similar to: ring group failure with "ExtensionState: 4"

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

Would it be possible to set the ringtone based on the number that was dialed? Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 We are currently using Digium D40, D50, D70 phones. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm writing a function and keep getting the following error message. myfunc <- function(lst) { lst <- list(roots = c("car insurance", "auto insurance"), roots2 = c("insurance"), prefix = c("cheap", "budget"), prefix2 = c("low cost"), suffix = c("quote", "quotes"), suffix2 = c("rate",

I'm trying to solve a mystery for the last couple of days. I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS. For several years, everything was working fine, no issues. A few days ago I started having problems at one particular site. NO CHANGES have been made to this office network - same router, switch and internet provider. No new equipment added or

Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -------------- next part -------------- An HTML attachment was

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried

Hello, I have setup a test server with asterisk/AMP and have several 7960's connected to it. The asterisk server has a public ip and all the 7960's are behind nat'd routers. When I try to call from extension to extension I get directed straight to voicemail. I do not have any cards installed and instead direct everything to an Ondo server. I have been told it's not an AMP

This is my first attempt to write software of any sort. What I am trying to is to use a .php page to query asterisk Manager and get the ExtensionState for each particular extension. Then when it has the answer it outputs an XML file for use as the directory on a Cisco 7960 phone. What I am thinking is that when the user hits the directory button to veiw the directory that is at this URL it will

I'm not having any luck getting the ExtensionState action of the Manager API to work. The response is always success but the Status is always -1 which to me means an error. Here is a typical telnet session. ----------------------------------- stockholm:~ # telnet localhost 5038 Trying ::1... telnet: connect to address ::1: Connection refused Trying 127.0.0.1... Connected to localhost. Escape

Hi, I am trying to get the extension status (weather it has dialed outgoing call via SIP or IAX2), using the following piece of code however it always returns -1 on all the extensions (valid/invalid). Am i missing something ? Any help. Thanks ----------------------------------- #!/usr/bin/perl use Asterisk::Manager; use lib './lib', '../lib'; $|++; my $astman = new

Hello List, I have been working on a PHP application in order to build a BLF style script. Until now everything is going Ok but something a little (in my oppinion) strange is going on with the 'ExtensionState' command; The problem is that it does not returns the 'Status' as it's suposed to, mentioned in the A.T.F.O.T book - version 2., where it sais something like:

I have a really long functions, and at the end of the function, I am using a if statement to tag certain keywords based on whether they have certain values contained in them. However, the if statement doesn't seem to work. When I had split up the commands into various functions, it worked fine, but I'm not sure what going on now that it's combined into a single function. myfunc

I have a repetative task in R and i'm trying to find a more efficient way to perform the following task. lst <- list(roots = c("car insurance", "auto insurance"), roots2 = c("insurance"), prefix = c("cheap", "budget"), prefix2 = c("low cost"), suffix = c("quote", "quotes"),

Hi all, Just installed Asterisk 13 on CentOS 7 and have run into a problem. The Scenario is this: Asterisk is on the internet the Phone, a D40, is behind NAT So someone calls the number and Asterisk routes the call to the D40 Everything works fine and the call is established, but then after 5 min. the caller stops getting audio from the D40 but there is still audio to the D40. using both

Hi all, I'm trying out a Digium D70 phone with Asterisk 11. My voicemail messages are listed in the "visual voicemail" app on the phone, but they do not successfully play back. The correct duration is shown, but the progress bar just jumps back to zero when I press the "Play" softbutton. I can hear my messages fine if I "manually" dial into my voicemail

Hello, is there a mailinglist where I can post questions regarding Digium IP-phones ? I have the following question : I'm trying to provision a Digium D70 IP-phone from a https provisioning server. The Digium D70 contacts the provisioning server correctly but seems to log in with the wrong credentials : /var/log/ssl_access_log : XX.XX.XX.46 - - [22/Jan/2014:12:15:09 +0100] "GET

Hi ALL Have installed asterisk@home 1.0 On FWD DID's, appears that 2 calls are generated to the inbound extention. I have confirmed this on a number of friends boxes also. Does anyone have a fix for this ? I set the DID simply to a custom context and it did the same... Anyone have a way to fix this ? Here is the output...... -- Accepting AUTHENTICATED call from 65.39.205.121, requested

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New

Skipped content of type multipart/alternative-------------- next part -------------- asterisk1*CLI> soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' --

OK, i've got asterisk @ home 1.3 up and running with Broadvoice. BUT I have one nagging problem to sort out. When you call my BV # the calling party gets no ring indication, just silence until either I answer the phone, or the call bounces over to voicemail. below is the console output when a call is recieved. what am i missing here? thanks Bernie -- Executing