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Questions tagged [asteriskami]

The Asterisk Manager Interface (AMI) is a socket-based protocol for controlling the Asterisk telephony engine.

The Asterisk Manager Interface (AMI) is a socket-based protocol for controlling the Asterisk telephony engine.

218 questions
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How to execute a dialplan by asterisk's AMI API?

Couldn't find a specific answer for this. I'm a newbie to asterisk and AMI. I need to auto generate calls using asterisk and pass parameters to an AGI program. Using a call file seems to generate the call first which is not wanted. So, how do I use…
Luchman
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Asterisk AMI events sometimes missing

I have a Python service which connects to Asterisk via AMI and listens for events to detect when a call has begun. This seems to work on most of the Asterisk servers I connect to. However, on a few of our servers we just don't see any of the AMI…
Eli Courtwright
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Originate a call with text to speech message

I use the Asterisk-Manager package for NodeJs https://www.npmjs.com/package/asterisk-manager and have a tape announcement as a text which must be translated via text to speech. When I try to call an outgoing phone number how can I setup the text to…
user9945420
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How to set AGI variable in Python?

I have next agi script test_agi.py: This script make checking is called number is Really number or not using HLR request (API) import urllib.request import json from pyagi.pyagi import AGI agi = AGI() dst = agi.env["agi_dnid"] url =…
SergeyMoroz
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How to make outgoing call via JAIN-SIP to asterisk

I try to migrate my project from asterisk-ami to SIP protocol. However, I couldn't find any examples to create outgoing call via jain sip. Some documentation mentions project sip-communicator but he is outdated. About project: it's notification…
Vladislav Kysliy
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Asterisk Originate Caller ID Display

Situation 1 We use this Originate command to dial out. Action: Originate Channel: DAHDI/g11/(Destination number) Context: queue-123 EXTEN: 123 Priority: 1 Timeout: 30000 Callerid: (Destination number) Async: yes It was success but we would like…
AkiEru
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Execute Asterisk client command with PHP

I'm trying to execute an Asterisk client command with PHP, to display with html. I used to do it normally. When the server was rebooted, I used to give 777 permissions to the file "/var/run/asterisk/asterisk.ctl" and then I could run a command and…
leirbag
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Why is dynamic real time not recommended as per asterisk?

In extconfig.conf they have mentioned that "However, note that using dynamic realtime extensions is not recommended anymore as a best practice; instead, you should consider writing a static dialplan with proper data abstraction via a tool like…
user3705456
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Asterisk: SIP Transfer (Refer) - Changing Headers / Caller ID of Original Caller

Basically I have been implementing a standard TRANSFER via PHP AGI (PAGI) to one of our internal PBX systems from Asterisks. The caller calls into our Asterisk application, we then do some searching and we change the caller id which works perfectly…
Warren Doyle
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Asterisk got hang CLI restart not work 12.8.1 on EC2 micro

I have FIAF with asterisk 12.8.1 running on Amazon EC2 instance. I have created 20 extensions. I have created a JAVA AMI connection which is having 3 thread. Each thread have connection to asterisk AMI with Event: system, command and OFF. Randomly…
Dhananjay Kashyap
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Asterisk queue: going to failover destination after 3 retries?

I am working on Asterisk 12.8.0. I want to know if it's possible to configure a queue in order if a caller has called 3 times 3 differents agents and these didn't answer him, the caller is going to failover destination? For example : -- Executing…
Pedro
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How to link Originate Action with Hangup event in JAVA AMI SIP

I am using Asterisk Java to make calls to various numbers via SIP channels originateAction.setChannel("SIP/10.0.66.118/0900044314"); But when it comes to event the Channel changes to SIP/10.0.66.118-0000003f Please suggest how to link the event…
Anuj Kumar
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CDR report does not show destination number (adhearsion - asterisk)

I have been using asterisk to connect to adhearsion for my system. System originate call to a sip number by using follow script. Adhearsion::OutboundCall.originate number, from: call_from, controller: controller , …
achyutdev
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Originate call using adhearsion

i have searched google, tried to read adhearsion docs and make sense of the examples. i understand parts of it but am not able to put it together. i can edit extensions.conf and do agi(agi://localhost) and handle incoming calls in the dialplan.rb.…
Kinjal Dixit
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How do I get the UniqueId and Channel Name from a call I have just originated?

I am trying to upgrade to the latest version of AsterNet from an old copy of Asterisk.Net. We used to originate calls with AsteriskManager.OriginateCall(), which returned a Call object. That call object had a UniqueId and ChannelName. Now I'm…
Matt
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