similar to: Asterisk@Home

Hello, You can do either, you can send multiple Originate actions in a long line without waiting for a response back(although the responses do usually come back very fast) or you can open multiple connections using each one to Originate a new call. We use the multiple connection method in the astGUIclient suite because if you get a pause or lag in Manager output on a single connection(which does

> This wiki should cover most of the basic stuff that gets asked over and >over again just to help reduce the amount of repetition that most of you >have probably noticed takes place here. Problem is, Wikis in general suck and voip-info.org in particular is quite useless except as a random clicky-clicky exercise. You ever use the search on voip-info.org? It's almost like someone

Can anyone provide me with a step by step on how to set up Windows Messenger on a Windows XP Pro box as a SIP client with asterisk? I'm interested in doing various tests of my asterisk server from the Windows perspective of the world. In the alternative if someone could provide information on another Windows based fully functional easy to configure iax or SIP client that would suffice as

I'm trying to set a channel variable and make it available to another channel: I thought that if I SetVar(_SomeVariable=SomeValue) or SetVar(__SomeVariable=SomeValue) then SomeVariable would be available in the destination channel. However __SomeVariable, _SomeVariable and SomeVariable are all blank. The scenario: Agents logon to the queue using callbacklogin. From what I can gather

<<I am getting "Message: Originate failed" even the phone is ringing on the other end of the line.>> Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see. Bill Seddon ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf

Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian

Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz

Hello all: Can anyone help me with finding the best locations for getting setup and other documentation for *. Thank you. Phil Menico www.xtend.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050110/6b26b235/attachment.htm

I appologize if this was already answered somwhere on http://www.voip-info.org/wiki-Asterisk, I'm sure it probably is. And if you wish to just point me to a link that would be appreciated. I am very new to asterisk and unix all around, so these questions may sound rather ignorant. First being, how do I setup asterisk to point to another asterisk server and make all the lines which should

I am running Asterisk at home on one of my systems here. It uses Centos 4.2. It does NOT have a GUI interface. Command line only at console or via SSH (yeah, I know that I was working on another server to get GUI tools working through SSH, but I am NOT going to muck with AAH build). yum update is potentially dangerous, when it replaces the kernel. You can loose all of your Zaptel driver

Hello every one. This is a question done by me, not yet answered. Please, help. I: 1. Run install-pdf from linux to support faxes on my asterisk. 2. Made the configurations throuhg AMP in a. Setup->Inbound Routing->(the only route I have)->fax extension->System b. Setup->Inbound Routing->(the only route I have)->fax email->(my email) c.

Hello all! Is there any possibility to determine the IP address of a caller in my dialplan? I would like to have a predefined channel variable like ${CALLER_IP} but it seems it doesn't exist (http://www.voip-info.org/wiki-Asterisk+Variables) .. is this list complete? Are there any other possibility to store the SIP/IAX caller's IP address on every call? Thanks Niels

Hey, this is happening to anyone who I try this with. We get into the DISA, then hear the dial tone. Dial 1 then start dialing the number, and it hangs up. I thought adding a wait time after the DISA may help, I was wrong. Here is what I have thus far in the DISA extentions. [DISA] exten => 7,1,DISA(no-password||"Scheda" <565> 455-1337) exten => 7,2,Wait(45) exten =>

I made the observation that I'm able to make a call with my SIP client (kphone) even when I'm not registered/authenticated. Of course, when I'm not registered at asterisk, people can't call me, but it's still a huge security hole, that unregistered Clients can make calls. Is there a way to tell asterisk to only allow registered clients making calls? I know about the

Hey, I recorded this intro, and changed it to a gsm file in the shell, and I'm getting an error saying that it isn't in the directory at all when it's sitting right there. I don't know why that is. If you want to hear it, it's http://scheda.underfireradio.com/astintro.mp3 I don't know what the matter is, I've tried renaming it, copy and pasting it in there, deleting

Does anyone know if SIP will/is support on handheld PCs such as the iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems like a solution to provide a wireless based sip phone for any user would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can

I am doing a yum install netconfig on the new 5.1 and nothing gets installed. Was it removed? replaced with something else? yum provides netconfig does not show a "bin" command just false matches. Is there a new command? Thanks, Jerry

Hello, When I am booting my gentoo domU I get the following output to the console and then it just hangs: (full boot output can be found at http://pastecode.com/19771) NET: Registered protocol family 1 NET: Registered protocol family 17 Using IPI Shortcut mode EXT2-fs warning (device sda1): ext2_fill_super: mounting ext3 filesystem as ext2 VFS: Mounted root (ext2 filesystem) readonly. Freeing

Hello all, Today after a reboot of my server, I encountered the following error: > # xm console hackbloc > xenconsole: Could not read tty from store: No such file or directory I have not changed anything and have never run into this problem before. I have no idea what to do here, I mounted the image and checked that /dev/tty* were still there and they are. xm show lists the server as

Hey, I'm currently using the GotoIf application to set it so if certain caller ID's call my number, it will transfer it to my cell phone, here is the code I have so far. I get an error message that states "call rejected by 198.22.67.70: No such context/extention." when I call the number from my house number. Anyway, here is the code I have. [inbound] exten =>