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Questions tagged [asterisk]

Asterisk is a software PBX used to route audio and video calls. QUESTIONS REGARDING THE USE AND CONFIGURATION OF ASTERISK ARE OFF-TOPIC. Only programming questions are on-topic for Stack Overflow. For example: dialplan configuration in extensions.conf, interfacing with Asterisk's C APIs, or working with AGI scripts.

Asterisk is a Private Branch eXchange (PBX) software; its main aim is to route audio and video calls, but it supports the creation of services like voicemail, Interactive Voice Responders (IVR), least-cost routing (LCR) etc.

Asterisk supports different Voice over IP (VoIP) protocols - like SIP, IAX2, H.323 - and can be integrated with traditional telephony equipment and networks - like analog networks, ISDN BRI/PRI, GSM/UMTS.

Its behavior can be programmed using Asterisk Extension Language (AEL) or a legacy dialplan syntax, and it exposes various APIs (including HTTP APIs) for run-time interaction with other programs and servers. External scripts can be called via the Asterisk Gateway Interface (AGI.)

As Stack Overflow is a programming site, questions tagged must relate to the topic of programming. Such topics may include (but are not limited to):

  • Dialplan programming using traditional syntax or AEL
  • Connecting to Asterisk via API interfaces
  • Writing and calling AGI scripts
  • Work on the Asterisk codebase itself

Questions concerning the following topics are off-topic for Stack Overflow and may be better suited for another site such as Server Fault or Super User:

  • Asterisk configuration
  • Hardware interface problems
  • Asterisk GUIs such as FreePBX
  • Call quality issues (e.g. one-way audio)

Important links:

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How can I set environment variables in my Linux service for Asterisk even though it doesn't have a real user?

I have created a linux service that runs as a deamon (and gets started from /etc/init.d/X). I need to set some environment variables that can be accessed by the application. Here's the scenario. The application is a bunch of Perl AGI scripts that…
domino
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getting iaxclient to send audio to/get audio from buffer instead of audio-device

I'm trying to write a C++ program (altough python would've been fine as well in case someone knows a better (IAX/SIP) alternative) which connects to an Asterisk server. After connecting, it should listen for audio and process that. It should also…
Folkert van Heusden
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Asterisk auto Call recording

We are running asterisk with 8 port FXO. FXO connects to our old PBX (Samsung Office Serv 100). Now we want to record all calls routed through FXO (if it was dialed to outside or comming from outside). Is there a simple way to do this?
Manjoor
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VoIP Integration in App & Web

I have a very general question on how to implement VoIP for our current mobile & Web App. (we have an Android+iOS App and a Web Application based on AngularJS/NodeJS). What we want to achieve In the first step we want to achieve inter Application…
Markus
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setup an IVR with Asterisk

I need to setup a simple IVR system for a friend's company that will let the caller navigate through the menu by pressing phone keys. Its kind of like a bus schedule. for today's schedule press '1', for tomorrow's schedule press '2' and so…
Sebastian
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FreePBX Twilio Outbound Ringtone

I have a Twilio SIP trunk connected to FreePbx, all users are using the webrtc module of FreePBX to make calls. They can make and receive calls fine with two way audio, however with outbound calls the caller does not hear ringtone (ringing) as the B…
Freddy Wetson
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Wildcards in variable path with Powershell

I would like in my script to use wildcard in variable like this : $TARGET = "\\MACHINE1\c$\ProgramData\Test\12.*\Data\" The problem is $TARGET returns \\MACHINE1\c$\ProgramData\Test\12.*\Data\ and…
robinwood13
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Asterisk,SIP Retransmission timeout

I have created a sip trunk from One Asterisk(version 11.2.1) say 'A' server to another Asterisk server(11.7.0) say 'B', and I am getting sip response 200 ok. But when I start calling on a DID on Asterisk A then the call is being routed to Asterisk…
Vivek Raj
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getting the group name to the according pri port in asterisk

I am using sagoma 8 port card My chan_dahdi.conf to configure the ports are ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2015-06-12 ;Dahdi Channels Configurations ;For detailed Dahdi options, view…
codegasmer
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How can I use Twilio as a SIP trunk for my Asterisk to make and receive calls?

I have a Twilio account which has a number (let's say 8881231234), and I have Asterisk box. I'd like to use Twilio as an Asterisk trunk to be able to make calls at their rates and receive calls from that number on my Asterisk. I haven't found any…
g3rv4
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Difference between tx and rx?

With asterisk I can set the volume of TX and RX. But what are those options? I've already googled this but can't find anything. Whats is the difference between TX and RX?
Jochem Gruter
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SIP command not found

I have installed asterisk on Ubuntu Natty, When I go into asterisk CLI & type in sip reload or any SIP related commands, it says SIP command not found. Anyone had a similar problem before? Thanks
krishna bhargavi
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Asterisk + Node.js + Browser Streaming

I would like to build a service that allows a user to listen to a call live from their browser. I have some experience with Asterisk and this seems to be flexible enough to do what I have described. Node.js sounds good because it is purported to…
Jonathan
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Are there parallels to Asterisk AMI and AGI in FreeSWITCH?

Asterisk has Asterisk Manager Interface (AMI) and Asterisk Gateway Interface (AGI), using which one can trigger PHP scripts at certain events from Asterisk. Using the same PHP scripts can also instruct Asterisk what to do next to a call/conference.…
jeff musk
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how can I make a sip call with twisted sip protocol?

I have an asterisk server and I want to know is this possible to make a sip call with twisted sip protocol? if yes how can I do this? unfortunatly I can't find any document about how to use twisted sip protocol or any example of how It works.
nim4n
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