similar to: How to access voicemail from deskphone

This system targets a different market... I like Olle's system. He did a good job. Olle's minivm is a great choice for those wishing to build customized voicemail systems, but as the name suggests, the systems are very basic. Large systems are difficult to maintain in the dial plan and some of the functionality we need would be difficult to implement with that approach. Justin -----

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

Friends in the Asterisk community, One thing I avoided working with for a long time is the Asterisk voicemail code. One module in Asterisk I've constantly been naming as one of the worst parts is voicemail. One part of Asterisk that I've been kind of avoiding during my trainings is voicemail. And there's where I've spent a lot of time recently... Life is strange. Instead

HI, I tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch Nov 26 15:34:59 Installed: asterisk18-core-1.8.0-1_centos5.i386 Nov 26 15:35:02 Installed: asterisk18-voicemail-1.8.0-1_centos5.i386 Nov 26

Hello All, I'm planning to use Asterisk only for voicemail Application and Recording will be done at different server. When user changing his personal greeting or leaving voicemail Call need to throw to external Voicemnail recording server over SIP til the time recording complete. While throwing Cal from Asterisk to application box i have to use SIP request which having some string in

On Mar 13, 2013 10:16 PM, "Amit Salunkhe" <amitsalunkhe21 at gmail.com> wrote: > Hi > > I want to know asterisk 1.8 as text to speech server. > > If we can use as TTS server then it support SSML. > > Any sample configuration available for this requirement. Plz help me with > support asterisk as tts server. > > Amit-- > -------------- next part

Has anyone here ever tried using a wifi bridge to place a deskset in someplace where there was no LAN drop? If so what hardware did you use and was it succesful? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005

Don, All you need to do is configure your VM button on the phone via the cfg files. You setup the phone to look to an FTP server for config scripts and then setup the scripts to work as needed. Here is an example of where you set this VM button from my file called <mac address>-phone.cfg... <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.subscribe="1006"

Hi, I am using Asterisk personally at home. My SIP client (SPA 3000) supports MWI with SIP NOTIFY messages. With a previous version of Asterisk I had no problems with MWI. But now I am using the following version which comes with Trixbox 2.8.0.1, and I have problems with MWI. Asterisk 1.6.0.9-samy-r27 Problem description: When a voicemail is left on the extension, a SIP NOTIFY message is sent

---------------------------------------------------------------------- Message: 1 Date: Tue, 6 Mar 2007 20:02:07 +0100 From: Olle E Johansson <oej@edvina.net> Subject: [asterisk-users] Building a new voicemail system... Testers needed! To: Asterisk Non-Commercial Discussion Users Mailing List - <asterisk-users@lists.digium.com> Message-ID:

Here's a link to it: http://forums.digium.com/viewtopic.php?t=4363&highlight= Regards, Scott -----Original Message----- From: Scott Keagy Sent: Monday, December 04, 2006 5:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. A while back I posted a fully functional though somewhat elaborate

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

Hello, I'm trying to supervise an existing Voicemail box with a BLF button on Debian's asterisk 13.14.1 system. I mostly found this [1] document. I added in a context a line like: exten = *7000,hint,MWI:31 at default With "core show hints", I can read this: *7000 at subs : MWI:31 at default State:Unavailable Presence:not_set Watchers 1 My questions

> -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Saturday, June 17, 2006 2:30 PM > To: asterisk-users@lists.digium.com; Douglas Garstang > Subject: Voicemail with NFS (working, I think) > > I'm using a stand-alone VM server and exporting the VM files ro for > MWI function only. All my registration servers mount the remote

Thanks for the info, but not really what I?m looking for. If possible, I?d like to modify the source and re-compile the existing voicemail to make it match what I have today. Thanks. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent: Thursday, August 31, 2017 4:13 PM To: Asterisk Users Mailing List -

Howdy I have a question regarding support for picking up when the telco sends a MWI message. My client's setup is a small office with three incoming lines on a TDM400P with iaxy's and a Grandstream as extensions. I am using CVS Head from yesterday. (I was resolving a different issue.) Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco.

Hi - We've got multiple offices with their own asterisk boxes (CVS HEAD 11/03/04-14:59:37) connecting to each other using IAX forwards. All users are on SIP phones. Voicemail is centralized to one location. Everything is hunky dory except that the users in the remote offices don't get MWI on their phones. I've seen the other posts to this list regarding this, and

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is doing some IVR work prior to forwarding calls to the PBX and it also acts as the voice mail server for the PBX, with Asterisk configured for IMAP storage. When a call comes in and the caller leaves a voice mail, the VoiceMail application calls the program configured in

I'm using a stand-alone VM server and exporting the VM files ro for MWI function only. All my registration servers mount the remote NFS share just to check MWI, all read-write functions to the VM files occur on the VM server only. On the registration servers, I mounted the remote VM share with this in my fstab.conf: 10.10.14.124:/var/spool/asterisk/voicemail /mnt/vmserver nfs