similar to: AGI Exec Voicemail

Has anyone set it up so that an inside call and an outside call get different unavailable messages? j

Hello, I am having some trouble with the Asterisk::AGI perl library. It seems that the AGI->Exec() command is causing me a problem. Here's the line in my AGI code: $AGI->exec('Record',"$vmfile:wav, 30"); I'm trying to record voicemail to the file name stored in $vmfile with a silence timeout of 30. However, this is not being parse by AGI or Asterisk correctly,

I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've read several of the Wiki sections on Voicemail and "Asterisk variables" but could not find an obvious answer to this question. I would like to run a script that post-processes the voicemail after the Voicemail application returns (with AGI or DeadAGI), but I cannot figure out how to easily determine the name of the file written by the Voicemail application. Does anyone

From an AGI script I'm setting two channel variables and then calling Voicemail on several mailboxes. I know the channel variables are set correctly because I can "GET VARIABLE" from within the script. In voicemail.conf (in mailsubject and emailbody) I reference the channel variables, but the values do not appear in the email. Does the Voicemail system have access to my channel

http://www.wunderground.com/weather/api/ -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of dk at donkelly.biz Sent: Wednesday, December 16, 2015 9:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] weather.agi -----Original Message----- From:

On 6/26/23 9:00 AM, Joshua C. Colp wrote: > On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: > > I am connecting to the ARI with subscribe all, so I can see > channels being created. I now want to extract a variety of header > variables (at the moment the from and to tag).  I tried to read > them from the ARI but Asterisk refuses since the

Anyone using this feature of asterisk's voicemail? I'd never heard of ADSI, and saw it as I was perusing the voicemail source this morning. Is it some kind of visual way of managing voicemail on your phone's display, or does it require a terminal of some kind? Thanks! j

On 6/26/23 5:19 PM, Jeff LaCoursiere wrote: > On 6/26/23 9:00 AM, Joshua C. Colp wrote: >> On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: >> >> I am connecting to the ARI with subscribe all, so I can see >> channels being created.  I now want to extract a variety of >> header variables (at the moment the from and to tag).  I

Here is a funny story. We mostly do hotels in the Caribbean, and one of our first clients (going on ten years now) used the sample "weather.agi" that used to be shipped with... asterisk at home? Trixbox? Can't even recall where we originally got it from. This perl script uses festival to speak a brief weather forecast to the caller. We told our hotels this was a feature for

Dear friends, I am having problem with running a sample php and I can't figure out why. I can run the sample.php using CLI but when I run it inside the dialplan it does not work. Can someone please suggest the config problem that I may have made? dommy:/var/lib/asterisk/agi-bin# php sample.php #!/usr/bin/php5 -q VERBOSE "Here we go!" 2 VERBOSE "Call from - Calling

Hi Guys.. I am making an anoucement machine that is not allowed to "answer" the call due to a billing issue. I found that Playback with "noanwser" is usefull in this case. $AGI->exec('Playback',"$message","noanswer")} But when i request some values to the user with get_data, i think there is an answer anywere. Is there a way to get_data

So I hacked up the Dial app to return a numeric return code instead of changing contexts based on a number being busy or unanswered. The purpose for this modified dial app, which I call AGIDial, is to help me concoct a "follow-me" type of application. The app returns -1 for a completed call, 0 for unanswered, or 1 for busy. Well, I hooked the thing up to an AGI script that uses perl and

Howdie- Can anyone please confirm that the BACKGROUND application does ***NOT*** return IMMEDIATELY when called from within an AGI EXEC command? It seems that EXEC waits until DTMF or THE END OF THE AUDIO FILE to return to the AGI script. This essentially prevents repeated calls to BACKGROUND (or I assume any other asterisk dialplan application) from within an AGI, and prevents

hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension

Hi everybody, I'm trying to implement a enhanced blacklist system using AGI and Perl, configuration in extension.conf is: exten =>_numbera,1,AGI,blacklist_2_in.agi exten =>_numbera,2,Answer exten =>_numbera,3,AGI,xisco_1.agi exten =>_numbera,4,AGI,blacklist_2_out.agi The problem that I have now, is that blacklist_2_out.agi doesn't execute. I think this is because in

actually, i do have a workaround which bypasses the exec command entirely: system("asterisk -r -x 'add extension s,3,Playback(demo-congrats) into local'"); but it's ugly. seems like it should be possible to do this with exec. .t ---------- Forwarded message ---------- Date: Sun, 23 Nov 2003 21:17:50 -0500 (EST) From: tad <tad@media.mit.edu> To:

Hello I need some advice regarding the Asterisk server that are located at different locations. Three asterisk servers are here each at different location. Suppose A,B,C be the three servers respectively. Server A is connected to server B and server C through a VPN. I have a developed an IVR service on server B and server C where users come and record their voice. On the same servers B and C

I’m in training, so I have to demonstrate something SIP related. I figure it would be cool to hack a call, hanging it up while in progress from outside Asterisk. Doing so will demonstrate use/knowledge of ARI, AMI, SIP, route-sets, UDP, etc. Practical value: zero :) Who knows, maybe this will have an actual application for someone someday. In practical terms I think building a proxy

On 2017-04-19 02:39 AM, Pete Mundy wrote: > Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail > show users' I can't see why the vm_authenticate function is > failing to read the username :( I can answer that one. It's because we can't enter 'stocktrans2' from a telephone so we just hang up. The question is, why does it ask for the