CVS commit: pkgsrc/comms/asterisk22

To: pkgsrc-changes%NetBSD.org@localhost
Subject: CVS commit: pkgsrc/comms/asterisk22
From: "John Nemeth" <jnemeth%netbsd.org@localhost>
Date: Mon, 22 Jun 2026 02:21:18 +0000

Module Name:    pkgsrc
Committed By:   jnemeth
Date:           Mon Jun 22 02:21:18 UTC 2026

Modified Files:
        pkgsrc/comms/asterisk22: Makefile PLIST distinfo
        pkgsrc/comms/asterisk22/patches:
            patch-build__tools_make__xml__documentation

Log Message:
Update to asterisk 22.10.0:

## Change Log for Release asterisk-22.10.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)

### Summary:

- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0

### User Notes:

- #### res ari: Add attachable states to Channels and Bridges
  Bridge variables now can be set and retrieved via the following paths:
  `/bridges/{bridgeId}/variable`
  `/bridges/{bridgeId}/variables`
  Both Bridge and Channel variables can now be set with an optional 'report_events'
  boolean flag that will cause those variables to be included on all events on that
  object. The 'report_events' flag will default to False if not set to maintain
  backwards capability.
  To allow this, variables can now be either name value pairs (the current format):
  `<variable_name>: '<value_string>'`
   - or -
  `<variable_name>: {value: '<value_string>', report_events: [true|false]}`

- #### ARI: Added paths to get and set multiple channel variables.
  Added new ARI paths for getting and setting multiple channel
  variables at a time. For GET, this takes in a single string of
  comma-separated variable names, while POST takes in a dictionary of key
  value pairs. The behavior is the same as passing in variables when
  originating a channel.

- #### res_rtp_asterisk: Add option to control stun host resolution when TTL = 0
  A new `stunaddr_reresolve_ttl_0` parameter has been added to rtp.conf
  that allows control over what happens when a STUN server hostname lookup
  returns a TTL of 0.  The values can be set as follows:
  - 'no': This is the historical (and current default) behavior of not doing
  any further lookups and continuing to use the last successful result until
  Asterisk is restarted or rtp.conf is reloaded.
  - 'yes': Use the last cached result for the current call but trigger
  re-resolution in the background for the benefit of future calls.
  If the result of the background lookup is a ttl > 0, periodic resolution
  will be restarted otherwise the next call will use the new cached value
  and will trigger a background lookup again.
  A new CLI command `rtp resolve stun hostname` has been added
- #### app_dial: Properly handle callee hangup while sending digits.
  If a called channel sends progress or wink and the caller begins
  sending digits but the callee answers and then hangs up before digit
  sending can finish, the call is now answered before being disconnected.
  If the callee hangs up without answering, the call now continues in
  the dialplan.

- #### Upgrade bundled pjproject to 2.17.
  Bundled pjproject has been upgraded to 2.17. For more
  information about what is included in this release, see the
  pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.17

- #### res_pjsip: Add per-endpoint RTP port range configuration
  PJSIP endpoints now support rtp_port_start and
  rtp_port_end options to configure a dedicated RTP port range per
  endpoint, overriding the global rtp.conf setting.

- #### stasis_broadcast: Add optional ARI broadcast with first-claim-wins
  New optional modules res_stasis_broadcast.so and
  app_stasis_broadcast.so enable broadcasting an incoming channel to multiple
  ARI applications. The first application to successfully claim (via
  POST /ari/events/claim) wins channel control. StasisBroadcast() dialplan
  application initiates broadcasts. CallBroadcast and CallClaimed events notify
  applications. When modules are not loaded, behavior is unchanged.

- #### chan_iax2: Add CHANNEL getter to retrieve auth method.
  CHANNEL(auth_method) can now be used to retrieve the
  auth method negotiated for a call on IAX2 channels.

- #### res_pjsip_maintenance: Add PJSIP endpoint maintenance mode
  New module res_pjsip_maintenance adds runtime maintenance
  mode for PJSIP endpoints. Use "pjsip set maintenance <on|off>
  <endpoint|all>" to enable or disable, and "pjsip show maintenance"
  to list affected endpoints. AMI actions PJSIPSetMaintenance and
  PJSIPShowMaintenance provide programmatic access. No configuration
  file changes required.

