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, 27 Oct 2025 04:35:13 +0000

Module Name:    pkgsrc
Committed By:   jnemeth
Date:           Mon Oct 27 04:35:13 UTC 2025

Modified Files:
        pkgsrc/comms/asterisk22: Makefile PLIST distinfo

Log Message:
Upgrade to Asterisk 22.6.0.

## Change Log for Release asterisk-22.6.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)

### Summary:

- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0

### User Notes:

- #### app_queue.c: Add new global 'log_unpause_on_reason_change'
  Add new global option 'log_unpause_on_reason_change' that
  is default disabled. When enabled cause addition of UNPAUSE event on
  every re-PAUSE with reason changed.

- #### pbx_builtins: Allow custom tone for WaitExten.
  The tone used while waiting for digits in WaitExten
  can now be overridden by specifying an argument for the 'd'
  option.

- #### res_tonedetect: Add option for TONE_DETECT detection to auto stop.
  The 'e' option for TONE_DETECT now allows detection to
  be disabled automatically once the desired number of matches have
  been fulfilled, which can help prevent race conditions in the
  dialplan, since TONE_DETECT does not need to be disabled after
  a hit.

- #### sorcery: Prevent duplicate objects and ensure missing objects are created on u..
  Users relying on Sorcery multiple writable backends configurations
  (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes
  in sorcery.conf to ensure missing objects are recreated after temporary backend
  failures. Default behavior remains unchanged unless explicitly enabled.

- #### chan_websocket: Allow additional URI parameters to be added to the outgoing URI.
  A new WebSocket channel driver option `v` has been added to the
  Dial application that allows you to specify additional URI parameters on
  outgoing connections. Run `core show application Dial` from the Asterisk CLI
  to see how to use it.

- #### app_chanspy: Add option to not automatically answer channel.
  ChanSpy and ExtenSpy can now be configured to not
  automatically answer the channel by using the 'N' option.

- #### cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
  Enabling the tracking of the
  STREAM_BEGIN and the STREAM_END event
  types in cel.conf will log media files and
  music on hold played to each channel.
  The STREAM_BEGIN event's extra field will
  contain a JSON with the file details (path,
  format and language), or the class name, in
  case of music on hold is played. The DTMF
  event's extra field will contain a JSON with
  the digit and the duration in milliseconds.

- #### res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
  Options are now available in the menuselect "Resource Modules"
  category that allow you to enable the AES_192, AES_256 and AES_GCM
  cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support
  them but modern versions do.  Previously, the only way to enable them was
  to set the CFLAGS environment variable when running ./configure.
  The default setting is to disable them preserving existing behavior.

- #### cdr: add CANCEL dispostion in CDR
  A new CDR option "canceldispositionenabled" has been added
  that when set to true, the NO ANSWER disposition will be split into
  two dispositions: CANCEL and NO ANSWER. The default value is 'no'

- #### func_curl: Allow auth methods to be set.
  The httpauth field in CURLOPT now allows the authentication
  methods to be set.

- #### Media over Websocket Channel Driver
  A new channel driver "chan_websocket" is now available. It can
  exchange media over both inbound and outbound websockets and will both frame
  and re-time the media it receives.
  See http://s.asterisk.net/mow for more information.
  The ARI channels/externalMedia API now includes support for the

### Developer Notes:

- #### ARI: Add command to indicate progress to a channel
  A new ARI endpoint is available at `/channels/{channelId}/progress` to indicate progress to a channel.

- #### options:  Change ast_options from ast_flags to ast_flags64.
  The 32-bit ast_options has no room left to accomodate new
  options and so has been converted to an ast_flags64 structure. All internal
  references to ast_options have been updated to use the 64-bit flag
  manipulation macros.  External module references to the 32-bit ast_options
  should continue to work on little-endian systems because the
  least-significant bytes of a 64 bit integer will be in the same location as a
  32-bit integer.  Because that's not the case on big-endian systems, we've
  swapped the bytes in the flags manupulation macros on big-endian systems
  so external modules should still work however you are encouraged to test.

