similar to: Voicemail question

Hello, I am struggling with what seems a common unresolved problem, changing the password from voicemailman when using a realtime engine (adaptive_odbc in my case, connected to mysql). I have seen messages dating back to 2007 with this problem and the last one was bug 5168, reported as closed, but without explaining the fix

What I'm trying to achieve is that a voicemail message should be at least 3 seconds long for it to be saved, but *after that* a prolonged silence (e.g. 10 seconds) should terminate the call and recording. My current settings (Asterisk 10.7.0 and 11.2.1) are: ; Minimum length of a voicemail message in seconds for the message to be kept ; The default is no minimum. minsecs=3 ;

In Asterisk 1.4, is there any way to "force" new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. I know I can publish docs that tell them how to use the "0" menu and do this by hand... but users are

Hello all, I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to send the voicemails as attachment to e-mails and delete the voicemails from my PBX once it has been sent. But, I don't have a running MTA here even on the PBX itself. I just want to send the e-mails to my GMail account from my PBX. Can I just use the mail or mailx command to send the e-mail and attach the

> -----Original Message----- > From: Mark Charlton [mailto:asterisk@mcwebtree.com] > Plus if you send your users to VoicemailMain(${CALLERIDNUM}) > they don't hear > it at all. > They just get "enter password". Yup. If you do that, the only time they hear it is during the initial setup call (if you have "forcename=yes" or

Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info.

Hello List. I am having problems retreiving voicemails on my system. I noticed when someone leaves a message through the pstn line I can't hear anything. I tested leaving a message from one of the extensions and that can be heard. I don't know if is the type of card I'm using for analog ( cheap X100p modem ) calls but, can't hear any message coming in from that line. Any

Hi everyone, I've set up a little Asterisk system with a Digium TDM400P and everything works splendidly except for the messages callers leave. Every message that a caller leaves is very faint. I've already set volgain=6.0 in voicemail.conf, and that seems better, but to be at a good volume I estimate I may need to go up to 40.0. Is that reasonable? One interesting artifact is that at

When I go to Directory() inside extensions.conf, I can set the dial-context and change the search option to be for first name instead of last by passing the appropriate options. How does one configure the directory inside voicemail to do the same thing (usedirectory=yes inside voicemail.conf). There doesn't seem to be a way to configure the directory that is presented to the voicemail user

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

Hi Everyone, I cannot seem to get the voicemail gain option g(#) work in Asterisk 1.4.11. I am using it like so... Voicemail(4444 at mycontext,bg(10)) ; for busy announce and 10dB record gain This has absolutely NO affect on the resulting voicemail wav file. I have also tried using "format=wav" instead of wav49 in voicemail.conf to increase the volume as well. This also has no affect

Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, "default" is always being used. The output of "sip show peers" shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the "default" context.

>From voicemail.conf: ; Maximum number of messages per folder. If not specified, a default value ; (100) is used. Maximum value for this option is 9999. ;maxmsg=100 ; Maximum length of a voicemail message in seconds ;maxmessage=180 ; Maximum length of greetings in seconds ;maxgreet=60 I would like to configure these parameters on a per mailbox basis using Realtime voicemail. I

Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI: uniqueid => 17 customer_id => 0 context => techmast mailbox => 14 password => 1234 fullname => Sales and Service email => user1@domain.com email =>

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Hello all, This is my first attempt at posting a patch. So if I screw this all up, my apologies and please someone let me know without beating me up too bad. To use this patch. You need to have an extra line in /etc/asterisk/voicemail.conf that looks like volgain=10.0 The 10.0 gets passed to sox which you will need installed on your system. 10.0 is what works for me, anything over 1.0 will

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The

Hi - I just deployed a system using IMAP Voicemail. During my testing, voicemail worked fine. I could check vm from the phone, and the messages would get marked as read, or I could read the messages in a mail client, and the phone's mwi light would turn off. Very neat. I'm not exactly sure when things got munged up, but something broke. I can record messages with Voicemail(), but now

I am blessed with this user forum and able to set my Dev-PCI Digium card working fine with the Asterisk server of mine (i)But today i just wanted to know if someone can help me to set X-Ten Lite to call PSTN line using my FX0 Currently , I am able to use X Lite to call another X lite user locally (LAN) I also has attached my setting together Thanking you all in advance --------------

My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. What I want to do is execute a remote command with the voicemail as an argument. The remote machine command would email the message. I'm thinking of: same =>n,VoiceMail(vm,u) same =>n,System(ssh myserver "emailVM