### Upgrade Notes:

- #### jansson: Upgrade version to jansson 2.15.0
  jansson has been upgraded to 2.15.0. For more
  information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.0

- #### res_pjsip: Add per-endpoint RTP port range configuration
  An alembic database migration has been added to add
  the rtp_port_start and rtp_port_end columns to the ps_endpoints
  table. Run "alembic upgrade head" to apply the schema change.

### Developer Notes:

- #### res_pjsip: Add per-endpoint RTP port range configuration
  New public API: ast_rtp_instance_new_with_port_range()
  creates an RTP instance with a per-instance port range.
  ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end()
  allow RTP engines to query the override. Third-party RTP engines can
  use these getters to support per-instance port ranges.

- #### stasis_broadcast: Add optional ARI broadcast with first-claim-wins
  New public APIs in stasis_app_broadcast.h:
  stasis_app_broadcast_channel(), stasis_app_claim_channel(),
  stasis_app_broadcast_winner(), and stasis_app_broadcast_wait(). New ARI event
  types (CallBroadcast, CallClaimed) added to events.json. All code is isolated;
  no existing ABI modified.

- #### res_pjsip_maintenance: Add PJSIP endpoint maintenance mode
  ast_sip_session_supplement gains a new optional
  callback - int (*session_create)(struct ast_sip_endpoint *endpoint,
  const char *destination). It is called from the global supplement
  list (not per-session) at the start of ast_sip_session_create_outgoing()
  via ast_sip_session_check_supplement_create(). Returning non-zero
  blocks the outgoing session. Modules that need to gate outbound
  SIP session creation should register a supplement with this callback
  set rather than hooking into chan_pjsip directly.

- #### build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build
  The pjsua and pjsystest application binaries, the deprecated
  Python pjsua bindings (`_pjsua.so`), and the `asterisk_malloc_debug.c` stub
  implementations are no longer built or installed as part of the bundled
  pjproject dev mode build. The `PYTHONDEV` (python2.7-dev) build dependency
  is also removed. Developers who relied on the pjsua binary for Test Suite
  SIP simulation should use SIPp instead, which is the current Asterisk Test
  Suite standard.
  Fixes: #1840

