search for: ghayyad

...w 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 lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of bilal ghayyad Sent: Wednesday, June 17, 2009 5:36 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] asterisk-gui: read/write in the conf files or db? Hi All; asterisk-gui read/write from the conf files or database? Any advise? Regards Bilal

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 updated for today's economy) at Yahoo! Games.

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

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

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

Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.

...ce 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 read something > this in the below > > This bug is referring to Zaptel, not dahdi. > > If things were working fine, and you haven't made any > recent changes, in > my experience it's always been provider (99%) or...

...------- > > http://www.linuxtutorialblog.com/post/introduction-using-diff-and-patch-tutorial > > > -----Original Message----- > > From: asterisk-users-bounces at lists.digium.com > > [mailto:asterisk-users-bounces at lists.digium.com] > On Behalf Of > > bilal ghayyad > > Sent: Tuesday, June 14, 2011 2:47 PM > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] sig_pri.c:985 > pri_find_dchan: > > Span 1: D-chanannel anyway! > > > > Dears; > > > > To patch libpri: I just place the patch fil...

Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? Regards Bilal

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

Hi All; My ISDN was working fine, and suddenly I start getting the below: sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway! There is a Yellow Alarm, so what it could be the problem? My configuration as following: system.conf span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 chan_dahi.conf context=IncomingPSTN group=0 signalling=pri_cpe switchtype=euroisdn

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 Yahoo! in your own image. Join our Network Research Panel today!

Hi All; Any one can suggest a TFTP server to be installed in Fedora (same machine that Asterisk is installed) to be used for Cisco IP Phones to download the required firmware and configuration files. Thanks for the help in advance. Regards Bilal

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 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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 access it to change it? Also where I can find information about it? Regards Bilal Ghayad

...en dialed on >> FXS ports. How does ignorepat help this guy? >> >> Al lists wrote: >>> ignorpat is your friend >>> >>> On 9/30/07, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: >>>> On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: >>>>> Dear List; >>>>> >>>>> How can I place a call via Zap/g1 (group) but need to >>>>> determine the line (FXO port) >>>>> that will go via it? >>>> Simply don't use groups. Use channels directly. To...

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

Hi All; Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? Regards Bilal

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.... and zaptel 1.4.... ? Regards ------------- ITS IP Telephony and Contact Center Engineer Eng.

Hello; Is there a method "way" to be able to dial the phone number through asterisk from the outlook email contact? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140128/17174762/attachment.html>