The company Digium announced the release of version 1.6.0 asterisk. This is the first official release of version 1.6, differs from the previous 1.4 branch substantial number of new features brief review below.
One of the most significant change is that now SIP signaling can be carried out not only on the UDP protocol, but also for TCP.
The mere use TCP instead of UDP does not provide many benefits, but allows the use of TLS encryption bandwidth that is claimed by companies that are making higher demands on security.
It is worth noting as well the possibility of changing the timing for the SIP protocol, the so-called timers, defined in RFC 4028. This may require, for example, in cases where the network is characterized by stable long delay packages, as well as avoiding the hang SIP sessions, when communication with the SIP client fetches up during the session itself.
Less substantial but facing mention changes in the SIP steke:
- Ability to specify the port number for bindaddr, externip and externhost parameters.
- Options “musiconhold” and “musicclass” in the file sip.conf no longer exists, they are now replacing “mohsuggest” and “mohinterpret”.
- Function SIPPEER gives the port number and call pickup groups.
- Restructuring Mechanism MWI (Message Waiting Indication – notification of new voice messages) led to the scrapping Options checkwmi file sip.conf and adding pollmailboxes in sip.conf and pollfreq in voicemail.conf.
- Added the ability to send text in real-time mode on a protocol T.140. Who does not know what it is – see http://en.wikipedia.org/wiki/Text_over_IP.
- During the registration of SIP clients, the answer is 100 Trying default will no longer be sent as According to RFC is not required. For inclusion, it is necessary to clearly specify registertrying.
- To better accountability needed for queues and SIP subscriptions added a new option callcounter. Earlier this requires was the option call-limit to deliberately inflated value.
- Added option qualifyfreq, which sets the frequency of inspections qualify.
- Added new option trunkmaxsize, regulating the maximum number of channels in IAX2 Trunk.
- Added option srvlookup allowing use DNS SRV records to determine the IP address. In SIP steke this functionality was implemented much earlier.
- Added support for OSP – «http://www.transnexus.com/OSP Toolkit / what_is_osp.htm».
Changes in Local Channel:
- Function DEVICE_STATE () indicates INUSE or NOT_INUSE for existing ekstenshenov.
- Ability to incorporate Jitter buffer for Local channels, working in conjunction with a team option j Dial and the suffix / n.
Change the channel DAHDI (formerly Zap):
Zap channel was renamed the DUHDI http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/.
Changes in 1.6. include:
- Support for SS7 protocol, using the library libss7 (yes neuzhto!).
- Team dialplan show now displays information coincidentally c CID (number of caller), which can be used in dialplane after ekstenshena through direct slash.
- New command CLI: dahdi show version.
- In the configuration file chan_dahdi.conf an option setvar.
- You can now include notification of new messages (MWI) for FXO lines through the options and mwimonitor mwimonitornotify.
- Added the type of alarm auto, with its use of Asterisk will try to use the alarm for the channel, which is listed in DAHDI.
- New command dahdi set dnd from the command line interface allows you to change the Do-Not-Disturb status for the channel. In this event DNDState Manager (AMI) is generated irrespective of how it changed DND state.
Changes in Dialplan functions:
- Added option DEVICE_STATE () to obtain information on the status of various devices. The function can also be used to establish their own state and control of dialplana.
- Team MailboxExists turned into a feature.
- A new option for the Dial, which IP phones do not count the call as missing in the case of non, or — the lifting of the Asterisk.
- Added option HINT (), provides information stored in dialplane in a special alphanumeric prioretete hint.
- Added option SYSINFO () to obtain information related to the operating system.
- Added features TOUPPER () and TOLOWER () name spoken for themselves and are well known to programmers.
- When connecting channels, Asterisk sets variable BRIDGEPVTCALLID equally unique ID call driver channel
(if the driver has it). For SIP calls, for example, the variable will be field SIP call-ID.
It should be noted that despite the addition of new capabilities and the emergence of more convenient methods for implementing the desired functionality, Digium developers decided not to remove obsolete functionality of dialplan to preserve backwards compatibility.
Changes in command line interface (CLI):
In Asterisk 1.6.0 appeared commands such as:
- Sore show hint.
- Core show settings.
- Core show channels count.
- Queue pause member and queue unpause member.
- Possibility to change the parameters ulimit without restarting Asterisk.
- When you switch agi debug before the conclusion of AGI message will name the channel that facilitates debugging
on the operating systems dialplan set extenpatternmatching true / false.
- Now you can be placed variable channel from the CLI commands through a core set chanvar.
- Added Section startup_commands in cli.conf, where you can specify command of the CLI, performed immediately after running Asterisk.
- Team devstate change allows a CLI done the same thing as a function DEVICE_STATE () in dialplane.
Changes in voice mail (VoiceMail):
- Now you can customization of what sounds to use for specific items on the voice menu.
- voicemail show users shows users configured realtime basis.
- MWI algorithm has been changed. Now it is not based on periodic inspection, and to generate events. Different modules can subscribe to MWI and Asterisk will inform them about changes in golosvyh boxes. This also means that if change the contents of the voice mail from outside, such as web intferfeysom to voice mail, Asterisk still have to be Periodically check the contents of voice mailboxes. To do this, set up options pollmailboxes and pollfreq.
- Added the ability to store greetings to the IMAP server.
- Ability to change the buttons for scrolling, pause and stop playing a voice message.
- Added option lockmode in asterisk.conf to specify the method of blocking file voice mail and other files in the future. By default method of creating lockfile. Now there is another method – flock, which can work on file systems — SMB / CIFS.
- Added the ability backup / delete messages it for the user to restore accidentally deleted message.