similar to: Is it possible to setup group voicemail inAsterisk?

Hi I have two small question, if you can help me ;=) Problems with Answer+Music my extension: [Cal-In] exten => _811XXXX20,1,Goto(C-Internal,100,1) exten => _811XXXX21,1,Goto(C-Internal,200,1) [C-Phibee] exten => 100,1,Ringing exten => 100,2,Wait,1 exten => 100,3,Answer exten => 100,4,Dial(SIP/201&SIP/200,30) exten => 100,5,Hangup exten =>

Hello there, I'm a new Asterisk user and I wonder if it is possible to associate a voicemail box with a group of users, i.e., a single recorded message is sent to everyone in that group. If so, where can I find more information about that? Thanks in advance, Leo Burd

Hello there, I'm writing an application to display asterisk voicemail on a webpage. Since Flash only handles MP3 files, I wonder if it is possible to configure asterisk voicemail in such a way that it would record and play MP3 files... Would anyone give me suggestions on how to do this? Is format_mp3 stable enough for something like this? Thanks in advance, Leo Burd

This 2 line code is doing what I wanted. exten => 200,1,voicemail(200) exten => 200,2,Hangup What I've been told is that they want the 20 year old phone system to light up the message bulb. (yea, a filament bulb, not an LED) To do this you pick up on the line that goes into Asterisk and do a: exten => 200,1,SendDTMF(200w#86) But I don't know the path to take to get that

Hi, Just made a simple test to see how the two extensions (s and i) worked but for some reason I can't seem to make then act as I would like them to. I pick up the phone and dials 100 or 200 - and in the CLI it prints out what ever I have put in the Noop() If i dial any other number, nothing happens - no indication in the CLI. Souldn't the s or i context be activated when I dial a

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

Hello there, I'm trying to configure my voicemail system and I have a couple of questions: * Is real-time voicemail already working? If so, where is it that I should specify the database name, user and password? Where can I get more information about the different options that exist and the different files that need to be changed? * Is it possible to setup the voicemail interface to

Thanks for the suggestion, C F, but the problem is there is a rather big database application behind with many users, so a static configuration is not suitable for my needs. i am working mostly with realtime and agi. regards, roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Sunday, July 03,

Hello! I want to use "asterisk -rx "show version"" from a php script called in the browser using the local apache, which runs as user "apache". Asterisk is running as root. I added the following line to /etc/sudoers using visudo: apache ALL = NOPASSWD: /usr/sbin/asterisk When i am on the command line of my linux box it looks like this:

Hello! Is there a way to use AVM Fritz!PCI as a ZAP interface and have it configured for ZAP channels? Thanx in advance! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

Hello! Is it possible to run Asterisk as a SMS Service Center and does it work over a directly connected ISDN (CAPI) interface card? Did anyone already use Asterisk for that? Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! Thanxxxx! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Hello everyone, I'm having all sorts of problems with my PHP AGI scripts... Basically, my scripts run fine from the command line and don't do anything well called from Asterisk. Here are my questions: a) Does Asterisk require PHP CLI or CGI? From the command line, my script seems to work fine with PHP 4.3.11 (cli) but not with PHP 4.3.9 (cgi) b) How to debug my script? According

Thank you, Robert! The announcementfile plays well, but at Dial-option "m" i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

Hi, I'm having a bit of a problem with storing voicemail messages in an odbc database. I *think* I've got everything configured correctly but messages are stored on the asterisk server instread of in the database. System info 64 bit redhat RHEL 5.1 Asterisk 1.4.26 unixODBC installed used makemenuselect to instal res_odbc and cdr_odbc Back end database DB2 Database name voiceml

Hello all, I wonder if somebody could provide me with some advice on how to track what looks like a bug to me: I've got a PHP AGI script that is called whenever I dial into the system and also whenever I issue a specific Originate() request via AMI. The script works fine when I dial in. However, when I run it via Originate(), it sometimes does not play anything, sometimes plays part

Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in