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Update for 21.0.0-rc1

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Asterisk Development Team
2023-09-06 18:21:42 +00:00
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Change Log for Release asterisk-21.0.0-rc1
========================================
Links:
----------------------------------------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0-rc1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0-rc1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0-rc1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
Summary:
----------------------------------------
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- Update config.yml
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
User Notes:
----------------------------------------
- ### sig_analog: Add Called Subscriber Held capability.
Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
- ### res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
- ### http.c: Minor simplification to HTTP status output.
For bound addresses, the HTTP status page now combines the bound
address and bound port in a single line. Additionally, the SSL bind
address has been renamed to TLS.
Upgrade Notes:
----------------------------------------
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
ast_gethostbyname() has been deprecated and will be removed
in Asterisk 23. New code should use `ast_sockaddr_resolve()` and
`ast_sockaddr_resolve_first_af()`.
- ### app_sla: Migrate SLA applications out of app_meetme.
The SLAStation and SLATrunk applications have been moved
from app_meetme to app_sla. If you are using these applications and have
autoload=no, you will need to explicitly load this module in modules.conf.
- ### users.conf: Deprecate users.conf configuration.
The users.conf config is now deprecated
and will be removed in a future version of Asterisk.
- ### app_osplookup: Remove deprecated module.
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
- ### res_monitor: Remove deprecated module.
This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
This also removes the 'w' and 'W' options
for app_queue.
MixMonitor should be default and only option
for all settings that previously used either
Monitor or MixMonitor.
- ### chan_sip: Remove deprecated module.
This module was deprecated in Asterisk 17
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
- ### chan_alsa: Remove deprecated module.
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
- ### chan_mgcp: Remove deprecated module.
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
- ### chan_skinny: Remove deprecated module.
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
- ### app_macro: Remove deprecated module.
This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
For most modules that interacted with app_macro,
this change is limited to no longer looking for
the current context from the macrocontext when set.
The following modules have additional impacts:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs to be re-written
app_queue - can no longer call a macro on the called party's
channel. Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected
line or redirection macro options
options - stdexten is deprecated to gosub as the default
and only options
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority
- ### pbx_builtins: Remove deprecated and defunct functionality.
The previously deprecated ImportVar and SetAMAFlags
applications have now been removed.
- ### translate.c: Prefer better codecs upon translate ties.
When setting up translation between two codecs the quality was not taken into account,
resulting in suboptimal translation. The quality is now taken into account,
which can reduce the number of translation steps required, and improve the resulting quality.
- ### app_cdr: Remove deprecated application and option.
The previously deprecated NoCDR application has been removed.
Additionally, the previously deprecated 'e' option to the ResetCDR
application has been removed.
Closed Issues:
----------------------------------------
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits
Commits By Author:
----------------------------------------
- ### Bastian Triller (1):
- res_pjsip_session: Send Session Interval too small response
- ### George Joseph (9):
- Remove unneeded CHANGES and UPGRADE files
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- rest-api: Run make ari-stubs
- download_externals: Fix a few version related issues
- alembic: Fix quoting of the 100rel column
- .github: Update workflow-application-token-action to v2
- ari-stubs: Fix broken documentation anchors
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ### Jason D. McCormick (1):
- install_prereq: Fix dependency install on aarch64.
- ### Joshua C. Colp (1):
- manager: Tolerate stasis messages with no channel snapshot.
- ### Matthew Fredrickson (1):
- Revert "app_stack: Print proper exit location for PBXless channels."
- ### Maximilian Fridrich (1):
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- ### Mike Bradeen (1):
- app_voicemail: Fix for loop declarations
- ### MikeNaso (1):
- res_pjsip.c: Set contact_user on incoming call local Contact header
- ### Naveen Albert (4):
- res_pjsip_header_funcs: Make prefix argument optional.
- sig_analog: Add Called Subscriber Held capability.
- pbx.c: Fix gcc 12 compiler warning.
- app_dial: Fix infinite loop when sending digits.
- ### Sean Bright (1):
- extconfig: Allow explicit DB result set ordering to be disabled.
- ### zhengsh (1):
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
Detail:
----------------------------------------
- ### Update master branch for Asterisk 21
Author: George Joseph
Date: 2022-07-20
- ### translate.c: Prefer better codecs upon translate ties.
Author: Naveen Albert
Date: 2021-05-27
If multiple codecs are available for the same
resource and the translation costs between
multiple codecs are the same, ties are
currently broken arbitrarily, which means a
lower quality codec would be used. This forces
Asterisk to explicitly use the higher quality
codec, ceteris paribus.
ASTERISK-29455
- ### chan_skinny: Remove deprecated module.
Author: Mike Bradeen
Date: 2022-11-16
ASTERISK-30300
- ### app_osplookup: Remove deprecated module.
Author: Mike Bradeen
Date: 2022-11-18
ASTERISK-30302
- ### chan_mgcp: Remove deprecated module.
Author: Mike Bradeen
Date: 2022-11-15
Also removes res_pktcops to avoid merge conflicts
with ASTERISK~30301.
ASTERISK-30299
- ### chan_alsa: Remove deprecated module.
Author: Mike Bradeen
Date: 2022-11-14
ASTERISK-30298
- ### pbx_builtins: Remove deprecated and defunct functionality.
Author: Naveen Albert
Date: 2022-11-29
This removes the ImportVar and SetAMAFlags applications
which have been deprecated since Asterisk 12, but were
never removed previously.
Additionally, it removes remnants of defunct options
that themselves were removed years ago.
ASTERISK-30335 #close
- ### chan_sip: Remove deprecated module.
Author: Mike Bradeen
Date: 2022-11-28
ASTERISK-30297
- ### app_cdr: Remove deprecated application and option.
Author: Naveen Albert
Date: 2022-12-22
This removes the deprecated NoCDR application, which
was deprecated in Asterisk 12, having long been fully
superseded by the CDR_PROP function.
The deprecated e option to ResetCDR is also removed
for the same reason.
ASTERISK-30371 #close
- ### app_macro: Remove deprecated module.
Author: Mike Bradeen
Date: 2022-12-12
For most modules that interacted with app_macro, this change is limited
to no longer looking for the current context from the macrocontext when
set. Additionally, the following modules are impacted:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs a re-write
app_queue - can no longer a macro on the called party's channel.
Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected line or
redirection macro options
options - stdexten is deprecated to gosub as the default and only
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority
ASTERISK-30304
- ### res_monitor: Remove deprecated module.
Author: Mike Bradeen
Date: 2022-11-18
ASTERISK-30303
- ### http.c: Minor simplification to HTTP status output.
Author: Boris P. Korzun
Date: 2023-01-05
Change the HTTP status page (located at /httpstatus by default) by:
* Combining the address and port into a single line.
* Changing "SSL" to "TLS"
ASTERISK-30433 #close
- ### app_osplookup: Remove obsolete sample config.
Author: Naveen Albert
Date: 2023-02-24
ASTERISK_30302 previously removed app_osplookup,
but its sample config was not removed.
This removes it since nothing else uses it.
ASTERISK-30438 #close
- ### say.c: Fix French time playback. (#42)
Author: InterLinked1
Date: 2023-05-02
ast_waitstream was not called after ast_streamfile,
resulting in "o'clock" being skipped in French.
Additionally, the minute announcements should be
feminine.
Reported-by: Danny Lloyd
Resolves: #41
ASTERISK-30488
- ### core: Cleanup gerrit and JIRA references. (#58)
Author: Sean Bright
Date: 2023-05-03
* Remove .gitreview and switch to pulling the main asterisk branch
version from configure.ac instead.
* Replace references to JIRA with GitHub.
* Other minor cleanup found along the way.
Resolves: #39
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
Author: Sean Bright
Date: 2023-05-11
Deprecate `ast_gethostbyname()` in favor of `ast_sockaddr_resolve()` and
`ast_sockaddr_resolve_first_af()`. `ast_gethostbyname()` has not been
used by any in-tree code since 2021.
