CVS commit: pkgsrc/comms/asterisk20

To: pkgsrc-changes%NetBSD.org@localhost
Subject: CVS commit: pkgsrc/comms/asterisk20
From: "Ryo ONODERA" <ryoon%netbsd.org@localhost>
Date: Sat, 26 Oct 2024 13:12:23 +0000

Module Name:    pkgsrc
Committed By:   ryoon
Date:           Sat Oct 26 13:12:23 UTC 2024

Modified Files:
        pkgsrc/comms/asterisk20: Makefile PLIST distinfo
        pkgsrc/comms/asterisk20/patches: patch-configure patch-main_utils.c
Added Files:
        pkgsrc/comms/asterisk20/patches: patch-main_config.c

Log Message:
comms/asterisk20: Update to 20.10.0

Changelog:
20.10.0:
Summary:

    Commits: 24
    Commit Authors: 9
    Issues Resolved: 18
    Security Advisories Resolved: 0

User Notes:

    res_pjsip_notify: add dialplan application

    A new dialplan application PJSIPNotify is now available
    which can send SIP NOTIFY requests from the dialplan.
    The pjsip send notify CLI command has also been enhanced to allow
    sending NOTIFY messages to a specific channel. Syntax:
    pjsip send notify channel
    channel: Add multi-tenant identifier.

    tenantid has been added to channels. It can be read in
    dialplan via CHANNEL(tenantid), and it can be set using
    Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
    use the new tenantid option for pjsip endpoints (e.g., tenantid=My
    tenant ID) so that it will show up in Newchannel events. You can set it
    like any other channel variable using set_var in pjsip.conf as well, but
    note that this will NOT show up in Newchannel events. Tenant ID is also
    available in CDR and can be accessed with CDR(tenantid). The peer tenant
    ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
    as well if it has been set.
    res_pjsip_config_wizard.c: Refactor load process

    The res_pjsip_config_wizard.so module can now be reloaded.

20.9.3:
Summary:

    Commits: 1
    Commit Authors: 1
    Issues Resolved: 0
    Security Advisories Resolved: 1
        GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming SIP request can cause Asterisk to crash when res_resolver_unbound is used

20.9.2:
Summary:

    Commits: 1
    Commit Authors: 1
    Issues Resolved: 0
    Security Advisories Resolved: 1
        GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code execution / System() dialplan

20.9.1:
Summary:

    Commits: 2
    Commit Authors: 1
    Issues Resolved: 2
    Security Advisories Resolved: 0

20.9.0:
Summary:

    Commits: 20
    Commit Authors: 9
    Issues Resolved: 8
    Security Advisories Resolved: 0

User Notes:

    app_voicemail_odbc: Allow audio to be kept on disk

    This commit adds a new voicemail.conf option
    'odbc_audio_on_disk' which when set causes the ODBC variant of
    app_voicemail_odbc to leave the message and greeting audio files
    on disk and only store the message metadata in the database.
    Much more information can be found in the voicemail.conf.sample
    file.
    app_queue: Add option to not log Restricted Caller ID to queue_log

    Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
    will be stored in the queue log.
    If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.
    pbx.c: expand fields width of "core show hints"

    The fields width of "core show hints" were increased.
    The width of "extension" field to 30 characters and
    the width of the "device state id" field to 60 characters.
    rtp_engine: add support for multirate RFC2833 digits

    No change in configuration is required in order to enable this
    feature. Endpoints configured to use RFC2833 will automatically have this
    enabled. If the endpoint does not support this, it should not include it in
    the SDP offer/response.
    Resolves: #699

20.8.1:
Summary:

    Commits: 1
    Commit Authors: 1
    Issues Resolved: 0
    Security Advisories Resolved: 1
        GHSA-qqxj-v78h-hrf9: res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests

20.8.0:
Summary:

    Commits: 44
    Commit Authors: 15
    Issues Resolved: 26
    Security Advisories Resolved: 0

User Notes:

    res_pjsip_logger: Preserve logging state on reloads.

    Issuing "pjsip reload" will no longer disable
    logging if it was previously enabled from the CLI.
    loader.c: Allow dependent modules to be unloaded recursively.

    In certain circumstances, modules with dependency relations
    can have their dependents automatically recursively unloaded and loaded
    again using the "module refresh" CLI command or the ModuleLoad AMI command.
    tcptls/iostream: Add support for setting SNI on client TLS connections

    Secure websocket client connections now send SNI in
    the TLS client hello.
    res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.

    set identify_by=transport for the pjsip endpoint. Then
    use the existing 'match' option and the new 'transport' option of
    the identify.
    Fixes: #672
    res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI

    this new feature let users match endpoints based on the
    indound SIP requests' URI. To do so, add 'request_uri' to the
    endpoint's 'identify_by' option. The 'match_request_uri' option of
    the identify can be an exact match for the entire request uri, or a
    regular expression (between slashes). It's quite similar to the
    header identifer.
    Fixes: #599
    res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA

    the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
    manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action

    When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call 
initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. 
Exemple of an context to perform the previus indication:
    [addautoanswer]
    exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
    exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
    exten => _s,n,Return()
    manager.c: Add CLI command to kick AMI sessions.

