similar to: Pager Notification Script

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

Egad, not again with Broadvoice! Anyhow, I recently installed AAH and configured my TDM11B and got that and some SIP phones working. I still have some issues to work out, etc, but my current problem is Broadvoice. I have checked out all of the online resources, including the recent list exchange about the recent changes made by Broadvoice. However, the one thing I have found to be consitent in

Hi; Before I tell a customer that this would require custom development I figured I would ask here. Does Asterisk support pager notification of new voicemails out of the box? Or do I need an AGI script to do that? Also, if I want to call a number from an automated program in Asterisk and get the DTMF tones entered by the user on the other side, is there an easy way to do this? Best

I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's? What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call. Thank you for your time. -- Tomislav Par?ina Lama Computers Split

I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is possible to customize the email message sent to the pager email address. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 agoss@ntad.com

Just installed Asterisk 1.2 on a brand new clean machine running RedHat 9.0. I have a TDM400 card inside. When I boot, the card seems dead. When I do: modprobe wctdm modprobe Zaptel the lights come on and all seems fine, until I reboot that is... After a reboot I have to repeat the modprobe. I shouldn't have to do a modprobe every re-boot should I? How do you get the drivers to load

Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten => s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly appreciated! -------------- next part -------------- An

Since the list community has done so much for me in my humble asterisk beginnings I have put together a simple little script written in php that serves as a paging reminder script. If anyone is interested in a copy of it contact me off list and I'll forward you a copy. The basics of the script are as follows: It queries an asterisk inbox of your choosing for the existence of a file that

Ahoy, I made this pager, " def list @item_pages = Paginator.new self, Item.count, 10, @params[''page''] @items = Item.find :all, :conditions => "category_id = #{params[:condition]}", :limit => @item_pages.items_per_page, :offset => @item_pages.current.offset @categories = Category.find_all

I noticed while reading some posts that people were looking for a complete NPA-NXX list for all area codes and prefixes. We happen to have the entire database. So I am making it available to the public. Help is available at: http://download.sixtel.net/npa/help.txt (Caution, 20meg files) Mysql Insert for this is available at: http://download.sixtel.net/npa/npainsert.txt CSV data for

Does anyone on-list have direct experience with the new analog cards from Sangoma? I'm thinking about FXOs with echo cans. Need 2-4 ports but don't want to go through another TDM400 style experience. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

R-users, (forgive my return adres) I've been breaking my head why R---which i find fabulous, by the way---does not pipe interactive output through a pager (more or less), like it does with help(), or like GNU Octave does with arrays with more than terminal height rows. Maybe it is my installation (Debian/GNU Linux). Maybe it is my configuration, but i don't think so because with

Yeah....I think that every install I have done the first thing that happens is "why is there a delay before the call connects?" and the answer is "you have to hit dial or wait 10 seconds". -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Rymes Sent: Thursday, August 11, 2005

I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4, and Zaptel 1.4 The Digium cards installed are TDM2400 and TE110P. Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9 Now when I run ztcfg I get the following error message: (CAS signalling on span 2 conflicts with Clear channel on channel 40) --NOTE: signaling was spelled wrong in the error

I have run into two programs offering Virtual users. This allows a person to enter a code and take over any extension, another code is used to release the feature. The two programs are Ipmanager and Scopserv. I hate using GUI's as I have not seen a truly good one. I would like to implement this feature without the GUI. Any Ideas! From what I gather they are using Macro's, of which I have

Hi list, I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am getting this error : WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such file or directory with the default musiconhold.conf file. When I change musiconhold.conf to this: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 (and have converted all the wav files to mp3 and put them

I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? -- Thank You Amit Nepal

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

I've experiencing some difficulty passing inbound calls from the PSTN, through a large Asterisk switch and down our network to a Cisco 1751 router. This router has 4 FXS ports and is running IOS 12.3. Outbound dialing from phones on the FXS ports of the router works flawlessly, but inbound calls fail as though the Asterisk server does not see the extensions representing the FXS ports as

Was wondering why R help in recent Fedora only provided --More-- prompt for viewing, rather than allowing arrow key navigation as on RHEL. It turns out it uses a system command for paging and the current default pager is more: grep PAGER /usr/lib64/R/etc/Renviron PAGER=${PAGER-'/usr/bin/more'} rpm -qf /usr/lib64/R/etc/Renviron R-core-4.4.1-3.fc41.x86_64 This primarily affects function