Asterisk

From SubmitYourSIP

Asterisk is a popular multi protocol open source PBX sponsored by Digium. In addition to supporting several other protocols, Asterisk has support for SIP.

Fundamentally, Asterisk is a SIP B2BUA and many people use it for SBC functionality. It also does quite well in it's original role as a PBX, providing phone services to local phones, trunks, and the myriad of applications it offers (conferencing, IVR, etc).

For a full list of Asterisk features, please see asterisk.org or one of the many other Asterisk resources available online.

Contents 1 General tips 2 Asterisk 1.2 3 Asterisk 1.4 4 Asterisk 1.6

General tips

- SIP configuration in /etc/asterisk/sip.conf

- RTP configuration in /etc/asterisk/rtp.conf (shared by all channels that use RTP - SIP, MGCP, SCCP, etc)

- On busy systems you may need to increase the available number of file descriptors

Asterisk 1.2

- No support of multipart SIP bodies (disable SIP-T, SIP-I, SIP-GTD in other equipment)

- Shady RFC2833 implementation, no support for duration/variable length DTMF

Asterisk 1.4

- Improved support for multipart SIP bodies

- Reworked RFC2833 implementation, support for duration

Asterisk 1.6

- Experimental support for SIP TCP, SIP TLS

- Support for SIP Session Timer