Displaying 20 results from an estimated 10000 matches similar to: "Entering code to restart the machine"
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
2008 Mar 05
6
Asterisk based UNIX
Hi All;
Anyone tried to install Asterisk based on UNIX (not
linux)? Which UNIX was good to work with Asterisk?
Regards
Bilal
____________________________________________________________________________________
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2008 Jan 20
6
IAX softphone
Hi All;
I tried Firefly softphone with IAX and it gave very
poor quality.
Any one advise a good strong softphone that can work
with IAX fine?
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
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
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
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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2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears;
I do not know if any had experience in using speex or
ilbc with IAX and got good results, because I am
facing a problem with GSM.
I am facing a noise problem when I am using GSM with
IAX trunk as following:
IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk
using GSM codec ---> Remote Asterisk Box ---> Digium
Card (FXO) to terminate the call to the destination
While no
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
2007 Nov 22
6
Digium and Asterisk
Hi List;
Is Digium the best telephony cards to be used with
Asterisk? The prices are some how high, any
suggestion?
Regards
Bilal
____________________________________________________________________________________
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2011 Mar 05
3
Prepaid Billing other than A2Billing
Hi All;
Any one advise for open source prepaid billing other than A2Billing that can work with Asterisk and it is rich by features (for large business)?
Regards
Bilal
2014 Jan 28
4
Integration with outlook
Hello;
Is there a method "way" to be able to dial the phone number through asterisk from the outlook email contact?
Regards
Bilal
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2011 Jun 14
3
sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!
Dear;
Thanks a lot for guiding me.
Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch?
Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5):
libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch
It gave me that patched detected as shown below (example of one file, and I got same for other files):
patching file
2011 Jun 14
1
sig_pri.c:985 pri_find_dchan: Span 1: D-chanannel anyway!
Dears;
To patch libpri: I just place the patch file in the libpri source directory and then I run make and make install?
Or I need to compile the dahdi and asterisk also?
If the problem stayed, do I have to go for previous libpri version? Or for previous dahdi version and asterisk version?
Regards
Bilal
-----------
> bilal ghayyad wrote:
> > But I am afraid it is a bug because I
2007 Sep 09
3
nat=yes
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 from the endpoint?
Any help.
Regards
Bilal
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
2008 Mar 05
1
Voice quality is bad from one side and good from another side
Hi all;
I have two asterisk boxes installed in two separated
sites, the internet bandwidth between them is very
good and I am using G729 codec to communicate with
them and IAX.
The problem that side A hears well side B while side B
does not hear well side A !!
I did one thing in side B that in iax.conf, I set the
bandwidth=high and it helped, but still side B is
complaining from the quality
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
2013 Mar 08
11
digium card and virualbox
Hi All;
How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution?
Regards
Bilal