GENERAL ASTERISK SUPPORT IS OFF-TOPIC. Support questions may be asked on https://superuser.com. Server support like "please read this output and say what is wrong" or configuration is OFF-TOPIC. Asterisk is a PBX software whose main aim is to route audio/video calls. It is released under a dual license: the GNU General Public License (GPL) and a commercial license.
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 language, and exposes various APIs (including HTTP APIs) for run-time interaction with other programs and servers.
As Stack Overflow is a programming site, questions tagged asterisk should relate to the topic of programming. Such topics may include (but are not limited to):
- Dialplan functionality using traditional syntax or AEL
- Connecting to Asterisk via API interfaces
- Work on the Asterisk codebase itself
Questions concerning the following topics could be 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- voip-info.org wiki – a wide-ranging, but very outdated, source of information for Asterisk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
