similar to: Verification number / code

Is it possible to have Calls incoming to different DIDs? I want an AA that handles 100s of businesses. [Incoming-pizza] Exten => 4045551212,1,Goto(pizza,s,1) [Incoming-hvac] Exten => 8085551212,1,Goto(hvac,s,1) [Incoming-gutter] Exten => 6175551212,1,Goto(gutter,s,1)

Where is everyone located? I am in Washington DC. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/7c73847d/attachment.htm

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current

Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples.

Does anyone know how to send a text message from Asterisk?

I have a 'time and attendance' application. Think janitorial or security kind of thing where an employee goes from location to location. They're supposed to 'clock in' when they get to a site using a phone at that site to prove they're there. Some employees have discovered 'fake caller ID' services can be used to say they're on site when they are not. How

This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? If I were to build a call file script (described in this link http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then how does it work if my Asterisk machine is running on Centos 5.5? I simply

Zitat von jg <webaccounts173 at jgoettgens.de>: > Yes, it is called "core set verbose 42", the other options is "core > set debug 42". Enjoy the show! OK, thanks, but with this option I can just debug what happens if I call an extension right now... I'd like to have a command to ask Asterisk how it will handle a call... > Once you are more familiar

I want to create a script for IVR that compiles responses, aggregates them to a total number. Then, run an equation based on the result. Press 1 for X (X is a positive number 500) Press 2 for Y (Y is a positive number 200) Press 3 for Z (Z is a positive number 300) Press 20 to calculate the results = 500+200+300 =1000 then, exten => s,n,Read(NUMBER,,1000) exten => s,n,SayDigits(${NUMBER})

Does anyone have a Dial script or a hint on how I can dial 10000 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me.

Do you see any syntax errors? Positive comments welcomed. The short version of the logic is as follows: create a file based on the NUMBER create a an audio file move to a new extension (label) and play the results exten => 621,1,Answer() exten => 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5) ; create a variable from a DTMF entry / 12 characters long exten =>

My client wants to use my service that I will host. It is an IVR application. I have the solution worked out on the server side. I will port his current POTS line phone number to a DID service where I can control it via SIP. Question relates to his current phones. Forgive me as I am new. Does he need his current phones? How will they ring if I port the number? Should I simply have him remove

I am running Asterisk 1.4.2 and recently we changed the SIP provider of our main incoming DID number. The new provider prefixes all CallerID records with a +1 in front of the number, whereas the previous SIP provider did not. Consequently now all my blacklisted numbers aren't matching in my Dialplan, so I'm getting tele-spammed. Is there a way that I can work with the blacklist

Please don't top post. > Am 29. Mai 2015 09:42:55 MESZ, schrieb Luca Bertoncello > <lucabert at lucabert.de>: >> Zitat von jg <webaccounts173 at jgoettgens.de>: >>> Yes, it is called "core set verbose 42", the other options is "core >>> set debug 42". Enjoy the show! I know you can specify a level to the verbose application,

also sprach Steve Edwards <asterisk.org at sedwards.com> [2015-05-16 23:22 +0200]: > I use a preprocessor > (http://software.hixie.ch/utilities/unix/preprocessor/) to tailor > dialplans and configuration files to each host based on the client > (or project) and the hostname. Yeah sure, templating works, but it introduces a layer of complexity that can make debugging hard(er). I

I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan. I have a successful SIP session registered: Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) Asterisk*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060

Hi gang, Since I'm getting no joy from device_Status or SIPPEER in 1.4.26-rc1, I thought I would do an AGI to read my hints and check for line in use that way. The AGI works fine from a prompt, but returns the dreaded "utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I try to run it from the dialplan. Here is my dialplan snippet;

Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files. CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include => stdexten exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks)) exten => s,n,Wait(2)

Does anyone have any short answers on how I can fix this problem: A calls B. B rings Says connected. But the call is not bridged and therefor no audio passes. very simple dial plan. Frustrated. v 1.8

I have a very lite layout and attempting to get the SIP configuration set up initially before proceeding into other areas. VMware is running my Asterisk 11 on Ubuntu 12. Shouldnt I be able to at least ping the SIP provider IP? I run command "sip show registry" and do not see it set up. I run sip show peers and I do see an entry. I have not configured anything other then entries in the