similar to: How can we block the calls for specific code

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.

Hi List; How to upgrade the Asterisk, Zaptel and LibPri and keep the configuration the same? I do not need to remove current asterisk, zaptel and libpri and download new one and write new configuration. Regards, -------------- ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460

Hi List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

Hi List; Asterisk does not have any kind of cards that can work with it to be used with Digital Phones (digital phones differ than analoge phone and differ than IP Phones). Anyone can advise about this as I did not find this on Diguim Regards Bilal Ghayad __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

Hi List; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ------------ ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/

Hello; I am installing asterisk and dahdi on ubuntu "and I used my username bghayad to login for ubuntu and do the installation, actually I feel my problem is related to the?username and permission but I am not able how to fix it", I am facing now mainly the following two problems: The first one, asterisk is not starting automatically although I did sudo make config (for asterisk and

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

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

Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

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; Can some one advise me an PC to Phone client software to be used under Windows OS at the client side, to be communicated with Asterisk PBX? Regards Bilal

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

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

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

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

Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination

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

Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW?

Hi List; Can someone advise me which IP Phone model that has buttons that can be assigned to do specific functionalities (call pickup, call formward, call appearance) and a transfer button and hold button? Which is the best of the following (that has buttons can be assigned to specific functions): Cisco 7970 or 7960 Polycom 501 Grandsream IP Phone Budge Tone 1001 or 1002 Linksys SPA 942 or 922