    The "manager kick session" CLI command now
    allows kicking a specified AMI session.
    chan_dahdi: Allow specifying waitfordialtone per call.

    "waitfordialtone" may now be specified for DAHDI
    trunk channels on a per-call basis using the CHANNEL function.
    Upgrade bundled pjproject to 2.14.1

    Bundled pjproject has been upgraded to 2.14.1. For more
    information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1

20.7.0:
Summary:

    res_pjsip_stir_shaken.c: Add checks for missing parameters
    app_dial: Add dial time for progress/ringing.
    app_voicemail: Properly reinitialize config after unit tests.
    app_queue.c : fix "queue add member" usage string
    app_voicemail: Allow preventing mark messages as urgent.
    res_pjsip: Use consistent type for boolean columns.
    attestation_config.c: Use ast_free instead of ast_std_free
    Makefile: Add stir_shaken/cache to directories created on install
    Stir/Shaken Refactor
    alembic: Synchronize alembic heads between supported branches.
    translate.c: implement new direct comp table mode
    README.md: Removed outdated link
    strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
    res_rtp_asterisk.c: Correct coefficient in MOS calculation.
    dsp.c: Fix and improve potentially inaccurate log message.
    pjsip show channelstats: Prevent possible segfault when faxing
    Reduce startup/shutdown verbose logging
    configure: Rerun bootstrap on modern platform.
    Upgrade bundled pjproject to 2.14.
    app_speech_utils.c: Allow partial speech results.
    utils: Make behavior of ast_strsep* match strsep.
    app_chanspy: Add 'D' option for dual-channel audio
    app_if: Fix next priority calculation.
    res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
    BuildSystem: Bump autotools versions on OpenBSD.
    main/utils: Simplify the FreeBSD ast_get_tid() handling
    res_pjsip_session.c: Correctly format SDP connection addresses.
    rtp_engine.c: Correct sample rate typo for L16/44100.
    manager.c: Fix erroneous reloads in UpdateConfig.
    res_calendar_icalendar: Print iCalendar error on parsing failure.
    app_confbridge: Don't emit warnings on valid configurations.
    app_voicemail: add NoOp alembic script to maintain sync
    chan_dahdi: Allow MWI to be manually toggled on channels.
    chan_rtp.c: MulticastRTP missing refcount without codec option
    chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
    func_frame_trace: Add CLI command to dump frame queue.

User Notes:

    app_dial: Add dial time for progress/ringing.

    The timeout argument to Dial now allows
    specifying the maximum amount of time to dial if
    early media is not received.
    app_voicemail: Allow preventing mark messages as urgent.

    The leaveurgent mailbox option can now be used to
    control whether callers may leave messages marked as 'Urgent'.
    Stir/Shaken Refactor

    Asterisk's stir-shaken feature has been refactored to
    correct interoperability, RFC compliance, and performance issues.
    See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
    information.
    Upgrade bundled pjproject to 2.14.

    Bundled pjproject has been upgraded to 2.14. For more
    information on what all is included in this change, check out the
    pjproject Github page: https://github.com/pjsip/pjproject/releases
    app_speech_utils.c: Allow partial speech results.

    The SpeechBackground dialplan application now supports a 'p'
    option that will return partial results from speech engines that
    provide them when a timeout occurs.
    app_chanspy: Add 'D' option for dual-channel audio

    The ChanSpy application now accepts the 'D' option which
    will interleave the spied audio within the outgoing frames. The
    purpose of this is to allow the audio to be read as a Dual channel
    stream with separate incoming and outgoing audio. Setting both the
    'o' option and the 'D' option and results in the 'D' option being
    ignored.
    chan_dahdi: Allow MWI to be manually toggled on channels.

