
How to upgrade the Asterisk, Zaptel and LibPri and
keep the configuration the same? I do not need to
remove current asterisk, zaptel and libpri and
download new one and write new configuration.
Regards,
--------------
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
____________________________________________________________________________________
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2007 Aug 02
1
H.323
Hi List;
Did any one tried the H.323 module? How much it is
stable and work fine?
Regards,
------------
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
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2005 Jan 28
2
IP Phone for IP PBX
Hi All;
Asterisk is IP PBX, so there should be special IP Phones to be used with
this IP PBX and this IP Phone should has special buttons working with
Asterisk IP PBX, like: Line1, Line2, Line3, ... Line8 (these to be able to
capture direct PSTN lines), also a button for Hold and transferring for
another extension ... So can some one advise me which is the recommended IP
Phone to be used with
2007 Jul 01
4
Not able to find the file zaptel.conf after compiling asterisk and zaptel
...he
file zaptel.conf under etc/asterisk, I did not find
this file. Any help?
By the way: How can I know the asterisk and zaptel
version extactly that I compiled them? In other words,
asterisk 1.4.... and zaptel 1.4.... ?
Regards
-------------
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: + (965) 9849460
Yahoo ID: bilmar_gh at yahoo.com
MSN ID: bghayad at hotmail.com
____________________________________________________________________________________
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2007 Sep 09
3
canreinvite
Hi List;
If I need traffic to be directly between the
endpoints, then I have to set the canreinvite = yes?
If I did not configure the canrenvite at all, then by
default it will pass the traffic via Asterisk and not
directly between the endpoints?
What if one endpoint was SIP and configured with
canreinvite=yes while other endpoint was IAX2 and
configured with canreinvite=yes, then they can send
2007 Aug 02
4
Receiving SIP calls without registeration and dynamic IP address
...hat if the voip gateway was
having dynamic IP and I do not need to register on
asterisk, but I need to send the call directly and
asterisk authenticate the device based on the username
and password or any possible string?
Regards,
-----------
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
____________________________________________________________________________________
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2007 Aug 27
2
Is it possible to register without sending the password
...endpoint does not
register and nothing appear on trace level 3. And the
amazing thing that if the endpoint send wrong username
(for example: bilal_sip100) then it does not register,
but we see the failed attempt of registration at
Asterisk CLI (with trace level = 3).
Please any help?
Regards
Bilal Ghayad
Mobile: 009659849460
> If secret enabled, then some endpoints can not
> register (maybe due to compatibility in reading the
> negotiation packets), so what is the solution?
I'm sure they can. Maybe you could tell the list which
endpoints don't work?
> Also in SIP registration...
2005 Jan 14
2
PC to Phone
Hi;
Can some one advise me an PC to Phone client software to be used under
Windows OS at the client side, to be communicated with Asterisk PBX?
Regards
Bilal
2007 Mar 30
1
Which IP Phones have buttons can be assigned to functions with Asterisk
...ed to specific functions):
Cisco 7970 or 7960
Polycom 501
Grandsream IP Phone Budge Tone 1001 or 1002
Linksys SPA 942 or 922
Aastra 9133 i or 480i
Anyone can advise?
I heared that polycom needs adaptor for the power as
it does not provide standard PoE, also I do not know
this.
Regards
ITS
Bilal Ghayad
Functional Consultant
Mobile: 00965 9849460
____________________________________________________________________________________
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2007 Jul 13
1
Media Proxy Mode in Asterik: SIP and H.323
...r example: in some
communications we do a proxy for media and signaling
while other communications we do a proxy for only
signaling.
Where I can determine these things in Asterisk if I am
using SIP and if I am using H.323?
Regards
--------------
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
____________________________________________________________________________________
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2007 Jul 13
1
Can Asterisk hear on two IP addresses? And can I do routing for calls from private to public or public to private IP addresses
...nty and another network)? If yes, then is there
some condition on this kind of call routing (for
example: the communication mode to be full proxy for
media and signaling or it can be a proxy only for
signaling)?
Any help?
Regards
---------------
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 0965 9849460
____________________________________________________________________________________
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2007 Jul 12
0
No subject
retreive the variable and did not write it directly
${CONSOLE} as already CONSOLE is configured in the
[global] or what is the storey :) - ?
Regards,
----------
Bilal Ghayad
____________________________________________________________________________________
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2007 Aug 03
0
CONSOLE=Console/dsp
Hi List;
In the extensions.conf file, at the [global] context,
there is a variable configured as:
CONSOLE=Console/dsp
What does it mean that? What dsp mean and it is
shortcut for what?
How can I use the core to get some data about such
thing ambiguous for me?
Regards,
----------
Bilal Ghayad
____________________________________________________________________________________
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2007 Aug 23
0
ASTCC and IVR
Hi list;
ASTCC supports IVR or there is a separate module for
IVR?
Can someone advise me a link to start download and
ready about ASTCC to do the configuration?
Regards,
---------
ITS
IP Telephony and Contact Center Engineer
Bilal Ghayad
Mobile: 00865 9849460
____________________________________________________________________________________
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2007 Oct 02
0
Selecting a specific line from Zap/g And secondary dial tone
Dear List;
Thanks alot for the help.
But how can I let the second dial tone (after pressing
the extension to select that FXO port) to be
difference than normal dial tone?
Regards
Bilal Ghayad
--------------------------
Correction, on FXO port not FXS,
second, read his email first:
"Also, how it will be possible to assign an dedicated
line (connected to FXO) to an
button on the Polycom IP Phone or Broadtel IP Phone,
so if user select that button
then he will be sure that his outside...