similar to: Direct dial

Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company

Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121.

The puppet-module-tool GIT pages says it''s been converted into a puppet face and merged into puppet core. Does this make puppet-module obsolete or does it still need to be installed? I''m running puppet 2.7.9 but this "face" is not included so obviously it''s not available. Can someone provide a little clarification please. -- Later, Darin -- You received

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4 FreePBX = 2.11.0.4 I am trying to understand flow control in Asterisk dial plans and not having very much luck. I have read the Asterisk book from O'Rielly, or at least those parts I believe might apply, but that has not helped me much on this particular issue. What I wish is to set three distinct ring tones on our Snom phones for

Dear all, I have Asterisk managed by a FreePBX web console, and I want to add an external dial code, in order to dial 9 to get external line/tone for outgoing calls to the GSM network through my GSM gateway. Where from Asterisk/FreePBX can I setup this feature ??? Thanks a lot. Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail.

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

Sehr geehrte Damen und Herren, Ab dem 23.12.2016 bis einschließlich 06.01.2017 sind wir im Weihnachtsurlaub. Ab dem 09.01.2017 stehen wir Ihnen wieder wie gewohnt zur Verfügung. Wir wünschen Ihnen und Ihrer Familie ein frohes und gesegnetes Weihnachtsfest und einen Guten Rutsch ins neue Jahr. Mit freundlichen Grüßen Ihr Hardy Barth - Team EDV- u. Elektrotechnik Hardy Barth GmbH

hello r development team, i'm building R 2.3.0 on solaris and when i run the 'make install' i'm getting a syntax error during the "installing etc ..." which causes the installation to fail. i get this error whether i use gnu-make of sun-make, see the error and reasons below. gmake[1]: Entering directory `/export/medusa/darin/build/R-2.3.0/etc' installing etc ...

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

This is a cross-list post. I''d like to announce the initial release of a puppet request-tracker module, darin-rt, for managing Request Tracker. The module will install request-tracker and database packages, install request-tracker extensions (if packages are available in the repo), and create basic request-tracker queues. This is also a request for help in extending the module to

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I am attempting this: [from-internal] include => set-alert-if-local [from-internal-original]

I am found and application called Visual Dialplan - And the idea seems good, apart from it didn't read the dial plans from a freepbx setup. Are there any other applications that I may try, that you guys can recommend? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081206/0bb3b52d/attachment.htm

Gordon, I did as you suggested, and that seems to have fixed it. Thanks for the suggestion However, this is contrary to the given docs on how to use CUPS with Samba, is it not? Darin Bawden -----Original Message----- From: Gordon Pritchard [mailto:gordonp@sfu.ca] Sent: Wednesday, January 08, 2003 11:38 AM To: Darin Bawden Subject: Re: [Private] RH8 and Samba 3 On Wed, 2003-01-08 at 09:06,

Hello all, Does Samba properly support 2012_R2 domains? If so, what is the earliest version of Samba AD that supports it? I see that the most recent versions support ad dc functional level = 2012_R2 in smb.conf but I am unsure if I can safely run 2012_R2 functional level on older versions of Samba. A little background: In my test environment I setup a Samba 4.20 AD Domain Controller with

When you are dialing a regular extension you might do something like this: exten => Dial(123) that would presumably dial extension 123. but when one is using freepbx to admin the asterisk, and building a custom piece of dial plan code, how do you access a ring group? exten => Dial (Local/601 at xxxx) how does one know what context to specify? and is the local keyboard needed? is Local

I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make calls from the Toshiba to Asterisk and internal calls from Asterisk to the Toshiba. What I can't do is make an call with an outside destination from Asterisk to the Toshiba. The Toshiba is looking for 9 to grab an outside line then it expects to see the 10 digits. In the FreePBX dial plan I use 9|. which sends 9 plus the 10

Hi now I am frustrated with configuration for setting up voicemail notification to my mobile using some sms gateway. I am not getting any info about this can anyone help me out how to do this and what all I have to do in Sip.conf, ext.conf, voicemail.conf. Please clarify me the complete procedures to complete this project Thanks Darin IV

Hi Darin, Le 05/06/2024 ? 17:34, Darin via samba a ?crit?: > > > Hello Havany, > I am just going to jump into this discussion. > Welcome! >> We try 2 scenarios : - A "Big bang" migration to an new domain made from scratch : but we need to migrate all users, computers, laptops, filers without loosing profiles, files server access... In a short time (1-2 weeks

Hello all, The Asterisk Management Portal (AMP) is now known as FreePBX. FreePBX 2.0.1 is now available for download. A **BIG** thank you goes out to the project developers for all their hard work, and to beta testers for running FreePBX through it's paces! This exciting new release boasts a better user experience, additional functionality, and a new module system. The module system is