    The 'dahdi set mwi' now allows MWI on channels
    to be manually toggled if needed for troubleshooting.
    Resolves: #440

20.6.0:
Summary:

    logger: Fix linking regression.
    Revert "core & res_pjsip: Improve topology change handling."
    menuselect: Use more specific error message.
    res_pjsip_nat: Fix potential use of uninitialized transport details
    app_if: Fix faulty EndIf branching.
    manager.c: Fix regression due to using wrong free function.
    config_options.c: Fix truncation of option descriptions.
    manager.c: Improve clarity of "manager show connected".
    make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
    general: Fix broken links.
    MergeApproved.yml: Remove unneeded concurrency
    app_dial: Add option "j" to preserve initial stream topology of caller
    ast_coredumper: Increase reliability
    logger.c: Move LOG_GROUP documentation to dedicated XML file.
    res_odbc.c: Allow concurrent access to request odbc connections
    res_pjsip_header_funcs.c: Check URI parameter length before copying.
    config.c: Log #exec include failures.
    make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
    app_voicemail.c: Completely resequence mailbox folders.
    sig_analog: Fix channel leak when mwimonitor is enabled.
    res_rtp_asterisk.c: Update for OpenSSL 3+.
    alembic: Update list of TLS methods available on ps_transports.
    func_channel: Expose previously unsettable options.
    app.c: Allow ampersands in playback lists to be escaped.
    uri.c: Simplify ast_uri_make_host_with_port()
    func_curl.c: Remove CURLOPT() plaintext documentation.
    res_http_websocket.c: Set hostname on client for certificate validation.
    live_ast: Add astcachedir to generated asterisk.conf.
    SECURITY.md: Update with correct documentation URL
    func_lock: Add missing see-also refs to documentation.
    app_followme.c: Grab reference on nativeformats before using it
    configs: Improve documentation for bandwidth in iax.conf.
    logger: Add channel-based filtering.
    chan_iax2.c: Don't send unsanitized data to the logger.
    codec_ilbc: Disable system ilbc if version >= 3.0.0
    resource_channels.c: Explicit codec request when creating UnicastRTP.
    doc: Update IP Quality of Service links.
    chan_pjsip: Add PJSIPHangup dialplan app and manager action
    chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
    chan_dahdi: Warn if nonexistent cadence is requested.
    stasis: Update the snapshot after setting the redirect
    ari: Provide the caller ID RDNIS for the channels
    main/utils: Implement ast_get_tid() for OpenBSD
    res_rtp_asterisk.c: Fix runtime issue with LibreSSL
    app_directory: Add ADSI support to Directory.
    core_local: Fix local channel parsing with slashes.
    Remove files that are no longer updated
    app_voicemail: Add AMI event for mailbox PIN changes.
    app_queue.c: Emit unpause reason with PauseQueueMember event.
    bridge_simple: Suppress unchanged topology change requests
    res_pjsip: Include cipher limit in config error message.
    res_speech: allow speech to translate input channel
    res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
    res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
    api.wiki.mustache: Fix indentation in generated markdown
    pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
    configs: Fix typo in pjsip.conf.sample.
    res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
    res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
    .github: PRSubmitActions: Fix adding reviewers to PR
    .github: New PR Submit workflows
    .github: New PR Submit workflows
    res_stasis: signal when new command is queued
    ari/stasis: Indicate progress before playback on a bridge
    func_curl.c: Ensure channel is locked when manipulating datastores.
    .github: Fix job prereqs in PROpenedUpdated
    .github: Block PR tests until approved
    Update config.yml
    logger.h: Add ability to change the prefix on SCOPE_TRACE output
    Add libjwt to third-party
    res_pjsip: update qualify_timeout documentation with DNS note
    chan_dahdi: Clarify scope of callgroup/pickupgroup.
    func_json: Fix crashes for some types
    res_speech_aeap: add aeap error handling
    app_voicemail: Disable ADSI if unavailable.
    codec_builtin: Use multiples of 20 for maximum_ms
    lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
    asterisk.c: Use the euid's home directory to read/write cli history
    res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
    cel: add publish user event helper
    chan_console: Fix deadlock caused by unclean thread exit.
    file.c: Add ability to search custom dir for sounds
    chan_iax2: Improve authentication debugging.
    res_rtp_asterisk: fix wrong counter management in ioqueue objects
    make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
    func_periodic_hook: Add hangup step to avoid timeout
    res_stasis_recording.c: Save recording state when unmuted.
    res_speech_aeap: check for null format on response
    func_periodic_hook: Don't truncate channel name
    safe_asterisk: Change directory permissions to 755
    chan_rtp: Implement RTP glue for UnicastRTP channels
    app_queue: periodic announcement configurable start time.
    variables: Add additional variable dialplan functio
    app_dial: Add option "j" to preserve initial stream topology of caller

    The option "j" is now available for the Dial application which
    uses the initial stream topology of the caller to create the outgoing
    channels.
    logger: Add channel action allows you
    to hang up an unanswered incoming PJSIP call with a specific SIP
    response code in the 400 -> 699 range.
    app_voicemail: Add AMI event for mailbox PIN changes.

