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Asterisk
The Open Source Linux PBX.
www.asterisk.org.
Software : Daemons
: Asterisk : Übersicht
12-Dec-2004/09-Jan-07
Übersicht
 Asterisk
ist eine GNU/Linux-basierte Telefonanlage, die sowohl mit dem
konventionellen analogen Telefonnetz gekoppelt werden, als auch als
VoIP-/ SIP-Gateway genutzt werden kann. Je nach Einsatzgebiet kommt
Asterisk ohne jegliche zusätzliche Hardware aus (SIP-Gateway), oder es
werden »Adapter« zum POTS oder ISDN etc. benötigt.
Das in Deutschland recht verbreitete
ISDN
lässt sich ebenso mit einer Asterisk-Anlage verbinden; voraussetzung
dafür ist ein ISDN-Adapter, der entweder von ISDN4Linux unterstützt wird
oder für den eine CAPI verfügbar ist.
Schwieriger wird, es, wenn man dann auch noch ein ISDN-Telefon an
einer Asterisk-Anlage betreiben will; dann wird eine ISDN-Karte mit
HFC-Chipsatz und Zaptel-BRI-Treiber (»bristuff«) sowie ggf. noch
ein NTBA zur Stromeinspeisung benötigt.
Asterisk wird über eine Komandozeilen-Schnittstelle (Command Line
Interface, CLI), bedient, die mit dem MySQL-Promt oder einer sehr
einfachen Shell vergleichbar ist; eine grafische Benutzerschnittstelle (Graphica
User Interface, GUI) existiert von Hause aus nicht, es befinden sich
allerdings zahlreiche GUIs- und web-basierte Interfaces und
Statusanzeigen in Vorbereitung.
Die Software wurde maßgeblich von Mark Spencer von Digium, Inc.,
entwickelt. Die Software selbst unterstützt u.a. die Protokolle SIP,
H.323, IAX2, MGCP, SCCP, Analog, ISDN (von BRI bis PRI) und kann die
Codecs G.711, GSM, G.729, G.726, iLBC und Speex nutzen. Weiterhin ist
Telefax, Text2Speech und AGI möglich.
Aus der Selbstdarstellung:
»Asterisk is a complete PBX in software. It runs on Linux
and provides all of the features you would expect from a PBX and
more. Asterisk does voice over IP in three protocols, and can
interoperate with almost all standards-based telephony equipment
using relatively inexpensive hardware.
Asterisk provides Voicemail services with Directory, Call
Conferencing, Interactive Voice Response, Call Queuing. It has
support for three-way calling, caller ID services, ADSI, SIP and
H.323 (as both client and gateway). [...]
Asterisk needs no additional hardware for Voice over IP.
For interconnection with digital and analog telephony equipment,
Asterisk supports a number of hardware devices, most notably all of
the hardware manufactured by Asterisk's sponsors, Digium™. Digium
has single and quad span T1 and E1 interfaces for interconnection to
PRI lines and channel banks as well as a single port FXO card and a
one to four-port modular FXS and FXO card« (Quelle:
www.asterisk.org; Zugriff:
22-Dec-2004).
Kommandozeilenparameter
Starten von Asterisk:
- -c: Console mode. Don't go into background mode, stay in
foreground with a command line interface (implies -f)
- -C <filename>: Start Asterisk with a specified configuration
file
- -d: Debug mode
- -f: Stay in foreground mode
- -g: Dump core in case of a crash
- -h: Help (list all command line options)
- -i: Initializie crypto keys at startup
- -p: Run as pseudo-realtime thread
- -q: Quiet mode
- -v: The verbose command. Add more 'v':s to get more messages.
Verbindung zu einem laufenden Asterisk:
- -r: Connect to Asterisk running in the background and present
a command line interface
- -x: In combination with -r, execute an Asterisk CLI command
- -n: Disable ANSI colour support
Beispiele:
- asterisk -vvvvvvdr: Start a very chatty Asterisk command line
interface and connect to asterisk running in the background
- asterisk -vvvvvvc: Start Asterisk PBX, don't go background
- asterisk -rx reload: Connect to a running Asterisk, force
reload of configuration with the CLI command "reload"
Quelle:
www.voip-info.org/wiki-Asterisk+options; Zugriff: 22-Dec-2004.
CLI
www.voip-info.org/wiki-Asterisk+CLI
GUIs
www.voip-info.org/wiki-Asterisk+GUI.
Asterisk GUI client (astGUIclient) - A GUI client application for the
Asterisk open-source PBX,
astguiclient.sourceforge.net,
sourceforge.net/projects/astguiclient.
Internationalisierung
www.voip-info.org/wiki-Asterisk+International.
Hardware
Zaptel Compatible Hardware
- Wildcard TE405P A Quad-Span T1/E1 half-length 5-volt PCI card that
supports both telephony and data modes in T1 or E1 format on a
port-selectable basis.
