similar to: Can I get info about email addresses from voicemail.conf in dialplan or variables ?

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

Don't be afraid about the info that I'm going to post in this mail, but I want you to give as much info as possible. Also I want to show you what I've tried. What do I want When a voicemail-message is left via the Voicemail()-application, I want the .wav-file send to my mail-address as an attachment. My mail-setup I'm not using sendmail as MTA. I have msmtp as MTA and mutt as

print to standard error output in your perl script: print STDERR "This is how perl-AGI prints to Asterisk CLI output\n"; MATT--- -----Original Message----- From: Robert Rozman [mailto:rozman@fri.uni-lj.si] Sent: Tuesday, August 24, 2004 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Perl AGI - no output from agi script to Asterisk Hi,

I can't seem to find a definitive answer on this, and I really don't want to risk breaking a production server to find out; so I am going to try asking this here, and maybe anyone else in the same situation searching the archives sometime in future will find the answer I get. Can you use variables in the target of a GoTo() statement? What I am specifically thinking of is this;

I am using Asterisk 11.17.1 with my program that uses AMI Originate calls to generate a bunch of calls for a callcenter. The PBX configuration is handled by FreePBX 2.11. I want to understand the dialplan behavior in order to figure out why the CDR(recordingfile) is blank on the CDR records despite the dialplan setting it. My program generates the calls by setting Channel=Local/NUMBERTODIAL at

Hi, I'm using AMP and its dialparties.agi as most important script in system. I'd like to port configuration to more embedded system, where I don't have Perl available. So I'd like to implement dialparties.agi functionality as closest as possible with dialplan language..... Are there any existing dialplan scripts-examples that are close related to dialparties.agi

hi list, I'm looking for a way to execute commands in my dialplan specifically when a caller has hung up. my curretn dialplan looks like this: exten => s,1,Answer exten => s,n(restart),BackGround(intro) exten => s,n,Read(Enter,4,4) exten => s,n,Voicemail(${Enter},u) exten => s,n,agi(process.php|${Enter}) exten => #,1,Playback(thanks) exten => #,n,Hangup it lets a user

Hi guys, This is a two part question about the Voicemail application: Firstly, is there a way built in to the current app which would allow me to have Asterisk call a phone number everytime a certain mailbox receives a new voicemail? I know about the email and pager notifications that are already built in to voicemail.conf, and I've used those but also have a need to do what I describe

Used the Directory application instead of the Directory AGI. -----Original Message----- From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Tuesday, January 23, 2007 11:29 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] * 1.0.9 Voicemail record name does not playback in Directory() On * 1.0.9 User logs into voicemail, dials Option Zero, then Option

Hello, I have been attempting to get the Monitor function to accept a loal variable substitution in order to use the same filename later in the same context. Monitor does not appear to like it, as it attempts to use wav|filename as the recording type, as opposed to just wav. Here is what I get if I just supply a filename directly (it works fine): --context----------------------------- exten

I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I

Hi, I've been noticing a lot of these messages lately: "NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?" Is something broken? I'm running asterisk-1.4.22.1. They seem to happen in a number of different places where a beep or recording is played, such as when someone leaves voicemail or when an AGI script I have plays a time announcement -- lots

Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten => 11,1,Macro(oneline,SIP/11) Calling 11 (this is the same with BT or iax softphones) doesn't give me a ring - what is missing ? Thanks, Rob. [macro-oneline] ; ; Standard extension

I've recently installed asterisk and am working with the email a voicemail function. When a voice msg is left 4 files are created in the /var/spool. directory. They are .gsm, .txt, .wav and .WAV. The .wav (lower case) has the actual audio in it, the .WAV is a short blank audio file. When * emails the message it is sending the .WAV and not the .wav file. Any thoughts would be appreciated.

Hello All, I have my own AGI script running and I am trying to push the call to voice mail when Busy, Unavailable and Not Answered. Everything is working fine but the only problem is voice mail greetings for Busy and Unavailable is not played. By default only "Temp Greetings" voice mail greetings is played. I am passing the correct parameters for Busy => 'b', Unavailable

Hello all, I've installed AAH 1.1 on my VIA C3 powered mini PC. I've made the necessary changes to the * makefile, so the compilation went well. The first thing I did was configuring two extensions from AMP, namely 200 and 201. Then I installed X-lite on two PC's and configured them with one of the extensions: System settings - SIP proxy - Default: Username: 200 Authorisation user:

Hi, I'd like to implement scenario to send user to operator's queue by default (if doesn't dial any extension) but only if there is operator agent logged, so user could get response. If not I'd like to send it to voicemail... Any quick advice ? Thanks in advance, Robert.

Hi, I just setup two voicemail boxes - but after they record the wav file, they don't seem to email it even though i have the correct format in my voicemail.conf file. Aterisk doesn't give me any feedback besides the recording of the voip file... and then stays there. Secondly, I can't get the Playback() function to work. Thanks, Imran

I would like to have my asterisk ring my cell phone and let me know when a new voicemail arrives. In fact if it would automatically put into the voicemail menu that would be cool too. In the future I will probably want it to IM me. Are there good examples somewhere of doing stuff automatically on the arrival of a new voicemail ? I noticed a place for the pager email address in voicemail.conf,

Hi I have a problem w/ voicemail, the subject message is corruption when used voicemail variables, e.g. : voicemail.conf emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR} Return: Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?= Expected: Subject: 1504|12|"Teste - Rafael" <1570>|16 Thank's Att, *Rafael dos Santos Saraiva* Tel: (51)