similar to: My Kernel

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: 2.6.20-1.2319.fc5 But the directory under the kernels is: 2.6.20-1.2319.fc5-i686 So do I have to

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 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 ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

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: [root at localhost 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-1.4.4/zaptel-1.4.2.1/menuselect' make[3]: Entering directory

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 to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

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 knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

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 customized easy, to be able to use it with Banking, Telecom, Oil, .. etc. Regards Bilal

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 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

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.

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 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

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 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 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; 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

I tried that approach a while ago and failed miserably. Marshalling the structs of structs in the flac lib turned out to be a nightmare (I don't pretend to be an expert, mind you...). I eventually switched to writing my own C# lib from scratch. Work is still under progress. It's no rocket science, I do this a G-job. It has definite limitations (no documentation, decodes only 16-bit files,

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!