Hi List;
Did any one see a SIP endpoint that can work without
need to do registeration on the gatekeeper side? If
this SIP endpoint existed, then I can configure the
host=static, but I am not able to find any SIP
endpoint accept to not register (all sip endpoints
request to register), but most of
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
Hi List;
If I set nat=yes, then asterisk will send the packets
to the public IP address or to the private IP address
(which will be for the endpoint that is behind the
nating)?
And by setting the nat=yes, then what exactly will be
ignored at asterisk side when reading the
registeration messages f
Hi List;
What is the difference between increasing the verbose
level and the debug level?
By increasing the verbose level, then I will get more
traces messages and by increasing the debug level, I
will also get more traces messages. So what is the
difference?
Any help?
Regards
Bilal Ghayad
Dear Ram;
You are able to send it for a file?
Regards
Bilal
> Dear Jared;
>
> I would like to ask if there is a method to let the
> output of "set sip debug ip" to be sent for a file?
hi
when iam doing this
i see the server is load is very high
how can i send this traffic or mirror traffic
Dear Guillermo;
Is there an english link that help me in configuration
other than:
http://www.ecualug.org/?q=2006/12/12/comos/configurar_a2billing_en_menos_de_10_minutos
Also, what about ASTCC?
Another issue: a2billing support prepaid billing (so
it can be used for calling cards)?
Regards,
IT
Dear Jared;
I would like to ask if there is a method to let the
output of "set sip debug ip" to be sent for a file?
Regards
Bilal
> Hello, I'm working with our SIP provider to nail
down some call
quality issues
> we're having, and they've asked me to provide SIP
debug log files
from our
> a
/index.php?option=com_content&task=view&id=73
Regards/Pagarbiai,
VoIP Billing Solutions
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of bilal
ghayyad
Sent: Monday, August 27, 2007 1:12 PM
To: asteri
Dear Ryan;
I am also facing a problem with my SIP endpoint, but I
need also to know what commands and tools you used to
do the below debug as I need to do such thing for my
cases, can u help?
Regards
Bilal
Hi,
I am trying to setup a simple home voip service w/ *
I have compiled and installed t
Dear Philipp;
Thanks for your kindly help.
The log was for the sip endpoint registeration as
following:
Really destroying SIP dialog
'[EMAIL PROTECTED]' Method:
REGISTER
And this message was coming peridically (maybe every
time the endpoint trying to register).
The endpoint was the Firefly in
Dear Philipp;
How can I add the verbose and debug to the consol
entry in the logger.conf to be able to take logging
about the attempt of registeration for the sip end
point?
Regards
Bilal
> If secret enabled, then some endpoints can not
> register (maybe due to compatibility in reading the
> neg
Dear Tzafrir;
Sorry I did not understand what do you mean by:
"Does it work with '-T' and 'use strict'?"
Do u mean the ASTPP or the prepaid billing? Where I
have to run '-T' and 'use strict'?
You do not think that I need to do download the
prepaid billing software or it come with Asterisk?
Re
Hi List;
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.
Dear Philipp;
Kindly find the part of the configuration as below:
[general]
allow=all
disallow is comment by ( ; ).
[bilal_sip]
type=friend
context=internal
host=dynamic
canreinvite=no
dtmfmode=rfc2833
So where is the problem? The endpoint does not
register and nothing appear on trace level
Hi List;
I noticed that if I disabled secret in the context by
placing ( ; ) before it, then at the asterisk the log
will be:
-- Registered SIP 'bilal_sip' at 0.0.0.0 port 5060
expired
The IP address of the endpoint was not captured!!!
Why?
If secret enabled, then some endpoints can not
registe
Hi List;
I have a problem when trying to let an SIP ATA
endpoint (got it from broadtel company), I am getting
the following message:
- Registered SIP 'bilal_sip" at 0.0.0.0 port 5060
expires 60
I do not know why it takes it 0.0.0.0 while it has an
IP address (192.168.8.3).
In the sip.conf, the
wrote:
> On Mon, 20 Aug 2007, bilal ghayyad wrote:
>
> > Dear Gordon;
> >
> > Thanks a lot for your email.