## Issue and Commit Detail:

### Closed Issues:

  - 1217: [bug]: INSERT INTO cdr query prepare statement issue on cdr_adaptive_odbc to control statement preparation manually
  - 1357: [bug]: MessageSend WARNING “not a valid SIP/SIPS URI” when using endpoint not URI
  - 1653: [bug]: Asterisk ODBC Voicemail Crash Caused by Voicemail Re-entry Loop and Unsafe BLOB Retrieval
  - 1736: app_queue: update_queue() may double-increment member->calls with shared_lastcall=yes (regression observed after 20.17; impacts fewestcalls routing)
  - 1761: func_talkdetect.c: TALK_DETECT docs wording mistake
  - 1762: [bug]: 100% CPU usage when entering BridgeWait after JITTERBUFFER(disabled)=
  - 1807: [new-feature]: translate.c: implement different types of sample frame inputs
  - 1812: [new-feature]: add tests/test_codec_translations.c
  - 1818: [bug]: func_odbc: possible use-after-free crash during reload with active calls
  - 1839: Crash in MDMF Caller ID parser due to signed char length field on DAHDI channels
  - 1840: [bug]: Asterisk fails to compile with --enable-dev-mode=yes due to INIT_RETURN undeclared in bundled pjproject Python bindings
  - 1855: [bug]: core reload deadlocks Asterisk (pjsip, CLI, etc.)
  - 1858: [bug]: DNS records with a TTL of zero are permanently cached
  - 1859: [bug]: res_pjsip_outbound_registration: No expires header set when triggered via CLI
  - 1861: [bug]: Possible heap corruption in audiohook/translate write path during bridged media
  - 1862: [bug]: Build fails with Building Documentation: line 210: /tmp/xmldoc.tmp.xml: Permission denied
  - 1865: [bug]: chan_iax2: Another code path that causes crashes on negative data lengths
  - 1867: [bug]: Massive [eventpoll] file-descriptor leak (hundreds of epoll fds) when TURN is enabled in rtp.conf
  - 1872: [bug]:  Deadlock in chan_pjsip_new when endpoint set_var invokes PJSIP_HEADER
  - 1878: [new-feature]: chan_iax2: Allow retrieving the auth method using the CHANNEL function
  - 1883: [bug]: fix: stdatomic.h false positive on GCC 4.8
  - 1885: [bug]: cdrel_custom :SQLite version too old: sqlite3_prepare_v3 / SQLITE_PREPARE_PERSISTENT undeclared
  - 1888: [improvement]: pjsip: Upgrade bundled version to pjproject 2.17
  - 1892: [bug]: Build failure with bundled pjproject on OpenSSL 1.0.x: undefined reference to TLS_method and SSL_CTX_set_ciphersuites
  - 1894: [bug]: Outbound ARI websockets don't always clean up completely
  - 1896: [bug]: asterisk.c fails to compile when HAVE_LIBEDIT_IS_UNICODE isn't defined
  - 1901: [bug]: QUEUE_RAISE_PENALTY=rN ignored when set via queue rules
  - 1903: [bug]: g++ 16 no longer defines __STDC_VERSION__ causing channelstorage_cpp_map_name_id.cc to fail
  - 1907: [bug]: Deadlock between bridge and setting of RTP stats variables at hangup
  - 1910: [improvement]: Add attachable state variables to Channels and Bridges.
  - 1915: [bug]: app_dial: Channel not handled properly if callee disconnects while caller is sending it digits prior to answer
  - 1921: [bug]: Memory error in crypto_get_cert_subject when using malloc_debug
  - 1928: [bug]: Calling ast_softhangup with channel lock held can cause deadlock
  - 1931: [improvement]: jansson: Upgrade version to jansson 2.15.0
  - 1936: [bug]: Calling set_variable on PJSIP channel when originating with ARI with PJSIP_HEADER can result in deadlock
  - 1938: [bug]: res_rtp_asterisk: Copy/paste error in ast_rtp_get_stat()
  - 1941: [bug]: chan_websocket doesn't handle CONTINUATION websocket frames
  - 1947: [bug]: chan_dahdi fails to build with gcc-16 when openr2 is installed
  - 1950: [bug]: app_record does not detect channel hangup during beep playback
  - 1952: [bug]: OpenSSL 4.0.0
  - 1957: [bug]: Calendar module fails to build with libical 4.X
  - 1970: [bug]: Startup or shutdown segfault in res_ari_model under certain conditions with DEVMODE and persistent outbound websockets.

