similar to: [Fwd: using AMP custom extensions]

Hi all, I'm trying to setup a custom extension in AMP (yes i can code it by hand but the on-site admin that does moves & changes cannot). I've tried the following > add cutom extension 600 in the dial box i have Dial(IAX2/username:password@host/$EXTEN@from-internal) this doesnt work as these lines are added to extensions_additional.conf exten =>

Hello all, A new version of the Asterisk Management Portal is available for download. Please visit the AMP homepage at http://amp.coalescentsystems.ca Upgrade instructions are at http://amp.coalescentsystems.ca/UPGRADE Use our Sourceforge mailing list and forum for discussions about AMP. 1.10.006 ChangeLog: - Use extensions_custom.conf for customizations. Sample included. - Added option

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

Greetings *`s, I am having what appears to be a small problem, but the frustration is erally getting to me, what am I doing wrong here ? I used AMP to set up a custom menu, so if caller presses 1 it goes to ext200, if caller presses 2 it goes to ext201 etc etc... Now I have created a third option that when the caller presses 3 it must play a sound and hang up. No rocket science yet. When

Hi All, I've installed asterisk and manually configured IAX/SIP users. Everything works fine, I'm able to call other extensions. But when I installed AMP and created new extensions, I'm not able to call those extensions. I get the message that the extension is busy and it is forwarded to voicemail. What am I missing here? The workaround I found is by modifying the

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

----- Original Message ----- > From: corngood at gmail.com > To: llvmdev at cs.uiuc.edu > Sent: Saturday, April 13, 2013 9:13:57 PM > Subject: [LLVMdev] C++AMP -> OpenCL (NVPTX) prototype > > After reading about Intel's 'Shevlin Park' project to implement > C++AMP in > llvm/clang, and failing to find any code for it, I decided to try to > implement >

For all you AMP users: Why the heck is there no area in AMP to manage agents? I see you can add static agents under the queue settings, but there is no area to assign dynamic agents. Is every extension (or user) considered a dynamic agent? Is it setup up such that, as default, every extension is considered a dynamic agent that can log into any queue? (Yes, I realize you can add a password to

>From the command prompt type: help-aah This will give you a list of commands to change passwords. For example: Commands Descriptions ----------------------------------------------------------------------- config set the local time zone and keyboard type netconfig configure ethernet interface genzaptelconf autoconfig Zaptel

After reading about Intel's 'Shevlin Park' project to implement C++AMP in llvm/clang, and failing to find any code for it, I decided to try to implement something similar. I did it as an excuse to explore and hack on llvm/clang, which I hadn't done before, but it's now at the point where it will run the simplest matrix multiplication sample from MSDN, so I thought I might

Hi all, After pinging the AMP userlist at SourceForge, I got a great step by step explanation as to how to set up AMP for Apache2 (some maybe obvious stuff that wasn't in the Newbie Guide). Thanks to Jason Becker of Coalescent Systems. If anyone needs me to post Jason's instructions here, I can, but they are in a thread called "AMP noob issues with Apache2/Suse9.2" at

Ladies and Gentleman, I have setup Asterisk and AMP. They are working with out error. But now I need to get everything going. I have Voicepluse and they give sample iax.conf and extensions.conf files but that does me a little good as I am using AMP. Is there some docs somewhere on how to confgure once I have AMP up and running? I am not a telephony guy and alot of this looks like greek to

I've just started using AMP and found that I have a problem with escaped characters in config files. In particular, I have a custom config item that needs a semicolon in... SetVar(_ALERT_INFO=info=auto-answer;delay=1) To get the part of the line after the ; to be accepted by Asterisk as a non-comment it needs to be escaped with a backslash, but I have found that I need to put two

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I'm in charge of two Asterisk servers. One of these servers is running an Asterisk@Home version under CentOS, the other one is running Debian Sarge. The Debian server has been installed by me using this howto: http://www.astpp.org/index.php?n=ASTPP.DebianSarge-AMP On both servers, the AMP version is 1.10.010. However, the

I'm using Asterisk@Home with the AMP interface and I'm having troubles getting incoming calls working properly. In AMP, I have it set to take incoming calls from PSTN, during regular business hours, to be sent to extension 201. The include statement for extentions-additional.conf is uncommented in extensions.conf; and I've verified AMP successfully changes the config files. However

Hi all, I would like to point out a problem I observed. I installed a new asterisk server, very similar to another. So, after complete installation of asterisk and AMP, i tried to import back a ful AMPl backup from the first AMP / Asterisk box Everything was (quite) OK. the only problem was that when I dialed an internal extension i got always "Returned from dialparties with no extensions

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension