similar to: Asterisk Setup Documentation

Displaying 20 results from an estimated 500 matches similar to: "Asterisk Setup Documentation"

2006 Aug 25
2
[RESOLUTION] Polycom microbrowser issue Error HTTP 406 withIIS
I found this solution from the web and figured I'd share it because it affects all phones getting input from IIS. Map .gif, .jpg, .css etc (in my case I used .xhtml for the Polycom 601) in IIS under your sites: Properties -> Virtual directory tab-> Configuration -> Application configuration -> Mappings tab. Make ASP DLL [..\inetsrv\asp.dll] to handle these files. This
2007 Apr 17
3
Transfer via CTI
I used autodial to allow a user to make a call by clicking on a web directory and placing a call file into the Asterisk "outgoing" directory. That works perfectly for me. What if I want to click on the web directory and transfer my existing call? Is there a comparable interface? Thank you. Phil New York
2003 Aug 10
4
Windows Messenger
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
2005 Feb 24
3
Inheriting variables
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
2005 Mar 02
3
Asterisk Manager API - multi "Originate" cal ls
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
2005 Feb 28
1
Manager "Message: Originate failed" beinggenerated when callee does not pick up
<<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
2004 Dec 09
11
Asterisk@Home
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.
2005 Feb 26
1
Determine IP addres of a AIP/IAX user
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
2004 Sep 08
3
Newbie: Only allow authenticated users to call
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
2003 Sep 23
3
New kid on block
Hi, I am an experienced developer with Windows and familiar with Linux. I am looking for a SIP solution. 1) How does Asterisk compare to VOCAL in terms of support. 2) Is Asterisk free? 3) Where are the docs? Or even better. Where do I start? 4) Will it run on RH9? Thanks in advance. Costas -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --
2003 Nov 09
4
Multi phone presentation
Hi, Does anyone have sample * configuration on how I can get an incoming call to ring all SIP phones (small setup, say 4 phones) at the same time. 1) I would like to pickup up any phone and the ringing should stop (of course) 2) Put on hold and pick up on a different phone set. Do I need special phone features to achieve this? E.g. would the Grandstream 100 do it? Thanks -- Costas Menico
2004 Sep 10
4
SIP on Handhelds
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
2005 Feb 23
3
Able to tell if phone is registered?
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
2003 Oct 15
2
My Grandstream works, but my X-Lite doesn't:no sound after 5sec
This is troubling. Shouldn't your hubs/routers autosense the 10MBPS? ---------- Original Message ---------------------------------- From: WipeOut <wipe_out@lycos.co.uk> Reply-To: asterisk-users@lists.digium.com Date: Wed, 15 Oct 2003 07:53:13 +0100 >Steven J. Sobol wrote: > >>On Wed, 15 Oct 2003, Jon Pounder wrote: >> >> >>Nothing works. Call transfer
2003 Nov 25
4
How to demo * on a notebook
I want to be able to demo * on a notebook at a client's site. This means no FXO gateways; just 2 sip phones (like SNOM) and maybe a softphone (GnoPhone?). I already have RH9 running on my notebook. I would like to have one SIP phone dial and go through IVR before making a choice and ringing the other phone extensions. Of course the notebook would have to be running Asterisk. How can i setup
2003 Sep 28
9
Google newsgroup or Forum setup.
I am sure this has been asked before, but why not use Google newsgroup or at least some forum BBS software instead of this cumbersome mailing list process? -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --
2006 May 16
5
WiFi VoIP Handsets..
Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? Do they support things like WPA etc?? I have heard the battery life can be a problem.. Is this the case? Thanks..
2003 Oct 21
3
Asterisk with Gentoo
I am having serious issues with RH9 when it comes to speed. It may just be gnome or kde but it is slow launching apps. Does anyone know if Asterisk will compile under other distros. Many people are recommending http://www.gentoo.org/ and I am considering it. What is it about RH9 thats different when compiling Asterisk? Thanks -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com
2003 Dec 02
2
Does Asterisk overwrite any libraries?
I am using a brand new RH9.0 installation. I installed Asterisk afterwards so I am not sure if Asterisk caused the problem below. The ps doesn't work. It could also be something else. I also tried installing a some video package. But I thought to ask here first if someone has seen this before. [root@localhost asterisk]# ps ps: error while loading shared libraries: libproc.so.2.0.6: cannot
2003 Oct 24
4
Help with Dev Kit Lite
I installed Asterisk as per instructions in the FAQ on the digium.com site. Double checked it. I also think they have a bug in the zapata.conf where the context should be incoming and not internal. 1) I hear no dialtone when I pickup the phone on the S100U. Asterisk sees the event and displays the message on the screen. I tried dialing but nothing happens. I hangup and * shows the hangup event.