Displaying 20 results from an estimated 4000 matches similar to: "Asterisk based UNIX"
2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All;
I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile:
Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server?
Regards
Bilal
2009 May 26
8
Bandwidth management and ADSL router
Hi All;
I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX.
Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting?
Regards
2007 Jul 03
2
Putting a password on the international call
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 password?
As I might have mutliple passwords).
Regards
Bilal
2007 Sep 25
5
Do I need to run #modprobe zaptel for Digium
Hi List;
If I am configuring Diguim Analoge card, then I need
to run #modprobe wctdm, but the question why I need to
run #modprobe zaptel also?
What #modprobe zaptel does a things that #modprobe
wctdm does not do?
Any help?
Regards
Bilal
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2007 Jul 03
5
Determining the used codec for the IP Trunk (SIP Trunk)
Hi List;
Where I determine the codec to be used for the SIP
Trunk (between Asterik and another SIP softswitch)?
Regards
Bilal
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2011 May 05
4
SIP secruity: username and password
Hi All;
When the endpoint register on Asterisk or initiate a call, so they exchange the sip username and password. What is the possibility that this will be capture by the hacker and how to avoid this problem?
Regards
Bilal
2008 Feb 27
2
Entering code to restart the machine
Hi All;
How can I configure Asterisk in that way:
If I entered code (from my mobile when I call to the
Asterisk or from any Internal Phone), then the machine
do restart. I need this when I am far from the office
and I need to restart the machine and I do not have
Internet connection.
Any help?
Regards
Bilal
2009 Jun 11
2
In Dahdi: what we use instead of /sbin/ztcfg -vv
Hi All;
In dahdi: what we use instead of ztcfg -vv (that is existed /sbin/ztcfg -vv).
?
Regards
Bilal
2007 Jul 27
4
Asterisk Wiki
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 the command playbak()?
Using the backlines, it make the eyes feel hard by
keep reading without
2011 May 16
2
Reporting Tool: To show who is login, queue, ... etc
Hi All;
It look like there are some free (open source) tools that are used for Asterisk reporting special for call center (to see number of agents logged in, number of calls now, .. etc), and to be used as dashboard.
Can someone direct me for something really is suitable and stable?
Regards
Bilal
2007 Aug 02
4
Receiving SIP calls without registeration and dynamic IP address
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, but what if the voip gateway was
having dynamic IP and I do not need to register on
asterisk, but I
2012 Apr 04
2
Asterisk 1.8 and DeadAGI
Dears;
In asterisk 1.8, it is not more possible to use DeadAGI?
Also, I found the below commands in the a2billing and I would to ask why it set the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? How?
[a2billing-callingcard]
exten => _X.,1,NoOp(A2Billing Start)
exten => _X.,n,Answer()
exten => _X.,n,Wait(2)
exten => _X.,n,DeadAgi(a2billing.php,1)
exten =>
2007 Oct 19
1
Using register => to let Asterisk register to another softswitch via SIP
Hi All;
Alot of softswitches or PBX's does not accept to
manipulate any SIP call without being registered
firstly. So that means, I need asterisk to register
firstly then I can route my calls to that SIP trunk.
In IAX2, we use the register => , so what shall we do
in Asterisk? And how its format will be (if we will
use register)? Or what is the solution?
Regards
Bilal
2007 Oct 19
1
SIP to H323 translator
Hi All;
If I installed H.323 on asterisk, and the caller phone
was SIP endpoint while I need to route the call for a
destination via an H.323 trunk, so Asterisk will do
that SIP to H.323 translation automatically or I have
to do also a configuration to SIP to H.323
translation?
Regards
Bilal
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2007 Jul 13
1
Media Proxy Mode in Asterik: SIP and
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
we need H.323 endpoint to talk with SIP endpoint then
we use full
2007 Jul 24
10
What is the best softphone work with Asterisk
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
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2009 Jun 18
3
asterisk-gui: read/write in the conf files or db
Hi Danny;
Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager?
Regards
Bilal
-------------------------
It depends on how you are configured. The gui interfaces using Asterisk
Manager, so you get the Same IO from the gui that you would get from a
native manager session.
-----Original Message-----
From: asterisk-users-bounces at
2007 Oct 23
2
register => to let Asterisk register to another softswitch via SIP
Dear Alex;
Thanks alot for your nice help.
This is if I need to let Asterisk register with
another softswitch (so I used register =>), what if I
need asterisk to send call for the softswitch without
register to it (directly)? If I removed the register
=> then how it will distiguish the IP address in the
"host" at the [sip_trunk] is the IP address of the
softswitch that need to
2008 Dec 21
6
Asterisk and Dabatase
Hi All;
Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)?
Any advise?
Regards
Bilal
2009 Jul 05
1
SIP IP-Trunk to be authenticated based on username and password, not IP address
Hi List;
How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed