Overview
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
-Ioption and theasterisk.confinternal_timingoption 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
rasteriskis used to connect to an earlier version. If the newrasteriskconnects to an older Asterisk version then the root console verbose level is always affected by thecore set verbosecommand of the remote console even though it may appear to only affect the current console. If an older version ofrasteriskconnects to the new version of Asterisk then thecore set verbosecommand will have no effect. - The asterisk compatibility options in
asterisk.confhave been removed. These options enabled certain backwards compatibility features forpbx_realtime,res_agi, andapp_setthat 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.
Applications
ConfBridge
- The
sound_place_into_conferencesound 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 usesound_beginto play a sound to the conference whenwaitmarkedusers are moved into the conference.
SetMusicOnHold
- The SetMusicOnHold dialplan application was deprecated and has been removed. Users of the application should use the
CHANNELfunction'smusicclasssetting instead.
WaitMusicOnHold
- The WaitMusicOnHold dialplan application was deprecated and has been removed. Users of the application should use MusicOnHold with a
durationparameter instead.
Build System
- Sample config files have been moved from
configs/to a sub-folder of that directory,samples. - The
menuselectutility has been pulled into the Asterisk repository. As a result, thelibxml2development 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 therefslog 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,refcounter.py, in thecontribfolder that will process therefslog file. Note that this replaces therefcounterutility that could be built from theutilsdirectory.
CDR Backends
cdr_sqlite
- The
cdr_sqlitemodule was deprecated and has been removed. Users of this module should use thecdr_sqlite3_custommodule instead.
Channel Drivers
chan_dahdi
- SS7 support now requires
libss7v2.0 or later. - Added the
inband_on_setup_ackcompatibility option tochan_dahdi.confto deal with switches that don't send an inband progress indication in theSETUP ACKNOWLEDGEmessage. Default is nowno.
chan_gtalk
- This module was deprecated and has been removed. Users of
chan_gtalkshould usechan_motif.
chan_h323
- This module was deprecated and has been removed. Users of
chan_h323should usechan_ooh323.
chan_jingle
- This module was deprecated and has been removed. Users of
chan_jingleshould usechan_motif.
chan_pjsip
- Added a
force_avpoption tochan_pjsipwhich will force the usage ofRTP/AVP,RTP/AVPF,RTP/SAVP, orRTP/SAVPFas 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_transportoption tochan_pjsipwhich will cause the SDP answer to use the media transport as received in the SDP offer.
chan_sip
- Made set
SIPREFERREDBYHDRas inheritable for betterchan_pjsipinteroperability. - The
SIPPEERdialplan function no longer supports using a colon as a delimiter for parameters. The parameters for the function should be delimited using a comma. - The
SIPCHANINFOdialplan function was deprecated and has been removed. Users of the function should use theCHANNELfunction instead. - Added a
force_avpoption forchan_sip. When enabled this option will cause the media transport in the offer or answer SDP to beRTP/AVP,RTP/AVPF,RTP/SAVP, orRTP/SAVPFeven 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
dtlsverifyoption inchan_sipnow has additional values besidesyesandno. Ifyesis specified both the certificate and fingerprint will be verified. Ifnois specified then neither the certificate or fingerprint is verified. Ifcertificateis specified then only the certificate is verified. Iffingerprintis specified then only the fingerprint is verified. - A
dtlsfingerprintoption has been added tochan_sipwhich allows the hash to be specified for the DTLS fingerprint placed in SDP. Supported values aresha-1andsha-256withsha-256being the default. - The
progressinband=neveroption is now more zealous in the persecution of progress messages coming from Asterisk. Channels bridged with a SIP channel that hasprogressinband=neverset will not be able to forward their progress indications through to the SIP device.chan_sipwill now turn such progress indications into a 180 Ringing (if a 180 has not yet been transmitted) ifprogressinband=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:
- Our preferred codec
- Our configured codecs
- Any non-audio joint codecs
Now, in Asterisk 13, the preference order of codecs is:
- Our preferred codec
- Any joint codecs offered by the inbound offer
- All other codecs that are not the preferred codec and not a joint codec offered by the inbound offer
- chan_sip is now an extended support module.
chan_unistim
- The
unistim.confdateformathas changed the meaning of options values to conform to the values used inside Unistim protocol. - Added
dtmf_durationoption with changing default operation to disable received DTMF playback on a Unistim phone.
