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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

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

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

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

Thanks! Thanks! Thanks! I've got it work!!! :-) Message: 13 Date: Sun, 16 Jan 2005 12:17:21 -0000 From: "Bill Seddon" <bill.seddon@lyquidity.com> Subject: RE: [Asterisk-Users] failed to compile zaptel on redhat To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <ECOWS02MB8QDGMou3fY0007760a@smtp-out2.blueyonder.co.uk> Content-Type: text/plain;...

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

<<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

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 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.

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

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've installed Asterisk on Linux Red Had 9. Now, I was trying to set up a GUI based system for the PBX. I downloaded some packages, but I have to have Perl running CGI scripts through the webserver. It does not allow me to. I am able to run a basic script that just just prints out html messages and nothing else. However, when I try to run am-main.pl or config.pl or any other

Colin wrote: A lot of good sensible stuff. Well done Colin. Bill Seddon Lyquidity Solutions Limited -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Colin Anderson Sent: February 23, 2005 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-U...

Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -------------------------- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24

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

Having just run the command "sip show channels" I get a list of channels even though there is no one on the phone (we only have 4 so it's easy to tell). Here is what I get: Peer User/ANR Call ID Seq (Tx/Rx) Format 192.168.0.22 (None) 4c81ac8e90c 00101/00000 UNKN 192.168.0.22 (None) 984ee48048d 00101/00000 UNKN 192.168.0.22

How can Asterisk be configured to execute some number of dialplan commands when it is started or restarted? I want to be able to populate the registry (using DBPut() commands) to store some information each time Asterisk starts. Such information could, of course, be stored in a database and perhaps that will be the long term objective. In the meantime I'm hoping that it is possible to use

<< Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits.>> I have seen exactly this problem. Even IVR failed to work. Got an RMA from the supplier and they exchanged with no questions. Bill Seddon -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C