### Commit List:

-  res_ari: Add res_ari_model as an optional_module.
-  res ari: Add attachable states to Channels and Bridges
-  ARI: Added paths to get and set multiple channel variables.
-  res_stir_shaken: avoid direct ASN1_STRING accesses
-  tcptls.c: fix build with OpenSSL 4
-  res_calendar: Fix build with libical 4.X
-  app_record: Fix hangup handling during beep playback
-  odbc: Don't use prepared statements for distinct SQL statements
-  abstract_jb.c: Remove timerfd from channel when disabling jitter buffer
-  res_pjsip: Don't allow a leading period when wildcard matching
-  Ensure channel locks aren't held while calling ast_set_variables.
-  app_queue: fix double increment of member->calls with shared_lastcall
-  chan_dahdi: Fix set but not used in mfcr2_show_links_of().
-  tests: add tests/test_codec_translations.c
-  install_prereq: Add a 'minimal' mode for basic build dependencies
-  chan_websocket: Handle incoming CONTINUATION frames.
-  res_rtp_asterisk: Fix incorrect reference in ast_rtp_get_stat().
-  jansson: Upgrade version to jansson 2.15.0
-  channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup.
-  res_rtp_asterisk: Add option to control stun host resolution when TTL = 0
-  pjsip_configuration: Show actual dtls_verify config.
-  app_dial: Properly handle callee hangup while sending digits.
-  res_pjsip_messaging: Update To URI only if it is a SIP(S) URI
-  Upgrade bundled pjproject to 2.17.
-  res_stir_shaken: fix memory free crash when Asterisk is built with malloc_debug
-  manager: Eliminate unnecessary code, simplify sessions in stasis callbacks
-  res_stasis/resource_bridges: Split bridge playback control and wrapper cleanup
-  res_pjsip_outbound_publish.c: Add more verbose documentation for outbound_proxy usage
-  channel.c: Don't lock the channel in ast_softhangup while setting rtp instance vars
-  chan_pjsip: Fix deadlock when endpoint set_var uses PJSIP_HEADER
-  res_pjsip: Add per-endpoint RTP port range configuration
-  app_queue: Fix raise_respect_min lost in copy_rules() breaking rN queue rules
-  app_voicemail_odbc: fix msgnum race and crash on failed STORE
-  ari_websockets: Fix two issues in the cleanup of outbound websockets.
-  compat.h: Ensure check for `__STDC_VERSION__` is not attempted for c++.
-  pjproject: Backport fix for OpenSSL < 1.1.0 build failure in ssl_sock_ossl.c
-  asterisk.c: Fix #if HAVE_LIBEDIT_IS_UNICODE.
-  cdrel_custom: fix SQLite compatibility for versions < 3.20.0
-  translate.c: implement different sample_types for translation computation.
-  stasis_broadcast: Add optional ARI broadcast with first-claim-wins
-  res_audiosocket: Tolerate non-audio frame types
-  pbx_functions: Save module pointer before calling read and write callbacks.
-  chan_iax2: Add CHANNEL getter to retrieve auth method.
-  fix: backport pjproject stdatomic.h GCC 4.8 build failure patch
-  res_rtp_asterisk: Destroy ioqueue in rtp_ioqueue_thread_destroy.
-  res_pjsip_maintenance: Add PJSIP endpoint maintenance mode
-  chan_iax2: Add another check to abort frame handling if datalen < 0.
-  res_pjsip_outbound_registration: only update the Expires header if the value has changed
-  func_talkdetect.c: Clarify dsp_talking_threshold documentation.
-  make_xml_documentation: Remove temporary file on script exit.
-  res_pjsip_config_wizard: Trigger reloads from a pjsip servant thread
-  build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build
-  callerid: fix signed char causing crash in MDMF parser


To generate a diff of this commit:
cvs rdiff -u -r1.22 -r1.23 pkgsrc/comms/asterisk22/Makefile
cvs rdiff -u -r1.8 -r1.9 pkgsrc/comms/asterisk22/PLIST
cvs rdiff -u -r1.9 -r1.10 pkgsrc/comms/asterisk22/distinfo
cvs rdiff -u -r1.3 -r1.4 \
    pkgsrc/comms/asterisk22/patches/patch-build__tools_make__xml__documentation

Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.

Modified files:

Index: pkgsrc/comms/asterisk22/Makefile
diff -u pkgsrc/comms/asterisk22/Makefile:1.22 pkgsrc/comms/asterisk22/Makefile:1.23
--- pkgsrc/comms/asterisk22/Makefile:1.22       Thu May 14 16:40:32 2026
+++ pkgsrc/comms/asterisk22/Makefile    Mon Jun 22 02:21:18 2026
@@ -1,12 +1,11 @@
-# $NetBSD: Makefile,v 1.22 2026/05/14 16:40:32 ryoon Exp $
+# $NetBSD: Makefile,v 1.23 2026/06/22 02:21:18 jnemeth Exp $
 #
 # NOTE: when updating this package, there are two places that sound
 #       tarballs need to be checked; look in ${WRKSRC}/sounds/Makefile
 #       to find out the current sound file versions
 #       Also look in ${WRKSRC}/third-party/versions.mak for pjproject
 
-DISTNAME=      asterisk-22.9.0
-PKGREVISION=   1
+DISTNAME=      asterisk-22.10.0
 CATEGORIES=    comms net audio
 MASTER_SITES=  https://downloads.asterisk.org/pub/telephony/asterisk/
 MASTER_SITES+= https://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
@@ -141,7 +140,7 @@ CONFIGURE_ARGS+=    --without-timerfd
 DISTFILES+=    asterisk-extra-sounds-en-gsm-1.5.2.tar.gz
 