This function will be removed entirely in Asterisk 23.
Resolves: #78
UpgradeNote: ast_gethostbyname() has been deprecated and will be removed
in Asterisk 23. New code should use `ast_sockaddr_resolve()` and
`ast_sockaddr_resolve_first_af()`.
- ### res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
Author: InterLinked1
Date: 2023-05-18
The existing res_pjsip_pubsub APIs are somewhat limited in
what they can do. This adds a few API extensions that make
it possible for PJSIP pubsub modules to implement richer
features than is currently possible.
* Allow pubsub modules to get a handle to pjsip_rx_data on subscription
* Allow pubsub modules to run a callback when a subscription is renewed
* Allow pubsub modules to run a callback for outgoing NOTIFYs, with
a handle to the tdata, so that modules can append their own headers
to the NOTIFYs
This change does not add any features directly, but makes possible
several new features that will be added in future changes.
Resolves: #81
ASTERISK-30485 #close
Master-Only: True
- ### app_sla: Migrate SLA applications out of app_meetme.
Author: Naveen Albert
Date: 2023-05-02
This removes the dependency of the SLAStation and SLATrunk
applications on app_meetme, in anticipation of the imminent
removal of the deprecated app_meetme module.
The user interface for the SLA applications is exactly the
same, and in theory, users should not notice a difference.
However, the SLA applications now use ConfBridge under the
hood, rather than MeetMe, and they are now contained within
their own module.
Resolves: #50
ASTERISK-30309
UpgradeNote: The SLAStation and SLATrunk applications have been moved
from app_meetme to app_sla. If you are using these applications and have
autoload=no, you will need to explicitly load this module in modules.conf.
- ### Update config.yml
Author: Joshua C. Colp
Date: 2023-06-15
- ### rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
Author: George Joseph
Date: 2023-06-27
- ### .github: Update AsteriskReleaser for security releases
Author: George Joseph
Date: 2023-07-07
- ### users.conf: Deprecate users.conf configuration.
Author: Naveen Albert
Date: 2023-06-30
This deprecates the users.conf config file, which
is no longer as widely supported but still integrated
with a number of different modules.
Because there is no real mechanism for marking a
configuration file as "deprecated", and users.conf
is not just used in a single place, this now emits
a warning to the user when the PBX loads to notify
about the deprecation.
This configuration mechanism has been widely criticized
and discouraged since its inception, and is no longer
relevant to the configuration that most users are doing
today. Removing it will allow for some simplification
and cleanup in the codebase.
Resolves: #183
UpgradeNote: The users.conf config is now deprecated
and will be removed in a future version of Asterisk.
- ### Update version for Asterisk 21
Author: George Joseph
Date: 2023-08-09
- ### Remove unneeded CHANGES and UPGRADE files
Author: George Joseph
Date: 2023-08-09
- ### ari-stubs: Fix more local anchor references
Author: George Joseph
Date: 2023-09-05
Also allow CreateDocs job to be run manually with default branches.
- ### ari-stubs: Fix more local anchor references
Author: George Joseph
Date: 2023-09-05
Also allow CreateDocs job to be run manually with default branches.
- ### ari-stubs: Fix broken documentation anchors
Author: George Joseph
Date: 2023-09-05
All of the links that reference page anchors with capital letters in
the ids (#Something) have been changed to lower case to match the
anchors that are generated by mkdocs.
- ### res_pjsip_session: Send Session Interval too small response
Author: Bastian Triller
Date: 2023-08-28
Handle session interval lower than endpoint's configured minimum timer
when sending first answer. Timer setting is checked during this step and
needs to handled appropriately.
Before this change, no response was sent at all. After this change a
response with 422 Session Interval too small is sent to UAC.
- ### .github: Update workflow-application-token-action to v2
Author: George Joseph
Date: 2023-08-31
- ### app_dial: Fix infinite loop when sending digits.
