similar to: AMI timeouts

Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input.

Hi folks, We are upgrading from AMI 1.0 to AMI 1.3 and looking for any documents or AMI 1.3 Specifications. I found AMI 1.4 Specification in wiki.asterisk.org but not for AMI 1.3. Can someone provide me the link for AMI 1.3 specification ? Thanks in advance Shishir -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm working on building a cluster on AWS atop CentOS 7. For development, I've been working in the eu-west-1 (Ireland) region, where the AWS MarketPlace provides an official CentOS 7 AMI (ami-e4ff5c93). However, the production deployment is taking place in AWS's GovCloud region for regulatory reasons, and there, I couldn't find an official CentOS 7 AMI. Are there plans to

https://docs.asterisk.org/latest/Configuration/Interfaces/Asterisk-Manager-Interface-AMI/Asterisk-Manager-Interface-AMI-Changes/ On Tue, Jul 11, 2023 at 11:54 AM, TTT <[lists at telium.io](mailto:On Tue, Jul 11, 2023 at 11:54 AM, TTT <<a href=)> wrote: > Is there a web page that lists the AMI versions mapped to Asterisk versions? > > I noticed that the AMI version increased

Hi, I've been developing some CTI software around asterisk for a while, mainly with the help of AMI and fast AGI. It works quite fine, but I have some trouble sometimes with the un-synchronized property of these 2. Let me explain, we have a dialplan like this one : exten = s,n,UserEvent(useful_input_data) (...) a few actions exten = s,n,AGI(agi://127.0.0.1:3333/fetch,queuename) The idea is

Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple

I'm using a perl daemon i wrote to connect to AMI and perform actions. The daemon connects to asterisk via AMI at start up. Is there anyway to check if the AMI connection is still alive, for example every 2 seconds. if the connection is not alive, re-connect to AMI? Also, does AMI timeout after a certain amount of time of not sending commands? -------------- next part -------------- An HTML

>> There are SOME protocol level things accessible using CHANNEL[1] but that's it. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL I am trying to use the CHANNEL function listed above from the AMI. Since it is not an AMI “action”, but rather a dialplan “function”, I’m trying to figure out how to call this from the AMI. Using a telnet session

Hi, I'm seeing some strange behavior when trying to use the latest CentOS 7 AMI from the AWS marketplace. The AMI that we've been using previously is "ami-02eac2c0129f6376b" released January 30, 2019 at 6:40:58 PM Today I saw a new AMI with ID "ami-0c3b960f8440c7d71" that was released February 21, 2020 at 3:50:07 Both these AMIs are owned by AWS account

While doing a security audit on a system I maintain, I stumbled upon an unvalidated use of a variable to compose an Originate request to the local Asterisk instance via AMI. Taking as an example an earlier exploit for FreePBX, I realized that this, combined with Application: System as an injected value, could allow arbitrary code execution. I am in the process of fixing all instances of this bug

Hello List, I was writing something in PHP that connects to AMI and sends a data stream ( example of it: http://slackware-es.com/ami-input.txt ), but the file (voicemail.conf , in this case) does not get fully written. I tried pasting the stream directly through telnet to AMI, and everything *appears* to be OK, but the file is not being completely written. No errors on CLI No errors on AMI

Lets say I have to make 40 phone calls quickly. If I use the AMI interface to originate a call, close the connection, open another connection etc... This works. but is slow... If I open the AMI interface and originate a call - DONT close the interface , get the response, originate another call, some of the calls are missed. even though I get OK response. (All calls are Async mode). If I open

Hi, folks. I'm new to both Ubuntu Linux and Wine. I've been running Lotus Ami Pro for years under Windows 98 and have about 70MB of Ami Pro files. Nowadays, though, printers are not made to support Windows 98 anymore. I had heard about WINE, and rather than buy another Microsoft product, I decided to see if I could run Ami Pro under another OS. My new printer works perfectly under

Hello, I am using a perl script to pull call info from a DB and place calls via telnet and AMI, all on local machine of course. My problem is that I need to capture any response from the carier, such as this taht appears in the CLI: [Mar 1 12:55:50] == Using SIP RTP CoS mark 5 [Mar 1 12:55:50] -- Got SIP response 503 "No Circuit Available" back from xxx.xxx.xxx.xxx:5060 [Mar

Hello all, I am running Asterisk 10.10.0 and I can send Message between SIP's no problem. However, I would like to be able to send send Message to a SIP from AMI Or CLI. I check the ListCommands On the AMI and it don't have MessageSend. Therefore, I try the SendText. AMI: Action: SendText" Channel: SIP/600" Body: This is a test. Message: This is a test. Extension: 600";

Hi. Is there any way to eliminate AMI manager logins from the logging output (without just turning the log level down and thereby losing lots of other stuff as well)? I'm running Asterisk 13.14.1 as a backend service to LVS/IPVS, and using the AMI login as the "service alive" check to see which backend servers are available to take new commands. This results in lots of [Jun

I’m in training, so I have to demonstrate something SIP related. I figure it would be cool to hack a call, hanging it up while in progress from outside Asterisk. Doing so will demonstrate use/knowledge of ARI, AMI, SIP, route-sets, UDP, etc. Practical value: zero :) Who knows, maybe this will have an actual application for someone someday. In practical terms I think building a proxy

Hi, the updated Amazon Web Service AMI (ami-4ac6653d) has a serious bug. It is not possible to resize a disk after creating a new volume with a size greater than 8GB. resize2fs does nothing. The previous AMI worked perfectly but is no longer available. This bug is a show stopper for using CentOS on AWS. Regards, Thorsten PS: It would also be great to have a CentOS 6 AMI with HVM

Hi, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...

Hi, i'm currently using Originate command on AMI, i can call a certain channel like a SIP user SIP/1000 then once 1000 is answered it dials out to amobile or landline. Would just like to know if i can use AMI to dialout to a mobile or landline first (instead of SIP user) and once answered, dial another mobile or landline again. If not is it possible to call a macro from the AMI? i think