> >
> > I need one more tracing tool, how can I know the
> used
> > port of the IAX on teh Asterisk and wethor the
> > listening on that port
rt ASTERISK_PROMPT="new prompt >"
then, what you access the CLI, instead of:
hostname*CLI>
you get
new prompt >
Moj
bilal ghayyad wrote:
> Hi List;
>
> I read the following sentence:
>
> "The CLI prompt is set with the ASTERISK_PROMPT UNIX
> environmen
Hi List;
I read the following sentence:
"The CLI prompt is set with the ASTERISK_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 acce
Hi List;
I saw this is written in that link:
http://www.voip-info.org/wiki/view/Asterisk+options
And really I was not able to understand for what is
that and where I can learn about it and how to write
such thing? Can some one advise me?
!/bin/bash
asterisk Startup script for the asterisk PB
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
_
Dear Gordon;
Thanks a lot for your email.
I need one more tracing tool, how can I know the used
port of the IAX on teh Asterisk and wethor the
listening on that port is successully done (ready to
receive on that port)?
About the firewall, actually the client PC and
Asterisk on the same LAN (my P
Hi List;
I am using Firefly softphone Version 1.9.9 Build 4521
and I select IAX protocol and did the configuration in
Network1 (and I checked the Active checkbox) as
following:
Server: 192.168.8.4
username: iax2user1
password: password
In the Asterisk, I did the following configuration on
the /e
Hi List;
If I configured one SIP account or one IAX account
[sipuser1] or [iaxuser1] then how many calls can be
originate/terminate using the same account [sipuser1]
or [iaxuser1]?
In other words, can 10 IP Phones (users) do a calls
via Asterisk using the same account (SIP or IAX2)?
If yes, how
,
or
fell asleep.
Clear as mud?
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of bilal
ghayyad
Sent: Friday, August 03, 2007 8:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-
Dear James;
Thanks a lot for your kindly help and reply.
Here is the question: what is the CONSOLE variable
that is related to the channel? What it means and to
what it indicates this variable?
So when we say that Console/dsp, what does that mean?
Because console term is related to the console
Hi List;
At the extensions.conf file, at [demo] context, there
is a line:
exten => 1234,n,Macro(stdexten, 1234,
${GLOBAL(CONSOLE)})
In this line, I understand that it calls the macro
name stdexten [macro-stdexten] but about the other
variables, do we consider 1234 is ARG1 and the
${GLOBAL(CONSOL
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
Hi List;
What is the difference between WaitExten function and
TIMEOUT (response)? As I see that both are used to
determine the allowed time to enter the digits, any
one can advise?
Regards
Bilal
Shape Y
Hi Alex;
Kindly find my answers below preceeded by ( * ).
Bilal,
The purpose of registration is to establish a
contactability/reachability URI information in the
registrar dynamically.
* What is the URI?
If you have a static IP on both ends you can nail up
an IP-trusted peer session / SIP trun
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
Ready
for the
Hi List;
How can I configure asterisk to receive a call from
SIP end point without being registered at asterisk and
its IP address is dynamic, and authentication to be
based on the username and password or any other
string?
I know that if I place the host with static IP then no
need to register,
Hi List;
I know that we can use SIP/john and IAX2/jack/613 but
I do not know what are these:
Phone/phone0
Console/dsp
Any advise?
Regards
Bilal
Get the free Yahoo! toolbar and rest assured with the ad
Dear Jared;
Thanks a lot for your kindly answer.
Yes, but what does it mean:
Phone/phone0 and Consol/dsp?
Regards
Bilal
On Fri, 2007-07-27 at 06:46 -0700, bilal ghayyad
wrote:
> What the following mean:
>
> CONSOLE=Phone/phone0
> CONSOLE=Console/dsp
> TRUNK=Zap/g2
These are g
Hi List;
What the following mean:
CONSOLE=Phone/phone0
CONSOLE=Console/dsp
TRUNK=Zap/g2
I know SIP/John and Zap/1 but I do not know above (I
do not know also the difference between Zap/2 and
Zap/g2)?
Any advise?
Regards
Bilal
Hi List;
I am trying to use wiki via the link
(http://www.voip-info.org/wiki/index.php?page=Asterisk)
in effective way to find the needed resource for me,
but still it is hard to arrive for the needed
information.
For example: what is the best (shortest) way to search
for information related to t
when
the person
answered.
>Both IaxComms would crash. I'm sure there is one out
there but I
have
not
>found it, although I have not yet tried the SIP soft
phones.
>
>--On Tuesday, July 24, 2007 2:09 PM -0700 bilal
ghayyad
><[EMAIL PROTECTED]> wrote:
>
>> Hi
.
On 25/07/07, bilal ghayyad <[EMAIL PROTECTED]>
wrote:
>
> Hi All;
>
> Thanks for all replies :) -
>
> But that means, softphone in Asterisk is not that
> good, I see all complains. Any advise?