Author: Naveen Albert
Date: 2023-08-28
If the called party hangs up while digits are being
sent, -1 is returned to indicate so, but app_dial
was not checking the return value, resulting in
the hangup being lost and looping forever until
the caller manually hangs up the channel. We now
abort if digit sending fails.
ASTERISK-29428 #close
Resolves: #281
- ### app_voicemail: Fix for loop declarations
Author: Mike Bradeen
Date: 2023-08-29
Resolve for loop initial declarations added in cli changes.
Resolves: #275
- ### alembic: Fix quoting of the 100rel column
Author: George Joseph
Date: 2023-08-28
Add quoting around the ps_endpoints 100rel column in the ALTER
statements. Although alembic doesn't complain when generating
sql statements, postgresql does (rightly so).
Resolves: #274
- ### pbx.c: Fix gcc 12 compiler warning.
Author: Naveen Albert
Date: 2023-08-27
Resolves: #277
- ### app_audiosocket: Fixed timeout with -1 to avoid busy loop.
Author: zhengsh
Date: 2023-08-24
Resolves: asterisk#234
- ### download_externals: Fix a few version related issues
Author: George Joseph
Date: 2023-08-18
* Fixed issue with the script not parsing the new tag format for
certified releases. The format changed from certified/18.9-cert5
to certified-18.9-cert5.
* Fixed issue where the asterisk version wasn't being considered
when looking for cached versions.
Resolves: #263
- ### main/refer.c: Fix double free in refer_data_destructor + potential leak
Author: Maximilian Fridrich
Date: 2023-08-21
Resolves: #267
- ### sig_analog: Add Called Subscriber Held capability.
Author: Naveen Albert
Date: 2023-08-09
This adds support for Called Subscriber Held for FXS
lines, which allows users to go on hook when receiving
a call and resume the call later from another phone on
the same line, without disconnecting the call. This is
a convenience mechanism that most real PSTN telephone
switches support.
ASTERISK-30372 #close
Resolves: #240
UserNote: Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
- ### Revert "app_stack: Print proper exit location for PBXless channels."
Author: Matthew Fredrickson
Date: 2023-08-10
This reverts commit 617dad4cba1513dddce87b8e95a61415fb587cf1.
apps/app_stack.c: Revert buggy gosub patch
This seems to break the case when a predial macro calls a gosub.
When the gosub calls return, the Return function outputs:
app_stack.c:423 return_exec: Return without Gosub: stack is empty
This returns -1 to the calling macro, which returns to app_dial
and causes the call to hangup instead of proceeding with the macro
that invoked the gosub.
Resolves: #253
- ### install_prereq: Fix dependency install on aarch64.
Author: Jason D. McCormick
Date: 2023-04-28
Fixes dependency solutions in install_prereq for Debian aarch64
platforms. install_prereq was attempting to forcibly install 32-bit
armhf packages due to the aptitude search for dependencies.
Resolves: #37
- ### res_pjsip.c: Set contact_user on incoming call local Contact header
Author: MikeNaso
Date: 2023-08-08
If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.
Resolves: #226
- ### extconfig: Allow explicit DB result set ordering to be disabled.
Author: Sean Bright
Date: 2023-07-12
Added a new boolean configuration flag -
`order_multi_row_results_by_initial_column` - to both res_pgsql.conf
and res_config_odbc.conf that allows the administrator to disable the
explicit `ORDER BY` that was previously being added to all generated
SQL statements that returned multiple rows.
Fixes: #179
- ### rest-api: Run make ari-stubs
Author: George Joseph
Date: 2023-08-09
An earlier cherry-pick that involved rest-api somehow didn't include
a comment change in res/ari/resource_endpoints.h. This commit
corrects that. No changes other than the comment.
- ### res_pjsip_header_funcs: Make prefix argument optional.
Author: Naveen Albert
Date: 2023-08-09
The documentation for PJSIP_HEADERS claims that
prefix is optional, but in the code it is actually not.
However, there is no inherent reason for this, as users
may want to retrieve all header names, not just those
beginning with a certain prefix.
This makes the prefix optional for this function,
simply fetching all header names if not specified.
