similar to: Asterisk Manager: equivalent of 'show channels'?

Hello Has someone written a web page (preferably PHP) that simply shows what extensions are currently online? Thank you.

Hello list, we are developing a new application that uses the Manager API in order to find a set of channels where variables are set in a predefined way. To do this, we currently send a Status command to obtain all available channels and then query them all, one by one, fot the status of a certain dialplan variable. As you can imagine, this gets rapidly pretty tedious as the number of

Given a 1.6TB ZFS Z-Raid consisting 6 disks: And a system that does an extreme amount of small /(<20K) /random reads /(more than twice as many reads as writes) / 1) What performance gains, if any does Z-Raid offer over other RAID or Large filesystem configurations? 2) What is any hindrance is Z-Raid to this configuration, given the complete randomness and size of these accesses? Would

I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like

Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. *AMI login:- * *login.php* <?php $socket = fsockopen("127.0.0.1","5038",

Hello group, I put together a simple PHP based conferencing manager for Asterisk 11.3 I used ODBC MYSQL for conference IDs and PINs. All this is working as desired but I would love to add an active conferences display to the front end. It seems to me that AMI is the way to go but I have no idea how to accomplish this or even where to begin. Any guidance is appreciated. Pat...

Hi Guys, I am trying to make a web form where a person is allowed to put in $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller ID. There are a few problems that I am facing with Asterisk AMI Originate command. The reason why I want to use the darn AMI Originate is because I am sending calls to mobile phones and I want to have some accountability and to know if a call was

I'm looking to hear from any application developers in Argentina specifically or other South/Central American countries. Please understand this isn't a dial plan or remote installation I'm looking for but an actual application developer. If this fits your description please email me details on; Size of company (number of full time/versus contract staff) Location Previous

Have a question , Suppose i had a client tell me that he can access the web page but it takes long time to view the pages the website is a static website ( suppose this website does not server dynamic data or does not connect to a database )... what would one check other than : the server load ( cat /proc/loadaverage ) , the Apache logs , the number of client connection ( netstat

Wow! I've been *-ing for about 6 years and had literally no idea about that! I can see a way I could put it to a different use, but it seems to be a bit of a sledgehammer to crack the walnut of "how many current callers" compared to one line of (albeit hacky) dialplan. That's making me sound ungrateful. I don't mean to be! On Sun, 14 Jun 2020, 22:39 Steve Edwards,

Hi, I'm getting errors while originating a call through AMI. [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe Asterisk version 11.0.1

Hello Everyone, I am making a simple index.php file which will allow a web user to enter his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged. Following is the index.php and the contents of extensions_custom.conf. When I submit the form nothing happens. I don't even see Manager Connected msg. Your input will be much appreciated. I am thinking I have some syntax

This patch fixes two spelling mistakes in code comments, which means no functional change: still-extant -> still-existant the -> then Okay? Index: sshbuf.c =================================================================== RCS file: /cvs/src/usr.bin/ssh/sshbuf.c,v diff -u -p -u -p -r1.19 sshbuf.c --- sshbuf.c 2 Dec 2022 04:40:27 -0000 1.19 +++ sshbuf.c 13 Aug 2024 16:39:12 -0000 @@

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data)" IAX2/FONEMITEL123456 1296197222 at entryhome<mailto:1296197222 at entryhome> Ringing

Hello, I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success. I have an application that sends an action Originate to AMI for calling, it's working well, but when i see to Asterisk's CLI, i see 2 calls for just one originate: pftestes40copiabh*CLI> core show channels verbose Channel Context Extension Prio State Application

Has anyone used the G option with the Dial app? I'm looking for a way to control the called party leg. Specifically, I'd like to pass a few variables to the called side for some call control. Here's a synopsis of what I'm doing: Make outbound call w/ AMI Originate action. Called party answers ("Customer") Customer identifies himself, and now I use Dial w/ the G

Hello again List! I wrote something to allow me to easily view the state of Asterisk SIP devices, ZAP channels and Agents through a WEB interface. The pages are AJAX enabled to watch events and automatically update the status buttons. It is an Asterisk Switchboard created solely from Asterisk Events. It does not poll the Asterisk Manager (although it does have functions to run manager

On Tue, 13 Aug 2024, Tobias Stoeckmann wrote: > This patch fixes two spelling mistakes in code comments, > which means no functional change: > > still-extant -> still-existant extant is a valid word > the -> then ok