Asterisk 1.6

Asterisk 1.6 emerged with the several features of version 1.2 and 1.4, and also it features new dial plan functions. Further is the list of the recent updates on functionality changes.

AMI - (TCP/TLS/HTTP) The manager
The new features are: TLS support of the manager interface, HTTP server, built-in HTTP server URI redirect, GetCofnigJSON, a possibility to bridge two channels active on the system („Bridge”) and the action that allows you to retrieve of voicemail setup of all users. (”ListAllVoicemailUsers)
Dialplan functions
The added functionalities have a new feature to Dial() to tell IP phones to stop counting timed out or cancelled calls as a "missed" calls, a DEVSTATE() dialplan function retrieves dialplans device state, , LOCK(), TRYLOCK(), and UNLOCK().
Modifications in CLI
New functionalities include the feature of setting the verbose and debug values on a file basis, channels count CLI command.
SIP modifications
New functionalities include: improved DNAT and STUN support, the new way to match incoming requests, Asterisk reports of a busy device when reaches the certain level of calls „busy-level”, new real-time „sipregs” (store SIP registration data), extra support for T.140 real-time text in SIP/RTP, sets new values for call transfers, and a new header displayed for cancelled calls when answered by other phone.
IAX2 modifications
Added functionalities: configuration of trunkmaxsize, chan_iax2, srvlookup, iax.conf, and OSP support.
DUNDi modifications
The new modifications allow to specify arguments to the Dial application, while using the DUNDi switch, option to dynamically set weights for responses, dialplan functions ( DUNDIQUERY and DUNDIRESULT) allowing to start a DUNDi query from the dialplan and help to find out the number of results, allowing to access each of them.
ENUM modifications
Added functionalities comprise couple of new dialplan functions (ENUMQUERY and ENUMRESULT) allowing one to start an ENUM search from the dialplan to access the results without having to send multiple DNS queries.
Voicemail modifications
Added functionalities include: the possibility of customization of sound files, which are used for some prompts in the the Voicemail application, the possibility for the “voicemail show users” CLI command, “tw” language support, greeting storage support, and the ability to customize keys for message playback.
Another update expected is an Open Source Asterisk PBX 1.8 release within Long-Term Support (LTS) commitment.

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SDP Protocols Are Used To Transmit Media Session Information

In 1998, the Internet Engineering Task Force (IETF) published the specification for a Session Description Protocol or SDP as a format that describes parameters for streaming media. The original IETF Proposed Standard was updated in 2006 as RFC 4566. Although the SDP was created as a feature of Session Announcement Protocol (SAP), it can be used with Real-time Transport Protocol, Real-time Streaming Protocol (RTSP), and Session Initiation Protocol, as well as a standalone protocol. Parameter negotiation, session announcement and session invitation are included in the descriptive sessions from the SDP protocol. Rather than transmit data like other types of protocols, an SDP negotiates between media type endpoints, format and properties involved. A session begins when a connection is established, and the session is terminated only after every endpoint is no longer participating.

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