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 asterisk 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:
- The Asterisk home page
- The official Asterisk wiki
- Asterisk Forums
- Digium home page – the creators and financial supporters of the Asterisk project
- voip-info.org wiki – a wide-ranging, but very outdated, source of information for Asterisk
- Asterisk entry on Wikipedia
