similar to: Paging Phones stay off the hook if you dont wait long enough.

Does anyone know the secret to get the GXP-2000 Message waiting lamp to illuminate? Or can point me toward some docs that might explain it? Thanks! --Jeffrey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060501/7ffc668f/attachment.htm

Hi, I am desperately trying to make a little application work that is only available for Win or Linux machines. My computer is a G4 Power PC Mac running Tiger. Darwine is not compatible PPC. There was an old version, but in the readme file it said that I would not be able to run .exe files on a PPC, so I don't understand the point of the application... Qemu doesn't work. OpenLina is

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

I've had 50+ people download the web components, and other than reports of compile issues, I have not heard if this collection has worked for anyone. I do plan to keep updating the * applications and the web pages, but I have almost meet all of our internal requirements and wonder if anyone else is finding it usefull. My focus has been and will likely stay on the user interface, since I have

Okay...here's one for all of you.... 3 party meet-me conference: Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM, no VoIP at all involved. No echo at all. Call 2: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk. Caller immediately hears his own echo Call 3: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk.

Hello everybody, I want to use the TIMEOUT() function, but in the CLI the "show functions" command only shows 7 custom functions: QUEUEAGENTCOUNT SORT CUT CHECKSIPDOMAIN SIPCHANINFO SIPPEER SIPHEADER In addition, sometimes I get the debug message "function LANGUAGE not registered". How can I install those functions? I'm using Asterisk 1.2.10. Thanks in advance, --

Hi all, I am using asterisk1.2(vicidial). I am using like pbx . In this how can I confugure the internal conference calls. suppose I have A,B,C,D,E users these all peoples should be internal conferece . for them i was give 101,102,103,104,105 extensions. For this scenario what can I do exact configuration in dialplan and any to edit confugration files please help me . and how can they cut the

hallo, i tried to setup my extentions,conf like this but it never jumps to the busy part (102) asterisk always plays the unavail msg, also when i am connected to another iax channel (conferece room) and no more channel on my client is available. could sombody give me a hint what could be wrong? thanks , alex snd*CLI> -- Accepting AUTHENTICATED call from 81.135.10.114, requested

Hi all, I want to initiate a call from the asterisk to an extension, where I will forward the asterisk side to another extension later (to the conference extension). I can initiate a call uning originate call from an extension to the desired extension, but it would need someone from the originator extension to answer the phone. How can i register an extension to asterisk where it

Hello list, is it possible to extract the Remote-Party-ID from an incoming call in the dialplan ? Is there some kind of function for this ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110316/bda02b4d/attachment.htm>

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

Hi, I have a main Asterisk server, and a server at a branch location connected via a IAX2 trunk. I want to have a queue at the main location that has people from both locations as members. I got this working, but the trouble comes when the round-robin logic selects a member at the branch office to call. If that user is unavailable, their voicemail answers the call, and the main server

Hello, I come with more impossible tasks; I''m getting XML results from a Mono program, but they''re in UTF-16 format. I''m running Lighttpd /w FastCGI, and just need to pass the results through without manging the unicode. Help? Please? Rektide -------------- next part -------------- An HTML attachment was scrubbed... URL:

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want

I have been testing * with some Cisco 7912G's, in hope to trash our Nortel system. One feature our Nortel system has that I will need to fiqure out on the * system is paging. Is it possible to page a group of phones (all phones) with announcements? We are a k-12 school and we use our current phone system to make announcements on the phones monitor speaker. Any direction I can be pointed in

Hi i installed samba 4.9.1 from source on centos 7. I need to setup backup script . I cannot use samba-tool: Command: samba-tool domain backup offline --targetdir=/backupdir Gives me : Usage: samba-tool domain backup <subcommand> samba-tool domain backup: error: no such option: --targetdir Command: Samba-tool domain backup -h Gives me: Usage:

Does anyone know if its possible to have more than one sound card in an Asterisk box and use each one as a paging zone? How about left and right channels of a single sound card? I'm looking to have 2 paging outputs if possible - I've read about using a Grandstream phone on autoanswer but I'd prefer to have the feeds come directly from the * box and go into a stereo amp feeding 2

Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/82ae09ec/attachment.htm

I'm running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9 and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports). Everything seems to work except threeway calling. I can establish a threeway call, but it uses up BOTH FXO lines. Note that I DO have threeway calling active with my Bell service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No