Core
- The behaviour of
accountcodehas changed somewhat to supportpeeraccount. The main change is that Local channels now crossaccountcodeandpeeraccountsettings across the special bridge between the;1and;2channels just like channels between normal bridges. See New in 13 for more information.
ARI
- 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.
AMI
- 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:
- MixMonitor -
system - MixMonitorMute -
callorsystem - StopMixMonitor -
callorsystem
- MixMonitor -
- The undocumented
manager.confsettingblock-socketshas 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.confsettingeventfilternow takes an "extended" regular expression instead of a "basic" one.
CDR
- The
endbeforehextensetting now defaults toyes, instead ofno. When set tono, 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.
CLI
core show settingsnow lists the current console verbosity in addition to the root console verbosity.core set verbosehas 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.
HTTP
- Added
http.confsession_inactivitytimer 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.confwith thesession_keep_aliveparameter.
Logging
- The
verbosesetting 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 withcore set verbosecould again set the root console verbose level and affect the verbose level logged.
RealTime
- A number of Alembic scripts have been updated between Asterisk 12 and Asterisk 13. These include the following:
- For the
configRealTime schemas:1758e8bbf6b_increase_useragent_column_size.py- increase the size of theuseragentcolumn insippeersfrom20characters to255characters.1d50859ed02e_create_accountcode.py- add theaccountcodecolumn to theps_endpointstable.21e526ad3040_add_pjsip_debug_option.py- add thedebugcolumn to theps_globalstable.28887f25a46f_create_queue_tables.py- creates the various Queue related tables.2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py- adds theps_systems,ps_globals,ps_transports, andps_registrationstables. Adds several new columns forps_endpoints,ps_contacts, andps_aors.3855ee4e5f85_add_missing_pjsip_options.py- adds themessage_contextcolumn for theps_endpointstable and theuser_agentcolumn for theps_contactstable.4c573e7135bd_fix_tos_field_types.py- changes the type of theps_endpoints.tos_audio,ps_endpoints.tos_video, andps_transports.toscolumns.5139253c0423_make_q_member_uniqueid_autoinc.py- modifies theuniqueidcolumn on thequeue_memberstable to be a unique auto-incrementing index, if the database supports it.51f8cb66540e_add_further_dtls_options.py- adds theforce_avpandmedia_use_received_transportcolumns to theps_endpointstable.c6d929b23a8_create_pjsip_subscription_persistence_.py- adds theps_subscription_persistencetable.e96a0b8071c_increase_pjsip_column_size.py- increases the size of the columnsps_globals.user_agent,ps_contacts.id,ps_contacts.uri,ps_contacts.user_agent,ps_registrations.client_uri, andps_registrations.server_uri.
For the
voicemailODBC backend schemas:39428242f7f5_increase_recording_column_size.py - changed the type of the
voicemail_messages.recordingcolumn toLargeBinary, with a max size of4294967295.
Added a new family of schemas for CDR backends,
cdr.
- For the
Resources
res_http_websocket
- Added a compatibility option to
ari.conf,sip.conf, andpjsip.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.
res_odbc
- The compatibility setting,
allow_empty_string_in_nontext, has been removed. Empty column values will be stored as empty strings during RealTime updates.
res_jabber
- This module was deprecated and has been removed. Users of this module should use
res_xmppinstead.
Scripts
safe_asterisk
- The
safe_asteriskscript was previously not installed on top of an existing version. This caused bug-fixes in that script not to be deployed. If yoursafe_asteriskscript is customized, be sure to keep your changes. Custom values for variables should be created in*.shfile(s) insideASTETCDIR/startup.d/. For more information, see the original bug report that necessitated this change, ASTERISK-21965. - Changed a log message in
safe_asteriskand the$NOTIFYmail 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)"
Utilities
refcounter
- The
refcounterprogram has been removed in favour of therefcounter.pyscript incontrib/scripts.