 # pjproject
-PJPROJ_VERSION=        2.16
+PJPROJ_VERSION=        2.17
 SITES.pjproject-${PJPROJ_VERSION}.tar.bz2= \
        -https://raw.githubusercontent.com/asterisk/third-party/master/pjproject/${PJPROJ_VERSION}/pjproject-${PJPROJ_VERSION}.tar.bz2
 SITES.pjproject-${PJPROJ_VERSION}.md5= \
@@ -269,6 +268,7 @@ post-install:
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.8.1.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.8.2.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.9.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
+       ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.10.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.3.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.4.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.5.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
@@ -280,6 +280,7 @@ post-install:
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.8.1.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.8.2.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.9.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
+       ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.10.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/LICENSE ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/README-SERIOUSLY.bestpractices.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/README.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}

Index: pkgsrc/comms/asterisk22/PLIST
diff -u pkgsrc/comms/asterisk22/PLIST:1.8 pkgsrc/comms/asterisk22/PLIST:1.9
--- pkgsrc/comms/asterisk22/PLIST:1.8   Mon Apr 13 02:50:22 2026
+++ pkgsrc/comms/asterisk22/PLIST       Mon Jun 22 02:21:18 2026
@@ -1,4 +1,4 @@
-@comment $NetBSD: PLIST,v 1.8 2026/04/13 02:50:22 jnemeth Exp $
+@comment $NetBSD: PLIST,v 1.9 2026/06/22 02:21:18 jnemeth Exp $
 lib/asterisk/libasteriskpj.so
 lib/asterisk/libasteriskpj.so.2
 lib/asterisk/modules/app_adsiprog.so
@@ -62,6 +62,7 @@ lib/asterisk/modules/app_softhangup.so
 lib/asterisk/modules/app_speech_utils.so
 lib/asterisk/modules/app_stack.so
 lib/asterisk/modules/app_stasis.so
+lib/asterisk/modules/app_stasis_broadcast.so
 lib/asterisk/modules/app_stream_echo.so
 lib/asterisk/modules/app_system.so
 lib/asterisk/modules/app_talkdetect.so
@@ -269,6 +270,7 @@ lib/asterisk/modules/res_pjsip_exten_sta
 lib/asterisk/modules/res_pjsip_header_funcs.so
 lib/asterisk/modules/res_pjsip_history.so
 lib/asterisk/modules/res_pjsip_logger.so
+lib/asterisk/modules/res_pjsip_maintenance.so
 lib/asterisk/modules/res_pjsip_messaging.so
 lib/asterisk/modules/res_pjsip_mwi.so
 lib/asterisk/modules/res_pjsip_mwi_body_generator.so
@@ -313,6 +315,7 @@ lib/asterisk/modules/res_speech_aeap.so
 ${PLIST.srtp}lib/asterisk/modules/res_srtp.so
 lib/asterisk/modules/res_stasis.so
 lib/asterisk/modules/res_stasis_answer.so
+lib/asterisk/modules/res_stasis_broadcast.so
 lib/asterisk/modules/res_stasis_device_state.so
 lib/asterisk/modules/res_stasis_playback.so
 lib/asterisk/modules/res_stasis_recording.so
@@ -2320,6 +2323,8 @@ share/doc/asterisk/CREDITS
 share/doc/asterisk/ChangeLog-22.0.0.md
 share/doc/asterisk/ChangeLog-22.1.0.md
 share/doc/asterisk/ChangeLog-22.1.1.md
+share/doc/asterisk/ChangeLog-22.10.0.html
+share/doc/asterisk/ChangeLog-22.10.0.md
 share/doc/asterisk/ChangeLog-22.2.0.md
 share/doc/asterisk/ChangeLog-22.3.0.html
 share/doc/asterisk/ChangeLog-22.3.0.md

Index: pkgsrc/comms/asterisk22/distinfo
diff -u pkgsrc/comms/asterisk22/distinfo:1.9 pkgsrc/comms/asterisk22/distinfo:1.10
--- pkgsrc/comms/asterisk22/distinfo:1.9        Mon Apr 13 02:50:22 2026
+++ pkgsrc/comms/asterisk22/distinfo    Mon Jun 22 02:21:18 2026
@@ -1,17 +1,17 @@
-$NetBSD: distinfo,v 1.9 2026/04/13 02:50:22 jnemeth Exp $
+$NetBSD: distinfo,v 1.10 2026/06/22 02:21:18 jnemeth Exp $
 
