similar to: AMP/A@H (asterisk at home) custom incoming routing

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- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: "Thank you for calling xxxx". I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the "phpconfig for Asterisk PBX" page, and selected

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 know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an <mailto:Asterisk@home> Asterisk@home installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant

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

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

do you have selinux enabled? It should not be. p p.s. - if it comes to re-installing, you can backup all your settings with the freepbx backup utility and then restore so that you don't have to re-enter everything. From: "Lachek Butalek" <lachek@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date:

Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

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

Egad, not again with Broadvoice! Anyhow, I recently installed AAH and configured my TDM11B and got that and some SIP phones working. I still have some issues to work out, etc, but my current problem is Broadvoice. I have checked out all of the online resources, including the recent list exchange about the recent changes made by Broadvoice. However, the one thing I have found to be consitent in

Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system

Is there a configuration difference for clone X100P cards versus "compatible"? I have a similar problem to what David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel => 1 without the semicolon. Any outgoing call attempt returns "all circuits are busy"

Hi All, I've installed asterisk and manually configured IAX/SIP users. Everything works fine, I'm able to call other extensions. But when I installed AMP and created new extensions, I'm not able to call those extensions. I get the message that the extension is busy and it is forwarded to voicemail. What am I missing here? The workaround I found is by modifying the

Hi all, I'm trying to setup a custom extension in AMP (yes i can code it by hand but the on-site admin that does moves & changes cannot). I've tried the following > add cutom extension 600 in the dial box i have Dial(IAX2/username:password@host/$EXTEN@from-internal) this doesnt work as these lines are added to extensions_additional.conf exten =>

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:

Hi, I have configured an Asterisk server with TDM01P (1FXO) for testing purpose. The interface I'm using is AMP. I want to configure my extension so that when I dial from my mobile phone to the asterisk line, I want it to transfer the call to any extension, say 3042 and after a particular number of rings, transfer the call to voice mail so that I can record my message. My Zaptel.conf is as

Hello every one. This is a question done by me, not yet answered. Please, help. I: 1. Run install-pdf from linux to support faxes on my asterisk. 2. Made the configurations throuhg AMP in a. Setup->Inbound Routing->(the only route I have)->fax extension->System b. Setup->Inbound Routing->(the only route I have)->fax email->(my email) c.

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to