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As Asterisk 13 is built on the architecture introduced in Asterisk 12, users upgrading to Asterisk 13 from an older version of Asterisk should be aware of the architectural changes that were made in the previous Standard release. It is recommended that you review:

  • The upgrade notes on this page
  • The New in 13 information, which lists the major new features in Asterisk 13
  • The notes on Upgrading to Asterisk 12 if you are upgrading from a version of Asterisk prior to Asterisk 12
  • The notes on what is New in 12 if if you are upgrading from a version of Asterisk prior to Asterisk 12.

General Asterisk Updates

  • The asterisk command line -I option and the asterisk.conf internal_timing option have been removed. Internal timing is always enabled if any timing module is loaded.
  • The per console verbose level feature as previously implemented in Asterisk 11 caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the core set verbose command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version of Asterisk then the core set verbose command will have no effect.
  • The asterisk compatibility options in asterisk.conf have been removed. These options enabled certain backwards compatibility features for pbx_realtime, res_agi, and app_set that made their behaviour similar to Asterisk 1.4. Users who used these backwards compatibility settings should update their dialplans to use ',' instead of '|' as a delimiter, and should use the Set dialplan application instead of the MSet dialplan application.



  • The sound_place_into_conference sound used in ConfBridge is now deprecated and is no longer functional. It has technically been broken since its inception and - to meet its documented use case - a different method is used to achieve the same goal. The new method is to use sound_begin to play a sound to the conference when waitmarked users are moved into the conference.


  • The SetMusicOnHold dialplan application was deprecated and has been removed. Users of the application should use the CHANNEL function's musicclass setting instead.


  • The WaitMusicOnHold dialplan application was deprecated and has been removed. Users of the application should use MusicOnHold with a duration parameter instead.
On this Page

Build System

  • Sample config files have been moved from configs/ to a sub-folder of that directory, samples.
  • The menuselect utility has been pulled into the Asterisk repository. As a result, the libxml2 development library is now a required dependency for Asterisk.
  • A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference counted objects will emit additional debug information to the refs log file located in the standard Asterisk log file directory. This log file is useful in tracking down object leaks and other reference counting issues. Prior to this version, this option was only available by modifying the source code directly. This change also includes a new script,, in the contrib folder that will process the refs log file. Note that this replaces the refcounter utility that could be built from the utils directory.

CDR Backends


  • The cdr_sqlite module was deprecated and has been removed. Users of this module should use the cdr_sqlite3_custom module instead.

Channel Drivers


  • SS7 support now requires libss7 v2.0 or later.
  • Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to deal with switches that don't send an inband progress indication in the SETUP ACKNOWLEDGE message. Default is now no.


  • This module was deprecated and has been removed. Users of chan_gtalk should use chan_motif.


  • This module was deprecated and has been removed. Users of chan_h323 should use chan_ooh323.


  • This module was deprecated and has been removed. Users of chan_jingle should use chan_motif.


  • Added a force_avp option to chan_pjsip which will force the usage of RTP/AVP, RTP/AVPF, RTP/SAVP, or RTP/SAVPF as the media transport type in SDP offers depending on settings, even when DTLS is used for media encryption. This option can be set to improve interoperability with WebRTC clients that don't use the RFC defined transport for DTLS.
  • Added a media_use_received_transport option to chan_pjsip which will cause the SDP answer to use the media transport as received in the SDP offer.


  • Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip interoperability.
  • The SIPPEER dialplan function no longer supports using a colon as a delimiter for parameters. The parameters for the function should be delimited using a comma.
  • The SIPCHANINFO dialplan function was deprecated and has been removed. Users of the function should use the CHANNEL function instead.
  • Added a force_avp option for chan_sip. When enabled this option will cause the media transport in the offer or answer SDP to be RTP/AVP, RTP/AVPF, RTP/SAVP, or RTP/SAVPF even if a DTLS stream has been configured. This option can be set to improve interoperability with WebRTC clients that don't use the RFC defined transport for DTLS.
  • The dtlsverify option in chan_sip now has additional values besides yes and no. If yes is specified both the certificate and fingerprint will be verified. If no is specified then neither the certificate or fingerprint is verified. If certificate is specified then only the certificate is verified. If fingerprint is specified then only the fingerprint is verified.
  • dtlsfingerprint option has been added to chan_sip which allows the hash to be specified for the DTLS fingerprint placed in SDP. Supported values are sha-1 and sha-256 with sha-256 being the default.
  • The progressinband=never option is now more zealous in the persecution of progress messages coming from Asterisk. Channels bridged with a SIP channel that has progressinband=never set will not be able to forward their progress indications through to the SIP device. chan_sip will now turn such progress indications into a 180 Ringing (if a 180 has not yet been transmitted) if progressinband=never.
  • The codec preference order in an SDP during an offer is slightly different than previous releases. Prior to Asterisk 13, the preference order of codecs used to be:
    1. Our preferred codec
    2. Our configured codecs
    3. Any non-audio joint codecs

Internal Implementation Details Ahead


One of the ways the new media format architecture in Asterisk 13 improves performance is by reference counting formats, such that they can be reused in many places without additional allocation. To not require a large amount of locking, an instance of a format is immutable by convention. This works well except for formats with attributes. Since a media format with an attribute is a different object than the same format without an attribute, we have to carry over the formats with attributes from an inbound offer so that the correct attributes are offered in an outgoing INVITE request. This requires some subtle tweaks to the preference order to ensure that the media format with attributes is offered to a remote peer, as opposed to the same media format (but without attributes) that may be stored in the peer object. 