    The VoicemailPasswordChange event is
    now emitted whenevernow supports translation of an input channel
    to a format supported by the speech provider, provided a translation
    path is available between the source format and provider capabilites.
    res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

    With this update, the PJSIP realm lengths have been extended
    to support up to 255 characters.
    res_stasis: signal when new command is queued

    Call setup times should be significantly improved
    when using ARI.
    lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

    You no longer need to select DEBUG_THREADS to use
    DETECT_DEADLOCKS. This removes a significant amount of overhead
    if you just want to detect possible deadlocks vs needing full
    lock tracing.
    file.c: Add ability to search custom dir for sounds

    A new option "sounds_search_custom_dir" has been added to
    asterisk.conf that allows asterisk to search
    AST_DATA_DIR/sounds/custom for sounds files before searching the
    standard AST_DATA_DIR/sounds/ directory.
    make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

    The "Build Options" entry in the "core show settings"
    CLI command has been renamed to "ABI related Build Options" and
    a new entry named "All Build Options" has been added that shows
    both breaking and non-breaking options.
    chan_rtp: Implement RTP glue for UnicastRTP channels

    The dial string option 'g' was added to the UnicastRTP channel
    which enables RTP glue and therefore native RTP bridges with those
    channels.
    app_queue: periodic announcement configurable start time.

    Introduce a new queue configuration option called
    'periodic-announce-startdelay' which will vary the normal (historic)
    behavior of starting the periodic announcement cycle at
    periodic-announce-frequency seconds after entering the queue to start
    the periodic announcement cycle at period-announce-startdelay seconds
    after joining the queue. The default behavior if this config option is
    not set remains unchanged.
    Signed-off-by: Jaco Kroon jaco%uls.co.za@localhost
    variables: Add additional variable dialplan functions.

    Four new dialplan functions have been added.
    GLOBAL_DELETE and DELETE have been added which allows
    the deletion of global and channel variables.
    GLOBAL_EXISTS and VARIABLE_EXISTS have been added
    which checks whether a global or channel variable has
    been set.

20.5.2:
Summary:

    res_rtp_asterisk: Fix regression issues with DTLS client check


To generate a diff of this commit:
cvs rdiff -u -r1.9 -r1.10 pkgsrc/comms/asterisk20/Makefile
cvs rdiff -u -r1.4 -r1.5 pkgsrc/comms/asterisk20/PLIST
cvs rdiff -u -r1.3 -r1.4 pkgsrc/comms/asterisk20/distinfo
cvs rdiff -u -r1.1 -r1.2 pkgsrc/comms/asterisk20/patches/patch-configure \
    pkgsrc/comms/asterisk20/patches/patch-main_utils.c
cvs rdiff -u -r0 -r1.1 pkgsrc/comms/asterisk20/patches/patch-main_config.c

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

Modified files:

Index: pkgsrc/comms/asterisk20/Makefile
diff -u pkgsrc/comms/asterisk20/Makefile:1.9 pkgsrc/comms/asterisk20/Makefile:1.10
--- pkgsrc/comms/asterisk20/Makefile:1.9        Sun Oct 20 14:03:30 2024
+++ pkgsrc/comms/asterisk20/Makefile    Sat Oct 26 13:12:23 2024
@@ -1,11 +1,10 @@
-# $NetBSD: Makefile,v 1.9 2024/10/20 14:03:30 wiz Exp $
+# $NetBSD: Makefile,v 1.10 2024/10/26 13:12:23 ryoon 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
 
-DISTNAME=      asterisk-20.5.1
-PKGREVISION=   2
+DISTNAME=      asterisk-20.10.0
 CATEGORIES=    comms net audio
 MASTER_SITES=  https://downloads.asterisk.org/pub/telephony/asterisk/
 MASTER_SITES+= https://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
@@ -16,7 +15,7 @@ DISTFILES=    ${DEFAULT_DISTFILES}
 COMMENT=       The Asterisk Software PBX
 LICENSE=       gnu-gpl-v2
 
-EXTRACT_ONLY=  ${DISTNAME}.tar.gz
+EXTRACT_ONLY=  ${DISTNAME}${EXTRACT_SUFX}
 MAINTAINER=    ryoon%NetBSD.org@localhost
 HOMEPAGE=      https://www.asterisk.org/
 
@@ -163,7 +162,7 @@ CONFIGURE_ARGS+=    --without-unbound
 DISTFILES+=    asterisk-extra-sounds-en-gsm-1.5.2.tar.gz
 