>
> Please Mr. "Time Bandit": What do u mean by "my
IAX2&
> Milos
>
> 2007/7/24, bilal ghayyad <[EMAIL PROTECTED]>:
> > Hi List;
> >
> > I need to configure a softphone to be client and
use
> > it with Asterisk, which is the r
e been able to connect
reliably and make calls.
-Ryan
bilal ghayyad wrote:
> Hi List;
>
> I need to configure a softphone to be client and use
> it with Asterisk, which is the recommended one? Is
it
> iax2?
>
> Regards
> Bilal
___
Hi List;
Is there a link that help me to configure asterisk to
register to another gatekeeper as client?
Regards
Bilal
Ready
for the edge of your seat?
Check out tonight's top picks on Yahoo! TV.
http
Hi List;
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?
Regards
Bilal
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's
Dear List;
I am trying to create a link between Asterisk and My
softswitch, the link to be SIP Trunk.
I did the below configuration and I do not know if any
one can help me and advise me to have better
configuration to be sure that link is fine. But I do
not know how to determine the SIP username
Dear Noah;
Thanks a lot. It is the sense :) -
Regards
Bilal
Ready
for the edge of your seat?
Check out tonight's top picks on Yahoo! TV.
http://tv.yahoo.com/
_
Dear Tim;
What is folks? Where I can find it about VPN solution?
Regards
Bilal
> Hi,
>
> Greetings to All,
>
> Im looking for some help on configuring VPN on the
Asterisk PBX that
I
> have hosted in US. Im currently in Middle East and
as everyone knows
> some countries here has taboo to VOIP. Im
Dear Edgar;
I am little bit confused, do u mean that asterisk does
not work in that way:
RTP (media) to be from the sournce to the destination
directly while signaling to be via asterisk?
So, what he parameter canreinvite is doing?
Regards,
ITS
Ip Telephony and Contact Center Engin
Dear Alex;
Thanks for your kindly help and answer.
The question here is: how asterisk will be able to
receive calls at two network cards where each network
card has a different IP address.
Maybe we need to know if asterisk is doing a hear on
the ports only without caring for IP or it is doing a
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:
Dears;
If I need to do an SIP Trunk between Asterisk and
another IP PBX, then no need to do registeration to
that IP PBX (it the other IP PBX support this)?
In this case, do I need to make the host an static IP
address? Or what is the method to determine that no
registeration?
>From the other si
; line, is for call parking. The others are
NOT for call
parking and are unrelated -- They are just for dialing
charlie and bob
directly.
Jared Smith wrote:
> On Sat, 2007-07-14 at 13:20 -0700, bilal ghayyad
wrote:
>> [incoming]
>> include => parkedcalls
>> exten=103,1,Dia
Hi List;
[incoming]
include => parkedcalls
exten=103,1,Dial(SIP/Bob,,tT)
exten=104,1,Dial(SIP/Charlie,,tT)
When we use tT and when we use t alone or T alone, I
know this for call parking, but I do not know what the
tT does?
Regards
Bilal
__
Dear Mojo;
Looking to the below example again, there are two
lines for s-NOANSWER and s-BUSY, one line with priorty
1 and other with priority 2, and both lines are
calling the Voicemail application, so the question is:
when it will jump to priority 2 for s-NOASNWER and
s-NOBUSY?
Last thing, like
Dear Alex;
Thanks for your kindly reply.
Please explain for me what do u mean exactly in "a la"
in the following sentence u wrote it below?
" in SIP, this can be done via
"re-INVITEs" a la the canreinvite= option for SIP
peers in sip.conf"
Another thing, do u mean that it is easier (better) if
Dear Jared;
Thanks for your kindly help.
But what do u mean by more characters? What that
pattern that will contain a character?
Also, what that pattern that will contain a dash (-)?
Regarding to the s-CONGESTION then what it means by
"CONGESTION" word? Why u used here "CONGESTION"?
Regards,
-
Hi List;
Can asterisk hear (receive) calls on two IP addresses?
How?
If yes, then:
If I have a VPN router, and my Asterisk server
connected to two network cards, one has a private IP
address (192.168.0.2) connected to the VPN router
(192.168.0.1) and another network card has a private
IP address
Hi List;
All we know that in voice, there are a type of
communications between endpoints, for 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
Hi List;
I have this example for Macro and I am not able to
understand some line, if any one can help me plz :)-
[macro-voicemail]
exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,2,Goto(incoming,s,1)
exten =>
Dear Alex;
I am asking about:
What is the configuration that I can do it to let the
traffic between the two Asterisk PBX and another IP BX
to be g729 or G711 or g723?
