Asterisk Blog - Page 18

Open source Many people use Asterisk because they want an open source solution. They are frustrated when they find that the G.729 module from Digium (the Asterisk creator) is not open source. In fact, it is technically possible…

How to use the G.729 codec in pass-thru mode in Asterisk “Pass-thru” means that if you were, for example, using two phones which both have inbuilt support for g729 codec. You can let them talk to each other…

Asterisk Codecs Asterisk supports the following narrow-band and wideband (HD audio) codecs: G.711 ulaw (as used in US) G.711 alaw (as used in Europe) G.722 – 16 kHz wideband codec; passthrough, playback and recording in Asterisk 1.4; full…

Each Asterisk contexts in extensions.conf may have an ‘i’ extension that is used for invalid extensions, i e, non-existing extensions dialled by clients. Not Automatic You would think that Asterisk would automatically jump to the ‘i’ extension if…

DTMFmode for SIP client configuration Syntax dtmfmode=inband Choices are inband, rfc2833, info or auto inband: This send tones as inband audio within the voice stream. The device that you press the key on generates the DTMF tones. –…

Asterisk is an open source software PBX, created by Digium, Inc. and a continuously growing user and developer base. Digium invests in both developing the Asterisk source code and low cost telephony hardware that works with Asterisk. Asterisk…

The following list of events was discovered by reading through the Asterisk source tree, with liberal applications of grep to help out. I will hopefully come back someday and document these. This is a definitive list of all…

Please note that AEL is still considered EXPERIMENTAL at this time.   The Asterisk Extension Language Over time, people have been pushing to add features to extensions.conf to make it more like a programming language. AEL is intended…

EDIT 04-24-09: I do not believe this is true. On asterisk 1.4 I have sip.conf, as well as SIP extensions defined within a MySQL database. Both means of authentication work. Even using #include “/etc/asterisk/sip.d/” lines within sip.conf work…

chan_mobile (used to be chan_cellphone) — Use Bluetooth cell / mobile phones as FXO devices Asterisk Channel Driver to allow Bluetooth Cell/Mobile Phones to be used as FXO devices and Bluetooth Headsets as FXS devices The official Homepage…

Status: 2007-03-07 Asterisk 1.4.1 2006-12-26 Asterisk 1.4.0 2006-12-18 Asterisk 1.4.0beta4 and Zaptel 1.4.0beta3 2006-11-19 Asterisk 1.4.0beta3 and Zaptel 1.4.0beta2 2006-11-09 Zaptel 1.2.11 2006-10-19 Asterisk 1.2.13 2006-10-19 Asterisk 1.0.12 2006-09-21 Asterisk 1.4.0beta2, Zaptel 1.4.0beta1 and libpri 1.4.0beta1 2006-09-15 Asterisk…

Current CVS -HEAD provides some support of SIP presence as defined in RFC 3856 – A Presence Event Package for the Session Initiation Protocol (SIP). The release version (1.0.9) has some support too. To support this, you have…

sip.conf Setup You can keep any sip users in the flatfile AND use RealTime. How cool is that? Extconfig.conf Setup with Asterisk 1.6.1.1 Add the following line, swapping your own personal values if you wish: sipusers => mysql,general,sip_buddies…

ver: CVS HEAD >= 08-24-05 | Asterisk 1.2 Requirements unixODBC (app and dev required) – Check your local RPM provider or http://www.unixodbc.org/ Setup Asterisk 1.4 1) Run “make menuselect” 2) Go down to “Voicemail Build Options” 3) Select…