Welcome to Asterisk Up-to-Speed
Asterisk Up-to-Speed is the essential reference for any Asterisk administrator. Easily and quickly educate yourself with the latest new features, upgrades, and changes to the world's most popular open-source PBX, Asterisk.
Say your current Asterisk system uses version 1.4 and you're looking into rewriting your dialplan for future versions. You want to know how the DIAL() application changed since then. Instead of mulling over changelogs and upgrade files to discover the new features, quickly get the immediate pertinent information, such as:
DIAL Function:
Asterisk 1.4 -> Asterisk 1.6.0
- A new option to Dial() for telling IP phones not to count the call
as "missed" when dial times out and cancels.
Asterisk 1.6.0 -> Asterisk 1.6.1
- A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
invalid input and will be assumed to mean that no timeout is desired.
- Dial has a new option: F(context^extension^pri), which permits a callee to
continue in the dialplan, at the specified label, if the caller hangs up.
- The Dial() application no longer copies the language used by the caller to the callee's
channel. If you desire for the caller's channel's language to be used for file playback
to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
Asterisk 1.6.2 -> Asterisk 1.8
- Added progress option to the app_dial D() option. When progress DTMF is
present, those values are sent immediately upon receiving a PROGRESS message
regardless if the call has been answered or not.
- Added functionality to the app_dial F() option to continue with execution
at the current location when no parameters are provided.
- Added the 'a' option to app_dial to answer the calling channel before any
announcements or macros are executed.
- Modified app_dial to set answertime when the called channel answers even if
the called channel hangs up during playback of an announcement.
- Modified app_dial 'r' option to support an additional parameter to play an
indication tone from indications.conf
- The 'f' option to Dial has been augmented to take an optional argument. If no
argument is provided, the 'f' option works as it always has. If an argument is
provided, then the connected party information of all outgoing channels created
during the Dial will be set to the argument passed to the 'f' option.
- Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
Gosub on the peer.
Version changes span from 1.4 -> 1.6.0 -> 1.6.1 -> 1.6.2. -> Asterisk 1.8 If you haven't already, we highly recommend to familiarize yourself with the remarkable book, Asterisk, The Future of Telephony
New features are organized alphabetically into eight areas of
Asterisk:
Applications and Functions:
Lists all the Dialplan applications, such as Voicemail, and functions, such as Dial, along with any new or changed variables and operators
Channels:
Everything you want to know about new features and changes with SIP, IAX2, SKINNY, etc. right here
Codecs:
G.711a, G.729, GSM etc. etc.
Core Asterisk:
This sections contains information about the core workings of Asterisk, including AEL, Language support, CLI interface, etc.
Formats:
Wave, Mp3 formats, etc. all explained here
PBX Modules:
See the latest about PBX modules, such as DUNDI, and Dialplan Behavior
Resource Modules:
Realtime support, Parked calls, are all explained here.
The Asterisk Up-to-Speed team completely reorganized the various changelog files located in source
code of Asterisk according to an Asterisk administrator's perspective (as opposed to Asterisk programmers' and developers') in an intiutive and straightforward fashion.