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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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asterisk anti ex-girlfriend Dial plan?

I wrote simple dial plan in asterisk. This dial-plan target is to check caller-id of incoming call and for specific hangup :) ! but this dial-plan hangup all incoming call with diffrent caller-id. So what do i do? ;( [general] static=yes …
Rev
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ARI JS client mute error

Am currently developing a mute function for asterisk which I can run from my web front end using asterisk ARI. But every time I try to run/call the mute function it gives me the following error: Error: { "message": "Channel not in Stasis…
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Set waiting tone for asterisk agi function processing

I am using asterisk with normal PHP AGI following this link the problem is that my PHP AGI takes 5 seconds to execute .I just want to set some waiting tone for the user to wait until the AGI is been processing. On the same link I found…
codegasmer
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How to get all dialer events from Asterisk REST API (ARI)?

I'm making a web application which should be able to monitor calls on my Asterisk server. I can connect to ARI with Javascript WebSocket on URL ws://(host):8088/ari/events?app=dialer and it works. The problem is that I only get events from calls…
demian
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Unit/integration testing Asterisk configuration

Unit and integration testing is usually performed as part of a development process, of course. I'm looking for ways to use this methodology in configuration of an existing system, in this case the Asterisk soft PBX. In the case of Asterisk, the…
Jakob Borg
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Connecting Skype to Asterisk

I have an Asterisk PBX under development, that I would like to link to a Skype account if possible. The idea is that people would call a particular Skype username, and be redirected to my SIP and through that to Asterisk. Is this doable? I have…
Philip Bennefall
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SIP to PSTN gateway connection from asterisk?

We are working on a web phone application that can make sip calls to other devices and make PSTN calls as well. We use Asterisk 1.8 as our sip server. The SIP calls from the web phone is working fine. We want to be able to provide SIP to PSTN…
over.drive
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Java - Asterisk, what does it mean

I'm looking through the Alexa SDK. In the documentation here, they have the following: directivesServices.enqueue(SendDirectiveRequest.\*builder\*().build()); What on earth does that asterisk do? I have never seen one in my life, and I'm curious…
Black Dynamite
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Asterisk gives "Strict RTP learning" message and no audio for Chrome WebRTC but works in Firefox

I've been experimenting with WebRTC with an Asterisk server (v13.18) on the same LAN as my computer. I configured the Asterisk extension 6003 to automatically answer and play a certain notorious sound file whenever it's dialed, then confirmed that…
Eli Courtwright
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Hangup notification sound

Background My client uses a Asterisk 1.6 based PBX telephone system as their call center solution. They use a soft phone application to pick up all calls from the inbound queue. To reduce their work load, the soft phone application they use has a…
AkiEru
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Whats the best architecture for CDR integration to a Asterisk based Application

Iam developing a VOIP platform which would allow users to make 100s of calls concurrently using my service. Asterisk stores all call detail records in the CDR table. I would like to know where is best place to keep this table for the best possible…
Sumit Ghosh
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Node ARI Client | Connect method not firing callback?

So, I've started playing with the Asterisk Restful Interface (ARI). I have created a separate express app to do this. I have a correctly configured instance of Asterisk 13 running. I know this because When I go to…
An0nC0d3r
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Asterisk unable to connect to database using odbc connector

I am pretty new to Asterisk and this is my first attempt to connect to database (MySQL). But I keep getting this error [unixODBC][Driver Manager]Data source name not found, and no default driver specified. Here is my…
Paullo
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SRTP output wanted, but no crypto suite was negotiated from kamailio rtpengine

I am trying to integrate webrtc->kamailio->asterisk to call from web browser. I am using kamailio configuration file from caruizdiaz and chrome browser with sipml5 and asterisk as media server. Till now I have achieved to call to pstn numbers…
dkakoti
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how to configure asterisk instant messaging

Does asterisk support instant messages? I have tried to configure asterisk for IM (from this example), but when I'm trying to send IM to another sip account asterisk returns warning: WARNING[20128]: chan_sip.c:16379 receive_message: Received message…
Dr Glass
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