similar to: Asterisk Manager API - multi "Originate" cal ls

Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read in places that you use "originate" command and wait for an event back, does that mean you cannot place another "originate" until the event comes back ? Is it true that multiple API connections to Asterisk

<<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

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

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

Hi all, When the 'Originate' command is issued with 'Async' open set to 'yes', I got the response right away with the correct 'ActionID'. What follows is the 'Newchannel' event with a 'Channel' ID, but their is no 'ActionID' to tie it back to the command. How do you guys deal with this? -------------- next part -------------- An HTML

Hello all Has anyone had any success with the Manager API ? I am trying to check an extension status without too much luck I have the following <?php $fp = fsockopen("127.0.0.1", 5038, $errno, $errstr, 30); if (!$fp) { echo "$errstr ($errno)<br />\n"; } else { $out = "Action: Login\r\n"; $out .=

I have started to receive a lot of positive response for the Asterisk@Home project. For those of you unfamiliar with this project the goal of Asterisk@Home is to make a full featured version of Asterisk very easy to install. We have created a 1 step .iso that installs RHEL (RedHat Enterprise Linux) and Asterisk. It includes a web GUI that allows easy editing of the Asterisk Config files.

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

Hello, all! I'm looking at the wiki page and info on the mailing list and I'm getting conflicting info... I am using the manager API from the telnet CLI and I am testing creating calls with it. I login with events: on and I can originate calls just fine. However, when I set ActionID on an Originate, I cannot see anywhere where that actionid carries into the Event output. But I found

Hi, I'm currently trying to originate a call with 2 variables set. I tried doing it via manager API and call File and both failed, because the vars were not separated. I'm using Asterisk 1.2_beta1 on this machine Can anyone here verify wether this is a bug or just a stupid error on my part? This is the callfile I tried to use, after the manager way failed: Channel:

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

Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use these settings to originate calls using the sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel:

Hello User-List, is it possible to play a sound file directly to a caller channel? Like this AMI command Action: Originate Channel: SIP/20-00001d41 Application: Playback Data: /path/to/audio/file I get an Error Message. My intension is to play a sound file to a caller and the other callers don't hear this. Can someone help me ? Thanks a lot Bye Daniel

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

Hi all, How can I playback a file within an active call? I've tried with ChanSpy whisper mode like this (using AMI): Action: Originate Channel: Local/9999 at default Priority: 0 Variable: MSG=test Application: ChanSpy Data: SIP/1234-123 Async: 1 and in the dialplan: [default] exten => 9999,1,Answer() exten => 9999,n,Wait(2) exten => 9999,n,Playback(${MSG}) Where

Is it well known that setting the ActionID when connecting to AMI has absolutely no effect? Is this fixed in Asterisk 1.4? If you add an ActionID to your Originate command for example, it looks like the only events that come back with an ActionID associated are the initial response, OriginateSuccess and OriginateFailure. That's it. No other events have an ActionID associated. This pretty much

Hi All, I have a new asterisk setup running at home and am very happy with it. One thing that I am trying to do is to take various actions in the dialplan *if* a particular phone is registered/authenticated/connected. For example, if someone dials *me* and is shows that I am connected via my softphone, to try it instead of my deskphone (and possibly notifiy the user in advance that it is

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Thanks in advance... Anahi Ludue?a