-BLAKE2s (asterisk-22.9.0/asterisk-22.9.0.tar.gz) = 063cd232804bd0c27a3aaa3c637ff23dc03659ef584fe54d612a902643ed69a6
-SHA512 (asterisk-22.9.0/asterisk-22.9.0.tar.gz) = a8160b7044beedbcd1f0374a9e49aab7ed2b13d3475605f3dfa1deaed98fd959fc7c6e211b7da66ec7a15e6d7e092d7a8050f4635f934bd81d6cbb7518771681
-Size (asterisk-22.9.0/asterisk-22.9.0.tar.gz) = 26585391 bytes
-BLAKE2s (asterisk-22.9.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
-SHA512 (asterisk-22.9.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
-Size (asterisk-22.9.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
-BLAKE2s (asterisk-22.9.0/pjproject-2.16.md5) = 1db3bea58adb58231de14971b28b7de2691da688e928b86eb75aa1828442de7c
-SHA512 (asterisk-22.9.0/pjproject-2.16.md5) = eb099d845b774dcad86a53694a488055c7bf5acb172fdbe904682a780ab46ef9369f9c9dcd462af477359aa1d72e37256be4c92be612ca7b62f03d6d0bd727c7
-Size (asterisk-22.9.0/pjproject-2.16.md5) = 166 bytes
-BLAKE2s (asterisk-22.9.0/pjproject-2.16.tar.bz2) = 7f24150c9b62be590aea7077ad932e0ea6e34ab755fa25a31a5b095b978ed0e8
-SHA512 (asterisk-22.9.0/pjproject-2.16.tar.bz2) = a4c693f95e28d3f917adb7f27519adcd0286b39bd79e8bd39a3f84a550f99f5ad7951081f94a83a6e6fdb3acf7bfe38692b3682852a5921fe52ad134e997e820
-Size (asterisk-22.9.0/pjproject-2.16.tar.bz2) = 8783258 bytes
+BLAKE2s (asterisk-22.10.0/asterisk-22.10.0.tar.gz) = d572a794d2aadfa480b2a1efb0b3bca1b4bb5db5fe81f901412ad53eb1acfed2
+SHA512 (asterisk-22.10.0/asterisk-22.10.0.tar.gz) = c05ff7b6c58fce5627fa71306f0cd1976c65828d590992331d34946f5437c6019fdcbc2621445d4a10f6dbdadec371d1d0fb41cc0a0250b94de92e0f8f668f91
+Size (asterisk-22.10.0/asterisk-22.10.0.tar.gz) = 26662183 bytes
+BLAKE2s (asterisk-22.10.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
+SHA512 (asterisk-22.10.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
+Size (asterisk-22.10.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
+BLAKE2s (asterisk-22.10.0/pjproject-2.17.md5) = 40b4cd26a5101f81b6d57f5d8b8a74fe4b296292b21677e02a1bcc73d0d3a3c3
+SHA512 (asterisk-22.10.0/pjproject-2.17.md5) = 8946eebc9d7d472e001d03602a8a6b12632734dbc665acb5aca60c0f83ebbd4d69d31c06096fde44907ce2040be5fd4145ff7fcaaa2f681f90600faa0ed5f294
+Size (asterisk-22.10.0/pjproject-2.17.md5) = 166 bytes
+BLAKE2s (asterisk-22.10.0/pjproject-2.17.tar.bz2) = 3909b745c89a119f41126ee49ea196d5bf6889fdcbeb22d266dabc3fd339dd96
+SHA512 (asterisk-22.10.0/pjproject-2.17.tar.bz2) = b523b3761aab8931ad2a940d107861caf346e3c5cf9b12d23b5b267b8f3b4758ee842b401f1199c3ceb7e78bbaf0d2a167d2574222c51e6016586eb2caa1e33d
+Size (asterisk-22.10.0/pjproject-2.17.tar.bz2) = 8975382 bytes
 SHA1 (patch-Makefile) = 5cf3b6937ec23a82e4d056b91e493a36bc1089b9
 SHA1 (patch-addons_chan__ooh323.c) = 1775da7ca2129a962ed460bd1e78ba3ce6afa62c
 SHA1 (patch-apps_app__adsiprog.c) = 031139e5cd1ef6bb2afb0a74fee3d752eded0a2c
@@ -23,7 +23,7 @@ SHA1 (patch-apps_app__minivm.c) = 22ee6e
 SHA1 (patch-apps_app__queue.c) = fdf7cf202b60e24cd9227f7e461bbd541565d602
 SHA1 (patch-apps_app__sms.c) = ad65b3cb2a30489551101f7534c691cd1155d18f
 SHA1 (patch-apps_app__voicemail.c) = 5276457466fde27494bf43fd6d306397bc4ff97f
-SHA1 (patch-build__tools_make__xml__documentation) = 9e74959dc143cac9a4163a76806e6ad79a777b1f
+SHA1 (patch-build__tools_make__xml__documentation) = ead74d338fe2f8254db40458f6e70ca5e5d715e6
 SHA1 (patch-build__tools_mkpkgconfig) = 7fab8fcf46d9f8a3b98455674fec6307ec472b23
 SHA1 (patch-cdr_cdr__pgsql.c) = 82b002a1f5ed3b7361a98e2bffb5cea8833949b8
 SHA1 (patch-cel_cel__pgsql.c) = b280efab2b035ce60be268bac9bc8824910b2b8f

