similar to: voicemail number of rings

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

Dear R-users, Attach with is my data..what i want to do is finding a suitable distribution for my data..I want to run a few test like the poisson and the exponential distribution. Please help me on how to find the p-value for poisson as well as the exponential distribution without knowing the parameter. Is it possible?? Thanks in advance. love, Anisah -------------- next

Hi, I'm trying to automatically have the dialplan assign an extension to a roaming phone on my network. I tried the following without success: exten => 3001,1(readop),BackGround(beep) exten => 3001,n,Read(digito,vm-youhave,3) exten => 3001,n,SayDigits(${digito}) exten => 3001,n,Set(ROAM=${digito}) exten => 3001,n,Set(DB(roam/ext)=${digito}) exten =>

Dear all i'm creating an outgoing call to number xxx with this command: http://host:port/mxml?action=Originate&Channel=Local/xxx at to-external &Exten=testDTMF&Context=cRETEUNICA&Priority=1 wich points correctly to this portion of dialplan: [cRETEUNICA] exten => testDTMF,1,Answer exten => testDTMF,n,Read(digito,,1) exten => testDTMF,n,SayDigits(${digito}) The

Hi list, I have searched through asterisk command lines but haven't found how to do this: - can I list the phones (callerid or IMSIs?) currently registered ? If I do "dialplan show" that lists the configuration I loaded, e.g [ Context 'sip-local' created by 'pbx_config' ] '2102' => 1. Macro(dialSIP|IMSI1) [pbx_config] '2103'

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

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

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:

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 -

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:

Hello, I have setup the n-way conferencing with Asterisk and it's working when I use with my budgetone 100 phone but it doesn't work for any of the voip software or other ATA that I have. When I turned the debug on, I see that the correct keys (*0) were entered but asterisk doesn't detect the signal to trigger the features event. I have set a test extension to get the input dtmf key

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

Hello We would like to customize the voicemail menues. So the intro should not be played if some user has recorded an own greeting message and we would also like to remove some options from the menue. Is this all hardcoded or is it somehow possible to redefine the voice menues and the order how messages are played via voicemail.conf? Mit freundlichen Gr?ssen Benoit Panizzon -- I m p r o W

Has anyone set it up so that an inside call and an outside call get different unavailable messages? j

I am using 1.4, and have the above device, and it worked really well with monitoring 18 "hints" aka devices. Now, I've moved us to a hotdesking paradigm where the user is the "extension" not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ?

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 -----

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

---------------------------------------------------------------------- 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:

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

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin