search for: extenions

"Douglas Garstang" <dgarstang@oneeighty.com> wrote: >Boy, am I stuck... > >I'm officially ready to toss Asterisk out the window. I have to admit it isn't necessarily all the fault of Asterisk either. It just seems that every option I turn to suddenly ends in failure. I don't know if it's me that's bitten of more than I can chew with this project, or

Boy, am I stuck... I'm officially ready to toss Asterisk out the window. I have to admit it isn't necessarily all the fault of Asterisk either. It just seems that every option I turn to suddenly ends in failure. I don't know if it's me that's bitten of more than I can chew with this project, or maybe Asterisk just isn't mature enough yet. Nothing complicated really....

Hopefully someone here can save my sanity. I have been trying to solve this problem for days now, but just cant put my finger on it. Im new to * so I have probably done something stupid! I have a TDM400P with one FXO module and a FXS module. The main problem I have is not being able to get the extension attached to the FXS module to ring or be able to make calls. It gets a dialtone fine but I

On Mon, May 14, 2018, at 4:52 PM, Stephan Bosch wrote: > > > Op 25/04/2018 om 22:49 schreef James Cassell: > > On Wed, Apr 25, 2018, at 3:20 PM, Stephan Bosch wrote: > >> > >> Specify the ID used for duplicate checking explicitly using the > >> :uniqueid argument (https://tools.ietf.org/html/rfc7352#section-3.1). > >> Using the variables

On Wed, Apr 25, 2018, at 3:20 PM, Stephan Bosch wrote: > > > Op 23/04/2018 om 22:03 schreef Andr? Rodier: > > On 23/04/18 14:18, Stephan Bosch wrote: > >> > >> > >> Op 11-4-2018 om 23:58 schreef Andr? Rodier: > >>> Hello, > >>> > >>> I have tested the sieve duplicate script with success so far, but I > >>>

I am working on a project to unite several local school districts. We will have 14 different districts, every district would have their own asterisk box on location. We all have fiber lines running to a central location at our isd. This provides connectivity to all the districts. 1. would it be wiser to install a asterisk box at the central location of the ISD and have all asterisk boxes

...ake here, but I didn't find a solution. I also tried changing the value directly in trixbox web interface that says "Number of seconds to ring phones before sending callers to voicemail" and nothing happens. I know that trixbox does weird things to my configuration files, but I edited extenions.conf, since it does not get messed up by trixbox. If I use the dial command to dial out with my termination provider (runs on IAX2) the timeout option works just fine. All help is very welcome, Thiago Abra sua conta no Yahoo! Mail, o ?nico sem limite de espa?o para armazenamento! http://...

On 17/08/2018 09:14, Stephan Bosch wrote: > > > Op 14/05/2018 om 23:03 schreef James Cassell: >> >> On Mon, May 14, 2018, at 4:52 PM, Stephan Bosch wrote: >>> >>> Op 25/04/2018 om 22:49 schreef James Cassell: >>>> On Wed, Apr 25, 2018, at 3:20 PM, Stephan Bosch wrote: >>>>> Specify the ID used for duplicate checking explicitly using

If my extension is 22, and voice mail access number is 909, then with exten => 909,1,voicemailmain(s22) I can access voice mail 22, without number and password prompt. But, I want that every extension can access its voice mail without number and password. So, when I put exent => 909,1,voicemailmain(${calleridnum}) voicemail want only password. I want to eliminate password too, so when I

On 23/04/18 14:18, Stephan Bosch wrote: > > > Op 11-4-2018 om 23:58 schreef Andr? Rodier: >> Hello, >> >> I have tested the sieve duplicate script with success so far, but I have >> a question. > > Sieve duplicate script? You mean the Sieve duplicate extension (RFC 7352)? > >> I would like to know if the "duplicate" sieve flag in

Just wondering if anyone has tried MAC OS X and panther. I will like to do SIP to H323, not sure if this will be possible on the MAC because of the Libraries PWlib and OPenh32 for Linux.. Just curious.. Anyway, anyone has an easy guide (step by step) to setup oh323 with asterisk. I saw a guide but i am not very savy on linux. thanks, Francisco ----- Original Message ----- From:

Hi, I'd like to be able to create contexts in real-time when I add new clients to my asterisk box. Currently, I have to create a blank context in extensions.conf and add:- switch => Realtime/@ Is there any way to avoid the step of creating the blank context and simply include all the entries from the database? Thanks Dan -------------- next part -------------- An HTML attachment was

Op 23/04/2018 om 22:03 schreef Andr? Rodier: > On 23/04/18 14:18, Stephan Bosch wrote: >> >> >> Op 11-4-2018 om 23:58 schreef Andr? Rodier: >>> Hello, >>> >>> I have tested the sieve duplicate script with success so far, but I >>> have >>> a question. >> >> Sieve duplicate script? You mean the Sieve duplicate extension

Op 25/04/2018 om 22:49 schreef James Cassell: > On Wed, Apr 25, 2018, at 3:20 PM, Stephan Bosch wrote: >> >> Specify the ID used for duplicate checking explicitly using the >> :uniqueid argument (https://tools.ietf.org/html/rfc7352#section-3.1). >> Using the variables extenion, compose the uniqueid from the message-id >> and the mailbox name. >> > In my

Op 14/05/2018 om 23:03 schreef James Cassell: > > On Mon, May 14, 2018, at 4:52 PM, Stephan Bosch wrote: >> >> Op 25/04/2018 om 22:49 schreef James Cassell: >>> On Wed, Apr 25, 2018, at 3:20 PM, Stephan Bosch wrote: >>>> Specify the ID used for duplicate checking explicitly using the >>>> :uniqueid argument

I have a regular conference call that I usually listen to and record, but there are times when I can't be near an asterisk connected phone to call. In those cases I'd like to place the call from the CLI and have it record automatically. The problem I'm having is that once the call is connected, I need to dial a PIN number to connect to the conference. I've tried setting up

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> <font size="+1">Hi,<br> <br> I have the 4 port TDM card.

Hello everyone, I have an IAX server (asterisk@home) with a FXO card. I have a trunk connected to a voip provide, asteriskout. When I call my server from a public phone, I want to route this call to the asteriskOUT trunk so that I can make long distance calls. How can I setup a secret password in the extension.conf, so that my asterisk server can allow me to make long distance call ? Thanks

Hi Guys, This article was posted few days back. I thought i can get more info here. I am trying to bridge two outbound calls together. (have a program start a context, dial one party and then bridge another party) I thought that the G() flag in the dial application would work. I tried the the following test (continue down a dial plan). One station calls into a context ... in this case, dials

Dear List, I am new in this world (Asterisk) and facing a problem . i want to make some group, base on extensions .so that certain extensions could call to certain predefine number only. let me give u all a short example extensions 1,2,3,4 will be group A , extensions 5,6 will be group B . so group A only allowed to call one or certain predefine number but group B could call anywhere.