similar to: Asterisk 1.6.0.1, IMAP Voicemail storage and temporary greetings.

Hi all, I'm trying to build a little voicemail server based on asterisk here, using Asterisk's "Commedian Mail" application. Unfortunately, I'd expect some people to have trouble using the English prompts that come with asterisk. However, I can't imagine I'm the first person who has this problem, and Commedian Mail seems to support multilingual prompts fine, it's

Hello all, I'm running into an issue with Asterisk 12.4 and IMAP voicemail. I have asterisk set up to connect to my Dovecot IMAP server and I can leave and retrieve messages from my inbox and old messages. However, I am unable to move messages between folders. I get a message from asterisk stating "Sorry the users mailbox can't accept more messages". Here is my setup: In

I installed version 1.6.0.3-rc1 and my AMI application stopped working. I reinstalled 1.6.0.1 and it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel:

Hello, everyone. No need to read this message. I'm posting for documentation for other poor, ignorant slobs like me who are struggling to pull together the many technologies to make converged networks happen. Hopefully, this will help save someone else the time I spent. I started the below email until I realized I had solved multiple parts of a compound problem but not all at the same time.

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c -> manager.o manager.c: In function ?action_getvar?: manager.c:1732: error: ?SENTINEL? undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears

The Asterisk.org development team has released Asterisk versions 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, as well as Asterisk-Addons versions 1.6.0.1 and 1.6.1-rc2. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a fix for a regression introduced in Asterisk 1.2.30 and Asterisk 1.4.21.2 and has existed in the

In download dated 10/9. Bug fix? Mistake? sean

Hello everyone! I've just switched from Asterisk 1.6.0-beta9 to 1.6.0.1 and my mISDN is not working. Here's what happens, if I try to call the line: bach >> P[ 1] --> !! lib: No free channel! P[ 1] --> we have already send Release_complete I haven't changed the configuration fles. Should I change something there? If you need more info, just tell me and I'll

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

I have had two users on two separate systems indicate that they could "not hear a new message" When I investigate I find that the user has marked a message for deletion. The .WAV .wav and gsm files are gone but the .txt file remains thus giving asterisk and the user the impression that a new message exists when it does not. Has anyone else encountered this issue? Is there a fix?

All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help.

Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears?

19.08.2012, 00:39, "Julian Klappenbach" <jklappenbach at gmail.com>: >With this approach, the community would gain language independence for browsers Browser community is strongly opposed to the idea of having multiple web-faced languages > The first language I'd like to tackle is ECMAScript / Javascript. You can tale a look at llvm-lua project. However, speed of JIT

Hi all, 2 days ago, managed to install rhat and asterisk, starting with 2 sip phones, simple config. CLI reports : "playing 'vm-theperson' (language 'no')" when transferred to voicemail after timeout, or "playing 'vm-password' (language 'no')" when dialing into voicemail extension. But, no sound, nothing heard in phones. First time, first

Hello everyone, Is it possible to ask the VoiceMail command to play the user greeting before recording a new voice message? I've looked at the documentation and I don't see how to do that... Any help would be really appreciated! Best, Leo

I was trying to find out how I could put in a greeting when a caller ***first*** joins the queue. I searched high and low but could only find (in queues.conf): . "announce", which is announcement to the agent . "announce-frequency" which is announcement of queue position . "periodic-announcement-frequency", "periodic-announce" which may seem applicable to

Hi all. I've got an odd situation with my RT asterisk server. I've got a number of users who are reporting that their voicemail greeting isn't being played anymore. This used to work before a recent asterisk restart. The dialplan is in AGI, so it wasn't changed. I'm storing voicemail in a mysql database and that is working properly. It's just the greeting message

Hi all. I've got an odd situation with my RT asterisk server. I've got a number of users who are reporting that their voicemail greeting isn't being played anymore. This used to work before a recent asterisk restart. The dialplan is in AGI, so it wasn't changed. I'm storing voicemail in a mysql database and that is working properly. It's just the greeting message