similar to: Asterisk and gateway

Dear List, I am new in this world (Asterisk) and facing a problem . i want to make some group, base on extensions .so that certain extensions could call to certain predefine number only. let me give u all a short example extensions 1,2,3,4 will be group A , extensions 5,6 will be group B . so group A only allowed to call one or certain predefine number but group B could call anywhere.

Dear List, I am facing another strange problem . some of my envisions like to use other prepaid card (whatever they found in market) but when they dial that access number (phone number to put the pin) they get IVR (Please provide your pin number ) but when my user press pin its not going through, that IVR even can't get wrong pin number . just get disconnected as no pin number provided. what

Dear list i am using Asterisk 1.2.5 with A@H . here is my problem. if i dial a number (consider 79) i have to wait around 20 seconds before my Asteisk box response. now i want to decrease this waiting time . any idea how to do that ? thanks Salaque

Dear all, Anyone can point me how to soft hangup all channels using single command ? I am using Asterisk 1.4.15. thanks Salaque

Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ?

Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the

Hello all, I was wondering if anybody knows where one might obtain a PCI ISDN card supporting a single BRI for use with Asterisk in Australia (and using something like chan_capi). Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I'm after something like the AVM Fritz! cards. I found one place

I have a friend traveling overseas. I want to allow him to call a number which will give him a busy signal (so no charge), but will then send me an email that he has called. I know how to use a call file to trigger a call (I created a callback system for myself when I traveled overseas a few months ago), but I don't know how to trigger an email. All I want is a simple message like

Hi everybody. We have a samba4 domain deployed across serveral countries. Some of them (overseas) have a poor VPN connection with mainland. Since Samba4 does not support (yet) subtrees, we have deployed a DC in each location for domain validation. However users in mainland logon randomly at overseas location and sometimes this is a problem due to low bandwidth available. Is there any way to

I'm thinking of doing an app to work with the CID that's gotten from the Zap channel. All the CID's I see from within the US are 10 digit numbers. I'm out in the rural areas of the US, and no-one ever calls me from overseas. If they did, what would the CID look like? What does the CallerID look like overseas? How many countries provide it? murf

First, please forgive me if this is a total newbie question, I've only just begun to scratch the surface of asterisk. I currently have a dialplan set up to let me dial a specific extension, authenticate the user, then have * dial a hard coded/programmed overseas number. What I would like to do is set up my dialplan to have an extension that offers up an outbound dialtone allowing the

Hello, I would to allow users to place calls overseas such as India and Malaysia but only with a security code. if they don't have a security code I want to be able to drop the calls. can someone point me to a right direction to achieve this goal? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, We ran into a little problem recently with our phone provider (Telus Canada): we are unable to dial numbers outside North America. This is what happens: the phone number 011... is sent out over our T1, Telus sees the correct number on their switch. However the switch thinks it's a North American phone number (and thus has to have 10 digits) and rejects the overseas number as

On a BT line, how do I determine whether the number on an incoming call has been deliberately withheld (by dialling 141) or is merely unavailable (e.g. because it originated from overseas or passed through some ancient switching equipment) ? In the first case, I want the caller to be played a message to the effect that we are not at home to anonymous cowards but if their business is

Hi all. I've ordered a TDM400P with 4 FXO, but after using my X100P I'm thinking about returning the TDM400P because of bad echo issues. If I do get the echo issues I'll look at digital options. My question: Is anyone using ISDN (BRI) in the states? I've heard ISDN4LINUX devices suffer bad echo but chan_capi works great. All the chan_capi cards I find though are for overseas

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I got a call from an overseas call center telling me about the problems with the Windows machine I was using. They wanted to remote in and fix things for me ... (Ignore the fact I use a MacBook Pro or an ASUS laptop with Debian). What I found curious was the caller's name was Asterisk, and the caller's number was Asterisk at 10 or or Astrk at 10 similar. (I don't recall the exact

Hello. While trying to get to the bottom of a puppetmaster+unicorn+nginx rig that''s serving empty files, it seemed to be a relatively sensible idea to upgrade the puppetmaster to 2.6.1 This broke in a manner that seems to be the multiple-& bug. Running the puppetmaster standalone gives the following: [ ... ] info: mount[files]: allowing *.mysparedomain.com access info:

Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. 1. Does asterisk generate CDR? If yes how do I