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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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Not getting events with PAMI

class VoipEventStart implements IEventListener { public function handle(EventMessage $event) { $a = $event->getKeys(); if( ($a['event'] == "Hangup" || $a['event'] == "HangupRequest") && strpos($a['channel'], 'SIP/') !==…
Anu
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How to get events in PAMI

I successfully insttalled PAMI in my server where asterisk is setup. And i wrote a new php file which class A implements IEventListener { public function handle(EventMessage $event) { print_r("Inside"); } } $pamiClient = new…
Anu
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Asterisk Event to listen to for Transferred call

I'm building an application that listens to the Asterisk AMI to see when a user gets a call or makes a call out. I have got the events for when a user gets a call and when they make a call out. But what event do I listen for when a call gets…
Jason
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Originate call on multiple extensions and Hangup if one is answered

I am using Asterisk Manager Interface to originate call. I am making Outbound call and then originate calls on multiple extensions. The problem is that the other extensions doesn't get disconnected if call has been answered on one of the extensions.…
Code Name Jack
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install asterisk web interface

I have installed Asterisk on ubuntu 14.04 LTS with: sudo apt-get install asterisk and It is running on my system, some logs : root@<>:~# asterisk -rvv Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more…
Ashraf Shahzad
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asterisk PAMI Originate Call issue

Received an unknown call with DID [rawContent:protected] => Event: Newexten Privilege: call,all Channel: SIP/701-000056ff ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 701 CallerIDName: 701 …
psolz
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No ring back tone on outbound calls. [FreePBX]

I have a FreePBX 13 server set up with a SIP Trunk connection, however for some reason we are not getting the ring back tone for calls going out of the trunk connection. I was able to implement a work around for this by placing the "Tr" options…
Ox3
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asterisk confbridge - how to add japanese version

I'am Using asterisk conference When I join to conference it says Please Enter Pin Number How can I Change it to Japanese Version?
uno
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How to determine current PBX status (calls, ringing, etc. etc.) with asterisk-dotnet

I am using asterisk-dotnet, and I am looking for a way to get the current PBX "status". What I need is to determine: which extensions are busy in a call which extensions are ringing which number is calling the extension which number is called by…
Anthares
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How can I load test and performance test an asterisk sip server to find out how many concurrent calls a server can handle?

I have deployed asterisk server in AWS. 1) I would like to know how many calls it can handle concurrently at the same time? 2) Is it possible to do a load test with several phone numbers? How to simulate it? Any tools?
sofs1
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Can I have a new real time SIP family name other than the well defined family names in asterisk server?

In this link http://www.voip-info.org/wiki/index.php?page_id=1735 [settings] => ,~np~[~/np~,table_name~np~]~/np~ sippeers => mysql,asterisk,sip_peers sipusers => mysql,asterisk,sip_users queues =>…
user3705456
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Either the outbound or inbound call only work in asterisk setup, not both. Why?

This is my sip.conf ; inbound configuration [nexmo-sip] fromdomain=sip.nexmo.com type=friend context=nexmo insecure=port,invite nat=no ;Add your codec list here. ; Note: Use "ulaw" for US only, "alaw" for the rest of the…
user3705456
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Connecting to Asterisk AMI via shell script - Permission Denied

I am trying to create a shell script to listen for events from the Asterisk AMI, but when I attempt to connect, I get: Response: Error Message: Permission denied Here is the relevant code from my script: set username "username" set secret…
S.Nishizaki
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Linking bridged calls from queue

I made a PHP script that calls 10 numbers and if they answer they are put in a queue and if there are free agents they pick up each call. My problem is I cannot log which number is bridged to which extension. Can you please help? Thanks in advance!
Ivan Velkov
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Appending categories with template via AMI and PHP

I'm using a PHP library to update config files of asterisk. I add a user like this: $a->write('Action: updateconfig\r\nReload: yes\r\nSrcfilename: users.conf\r\nDstfilename: users.conf\r\nAction-000000: NewCat\r\nCat-000000:test\r\nAction-000001:…
Sergey Scopin
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