Asterisk is the world's most powerful and popular telephony development tool-kit. It is used by small businesses, large businesses, call centers, carriers and governments worldwide.
Asterisk is open source telephony project. Under development since 1999, Asterisk is free, open source software turns an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services.
Digium makes Asterisk available to the open source community under the GNU General Public License (GPL).
Some of the features of Asterisk is as follows;
- Supports various VoIP protocols including IAX, SIP, MGCP, SCCP and H323.
- Support for PSTN interface cards and devices.
- Routing and call handling for incoming calls.
- Outbound call generation and routing.
- Media management functions (record, play, generate tone, etc.).
- Call detail recording for accounting and billing.
- Transcoding (conversion from one media format to another).
- Protocol conversion (conversion from one protocol to another).
- Database integration for accessing information on relational databases.
- Web services integration for accessing data using standard internet protocols.
- LDAP integration for accessing corporate directory systems.
- Single and multi-party call bridging.
- Call recording and monitoring functions.
- Integrated "Dialplan" scripting language for call processing.
- External call management in any programming or scripting language through Asterisk Gateway Interface (AGI)
- Event notification and CTI integration via the Asterisk Manager Interface (AMI).
- Speech synthesis (aka "text-to-speech") in various languages and dialects using third party engines.
- Speech recognition in various languages using third party recognition engines.
This combination of components allows an integrator or developer to quickly create voice-enabled applications. Asterisk integrators have built everything from very small IP PBX systems to massive carrier media servers.
Asterisk IP PBX
Asterisk can be configured as the core of an IP or hybrid PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world over IP, analog (POTS), and digital (T1/E1) connections.
Asterisk runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and provides all of the features you would expect from a PBX including many advanced features that are often associated with high end (and high cost) proprietary PBXs. Asterisk's architecture is designed for maximum flexibility and supports Voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.
Asterisk SIP Configuration for VoiceTrunking
If you are not an advanced user of Asterisk, we highly recommend the use of one of GUI interfaces of Asterisk for SIP Configuraiton.
There are several GUI interfaces for Asterisk that simplify installation of Asterisk. These interfaces allow administrators to view, edit, and change most aspects of Asterisk via a web interface.
You can download free GUI versions of Asterisk from one the below links below:
While our goal is to make all Bring Your Own Device installations as easy as possible, this option is intended for advanced users. VoiceTrunking does not provide technical support for Asterisk SIP Configuration.
Below you can find Asterisk SIP Trunk configuration guide for VoiceTrunking SIP Trunk service.
Outgoing Settings
Peer Details
username=5551231234 (your VoiceTrunking account assigned while signing up)
type=peer
secret=XXXXX (your VoiceTrunking password)
nat=auto
insecure=very
host=sip.VoiceTrunking.com
fromuser=5551231234
(your VoiceTrunking SIP account assigned while signing up)
fromdomain=sip.VoiceTrunking.com
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ilbc
allow=ulaw
allow=alaw
Incoming Settings
USER Details
username=5551231234
(your VoiceTrunking SIP account assigned while signing up)
type=user
secret=XXXXXX (your VoiceTrunking password)
nat=auto
insecure=very
host=sip.VoiceTrunking.com
fromdomain=sip.VoiceTrunking.com
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=g729
allow=ulaw
allow=alaw
allow=ilbc
REGISTRATION STRING
5551231234:XXXXXXXX@sip.voicetrunking.com/55551231234
(for 5551231234 use your VoiceTrunking SIP account and for XXXXXXXX use your VoiceTrunking password)
NOTE: Asterisk does not support DNS SERVER lookups for inbound calls.
If you also have virtual phone number with your SIP Trunk service please add the following line to the "sip_general_custom.conf" file
srvlookup=no