In other words, how can I let the ued codec for the IP
Trunk between my Asterisk and the other IP PBX to be
g729 and not g711? Of
Hi List;
How can I convert some digits to another digits, and
how I can insert in the end or in the begining some
digits, for example:
If I have a number like 11336784888, then I need to
replace each digit of value 1 by 5, how?
Also how can I add digits to the numbers like adding
00 in the begin
Dear List;
To have better security, how can I put a password on
the international calls (if the user dialed the
international call, then it will be asked for password
to send the call outside)?
Can this password read from the CDR file to know whom
did these international calls (using which passw
Hi List;
Where I determine the codec to be used for the SIP
Trunk (between Asterik and another SIP softswitch)?
Regards
Bilal
Be a better Heartthrob. Get better relationship answers from someone who kno
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: [EMAIL PROTE
Hi List;
I compiled Zaptel 1.4 and Asterisk 1.4 after
downloading them using svn, but when I checked the
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.
So if the it is only outgoing then no need for context
but if it is incoming or incmoing & outgoing then I
need context. Correct?
Regards
Bilal
> Yes, you can only send calls to peers, not receive
>them, so no context=
needed.
Moj
___
etc/modprobe.d/zaptel,
*** they have been moved to
/etc/modprobe.d/zaptel.bak.
***
*** In the future, do not edit /etc/modprobe.d/zaptel,
but
*** instead put your changes in another file
*** in the same directory so that they will not
*** be overwritten by future Zaptel updates.
***
Regards
Bilal
Hi Noah;
The reason that I am asking wether I need to determine
the context is what I read in the documentation (about
configuring outbound IAX connections), it did not
mention the context at all, please read the below
paragraph (I copy it from the documentation and paste
it):
Configuring Outboun
Hi List;
If I need to do a trunk between Asterisk and another
SIP softswitch (so Asterisk will send a SIP calls to
that softswitch), then I have to configure this on the
sip.conf file or where exactly? And is it the same
when I configure iax trunk?
Should I determine the context in this case for
Hi List;
Maybe I have to resummarize my problem with Zaptel
Compilation:
I am getting the error while I am compilaing Zaptel
when I ran the command "make linux26", although I did
the: software symbolic link, ./configure, and I
checked my kernel is 2.6.20-1.2320.fc5 which typical
for the output of
Hi List;
I think my problem in Zaptel compilation is related to
autoconf: no input file, anyone has an advise?
Also, I did a change in the Makefile existed in the
following path:
/usr/src/kernels/2.6.20-1.2319.fc5-i686/
EXTRAVERSION = 2.6.20-1.2319.fc5
Now, if I run uname -r then I get output
Hi List;
I am facing a problem relaed to menuselect when I am
trying to compile zaptel -1.4.2.1, the error as
following:
[EMAIL PROTECTED] zaptel-1.4.2.1]# make linux26
make[1]: Entering directory
`/usr/src/asterisk-1.4.4/zaptel-1.4.2.1/menuselect'
make[2]: Entering directory
`/usr/src/asterisk-1
Hi List;
I saw sip.conf and iax.conf but I do not see a files
for H.323 IP Phones, does that mean Asterisk does not
support H.323 IP Phones?
Also, what if Asterisk need to talk with another IP
PBX that support H.323, so the IP Trunk in that case
should be H.323 IP Trunk, does Asterisk support suc
Dear Cohen;
In this link:
http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html
In the subject:
2.Installation, then in the sub title: Zaptel
Installation
Please advise.
___
You snooze, you lose. Get
Hi List;
Why I need zlib1g to do installation for Zaptel? Will
zlib1g do compression or it will what extactly do
during the installation process?
Regards
Bilal
Moody friends. Drama queens. Your life? No
Hi List;
My Question was:
>From where I can download the Asterisk GUI, a lot of
replies we received but I did not receive from where I
download it and how I compile it.
Regards
Bilal
Get the Yahoo! too
Hi List;
I did yum install kernel and yum install kernel-devel,
now when I type 'uname' -a I have the following:
[EMAIL PROTECTED] /]# 'uname' -a
Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1
SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386
GNU/Linux
And when I type rpm -q kernel, then I ha
Hi List;
Where I can download Asterisk GUI and what I can have
benifit from it?
