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asterisk 1.6

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. 

 asterisk 1.6

Whenever building your own corporate phone scheme using the Asterisk open source telephony suite could output in massive cost savings for your entrepreneur, s not for the faint hearted. Asterisk is a complex scheme. Consequently, personally I'd recommend taking a Asterisk quick Start course to get you up and running, when you fancy trying out sterisk in the lab to see when it should be suitable for your organization here's what you need to see to start.

Now please pay attention. You can run Asterisk on pretty much any Linux distro with a kernel version 6 or later, you'll have to search for a Linux box to install Asterisk onto -the one I have got is running Ubuntu Jaunty. You'll as well have to look for the GCC version x or later for compiling.

asterisk 1.6

Basically, head over to and up 4 tarballs, for Asterisk, libpri, dahdi linux, once you had a suitable rig set download. It's a nice concept at this stage to copy them to /usr/src/.

Do not worry in the event the versions aren't specifically just like the ones above.

And now here is a question. Where do I plug the phones in? That's where it starts getting very intriguing, right? Where do I plug the phone lines in? Nonetheless, what hardware should I need?

As a output, whenever building your own corporate phone method using the Asterisk open source telephony suite could output in massive cost savings for your business, s not for the 'fainthearted'. Essentially, asterisk is a complex structure. Ultimately, download or head over to 4 tarballs, for Asterisk, libpri, dahdilinux, once you have got a suitable rig set up.

Do not worry when the versions aren't really identical to the ones above.

Where do I plug the phones in? Furthermore, where do I plug the phone lines in? Virtually, what hardware should I need?

Ok, and now one of the most important parts. Zaptel was discontinued and is replaced with the help of DAHDI. See the next announcements. Please use the next directions to upgrade to any version of DAHDI from Zaptel.

This guide assumes that all essential packages required for building from source are installed. Essentially, all commands need to be run with root privileges. You must use Asterisk 22 or later and Asterisk 0 or later, in order to use DAHDI. An example for RedHat based systems are.

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Let me tell you something. An example for Debian based systems will be. Please note that when upgrading from Zaptel, you need to upgrade to Zaptel 12". In addition, this will help remove any quite old versions of modules from your setup prior to upgrading.

It is a decent concept to ensure that a lot of Zaptel userspace tools aren't still present on the setup, once DAHDI is installed. While the DAHDI userspace tools are installed to /usr/sbin/, most Zaptel userspace tools are installed to /sbin/. You need to modify your configuration files to enable your hardware in DAHDI, with the intention to make specific that all the aptel userspace tools are removed you can run the succeeding command Now when upgrading from Zaptel. Most notably.

a good real overlook here's newest option 'echocanceller'. This option is good when you going to configure program echocancellation on a channel by channel basis. Script echocancellation should't function, when the echocanceller straight line is not specified here. This is the case. Use the format, with an intention to configure the default echo cancellers. To use MG2 on the 1st 8 channels.

When required you must be able to merely rename an existing zapata. A well-known reality that is. For example, all zaptel userspace tools have in addition been replaced with a DAHDI counterpart. All DAHDI tools are prefixed with 'dahdi_' now. For instance.

It's a well you can set 'dahdichanname=no' in the '' section of your 'etc/asterisk/asterisk, in case you will like to not modify your Dialplan while still running Asterisk 4. This will tell Asterisk to keep looking in zapata. Asterisk channel configuration and to keep referring to hardware channels as 'Zap' channels. That's where it starts getting very entertaining, right? You will still need to configure '/etc/dahdi/scheme. It is advised to use 'dahdi_genconf' to configure your hardware once dahdi was installed and loaded. This will automatically configure the method. Lots of info can be found easily on the web. Asterisk for the channels or as a starting point for modifying configuration to suit your situation. Needless to say, in the simplest use case, add the succeeding threshold to /etc/asterisk/chan_dahdi. Stop asterisk or stop the Zaptel driver, start the DAHDI driver and start Asterisk, right after the configuration files are checked then you can just restart your machine to pick up the revisal. Top whitey Papers and Webcasts.

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