 # pjproject
-PJPROJ_VERSION=        2.13.1
+PJPROJ_VERSION=        2.14.1
 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= \
@@ -181,7 +180,6 @@ SUBST_FILES.configs+=       configs/samples/ia
 SUBST_FILES.configs+=  configs/samples/musiconhold.conf.sample
 SUBST_FILES.configs+=  configs/samples/osp.conf.sample
 SUBST_FILES.configs+=  configs/samples/phoneprov.conf.sample
-SUBST_FILES.configs+=  configs/samples/sla.conf.sample
 SUBST_SED.configs+=    -e 's|doc/|${PREFIX}/share/doc/${PKGBASE}/|'
 SUBST_SED.configs+=    -e 's|/etc/asterisk|${ASTETCDIR}|'
 SUBST_SED.configs+=    -e 's|/var/lib/asterisk|${ASTVARLIBDIR}|'

Index: pkgsrc/comms/asterisk20/PLIST
diff -u pkgsrc/comms/asterisk20/PLIST:1.4 pkgsrc/comms/asterisk20/PLIST:1.5
--- pkgsrc/comms/asterisk20/PLIST:1.4   Mon Apr 29 14:27:55 2024
+++ pkgsrc/comms/asterisk20/PLIST       Sat Oct 26 13:12:23 2024
@@ -1,4 +1,4 @@
-@comment $NetBSD: PLIST,v 1.4 2024/04/29 14:27:55 nia Exp $
+@comment $NetBSD: PLIST,v 1.5 2024/10/26 13:12:23 ryoon Exp $
 lib/asterisk/libasteriskpj.so
 lib/asterisk/libasteriskpj.so.2
 lib/asterisk/modules/app_adsiprog.so
@@ -290,7 +290,6 @@ lib/asterisk/modules/res_pjsip_sdp_rtp.s
 lib/asterisk/modules/res_pjsip_send_to_voicemail.so
 lib/asterisk/modules/res_pjsip_session.so
 lib/asterisk/modules/res_pjsip_sips_contact.so
-lib/asterisk/modules/res_pjsip_stir_shaken.so
 lib/asterisk/modules/res_pjsip_t38.so
 lib/asterisk/modules/res_pjsip_transport_websocket.so
 lib/asterisk/modules/res_pjsip_xpidf_body_generator.so
@@ -318,7 +317,6 @@ lib/asterisk/modules/res_stasis_playback
 lib/asterisk/modules/res_stasis_recording.so
 lib/asterisk/modules/res_stasis_snoop.so
 lib/asterisk/modules/res_statsd.so
-lib/asterisk/modules/res_stir_shaken.so
 lib/asterisk/modules/res_stun_monitor.so
 ${PLIST.kqueue}lib/asterisk/modules/res_timing_kqueue.so
 lib/asterisk/modules/res_timing_pthread.so
@@ -2438,7 +2436,3 @@ share/examples/asterisk/voicemail.conf
 share/examples/asterisk/xmpp.conf
 ${PLIST.webvmail}share/httpd/htdocs/_asterisk/animlogo.gif
 ${PLIST.webvmail}share/httpd/htdocs/_asterisk/play.gif
-@pkgdir libdata/asterisk/third-party/pjproject
-@pkgdir libdata/asterisk/keys/stir_shaken
-@pkgdir libdata/asterisk/firmware/iax
-@pkgdir libdata/asterisk/documentation/thirdparty

Index: pkgsrc/comms/asterisk20/distinfo
diff -u pkgsrc/comms/asterisk20/distinfo:1.3 pkgsrc/comms/asterisk20/distinfo:1.4
--- pkgsrc/comms/asterisk20/distinfo:1.3        Thu May 16 13:41:59 2024
+++ pkgsrc/comms/asterisk20/distinfo    Sat Oct 26 13:12:23 2024
@@ -1,21 +1,21 @@
-$NetBSD: distinfo,v 1.3 2024/05/16 13:41:59 gdt Exp $
+$NetBSD: distinfo,v 1.4 2024/10/26 13:12:23 ryoon Exp $
 
