similar to: AsteriskNOW generation of zapata.conf file

I am attempting to use an IAX2 channel between two Asterisk systems. This would seem to be a normal thing to do. I actually want to trunk traffic between the two that are in remote locations. However, I have started with what I think is a simple configuration, which should allow for one way calling. Attached are the pertinent parts of my configuration files. I am attempting to place a call on

I am new at this. I have read "Asterisk: The Future of Telephony" and have installed AsteriskNOW (beta 4, due to the dual processor problem in beta 5). The GUI interface does not seem to provide the capability that I need, although I have modified the *.conf files to successfully create what I need. Given this, I would like to install Asterisk on a distro. I am most familiar with

In looking at the safe_asterisk script, it would appear that it is encouraging the running of the Asterisk application as root user. My natural inclination is to run it as a non-privileged user. What is recommendation? +++++++++++++ This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415

Thanks to all that have helped me so far. I have made a lot of progress. I am able to make prilib and zaptel. Now to Asterisk... After installing the kernel source, I have: # cd /usr/src/linux # make cloneconfig # make prepare-all Then I have run ./configure in the asterisk-1.4.4 directory. I have: # make clean # make Which goes through a number of compiles and then ends

I am installing asterisk on a second box with OpenSuSE 10.2. I have installed libpri, run menuselect/configure and then make. The make stops at the last line shown below. Looking at the processes, the current process is running sed. Not sure from where. Any ideas? ... checking for ar... /usr/bin/ar checking for cp... /bin/cp checking for ln... /bin/ln checking for mkdir... /bin/mkdir

I have recently installed Asterisk Now Beta 4 (32-bit) with three Digium cards - 2 Wildcards TE400P (one with 4 FXS modules and one with 4 FXO modules) and a TE205P dual E1/T1 card. The problem is with the TE205P and configuring the /etc/zaptel.conf and /etc/asterisk/zapata.conf files. I go in and add the required lines to configure the TE205P and restart Asterisk. All is fine until I reboot the

What is the difference between the 1.2.x and the 1.4.x branches? I am starting to deploy asterisk from scratch. I expect that I should use the 1.4.x branch, but don't know the rationale for the two. +++++++++++++ This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415 or visit

Hi all, When i try to boot an ec2 instance using: *puppet node_aws bootstrap --server myserverOnAWS.com --image ami-61ad0108 --keyname test-malcom --login ubuntu --type m1.small --keyfile test-malcom.pem -g puppets --verbose --debug --puppetagent-certname test* I get this error: *notice: Signing certificate ...* *debug: certificate_status supports formats: b64_zlib_yaml pson raw yaml; using

Hi, I use asterisk branch 1.4 and gui 1.4 as well. I have the following situation: When I try to make gui configuration by http://localhost:8088/asterisk/static/config/setup/install.html I can see that application logs my user correctly but there is no browser window shift to the next page. it stays at the logging one. I get the following info in the console: [Feb 4 09:33:09] == Parsing

Anyone successfully implemented a solution for allowing ZapBarge call monitoring only for a specific group of agents calls? The issue I see is that the feature only works on zap channels, and all of the agents (in many cases) are IP phones. Allowing ZapBarge and ZapScan on the TDM PSTN (t100p) interface has privacy issues for senior managers, but would allow all outbound zap calls to be

Hi, Is it possible to monitor conversation of logged in Agents? Currently I am using ZapScan to monitor incoming calls, but I would like to monitor individual agents. raj

Hi all, It's my first time using usenet... Well, I work on an ISP and we have a linux box acting as a bridge+firewall. With this bridge+firewall we control the packet rate per second from each client and from our repeaters. But I can`t measure the packet rate per IP. Is there any tool for this? Actually, what I want is to measure the packet rate per IP and generate graphics with mrtg or

Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0

I am running version 3.2 of ferret and would like to upgrade to a more current version. will my existing 3.2 index files be compatible? thanks, AK -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/ferret-talk/attachments/20070910/742890f2/attachment.html

Hi, I currently have some systems on AsteriskNOW 1.7 & have been happy with its clean simplicity & reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick with their previous philosophy of keeping everything very vanilla & making it clean & simple for someone who understands how to manage CentOS, FreePBX, tftp, ntpd, etc. but

Hello, Is it possible to set things such as parts of config files are edited though AsteriskNOW GUI while other parts remain "hand editable" ? AsteriskNOW website include screenshots but not much information (such as user manual) beside that. This thing was the one that kept us from using freepbx (let me say I don't mean it's not possible with freepbx : I mean we couldn't

I just downloaded and installed the AsteriskNow appliance (http://www.asterisknow.org) . This looks like it has lots of promise. Anyone know what the secret is to being able to actually login to the root console? thanks, Geoff

I currently have a customer that a previous employee setup with Gentoo\Asterisk. I'm looking to migrate to AsteriskNOW. They have a custom menu, which I would assume is easily replicable in AsteriskNOW. The only other thing I can think of is the sound bites for the menus. Does anyone have any advise or migration recommendations for this move? -------------- next part

Hello Asteriskies, Has someone tried www.asterisknow.com ? What is the package manager used? And what is the added value compared to the well maintained debian based asterisk ? Thanks, -- Cheers, Maxim Veksler "Free as in Freedom" - Do u GNU ?

Guys, I've downloaded AsteriskNOW few days ago so I'm new to this product. The first issue is on service provider area. I've already used a VoIP account already configured with my ISP, it works fine! This configuration has been used until now with the client SJphone, Now I would use this profile as main VoIP service provider to setup in AsteriskNOW. Here are the profile detail as