Regards
Bilal
Be a better Globetrotter. Get better travel answers from someone who knows.
Yahoo! Answers - Check it out.
Hi List;
To compile the zaptel and libpri, do I have to have an
diguim card (hardware) fixed in the server?
Also, is there any problem if I compiled first
asterisk and then I tried to compile zaptel and
libpri?
Regards
Bilal
___
Hi List;
To compile the zaptel and libpri, do I have to have an
diguim card (hardware) fixed in the server?
Also, is there any problem if I compiled first
asterisk and then I tried to compile zaptel and
libpri?
Regards
Bilal
_
Hi list;
I am looking for an application that can be used with
call center, in this application we can integrate the
telephony part of the call center (like CTI Client ad
so on), any one can advise for a good application to
be used with Asterisk Call Center?
- Note: The application to be customiz
Hi List;
As I know from AVAYA (I am AVAYA certified) that
digital phones are connected to digital cards and it
does not go through ethernet switches at all, digital
phones should be independent on the ethernet network,
so if the network down, these phones will start
working, it will be totally iso
Hi;
Well, I understood now that Nortel has some digital
phones that can be used with astrisk, but the
question: what are the card models that should be
installed on Asterisk server? Digium? What these
models?
Regards
Bilal Ghayad
___
Dear Collins;
But what the cards that I can use it for these digital
phones (if available)?
Regards
Bilal
__
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Dear Andres;
How much it cost the 4 licenses of G729 and from where
I have to buy them?
Also, what if I need to do IP Trunk between Asterisk
and another IP PBX in another side (in case I need 30
ports for this IP Trunk, and I need to use G729 or
G723 codec), then also I need to buy a license for
Hi List;
When I need TC400B?
If I have a solution of 4 users (IP Phones) and 4
analoge lines, then I need TC400B?
Regards
Bilal
__
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_
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
___
Hi Noah;
ut TDM11B contains physically 4 ports, if it supports
only 1 FXS and 1 FXO, then what shall we do in the
other two ports already existed?
Regards
Bilal
__
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htt
Hi List;
Can someone advise me if Polycom support H323 that
work fine with Asterisk? And wether this H323 Polcyom
devices more costly than SIP Polycom.
Also, I am not able to know if new Polycom come with
PoE adaptor so no need for PoE Switch (can use normal
switch that does not support PoE)? Do
Hi List;
What is the difference between TDM11B and TDM04B? Why
the price of TDM11B cheaper than the price of TDM04B?
Regards
BIlal
__
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___
Hi List;
For TDM24 Cards, what it means that it support 6 FXS
and/or FXO modules for a total of 24 lines? Does it
means that this card can be divided to 6 modules (FXO
or FXS) where each module will support 4 ports?
Also, when it syas in the characteristics that:
Zero (0) FXS modules (green)
Si
Hi List;
Can someone advise me which IP Phone model that has
buttons that can be assigned to do specific
functionalities (call pickup, call formward, call
appearance) and a transfer button and hold button?
Which is the best of the following (that has buttons
can be assigned to specific functions)
Hi List;
I understand that I have to compile zaptel but what about asterisk? Is it
enough to extract it? Well, how I will run asterisk (without compilation and
installation)?
Any advise?
Regards
Bilal
Yah
Hi List;
To create the symbolic link, I read in the documenation that I have to type
this command:
# ln -s /usr/src/'uname -r' /usr/src/linux-2.4
1) What it means by 'uname -r'?
2) Why I have to create such symbolic link to do pointing for the kernel? For
what exctly will be used with asterisk
Hi List;
I am looking to use an good IP Phone working with
Asterisk and work based on PoE (so it takes the power
via the ethernet cable, no need to connect for it
separated power adaptor).
Can someone advise me for good one?
Regards
Bilal
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Hi List;
We need to apply Video conference, can asterisk
support this? What I need for that?
Regards
Bilal Ghayad
IP Telephony Engineer
Mobile: 00965 9849460
Office: 00965 2623174
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Hi List;
I would like to know where I can find the IP Phones
that can be used with Asterisk? Is there any link?
Regards
Bilal Ghayad
Mobile: 00965 9849460
Office: 00965 2623174
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Hi list;
Does asterisk work with fedora because redhat
enterprise is licensed and costly.
Regards
Bilal
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http://mail.yahoo.com
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Hi List;
Can someone advise me what is the email of the
administrator forum so I can send for him to fix my
account?
The forum that I am talking about it existed in the
following link:
http://forums.digium.com/
Regards
Bilal Ghayad
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