-BLAKE2s (asterisk-20.5.1/asterisk-20.5.1.tar.gz) = 2f4d9e5a95f62e57ae9e87956479e1d67e6d5545e2b1588b14f123e75b9c40e8
-SHA512 (asterisk-20.5.1/asterisk-20.5.1.tar.gz) = 80e4697a56e4115134cfc1d8cc22c20212e64b9ae7eba2bf722b704d7bbee2d7629208ab9c6276c9e965a11ea7566a34e288cf6661b21b769177794bb6f16375
-Size (asterisk-20.5.1/asterisk-20.5.1.tar.gz) = 28214779 bytes
-BLAKE2s (asterisk-20.5.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
-SHA512 (asterisk-20.5.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
-Size (asterisk-20.5.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
-BLAKE2s (asterisk-20.5.1/pjproject-2.13.1.md5) = 933be89ea03bc24a3a965d37a8985a2af3ea404e24a8fdd296a2be07d5390de0
-SHA512 (asterisk-20.5.1/pjproject-2.13.1.md5) = 5a8c35e79d10760f74d02332f35aad3517fc3c72e62e2b1c35fcb1d613bbad4d96fa07f1cea89514dba2d237a36a698e94b1242be938d492d20b76a130c0d0f1
-Size (asterisk-20.5.1/pjproject-2.13.1.md5) = 172 bytes
-BLAKE2s (asterisk-20.5.1/pjproject-2.13.1.tar.bz2) = 7c04ae740c66f92502684de834b4055e7f4842bcb35f0efc12f183c8d8c37f7f
-SHA512 (asterisk-20.5.1/pjproject-2.13.1.tar.bz2) = 2f1eb3cb8f52a5536c355b02b1abfa24c7d2263b5338568a7cd8167d349e007c34afafcdd7ed510bfc4fc859494383fac05e8325e46fd66219dbead86c5c3562
-Size (asterisk-20.5.1/pjproject-2.13.1.tar.bz2) = 7825201 bytes
+BLAKE2s (asterisk-20.10.0/asterisk-20.10.0.tar.gz) = ac656e3bfa530bd98a32d689b02da15ab514ef7f5f4c765596b61c64f669de22
+SHA512 (asterisk-20.10.0/asterisk-20.10.0.tar.gz) = b05a65423f6a203f3fca712fc8ee2c008deabfdaf05f1cd56b8bd0ffc1d1805c4dea58e4267d224f47777fbc1bcdc8aabe71cbe1e86e6e98fa597f08ac087cce
+Size (asterisk-20.10.0/asterisk-20.10.0.tar.gz) = 28309321 bytes
+BLAKE2s (asterisk-20.10.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
+SHA512 (asterisk-20.10.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
+Size (asterisk-20.10.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
+BLAKE2s (asterisk-20.10.0/pjproject-2.14.1.md5) = f384e59ad4f8227cd7131a5c07b68a83b75b319fa60c38d6f9d27af817a0f516
+SHA512 (asterisk-20.10.0/pjproject-2.14.1.md5) = 25ce388adcd7eaa2c21d95a58d9fc5e33a6cb96dd99c292574b8f11f6f1f985cf91f91ea252300bd1be192e396ac6c8a35a87b219864339798bf6195a7650c00
+Size (asterisk-20.10.0/pjproject-2.14.1.md5) = 172 bytes
+BLAKE2s (asterisk-20.10.0/pjproject-2.14.1.tar.bz2) = 4b22d553ddafc2d53d866b4936d465c161e2a095a6a75bd4b93be26e4803122c
+SHA512 (asterisk-20.10.0/pjproject-2.14.1.tar.bz2) = 996116df4a18fb28c8f68d122466f8664958226a38e432b6190b92fbf277b278d370a4b44fabeaf25691e3cdcde28a8879b2738ead5387d119229db01ce121d8
+Size (asterisk-20.10.0/pjproject-2.14.1.tar.bz2) = 8379251 bytes
 SHA1 (patch-Makefile) = 7d278c52ace3e972ee0e17b0aec137549d4b8d7c
 SHA1 (patch-build__tools_make__xml__documentation) = 9ee091af60ea480a88ce036dad2be7b257142ae0
 SHA1 (patch-build__tools_mkpkgconfig) = 7fab8fcf46d9f8a3b98455674fec6307ec472b23
-SHA1 (patch-configure) = 262036a7bcdf6e8e8431212a99dbcd65a7be8329
+SHA1 (patch-configure) = 8bfcc9774eeb107fb95a4d6695d5bdb62aa3cf8f
 SHA1 (patch-configure.ac) = 98690ecc0e1cc99b6d7a0cd13412bd1b2b05397a
 SHA1 (patch-contrib_scripts_vmail.cgi) = 7935ce96ea319eb19cc2ce999813eb837d5357c0
 SHA1 (patch-include_asterisk_autoconfig.h.in) = 2f8a986bd3b35ebb384280f0dbf81c32d689ddd2
@@ -23,10 +23,11 @@ SHA1 (patch-include_asterisk_sha1.h) = c
 SHA1 (patch-main_Makefile) = d8b2cdd76dbf4dfa3ae082ce39cc0f8396b368aa
 SHA1 (patch-main_acl.c) = 06a9d247b19d648e9ff54ac2a234dc8ac8c023bb
 SHA1 (patch-main_bridge__channel.c) = 2120e3d623843113066413a5980cef6ece81bd82
+SHA1 (patch-main_config.c) = 525c483af0c0bd826cc913522b5bf8254cae6aac
 SHA1 (patch-main_pbx__builtins.c) = 7c1d518f05afc1523b247f50b0363e25832c8636
 SHA1 (patch-main_stdtime_localtime.c) = d530bea8f93667a07b707e6fb01c6744aa264d40
 SHA1 (patch-main_taskprocessor.c) = f90805bd78fd4096beb9ee1cf9c794c50b87481a
-SHA1 (patch-main_utils.c) = ab85be3687dd7f39b742bd5e4036f9e297f3e272
+SHA1 (patch-main_utils.c) = 430be968b3a6f899cc9d541c378059fe0dd7f91e
 SHA1 (patch-pbx_pbx__dundi.c) = d2a50650a19463304c81fc19c460565b94f91b72
 SHA1 (patch-res_res__rtp__asterisk.c) = a5952d7e25538cdf9786684a05fb05b8f9e1872c
 SHA1 (patch-res_res__xmpp.c) = 8e8951b8a57eac168e9bf4429392ddfe808ef590

Index: pkgsrc/comms/asterisk20/patches/patch-configure
diff -u pkgsrc/comms/asterisk20/patches/patch-configure:1.1 pkgsrc/comms/asterisk20/patches/patch-configure:1.2
--- pkgsrc/comms/asterisk20/patches/patch-configure:1.1 Sun Dec 10 06:40:20 2023
+++ pkgsrc/comms/asterisk20/patches/patch-configure     Sat Oct 26 13:12:23 2024
@@ -1,38 +1,38 @@
-$NetBSD: patch-configure,v 1.1 2023/12/10 06:40:20 ryoon Exp $
+$NetBSD: patch-configure,v 1.2 2024/10/26 13:12:23 ryoon Exp $
 
---- configure.orig     2023-10-18 17:10:25.000000000 +0000
+--- configure.orig     2024-10-17 16:00:41.000000000 +0000
 +++ configure
-@@ -9229,7 +9229,7 @@ $as_echo_n "checking for clang -fblocks.
+@@ -10138,7 +10138,7 @@ else $as_nop
+ 
+                       { printf "%s\n" "$as_me:${as_lineno-$LINENO}: checking for clang -fblocks" >&5
+ printf %s "checking for clang -fblocks... " >&6; }
+-                      if test "`echo 'int main(){return ^{return 42;}();}' | ${CC} -o /dev/null -fblocks -x c - 2>&1`" = ""; then
++                      if test "`echo 'int main(){return ^{return 42;}();}' | ${CC} ${LDFLAGS} -o /dev/null -fblocks -x c - 2>&1`" = ""; then
+                               AST_CLANG_BLOCKS_LIBS=""
                                AST_CLANG_BLOCKS="-Wno-unknown-warning-option -fblocks"
-                               { $as_echo "$as_me:${as_lineno-$LINENO}: result: yes" >&5
- $as_echo "yes" >&6; }
--                      elif test "`echo 'int main(){return ^{return 42;}();}' | ${CC} -o /dev/null -fblocks -x c -lBlocksRuntime - 2>&1`" = ""; then
-+                      elif test "`echo 'int main(){return ^{return 42;}();}' | ${CC} ${LDFLAGS} -o /dev/null -fblocks -x c -lBlocksRuntime - 2>&1`" = ""; then
-                               AST_CLANG_BLOCKS_LIBS="-lBlocksRuntime"
-                               AST_CLANG_BLOCKS="-fblocks"
-                               { $as_echo "$as_me:${as_lineno-$LINENO}: result: yes" >&5
-@@ -15322,7 +15322,7 @@ fi
- done
- 
- 
--ac_fn_c_check_header_mongrel "$LINENO" "sys/poll.h" "ac_cv_header_sys_poll_h" "$ac_includes_default"
-+ac_fn_c_check_header_mongrel "$LINENO" "poll.h" "ac_cv_header_sys_poll_h" "$ac_includes_default"
- if test "x$ac_cv_header_sys_poll_h" = xyes; then :
+                               { printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: yes" >&5
+@@ -16641,7 +16641,7 @@ then :
+ fi
+ 
+ 
+-ac_fn_c_check_header_compile "$LINENO" "sys/poll.h" "ac_cv_header_sys_poll_h" "$ac_includes_default"
++ac_fn_c_check_header_compile "$LINENO" "poll.h" "ac_cv_header_sys_poll_h" "$ac_includes_default"
+ if test "x$ac_cv_header_sys_poll_h" = xyes
+ then :
  
- else
-@@ -17539,7 +17539,7 @@ rm -f core conftest.err conftest.$ac_obj
+@@ -19537,7 +19537,7 @@ rm -f core conftest.err conftest.$ac_obj
  LDFLAGS=${old_LDFLAGS}
  rm -f conftest.dynamics
  