Now, in Asterisk 13, the preference order of codecs is:

    1. Our preferred codec
    2. Any joint codecs offered by the inbound offer
    3. All other codecs that are not the preferred codec and not a joint codec offered by the inbound offer


  • The unistim.conf dateformat has changed the meaning of options values to conform to the values used inside Unistim protocol.
  • Added dtmf_duration option with changing default operation to disable received DTMF playback on a Unistim phone.


  • The behaviour of accountcode has changed somewhat to support peeraccount. The main change is that Local channels now cross accountcode and peeraccount settings across the special bridge between the ;1 and ;2 channels just like channels between normal bridges.  See New in 13 for more information.


  • The ARI version has been changed to 1.5.0. This is to reflect the backwards compatible changes listed in New in 13.
  • A bug fix in bridge creation has caused a behavioural change in how subscriptions are created for bridges. A bridge created through ARI, does not, by itself, have a subscription created for any particular Stasis application. When a channel in a Stasis application joins a bridge, an implicit event subscription is created for that bridge as well. Previously, when a channel left such a bridge, the subscription was leaked; this allowed for later bridge events to continue to be pushed to the subscribed applications. That leak has been fixed; as a result, bridge events that were delivered after a channel left the bridge are no longer delivered. An application must subscribe to a bridge through the applications resource if it wishes to receive all events related to a bridge.


  • The AMI version has been changed to 2.5.0. This is to reflect the backwards compatible changes listed in New in 13.
  • MixMonitor AMI actions now require users to have authorization classes:
  • The undocumented manager.conf setting block-sockets has been removed. It interferes with TCP/TLS inactivity timeouts.
  • The response to the PresenceState AMI action has historically contained two Message keys. The first of these is used as an informative message regarding the success/failure of the action; the second contains a Presence state specific message. Having two keys with the same unique name in an AMI message is cumbersome for some client; hence, the Presence specific Message has been deprecated. The message will now contain a PresenceMessage key for the presence specific information; the Message key containing presence information will be removed in the next major version of AMI.
  • The manager.conf setting eventfilter now takes an "extended" regular expression instead of a "basic" one.


  • The endbeforehexten setting now defaults to yes, instead of no. When set to no, this setting will cause a new CDR to be generated when a channel enters into hangup logic (either the 'h' extension or a hangup handler subroutine). In general, this is not the preferred default: this causes extra CDRs to be generated for a channel in many common dialplans.


  • core show settings now lists the current console verbosity in addition to the root console verbosity.
  • core set verbose has not been able to support the by module verbose logging levels since verbose logging levels were made per console. That syntax is now removed and a silence option added in its place.


  • Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything.
  • Added support for persistent HTTP connections. To enable persistent HTTP connections configure the keep alive time between HTTP requests. The keep alive time between HTTP requests is configured in http.conf with the session_keep_alive parameter.


  • The verbose setting in logger.conf still takes an optional argument, specifying the verbosity level for each logging destination. However, the default is now to once again follow the current root console level. As a result, using the AMI Command action with core set verbose could again set the root console verbose level and affect the verbose level logged.




The database migration script that adds the extensions table had to be modified due to an error when installing for MySQL. The extensions table's id column was changed to be a primary key. This could potentially cause a migration problem. If so, it may be necessary to manually alter the affected table/column to bring it back in line with the migration scripts.

  • A number of Alembic scripts have been updated between Asterisk 12 and Asterisk 13. These include the following:
    • For the config RealTime schemas:
      • - increase the size of the useragent column in sippeers from 20 characters to 255 characters.

      • - add the accountcode column to the ps_endpoints table.

      • - add the debug column to the ps_globals table.

      • - creates the various Queue related tables.

      • - adds the ps_systems, ps_globalsps_transports, and ps_registrations tables. Adds several new columns for ps_endpointsps_contacts, and ps_aors.

      • - adds the message_context column for the ps_endpoints table and the user_agent column for the ps_contacts table.

      • - changes the type of the ps_endpoints.tos_audiops_endpoints.tos_video, and ps_transports.tos columns.

      • - modifies the uniqueid column on the queue_members table to be a unique auto-incrementing index, if the database supports it.

      • - adds the force_avp and media_use_received_transport columns to the ps_endpoints table.

      • - adds the ps_subscription_persistence table.

      • - increases the size of the columns ps_globals.user_agentps_contacts.idps_contacts.urips_contacts.user_agent, ps_registrations.client_uri, and ps_registrations.server_uri.

    • For the voicemail ODBC backend schemas:

      • - changed the type of the voicemail_messages.recording column to LargeBinary, with a max size of 4294967295.

    • Added a new family of schemas for CDR backends, cdr.



  • Added a compatibility option to ari.conf, sip.conf, and pjsip.conf - websocket_write_timeout. When a websocket connection exists where Asterisk writes a substantial amount of data to the connected client, and the connected client is slow to process the received data, the socket may be disconnected. In such cases, it may be necessary to adjust this value. Default is 100 ms.


  • The compatibility setting, allow_empty_string_in_nontext, has been removed. Empty column values will be stored as empty strings during RealTime updates.


  • This module was deprecated and has been removed. Users of this module should use res_xmpp instead.



  • The safe_asterisk script was previously not installed on top of an existing version. This caused bug-fixes in that script not to be deployed. If your safe_asterisk script is customized, be sure to keep your changes. Custom values for variables should be created in *.sh file(s) inside ASTETCDIR/startup.d/. For more information, see the original bug report that necessitated this change, ASTERISK-21965.
  • Changed a log message in safe_asterisk and the $NOTIFY mail subject. If you use tools to parse either of them, update your parse functions accordingly. The changed strings are:
    • "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
    • "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"



  • The refcounter program has been removed in favour of the script in contrib/scripts.


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