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Overview

Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such, the focus of development for this release of Asterisk was on improving the usability and features developed in the previous Standard release, Asterisk 12. Beyond a general refinement of end user features, development focussed heavily on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the new features include:

  • Asterisk security events are now provided via  AMI, allowing end users to monitor their Asterisk system in real time for security related issues.
  • External control of Message Waiting Indicators (MWI) through both AMI and ARI.
  • Reception/transmission of out of call text messages using any supported channel driver/protocol stack through ARI.
  • Resource List Server support in the PJSIP stack, providing subscriptions to lists of resources and batched delivery of NOTIFY requests.
  • Inter-Asterisk distributed device state and mailbox state using the PJSIP stack.

And much more!

It is important to note that Asterisk 13 is built on the architecture developed during the previous Standard release, Asterisk 12. Users upgrading to Asterisk 13 should read about the new features documented in New in 12, as well as the notes on upgrading to Asterisk 12. In particular, users upgrading to Asterisk 13 from a release prior to Asterisk 12 should read the specifications on AMI, CDRs, and CEL, as these also apply to Asterisk 13:

Finally, all users upgrading to Asterisk 13 should read the notes on upgrading to Asterisk 13.

 

Tip
titleAsterisk 12 was different

Some of the new features listed below were released in point releases of Asterisk 12. Per the Software Configuration Management Policies laid out for Asterisk 12, new features were periodically merged and released in that branch of Asterisk. This was done to help users of Asterisk migrating to the new platform develop features in preparation for Asterisk 13.

While some of the features listed below were released under an Asterisk 12 release, they are all listed here as "new in 13", for two reasons:

  1. If you are upgrading from a previous LTS release (such as Asterisk 11), all of these features are new.
  2. If you are upgrading from some version of Asterisk 12, some of the previously released features may be new (as they may not have been in your version of Asterisk 12).

Applications

AgentRequest

  • The application will now return a new AGENT_STATUS value of NOT_CONNECTED if the agent fails to connect with an incoming caller after being alerted to the presence of the incoming caller. The most likely reason this would happen is the agent did not acknowledge the call in time.

ChanSpy

  • ChanSpy now accepts a channel uniqueid or a fully specified channel name as the chanprefix parameter if the 'u' option is specified.

ConfBridge

  • The ConfBridge dialplan application now sets a channel variable, CONFBRIGE_RESULT, upon exiting. This variable can be used to determine how a channel exited the conference. Valid values upon exiting are:

    ValueReason
    FAILED
    		The channel encountered an error and could not enter the conference.
    HANGUP
    		The channel exited the conference by hanging up.
    KICKED
    		The channel was kicked from the conference.
    ENDMARKED
    		The channel left the conference as a result of the last marked user leaving.
    DTMF
    		The channel pressed a DTMF sequence to exit the conference.
  • Added conference user option 'announce_join_leave_review'. This option implies 'announce_join_leave' with the added effect that the user will be asked if they want to confirm or re-record the recording of their name when entering the conference.

DAHDIBarge

  • The module app_dahdibarge was deprecated and has been removed. Users of DAHDIBarge should use ChanSpy instead.

Directory

  • At exit, the Directory application now sets a channel variable DIRECTORY_RESULT to one of the following based on the reason for exiting:

    ValueReason
    OPERATOR
    		user requested operator by pressing '0' for operator
    ASSISTANT
    		user requested assistant by pressing '*' for assistant
    TIMEOUT
    		user pressed nothing and Directory stopped waiting
    HANGUP
    		user's channel hung up
    SELECTED
    		user selected a user from the directory and is routed
    USEREXIT
    		 user pressed '#' from the selection prompt to exit
    FAILED
    		directory failed in a way that wasn't accounted for. Dang.
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Table of Contents

 MusicOnHold

  • MusicOnHold streams (all modes other than "files") now support wide band audio.

...

  • A new module, res_pjsip_multihomed handles situations where the system Asterisk is running out has multiple interfaces. res_pjsip_multihomed determines which interface should be used during message sending.

res_pjsip_outbound_publish

res_pjsip_outbound_registration

  • A new CLI command has been added: pjsip show registrations, which lists all configured PJSIP registrations.

res_pjsip_pidf_digium_body_supplement

  • A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY request body formatting for presence support in Digium phones.

Anchor
res_pjsip_pubsub
res_pjsip_pubsub
res_pjsip_pubsub

  • Subscriptions can now be persisted via the subscription_persistence object in pjsip.conf. Note that it is up to the configuration in sorcery.conf to determine how the subscription is persisted.
  • The publish/subscribe core module has been updated to support RFC 4662 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS). Resource lists are configured in pjsip.conf under a new object type, resource_list. Resource lists can contain either message-summary or presence events, can be composed of specific resources that provide the event, or other resource lists.
  • Inbound publication support is provided by a new object, inbound-publication. This configures res_pjsip_pubsub to accept PUBLISH requests from a particular resource. Which events are accepted is constructed dynamically; see res_pjsip_publish_asterisk for more information and Exchanging Device and Mailbox State Using PJSIP for examples on configuring this feature.

res_pjsip_pidf_digium_body_supplement

  • A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY request body formatting for presence support in Digium phones.

res_pjsip_send_to_voicemail

  • A new module, res_pjsip_send_to_voicemail allows for REFER requests with particular headers to transfer a PJSIP channel directly to a particular extension that has VoiceMail. This is intended to be used with Digium phones that support this feature.

res_pjsip_outbound_publish

res_pjsip_outbound_registration

  • A new CLI command has been added: pjsip show registrations, which lists all configured PJSIP registrations.

Anchor
res_pjsip_publish_asterisk
res_pjsip_publish_asterisk
res_pjsip_publish_asterisk

  • A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of Asterisk information to other Asterisk servers. This module is intended only for Asterisk to Asterisk exchanges of information. Currently, this includes both mailbox state and device state information. See Exchanging Device and Mailbox State Using PJSIP for examples on configuring this feature.

res_pjsip_send_to_voicemail

  • A new module, res_pjsip_send_to_voicemail allows for REFER requests with particular headers to transfer a PJSIP channel directly to a particular extension that has VoiceMail. This is intended to be used with Digium phones that support this feature.