As a result, the documentation is now correct.
Resolves: #230
UserNote: The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
- ### pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
Author: George Joseph
Date: 2023-08-11
The default is 32 with 8 being used by pjproject itself. Recent
commits have put us over the limit resulting in assertions in
pjproject. Since this value is used in invites, dialogs,
transports and subscriptions as well as the global pjproject
endpoint, we don't want to increase it too much.
Resolves: #255
- ### manager: Tolerate stasis messages with no channel snapshot.
Author: Joshua C. Colp
Date: 2023-08-09
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
- ### Remove unneeded CHANGES and UPGRADE files
Author: George Joseph
Date: 2023-08-09

0
CHANGES → ChangeLogs/historical/CHANGES
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3
UPGRADE.txt
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@@ -1,3 +1,6 @@
===== WARNING, THIS FILE IS OBSOLETE AND WILL BE REMOVED IN A FUTURE VERSION =====
See 'Upgrade Notes' in the CHANGES file
===========================================================
===
=== THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE

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CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start DATETIME,
answer DATETIME,
end DATETIME,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
-- Running upgrade 210693f3123d -> 54cde9847798
ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL;
ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL;
UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';

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@@ -0,0 +1,29 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 4105ee839f58
CREATE TABLE queue_log (
id BIGSERIAL NOT NULL,
time TIMESTAMP WITHOUT TIME ZONE,
callid VARCHAR(80),
queuename VARCHAR(256),
agent VARCHAR(80),
event VARCHAR(32),
data1 VARCHAR(100),
data2 VARCHAR(100),
data3 VARCHAR(100),
data4 VARCHAR(100),
data5 VARCHAR(100),
PRIMARY KEY (id),
UNIQUE (id)
);
INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58');
COMMIT;

35
contrib/realtime/mysql/mysql_voicemail.sql Normal file
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@@ -0,0 +1,35 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR(255) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR(80),
macrocontext VARCHAR(80),
callerid VARCHAR(80),
origtime INTEGER,
duration INTEGER,
recording BLOB,
flag VARCHAR(30),
category VARCHAR(30),
mailboxuser VARCHAR(30),
mailboxcontext VARCHAR(30),
msg_id VARCHAR(40)
);
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';

45
contrib/realtime/postgresql/postgresql_cdr.sql Normal file
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@@ -0,0 +1,45 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start TIMESTAMP WITHOUT TIME ZONE,
answer TIMESTAMP WITHOUT TIME ZONE,
"end" TIMESTAMP WITHOUT TIME ZONE,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
-- Running upgrade 210693f3123d -> 54cde9847798
ALTER TABLE cdr ALTER COLUMN accountcode TYPE VARCHAR(80);
ALTER TABLE cdr ALTER COLUMN peeraccount TYPE VARCHAR(80);
UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
COMMIT;

1542
contrib/realtime/postgresql/postgresql_config.sql Normal file
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29
contrib/realtime/postgresql/postgresql_queue_log.sql Normal file
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BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 4105ee839f58
CREATE TABLE queue_log (
id BIGSERIAL NOT NULL,
time TIMESTAMP WITHOUT TIME ZONE,
callid VARCHAR(80),
queuename VARCHAR(256),
agent VARCHAR(80),
event VARCHAR(32),
data1 VARCHAR(100),
data2 VARCHAR(100),
data3 VARCHAR(100),
data4 VARCHAR(100),
data5 VARCHAR(100),
PRIMARY KEY (id),
UNIQUE (id)
);
INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58');
COMMIT;

39
contrib/realtime/postgresql/postgresql_voicemail.sql Normal file
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BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR(255) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR(80),
macrocontext VARCHAR(80),
callerid VARCHAR(80),
origtime INTEGER,
duration INTEGER,
recording BYTEA,
flag VARCHAR(30),
category VARCHAR(30),
mailboxuser VARCHAR(30),
mailboxcontext VARCHAR(30),
msg_id VARCHAR(40)
);
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
COMMIT;