## Issue and Commit Detail:

### Closed Issues:

  - 401: [bug]: app_dial: Answer Gosub option passthrough regression
  - 927: [bug]: no audio when media source changed during the call
  - 1176: [bug]: ast_slinear_saturated_multiply_float produces potentially audible distortion artifacts
  - 1259: [bug]: New TenantID feature doesn't seem to set CDR for incoming calls
  - 1260: [bug]: Asterisk sends RTP audio stream before ICE/DTLS completes
  - 1269: [bug]: MixMonitor with D option produces corrupt file
  - 1273: [bug]: When executed with GotoIf, the action Redirect does not take effect and causes confusion in dialplan execution.
  - 1280: [improvement]: logging playback of audio per channel
  - 1289: [bug]: sorcery - duplicate objects from multiple backends and backend divergence on update
  - 1301: [bug]: sig_analog: fgccamamf doesn't handle STP, STP2, or STP3
  - 1304: [bug]: FLUSH_MEDIA does not reset frame_queue_length in WebSocket channel
  - 1305: [bug]: Realtime incorrectly falls back to next backend on record-not-found (SQL_NO_DATA), causing incorrect behavior and delay
  - 1307: [improvement]: ast_tls_cert: Allow certificate validity to be configurable
  - 1309: [bug]: Crash with C++ alternative storage backend enabled
  - 1315:  [bug]: When executed with dialplan, the action Redirect does not take effect.
  - 1317: [bug]: AGI command buffer overflow with long variables
  - 1321: [improvement]: app_agent_pool: Remove obsolete documentation
  - 1323: [new-feature]: add CANCEL dispostion in CDR
  - 1327: [bug]: res_stasis_device_state: can't delete ARI Devicestate after asterisk restart
  - 1332: [new-feature]: func_curl: Allow auth methods to be set
  - 1349: [bug]: Race condition on redirect can cause missing Diversion header
  - 1352: [improvement]: Websocket channel with custom URI
  - 1353: [bug]: AST_DATA_DIR/sounds/custom directory not searched
  - 1358: [new-feature]: app_chanspy: Add option to not automatically answer channel
  - 1364: [bug]: bridge.c: BRIDGE_NOANSWER not always obeyed
  - 1366: [improvement]: func_frame_drop: Handle allocation failure properly
  - 1369: [bug]: test_res_prometheus: Compilation failure in devmode due to curlopts not using long type
  - 1371: [improvement]: func_frame_drop: Add debug messages for frames that can be dropped
  - 1375: [improvement]: dsp.c: Improve logging in tone_detect().
  - 1378: [bug]: chan_dahdi: dialmode feature is not properly reset between calls
  - 1380: [bug]: sig_analog: Segfault due to calling strcmp on NULL
  - 1384: [bug]: chan_websocket: asterisk crashes on hangup after STOP_MEDIA_BUFFERING command with id
  - 1386: [bug]: enabling announceposition_only_up prevents any queue position announcements
  - 1390: [improvement]: res_tonedetect: Add option to automatically end detection in TONE_DETECT
  - 1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled
  - 1396: [new-feature]: pbx_builtins: Make tone option for WaitExten configurable
  - 1401: [bug]: app_waitfornoise timeout is always less then configured because of time() usage
  - 1457: [bug]: segmentation fault because of a wrong ari config
  - 1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.
  - 1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk change to stop media flow before DTLS completes

### Commit List:

-  res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
-  chan_websocket: Fix codec validation and add passthrough option.
-  res_ari: Ensure outbound websocket config has a websocket_client_id.
-  chan_websocket.c: Add DTMF messages
-  app_queue.c: Add new global 'log_unpause_on_reason_change'
-  app_waitforsilence.c: Use milliseconds to calculate timeout time
-  Fix missing ast_test_flag64 in extconf.c
-  pbx_builtins: Allow custom tone for WaitExten.
-  res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-  app_queue: fix comparison for announce-position-only-up
-  sig_analog: Skip Caller ID spill if usecallerid=no.
-  chan_dahdi: Fix erroneously persistent dialmode.
-  chan_websocket: Fix buffer overrun when processing TEXT websocket frames.
-  sig_analog: Fix SEGV due to calling strcmp on NULL.
-  ARI: Add command to indicate progress to a channel
-  dsp.c: Improve debug logging in tone_detect().
-  res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.
-  app_chanspy: Add option to not automatically answer channel.
-  xmldoc.c: Fix rendering of CLI output.
-  func_frame_drop: Add debug messages for dropped frames.
-  test_res_prometheus: Fix compilation failure on Debian 13.
-  func_frame_drop: Handle allocation failure properly.
-  pbx_lua.c: segfault when pass null data to term_color function
-  bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel.
-  res_rtp_asterisk: Don't send RTP before DTLS has negotiated.
-  app_dial.c: Moved channel lock to prevent deadlock
-  file.c: with "sounds_search_custom_dir = yes", search "custom" directory
-  cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-  channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.
-  res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-  cdr: add CANCEL dispostion in CDR
-  func_curl: Allow auth methods to be set.
-  options:  Change ast_options from ast_flags to ast_flags64.
-  res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA)
-  app_agent_pool: Remove documentation for removed option.
-  res_agi: Increase AGI command buffer size from 2K to 8K
-  ast_tls_cert: Make certificate validity configurable.
-  cdr.c: Set tenantid from party_a->base instead of chan->base.
-  app_mixmonitor:  Update the documentation concerning the "D" option.
-  sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf.
-  chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA
-  chan_pjsip.c: Change SSRC after media source change
-  Media over Websocket Channel Driver
-  bundled_pjproject: Avoid deadlock between transport and transaction
-  utils.h: Add rounding to float conversion to int.
-  res_musiconhold.c: Ensure we're always locked around music state access.
-  res_musiconhold.c: Annotate when the channel is locked.
-  res_musiconhold: Appropriately lock channel during start.

