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AUDIOHOOK_INHERIT

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  • The AUDIOHOOK_INHERIT function has been deprecated. Audiohooks are now unconditionally

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  • inherited through masquerades. As a side benefit, more than one audiohook of a given type may persist through a masquerade now.

CONFBRIDGE

  • The CONFBRIDGE dialplan function is now capable of creating/modifying dynamic conference user menus.
  • The CONFBRIDGE dialplan function is now capable of removing dynamic conference menus, bridge settings, and user settings that have been applied by the CONFBRIDGE the CONFBRIDGE dialplan function.

JACK_HOOK

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  • The JACK_HOOK function now supports audio with a sample rate higher than 8kHz.

MIXMONITOR

  • A new function, MIXMONITOR, has been added to allow access to individual instances of MixMonitor on a channel.

PERIODIC_HOOK

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  • A new function

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  • , has been added which allows for running a periodic dialplan hook on a channel.  Any audio generated by this hook will be injected into the call.

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TALK_DETECT
TALK_DETECT
TALK_DETECT

A new function, TALK_DETECT, has been added. When set on a channel, this function causes events indicating the starting/stopping of talking on said channel to be emitted to both AMI and ARI clients.

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Added the ability to support PostgreSQL PostgreSQL application_name on connections. This allows PostgreSQL to display the configured name in the pgthe pg_stat_activity view and CSV log entries. This setting is configurable for configurable for res_config_pgsql via the the dbappname configuration setting in resin res_pgsql.conf.

res_pjsip

  • The endpoint configuration object now supports 'accountcode'. Any channel created for an endpoint with this setting will have its accountcode set to the specified value.
  • transport and endpoint ToS options (tos, tos_audio, and tos_video) may now be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
  • Added the following new CLI commands:
    • "pjsip show contacts" - list all current PJSIP contacts.
    • "pjsip show contact" - show specific information about a current PJSIP contact.
    • "pjsip show channel" - show detailed information about a PJSIP channel.

 

  • Path support has been added with the 'support_path' option in registration and aor sections.
  • A 'debug' option has been added to the globals section that will allow sip messages to be logged.
  • A 'set_var' option has been added to endpoints that will automatically set the desired variable(s) on a channel created for that endpoint.
  • DNS functionality will now automatically be enabled if the system configured nameservers can be retrieved. If the system configured nameservers can not be retrieved the functionality will resort to using basic system resolution. Functionality such as SRV records and fail-over will not be available if the basic system resolution is in use.
  • Several new tables and columns have been added to the realtime schema for the res_pjsip related modules. See the UPGRADE.txt notes for updating the database schema.

 

res_pjsip_multihomed

  • 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_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

asdf

res_pjsip_outbound_registration

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

res_pjsip_publish_asterisk

asdf

res_hep

 * A new module, res_hep, has been added that acts as a generic packet capture agent for the Homer Encapsulation Protocol (HEP) version 3. It can be configured via hep.conf. Other modules can use res_hep to send message traffic to a HEP capture server.

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res_mwi_external
res_mwi_external

res_mwi_external

A new module, res_mwi_external, has been added to Asterisk. This module acts as a base framework that other modules can build on top of to allow an external system to control MWI within Asterisk. For implementations that make use of res_mwi_external, see res_mwi_external_ami and res_ari_mailboxes. Note that res_mwi_external canflicts with other modules that may produce MWI themselves, such as app_voicemail. res_mwi_external and other modules that depend on it cannot be built or loaded with app_voicemail present.

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 * Manager action 'Park' now takes an additional argument 'AnnounceChannel' which can be used to announce the parked call's location to an arbitrary channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two parties in a one to one bridge, 'TimeoutChannel' is treated as having parked 'Channel' like with the Park Call DTMF feature and will receive announcements prior to being hung up.