search for: extensions_customer

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

Greetings *`s, I am having what appears to be a small problem, but the frustration is erally getting to me, what am I doing wrong here ? I used AMP to set up a custom menu, so if caller presses 1 it goes to ext200, if caller presses 2 it goes to ext201 etc etc... Now I have created a third option that when the caller presses 3 it must play a sound and hang up. No rocket science yet. When

Hello Everyone, I am making a simple index.php file which will allow a web user to enter his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged. Following is the index.php and the contents of extensions_custom.conf. When I submit the form nothing happens. I don't even see Manager Connected msg. Your input will be much appreciated. I am thinking I have some syntax

Hi Daniel, Am 27.03.20 um 09:24 schrieb Administrator: > Hangup is h extension. your macro will never be executed. Solution: > > same = n,Dial(whatever) > same = n,[...]) > same = n,Hangup > > exten  = h,1,1,DumpChan() >  same = n,System(/home/asterisk/bash_test) I don't really understand your code… I think I don't have to edit the first part of the conf file

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

Hi, I am trying to change the dialplan to enable call recording (incoming and outgoing calls) on the "click of a button". Is it possible? All the documentation I found so far, enable recording for 'all calls' to an extension. Does this code look ok? Currently Recording "on" only for 1030 when user presses *44, start recording. *55 to stop recording

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running A@H 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on

I'm lerning to make my custom configurations. In extensions.conf, there is #include extensions_custom.conf [from-trunk] ; just an alias since VoIP shouldn't be called PSTN include => from-pstn [from-pstn] include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,

hi, i have a pronlem connecting call from pstn with the following confuguration, please advice extensions.conf [from-trunk] include => from-pstn [from-pstn] include => from-pstn-custom include => ext-did include => from-pstn-timecheck exten => fax,1,Goto(ext-fax,in_fax,1) extensions_custom.conf [from-pstn-custom] exten => s,1,Answer exten =>

I have this code which was taken from the phpagi project page along with the following in extensions_conf and the output from the asterisk CLI. When I call the 311 extension, I does nothing then hangs up. What am I doing wrong?? ----php code------------ #!/usr/local/bin/php -q <?php set_time_limit(30); require('phpagi.php'); $agi = new AGI(); $agi->answer(); $cid =

Hello, I am trying to make the following... Can someone tell me if it is possible? Is someone willing to do it from an asterisk@home box? 1. I send an http request to asterisk@home box. Ex: http://asterisk@home/call.php?phone=0033102030405&code=12345 2. Application will call phone number 0033102030405 (using a sip provider) 3. Application will play a pre-recorded voice prompt 4. Application

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

I'm looking for some help from any Asterisk "heavy" who might be doing something similar to what I'm trying to do... Background: I work for a research lab, testing telephony products and tools. Historically, we used Ameritec Crescendos and Fortissimos to act as load generators and call "sinks" when testing equipment. However, the equipment we are testing gets

Hi Friends, I am facing a strange problem with DISA. I have installed and configured Trixbox. I've created a secret extension i.e., 555 and called this extension in Digital Receptionist using custom extension i.e., created in extensions_custom.conf file. When I call from my mobile phone to my PSTN number, which is connected to FXO port, my IVR is responding. After entering my DISA

Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/7e1e68d1/attachment.htm

I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Basically, I can build the system but an looking for a card that will allow for upto 20 extensions to be wired into the back of the PC. Doeas anyone know of a solution to this Sean-- ICQ: 679813 FidoNet: 2:263/950 Jabber: