similar to: Digital Phones

Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels

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 List; In the outbound, I read in the documents the Wildcard match "by using the . (period)", but I did not understand how Wildcard will work (like what)? As I know that Wildcard is a term used with the Diguim TDM card (FXO and FXS), so what is the relation between such cards and the matching in the dial plan? Any help? Regards Bilal

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; 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; 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; 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

Hello; Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131218/7fbbc3c8/attachment.html>

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 All; How to increase the waiting time between entering the digits for the analoge phone device that is connected to fxs? Is it by DigitTimeout? But how it will be apply for analoge station if the user just pickup the handset and dialed the number? Any help? Regards Bilal

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 List; I need to do a call transfer using analoge phone connected to fxs, but I do not need this to be done using flash hook, let it to be using the # or * or any code, but how I can configure that this code is for transfer? Also, I do not need the flast hook to be used for trasfer as it cause usually a confusion to distinguish between the hangup and the call transfer. Any advise? Regards

Hi List; What is the command and where I can write it to block specific code from calls (then no one will be able to place call for any number start by that code)? --------------- Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: + (965) 9849460 Yahoo ID: bilmar_gh at yahoo.com MSN ID: bghayad at hotmail.com

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/

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

We have a client with a Nortel PBX with digital phone sets. Due to T1 problems (old firmware), we are interested in trying a FXO channel bank. Is there a channel bank (or equivalent) which emulates Meridian digital phone sets? In order words, an FXO channel bank that's Meridian digital? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Is using a Citel gateway with Asterisk a good solution for reusing of the old Nortel digital phones? Would love to get some input from actual users. Any/all opinions welcome. robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070510/bc5fc18f/attachment.htm

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

Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected