search for: cyhammoud

I appreciate any recomendation of a simple CallingCard Application and resources of users manual. __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/

Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

Hi: The phone company here use Europe telephone system. They offere ISDN lines with multiple channels, starting with two and they add channels on demand. Am I write to understand ISDN of more like E1 or T1 line. If not, what's the difference. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

Hi: Please advice me of a voip provider with reasonable reseller program. I was using voipjet and it has a lot of problems. Did anyone experienced asteriskout.com service? They have good prices. Regards; Chawki Hammoud ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

Hi: Astcc is working fine, except for one thing. It doesn't give the called party enough time to answer the phone. If nobody picks up in two rings, astcc reports back no answer and hangs-up. The only instant NOANSWER "value" was mentioned in astcc.agi script is: elsif ($res eq "NOANSWER") { $res = &mystreamfile("astcc-noanswer");

This is my second attempt trying to get help and I am hoping someone can. When the musiconhold extension is matched, Asterisk attempts to execute musiconhold and stops right away, this is what I gets: Executing MusicOnHold("OSS/dsp", "") in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Is there a file that

Hi: I had an fxo card from Digitnetworks and it was working fine on my Asterisk box. I then replaced it with TDM04B. I changed the zaptel and zapata to include the four channels. When I run ztcfg, I get configuration errors: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04:

Hi: It's the first time I have this problem. No matter how I configure my two IAX machines the connection is rejected. "chan_iax2.c:5550 socket_read: Call rejected by ****: No authority found" iax server A: [saad_out] type=peer host=hostip username=username secret=secret disallow=all allow=gsm iax server B: [guest] type=user username=username secret=secret context=tele

...users@lists.digium.com> Message-ID: <WorldClient-F200505190937.AA37330001@purecom.com> Content-Type: text/plain Do you have mpg123 installed? Is there a .mp3 file available to play in your /var/lib/asterisk/mohmp3 directory? -daryl -----Original Message----- From: chawki hammoud <cyhammoud@yahoo.com> To: Asterisk-Users@lists.digium.com Cc: Date: Thu, 19 May 2005 06:03:55 -0700 (PDT) Subject: [Asterisk-Users] MusicOnHold probelms >> This is my second attempt trying to get help and I am >> hoping someone can. When the musiconhold extension is >> matched, Asteris...

Hi every one: I appreciate the conrtibution every one is making and please forgive me for my question. I have Asterisk running on Linux Redhat9 dstr. I subscribed to a third party sip providers to make LD calls. Can I initiate a call sessions from asterisk CLI> command prompt after I configure extensions.conf and iax.conf? In case I need a soft phone, what is the necessary configuration that

My problem used to come and goe without knowing the cause and the remedy .But now it is consistent. Each time I boot my Redhat9, I get the error message ["gnome-sound-recorder" (process #) has crashed due to fatal error (Aborted)]. After I close the error window and open the application "Sound Recorder" from the start menu, I am able to record my voice and play it back. When

I INSTALLED FEDORA CORE AND ASTERISK YESTERDAY. THEN I MADE A THIRD PARTY PC-PHONE CALL THROUGH VOIPJET AND IT WENT FINE. TODAY I TRIED TO MAKE A CALL AGAIN, PEOPLE AT THE OTHER END CAN'T HEAR ANY THING. I TESTED MY SOUND CARD AND IT'S WORKING PROPERLY. IT SEEMS MY CALL IS GETTING LOST SOMEWHERE IN ASTERISK AND ISN'T LEAVING MY SERVER. PLEASE GIVE ME ANY ADVISE OF WHAT I AM DOING

i installed Asterisk on linux FC3 box and i was able to make third party calls from voipjet. I then installed an X100P from digitnetworks and i was able to execute modprobe zaptel and modprobe wcfxo and i had to add some lines to the file: 50-udev.rules before i was able to perform ztcfg without errors. I had no problems running Asterisk before i installed X100P, but this is not the case now.

I installed X100P from DigitNetworks. The system found the Wildcard X101P and i was able to modprobe zaptel and wcfxo and ztcfg. then, after i compile asterisk, i am able to make third party voip calls dialing from the asterisk cl and using a regular headset. but when i try to dial out from zap channel, i don't have a dial tone and i hear the same click when i make voip calls. when i dial a

can any body refere me please to a callingcard application that has user's manual or some clear documentation. i have installed areskicc and i have been struggling to make it work. i want to try something else. __________________________________ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250

Hi: I read many documents and I posted my question several times here without luck. I hope someone can help now please. Here is an example to demonstarte my problem: Suppose you manage the FWD server, how do you define an IAX client behind nat so he can receive calls from FWD. NAT client would register with FWD to let it know how to locate it. I just don't see how FWD finds the nat client.

Hi: I have just downloaded Asterisk on FC3 as aprt of my search for a stable distro for Asterisk. I faced few new problems. One of them is that asterisk is unable to compile because it fails to load pbx_ael.so. It reported the following error: [pbx_ael.so]Jul 30 11:42:57 WARNING[22741]: loader.c:301 __load_resource: /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol:

Hi: I unseccessfuly tried to place a call from one IAX behind a nat through another IAX with a real IP. I got the following error message on Asterisk Console: Executing Dial("OSS/dsp", "IAX2/wassim/01)") in new stack -- Called wassim/01) Call rejected by 195.112.214.98: No such context/extensionad: -- IAX2/wassim-1 is circuit-busy Aug 8 10:13:38 NOTICE[21759]:

Hi: I was running TDM12B. Both FXS and FXO were working fine. Then all of the sudden FXS had problems. When I pick-up the phone and dial any number, FXS doesn't respond. I just keep hearing the normal signaling line tone comming from the FXS. I changed the FXS module and it had the same problem. I changed the the TDM card and installed different FXS and nothing changed. I appreciate any

Hi: I was running TDM12B. Both FXS and FXO were working fine. Then all of the sudden FXS had problems. When I pick-up the phone and dial any number, FXS doesn't respond. I just keep hearing the normal signaling line tone comming from the FXS. I changed the FXS module and it had the same problem. I changed the the TDM card and installed different FXS and nothing changed. I appreciate any