--ac_fn_c_check_header_mongrel "$LINENO" "sys/poll.h" "ac_cv_header_sys_poll_h" "$ac_includes_default"
-+ac_fn_c_check_header_mongrel "$LINENO" "poll.h" "ac_cv_header_sys_poll_h" "$ac_includes_default"
- if test "x$ac_cv_header_sys_poll_h" = xyes; then :
+-ac_fn_c_check_header_compile "$LINENO" "sys/poll.h" "ac_cv_header_sys_poll_h" "$ac_includes_default"
++ac_fn_c_check_header_compile "$LINENO" "poll.h" "ac_cv_header_sys_poll_h" "$ac_includes_default"
+ if test "x$ac_cv_header_sys_poll_h" = xyes
+ then :
    HAS_POLL=1
- 
-@@ -18199,6 +18199,18 @@ if test "${ac_cv_pthread_once_needsbrace
- $as_echo "#define PTHREAD_ONCE_INIT_NEEDS_BRACES 1" >>confdefs.h
+@@ -20294,6 +20294,19 @@ printf "%s\n" "#define PTHREAD_ONCE_INIT
  
  fi
+ 
 +ac_fn_c_check_func "$LINENO" "strftime_l" "ac_cv_func_strftime_l"
 +if test "x$ac_cv_func_strftime_l" = xyes
 +then :
@@ -45,6 +45,7 @@ $NetBSD: patch-configure,v 1.1 2023/12/1
 +  printf "%s\n" "#define HAVE_STRPTIME_L 1" >>confdefs.h
 +
 +fi
- 
++
  
      if test "x${PBX_PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP}" != "x1"; then
+       { printf "%s\n" "$as_me:${as_lineno-$LINENO}: checking for PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP in pthread.h" >&5
Index: pkgsrc/comms/asterisk20/patches/patch-main_utils.c
diff -u pkgsrc/comms/asterisk20/patches/patch-main_utils.c:1.1 pkgsrc/comms/asterisk20/patches/patch-main_utils.c:1.2
--- pkgsrc/comms/asterisk20/patches/patch-main_utils.c:1.1      Sun Dec 10 06:40:20 2023
+++ pkgsrc/comms/asterisk20/patches/patch-main_utils.c  Sat Oct 26 13:12:23 2024
@@ -1,21 +1,8 @@
-$NetBSD: patch-main_utils.c,v 1.1 2023/12/10 06:40:20 ryoon Exp $
+$NetBSD: patch-main_utils.c,v 1.2 2024/10/26 13:12:23 ryoon Exp $
 
---- main/utils.c.orig  2019-12-23 16:43:56.000000000 +0000
+--- main/utils.c.orig  2024-10-17 16:00:41.000000000 +0000
 +++ main/utils.c
-@@ -39,6 +39,12 @@
- #elif defined(HAVE_SYS_THR_H)
- #include <sys/thr.h>
- #endif
-+#if defined(HAVE_GETIFADDRS)
-+#include <ifaddrs.h>
-+#if !defined(__linux__)
-+#include <net/if_dl.h>
-+#endif
-+#endif
- 
- #include "asterisk/network.h"
- #include "asterisk/ast_version.h"
-@@ -612,7 +618,7 @@ const char *ast_inet_ntoa(struct in_addr
+@@ -937,7 +937,7 @@ const char *ast_inet_ntoa(struct in_addr
  
  static int dev_urandom_fd = -1;
  

Added files:

Index: pkgsrc/comms/asterisk20/patches/patch-main_config.c
diff -u /dev/null pkgsrc/comms/asterisk20/patches/patch-main_config.c:1.1
--- /dev/null   Sat Oct 26 13:12:23 2024
+++ pkgsrc/comms/asterisk20/patches/patch-main_config.c Sat Oct 26 13:12:23 2024
@@ -0,0 +1,12 @@
+$NetBSD: patch-main_config.c,v 1.1 2024/10/26 13:12:23 ryoon Exp $
+
+--- main/config.c.orig 2024-10-26 12:10:13.346969822 +0000
++++ main/config.c
+@@ -44,6 +44,7 @@
+ #include <libgen.h>
+ #include <time.h>
+ #include <sys/stat.h>
++#include <sys/wait.h>
+ 
+ #include <math.h>     /* HUGE_VAL */
+ #include <regex.h>

Prev by Date: CVS commit: pkgsrc/math/R-VGAMdata
Next by Date: CVS commit: pkgsrc/doc
Previous by Thread: CVS commit: pkgsrc/math/R-VGAMdata
Next by Thread: CVS commit: pkgsrc/doc
Indexes:

Home | Main Index | Thread Index | Old Index