similar to: Re: asterisk-users Digest, Vol 32, Issue 21

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

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

Hi All we are using Asterisk 1.6.0.9 version.try to use Minivm for voicemail, but having following problem. 1.How any extension let's say 7001 can access his voicemail box from his deskphone, any config or dial plan example is there. What kind of config require in extension.conf,minivm.conf & sip.conf 2. same way for MWI on respective IP Phone what we require to config in

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

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 -

Hi, Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. ! ael agent agi cdr channel cli config console core database devstate dialplan dnsmgr dundi features file group hangup help http iax2 indication keys

I?m looking to change the TUI, the Telephone User Interface. In other words, instead of pressing ?1? to play a message, I want to press ?7?, etc., etc. 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 6:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Is there a way that I can modify the source code for the voicemail application? I need to change some of the options in the user?s interface to make it work like an existing system that I?m replacing. Thanks. Tim 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 6:30 PM To:

sorry, was only for users list... Hi Kevin, Hi list, you are right, acting now is not needed, when callbacklogin will be removed anywhere in future... But thinking how to realice alternatives can't be so wrong. Callbacklogin gives a very simple way to use more queues for one agent, which only has to logon to only one system. No need to make dbs or tables for saving, where the agent has to be

Hi! I want to configure voicemail to send emails with the date of the message in German/Austria, that means: "Montag, 26 J?nner 2009" instead of "Monday, 26 January 2009" voicemail.conf refers to "man strftime". This refers to the current locales. So, I tried export LANG=de export LC_ALL=de_DE before starting Asterisk. Unfortunately the date format is still

I'd be really happy if users could use the voicemail menu to change the number of rings until voicemail picks up. It seems like the current model of separate Dial and Voicemail commands would make that difficult, but is there any plan for such a feature in the future? How about a workaround or 3rd party add on? I store the dial timeout for each user in a database, so I know I could make

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Is there a way to customize the voicemail navigation system? Google shows some discussion of it in late 2006, but I see no references to it being implemented. Many thanks, -Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090126/d4da1aa4/attachment.htm

*lol* I think he was referring to a ASTERISK reboot, not a phone reboot. > -----Original Message----- > From: Olle E Johansson [mailto:oej@edvina.net] > Sent: Thursday, March 23, 2006 1:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: Subscription state after reload (New > subject) > > > > 23 mar 2006 kl.

How can a reload clear registrations? If I 'reload' without using realtime, I keep my sip peers (as well as astdb). I can still contact other phones... registration info is still there. If I `reload` with realtime, I lose my sip peers (but astdb remains). I can _STILL_ contact other phones.... registration info is still there and Asterisk must be referring to astdb to find the IP

Today is Swedens national day - since a few years a holiday too. We don't have a tradition on how to celebrate. Sweden has not been to war for a very long time, so there's no real spirit for the country here - it's been aroundfor such a long time, so what? :-) Guess we have to learn from abroad, to get a celebration feeling like July 4th in the US or May 17th in Norway (from

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Is there a way that I can modify the source code for the voicemail application? I need to change some of the options in the user's interface to make it work like an existing system that I'm replacing. Thanks. Tim -------------- next part -------------- An HTML attachment was scrubbed... URL:

I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be