## Change Log for Release asterisk-22.5.2

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j): A specifically malformed Authorization header in an incoming SIP request can cause Asterisk to 
crash

### Commit Authors:

- George Joseph: (1)

## Issue and Commit Detail:

### Closed Issues:

  - !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an incoming SIP request can cause Asterisk to crash

### Commit Details:

#### res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
  Author: George Joseph
  Date:   2025-08-28

  In the highly-unlikely event that get_authorization_hdr() couldn't find an
  Authorization header in a request, trying to get the digest algorithm
  would cauase a SEGV.  We now check that we have an auth header that matches
  the realm before trying to get the algorithm from it.

  Resolves: #GHSA-64qc-9x89-rx5j


To generate a diff of this commit:
cvs rdiff -u -r1.14 -r1.15 pkgsrc/comms/asterisk22/Makefile
cvs rdiff -u -r1.4 -r1.5 pkgsrc/comms/asterisk22/PLIST
cvs rdiff -u -r1.5 -r1.6 pkgsrc/comms/asterisk22/distinfo

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:1.14 pkgsrc/comms/asterisk22/Makefile:1.15
--- pkgsrc/comms/asterisk22/Makefile:1.14       Sun Oct  5 19:25:44 2025
+++ pkgsrc/comms/asterisk22/Makefile    Mon Oct 27 04:35:13 2025
@@ -1,12 +1,11 @@
-# $NetBSD: Makefile,v 1.14 2025/10/05 19:25:44 js Exp $
+# $NetBSD: Makefile,v 1.15 2025/10/27 04:35:13 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-22.5.1
-PKGREVISION=   2
+DISTNAME=      asterisk-22.6.0
 CATEGORIES=    comms net audio
 MASTER_SITES=  https://downloads.asterisk.org/pub/telephony/asterisk/
 MASTER_SITES+= https://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
@@ -260,14 +259,16 @@ post-install:
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.2.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.3.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.4.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
-       ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.4.1.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.5.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.5.1.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
+       ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.5.2.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
+       ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.6.0.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-22.4.1.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.5.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
        ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.5.1.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
+       ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.5.2.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
+       ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-22.6.0.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:1.4 pkgsrc/comms/asterisk22/PLIST:1.5
--- pkgsrc/comms/asterisk22/PLIST:1.4   Mon Aug 11 06:28:13 2025
+++ pkgsrc/comms/asterisk22/PLIST       Mon Oct 27 04:35:13 2025
@@ -1,4 +1,4 @@
-@comment $NetBSD: PLIST,v 1.4 2025/08/11 06:28:13 jnemeth Exp $
+@comment $NetBSD: PLIST,v 1.5 2025/10/27 04:35:13 jnemeth Exp $
 lib/asterisk/libasteriskpj.so
 lib/asterisk/libasteriskpj.so.2
 lib/asterisk/modules/app_adsiprog.so
@@ -101,6 +101,7 @@ ${PLIST.jabber}lib/asterisk/modules/chan
 lib/asterisk/modules/chan_pjsip.so
 lib/asterisk/modules/chan_rtp.so
 lib/asterisk/modules/chan_unistim.so
+lib/asterisk/modules/chan_websocket.so
 lib/asterisk/modules/codec_a_mu.so
 lib/asterisk/modules/codec_adpcm.so
 lib/asterisk/modules/codec_alaw.so
@@ -2323,12 +2324,14 @@ share/doc/asterisk/ChangeLog-22.3.0.html
 share/doc/asterisk/ChangeLog-22.3.0.md
 share/doc/asterisk/ChangeLog-22.4.0.html
 share/doc/asterisk/ChangeLog-22.4.0.md
-share/doc/asterisk/ChangeLog-22.4.1.html
-share/doc/asterisk/ChangeLog-22.4.1.md
 share/doc/asterisk/ChangeLog-22.5.0.html
 share/doc/asterisk/ChangeLog-22.5.0.md
 share/doc/asterisk/ChangeLog-22.5.1.html
 share/doc/asterisk/ChangeLog-22.5.1.md
+share/doc/asterisk/ChangeLog-22.5.2.html
+share/doc/asterisk/ChangeLog-22.5.2.md
+share/doc/asterisk/ChangeLog-22.6.0.html
+share/doc/asterisk/ChangeLog-22.6.0.md
 share/doc/asterisk/IAX2-security.pdf
 share/doc/asterisk/IAX2-security.txt
 share/doc/asterisk/LICENSE

Index: pkgsrc/comms/asterisk22/distinfo
diff -u pkgsrc/comms/asterisk22/distinfo:1.5 pkgsrc/comms/asterisk22/distinfo:1.6
--- pkgsrc/comms/asterisk22/distinfo:1.5        Mon Aug 11 06:28:13 2025
+++ pkgsrc/comms/asterisk22/distinfo    Mon Oct 27 04:35:13 2025
@@ -1,17 +1,17 @@
-$NetBSD: distinfo,v 1.5 2025/08/11 06:28:13 jnemeth Exp $
+$NetBSD: distinfo,v 1.6 2025/10/27 04:35:13 jnemeth Exp $
 
-BLAKE2s (asterisk-22.5.1/asterisk-22.5.1.tar.gz) = eb3deef5729713e6435cca17a0cadffb7ca2b57e37cfc30886260d90e1e710cd
-SHA512 (asterisk-22.5.1/asterisk-22.5.1.tar.gz) = 9ff03285f88a943e488c7a3bbb30d3200ac486b331213c771ecd8762dbe055c0befc8e6f23d087ff82b2f6c72b918e102d80a2549c1dda996a9c7fdba4473be9
-Size (asterisk-22.5.1/asterisk-22.5.1.tar.gz) = 26373168 bytes
-BLAKE2s (asterisk-22.5.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
-SHA512 (asterisk-22.5.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
-Size (asterisk-22.5.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
-BLAKE2s (asterisk-22.5.1/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
-SHA512 (asterisk-22.5.1/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
-Size (asterisk-22.5.1/pjproject-2.15.1.md5) = 172 bytes
-BLAKE2s (asterisk-22.5.1/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
-SHA512 (asterisk-22.5.1/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
-Size (asterisk-22.5.1/pjproject-2.15.1.tar.bz2) = 8492214 bytes
+BLAKE2s (asterisk-22.6.0/asterisk-22.6.0.tar.gz) = 673f441d10480852b54681d893e6a67453a236df27718cf41cd54273594eaf90
+SHA512 (asterisk-22.6.0/asterisk-22.6.0.tar.gz) = 17496da40a55b8ca54b3d578fcbc0eb032bcb01d3fcc71b6ace87341efc0adb54bdec961dbc25626d110768f72b26b2181f43aadcef1778f9b90e12a011987eb
+Size (asterisk-22.6.0/asterisk-22.6.0.tar.gz) = 26426677 bytes
+BLAKE2s (asterisk-22.6.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
+SHA512 (asterisk-22.6.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
+Size (asterisk-22.6.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
+BLAKE2s (asterisk-22.6.0/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
+SHA512 (asterisk-22.6.0/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
+Size (asterisk-22.6.0/pjproject-2.15.1.md5) = 172 bytes
+BLAKE2s (asterisk-22.6.0/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
+SHA512 (asterisk-22.6.0/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
+Size (asterisk-22.6.0/pjproject-2.15.1.tar.bz2) = 8492214 bytes
 SHA1 (patch-Makefile) = 5cf3b6937ec23a82e4d056b91e493a36bc1089b9
 SHA1 (patch-addons_chan__ooh323.c) = 1775da7ca2129a962ed460bd1e78ba3ce6afa62c
 SHA1 (patch-apps_app__adsiprog.c) = 031139e5cd1ef6bb2afb0a74fee3d752eded0a2c

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