- Wildcard TE410P A Quad-Span T1/E1 half-length 3.3-volt PCI card that
supports both telephony and data modes in T1 or E1 format on a
port-selectable basis.
- Wildcard T400P [Discontinued] A Quad-Span T1 half-length PCI card
which supports both telephony and data modes.
- Wildcard T100P A Single-Span T1 half-length (available with 2U
bracket) PCI card sporting the same features as the T400P.
- Wildcard E400P [Discontinued] A Quad-Span E1 half-length PCI card
which supports both telephony and data modes.
- Wildcard E100P A Single-Span E1 half-length (available with 2U
bracket) PCI card sporting the same features as the T400P.
- TDM400P A one-to-four port modular PCI 2.2 card currently featuring
FXS modules for interfacing with standard analog telephones.
- X100P A single port FXO PCI interface card for interfacing with a
standard analog phone line.
Dialogic Hardware: Analog
- D/41JCT-LS 4-port analog + voice
- VFX/41JCT-LS 4-port analog + voice + fax
- D/120JCT-LS 12-port analog + voice
Dialogic Hardware: Digital
- D/240JCT-1T1 T1 + 24 voice
- D/300JCT-1E1 E1 + 30 voice
- D/480JCT-2T1 (2) T1 + 48 voice
- D/600JCT-2E1 (2) E1 + 60 voice
Non-Zaptel & Non-Dialogic Hardware
- QuickNet: Internet PhoneJack Single FXS interface. Supports Linux
telephony interface. DSP compression built-in.
- QuickNet: Internet LineJack Single FXS or FXO interface. Supports
Linux telephony interface. DSP compression built-in.
- ISDN4Linux Any ISDN terminal adapter supported by isdn4linux should
provide connectivity.
- ALSA Any ALSA compatible full-duplex sound card.
- OSS Any OSS compatible full-duplex sound card.
Dokumentation
Es existiert ein Handbuch-Entwurf als PDF-Dokument, das vom Asterisk
Handbook Project erstellt wurde:
- Mark Spencer, Mack Allison, Christopher Rhodes, The Asterisk
Documentation Team: The Asterisk Handbook, Version 2,
www.digium.com/handbook-draft.pdf [PDF-Datei, ca. 70 Seiten,
englischsprachig].
Verschiedene inoffizielle Dokumentationen sind ebenfalls verfügbar:
Mailinglisten und IRC
Zu Asterisk existieren sechs »offizielle« Mailinglisten:
Neben den Mailinglisten kann Internet Relay Chat (IRC) zum
Erfahrungsaustausch genutzt werden:
- Server: irc.freenode.net
Port: 6667
Channel: #asterisk
Alternativen
SIP Proxies:
PBX-Projekte:
Literatur
Practical VoIP Using VOCAL,
von David G. Kelly, Cullen Jennings, Luan Dang
1st Edition July 2002
ISBN: 0-596-00078-2
512 pages, $44.95 US, $69.95 CA, £31.95 UK
Summary: »While many books describe the theory behind Voice
over IP, only Practical VoIP Using VOCAL describes how such a phone
system was actually built, and how you too can acquire the source code,
install it onto a system, connect phones, and make calls.
VOCAL (the Vovida Open Communication Application Library) is an
open source software project that provides call control, routing, media,
policy, billing information and provisioning on a system that can range
from a single box in a lab with a few test phones to a large, multi-host
carrier grade network supporting hundreds of thousands of users. VOCAL
is freely available from the Cisco Systems-sponsored Vovida.org
community web site (www.vovida.org).
A Silicon Valley start-up called Vovida Networks, Inc (think of
VOice, VIdeo, DAta) created VOCAL and invested over one hundred man
years into its development. Since Cisco acquired Vovida in 2000,
individuals representing every significant telecom company and service
provider in the world have downloaded the source code. Today, more and
more people are successfully building VOCAL into professional solutions,
while contributing fixes and new functionality back to Vovida.org.
Because VOCAL is open source, you can look "under the hood" to
the base code and protocol stack levels and discover not only how the
system works, but also how common problems are being worked out in the
development environment. We're hoping that you will be inspired to take
this system to another level by implementing a feature or functionality
that no one has thought of before« (Quelle:
www.oreilly.com/catalog/voip/desc.html; Zugriff: 22-Dec-2004).
Probekapitel:
Website:
www.oreilly.com/catalog/voip.
Siehe auch
Netmarks
Dokumentation und HOWTOs zu Asterisk:
Asterisk und VoIP:
Asterisk und ISDN:
Allgemein:
- SIP Product List - Lists of hardphones, softphones, and other
servers,
www.pulver.com/products/sip.
- IPTel.org - Lists of hardphones, softphones, and other servers,
www.iptel.org/info/products.
- Distributed Universal Number Directory (DUNDi) - The DUNDi
peer-to-peer system for connecting to the PSTN network via the Internet,
www.dundi.com.
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