Index: pkgsrc/comms/asterisk22/patches/patch-build__tools_make__xml__documentation
diff -u pkgsrc/comms/asterisk22/patches/patch-build__tools_make__xml__documentation:1.3 pkgsrc/comms/asterisk22/patches/patch-build__tools_make__xml__documentation:1.4
--- pkgsrc/comms/asterisk22/patches/patch-build__tools_make__xml__documentation:1.3     Mon Apr 13 02:50:22 2026
+++ pkgsrc/comms/asterisk22/patches/patch-build__tools_make__xml__documentation Mon Jun 22 02:21:18 2026
@@ -1,13 +1,13 @@
-$NetBSD: patch-build__tools_make__xml__documentation,v 1.3 2026/04/13 02:50:22 jnemeth Exp $
+$NetBSD: patch-build__tools_make__xml__documentation,v 1.4 2026/06/22 02:21:18 jnemeth Exp $
 
---- build_tools/make_xml_documentation.orig    2026-04-09 16:24:26.000000000 +0000
+--- build_tools/make_xml_documentation.orig    2026-06-11 14:27:27.000000000 +0000
 +++ build_tools/make_xml_documentation
-@@ -230,7 +230,7 @@ for subdir in ${mod_subdirs} ; do
+@@ -238,7 +238,7 @@ for subdir in ${mod_subdirs} ; do
                                        ${XMLSTARLET} val -e -d "${source_tree}/doc/appdocsxml.dtd" "${i}" || { echo "" ; exit 1 ; }
                        fi
                fi
--              ${SED} -r "/^\s*(<[?]xml|<.DOCTYPE|<.?docs)/d" "${i}" > /tmp/xmldoc.tmp.xml
-+              ${SED} -r "/^[[:space:]]*(<[?]xml|<.DOCTYPE|<.?docs)/d" "${i}" > /tmp/xmldoc.tmp.xml
+-              ${SED} -r "/^\s*(<[?]xml|<.DOCTYPE|<.?docs)/d" "${i}" > "${tmp_output_file}"
++              ${SED} -r "/^[[:space:]]*(<[?]xml|<.DOCTYPE|<.?docs)/d" "${i}" > "${tmp_output_file}"
                dirname=${i%/*}
                if [ "${dirname}" != "${subdir_path}" ] ; then
                        # If we're in a subdirectory like channels/pjsip, we need to check channels/Makefile

Prev by Date: CVS commit: pkgsrc/doc
Next by Date: CVS commit: pkgsrc/doc
Previous by Thread: CVS commit: pkgsrc/doc
Next by Thread: CVS commit: pkgsrc/doc
Indexes:

Home | Main Index | Thread Index | Old Index