Re: CVS import: pkgsrc/comms/asterisk18

[Thomas Klausner <wiz%NetBSD.org@localhost>]

Hi!

We have stopped using 'cvs import' for adding packages.

The easiest way is to use the import-package tool from pkgsrc/pkgtools/import-package.

Cheers,
 Thomas

On Sun, Jun 13, 2021 at 07:47:18AM +0000, John Nemeth wrote:
> Module Name:  pkgsrc
> Committed By: jnemeth
> Date:         Sun Jun 13 07:47:18 UTC 2021
> 
> Update of /cvsroot/pkgsrc/comms/asterisk18
> In directory ivanova.netbsd.org:/tmp/cvs-serv2627
> 
> Log Message:
> Import Asterisk 18.x as comms/asterisk18.
> 
> This is a long term support version.  It is scheduled to go to
> security fixes only on October 20th, 2024, and EOL on October 20th,
> 2025.
> 
> ------------------------------------------------------------------------------
> --- Functionality changes from Asterisk 18.3.0 to Asterisk 18.4.0 ------------
> ------------------------------------------------------------------------------
> 
> logger
> ------------------
>  * The dateformat option in logger.conf will now control the remote
>    console (asterisk -r -T) timestamp format.  Previously, dateformat
>    only controlled the formatting of the timestamp going to log
>    files and the main console (asterisk -c) but only for non-verbose
>    messages.
> 
>    Internally, Asterisk does not send the logging timestamp with
>    verbose messages to console clients. It's up to the Asterisk
>    remote consoles to format verbose messages.  Asterisk remote
>    consoles previously did not load dateformat from logger.conf.
> 
>    Previously there was a non-configurable and hard-coded "%b %e
>    %T" dateformat that would be used no matter what on all verbose
>    console messages printed on remote consoles.
> 
>    Example:
>    logger.conf
>     dateformat=%F %T.%3q
> 
>    # asterisk -rvvv -T
>    [2021-03-19 09:54:19.760-0400]  Loading res_stasis_answer.so.
>    [Mar 19 09:55:43]     -- Goto (dialExten,s,1)
> 
>    Given the following example configuration in logger.conf, Asterisk
>    log files and the console, will log verbose messages using the
>    given timestamp.  Now ensuring that all remote console messages
>    are logged with the same dateformat as other log streams.
> 
>    ---
>    [general]
>    dateformat=%F %T.%3q
> 
>    [logfiles]
>    console  => notice,warning,error,verbose
>    full     => notice,warning,error,debug,verbose
>    ---
> 
>    Now we have a globally-defined dateformat that will be used
>    consistently across the Asterisk main console, remote consoles,
>    and log files.
> 
>    Now we have consistent logging:
> 
>    # asterisk -rvvv -T
>    [2021-03-19 09:54:19.760-0400]  Loading res_stasis_answer.so.
>    [2021-03-19 09:55:43.920-0400]     -- Goto (dialExten,s,1)
> 
> res_pjsip
> ------------------
>  * PJSIP transports can now be partially reloaded safely. This
>    allows the local_net and external_* options to be updated without
>    restarting Asterisk.
> 
>  * PJSIP endpoints can now be configured to skip authentication
>    when handling OPTIONS requests by setting the
>    allow_unauthenticated_options configuration property to 'yes.'
> 
> ------------------------------------------------------------------------------
> --- Functionality changes from Asterisk 18.2.2 to Asterisk 18.3.0 ------------
> ------------------------------------------------------------------------------
> 
> app_mixmonitor
> ------------------
>  * app_mixmonitor now sends manager events MixMonitorStart,
>    MixMonitorStop and MixMonitorMute when the channel monitoring
>    is started, stopped and muted (or unmuted) respectively.
> 
> chan_iax2
> ------------------
>  * You can now specify a default "auth" method in the [general]
>    section of iax.conf
> 
> chan_pjsip, app_transfer
> ------------------
>  * Added TRANSFERSTATUSPROTOCOL variable.  When transfer is performed,
>    transfers can pass a protocol specific error code.  Example, in
>    SIP 3xx-6xx represent any SIP specific error received when
>    performing a REFER.
> 
> func_odbc
> ------------------
>  * Introduce an ARGC variable for func_odbc functions, along with
>    a minargs per-function configuration option.
> 
>    minargs enables enforcing of minimum count of arguments to pass
>    to func_odbc, so if you're unconditionally using ARG1 through
>    ARG4 then this should be set to 4.  func_odbc will generate an
>    error in this case, so for example
> 
>    [FOO]
>    minargs = 4
> 
>    and ODBC_FOO(a,b,c) in dialplan will now error out instead of
>    using a potentially leaked ARG4 from Gosub().
> 
>    ARGC is needed if you're using optional argument, to verify
>    whether or not an argument has been passed, else it's possible
>    to use a leaked ARGn from Gosub (app_stack).  So now you can
>    safely do ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of
>    thing.
> 
> res_srtp
> ------------------
>  * SRTP replay protection has been added to res_srtp and
>    a new configuration option "srtpreplayprotection" has been added
>    to the rtp.conf config file.  For security reasons, the default
>    setting is "yes".  Buggy clients may not handle this correctly
>    which could result in no, or one way, audio and Asterisk error
>    messages like "replay check failed".
> 
> ------------------------------------------------------------------------------
> --- Functionality changes from Asterisk 18.1.0 to Asterisk 18.2.0 ------------
> ------------------------------------------------------------------------------
> 
> Core
> ------------------
>  * The location where the media cache stores its temporary files
>    is no longer hardcoded to /tmp but can now be configured separately
>    via the astcachedir config variable in asterisk.conf. To retain
>    backwards compatibility, the default location remains /tmp.
> 
> app_voicemail
> ------------------
>  * The VoiceMail application can now be configured to send greetings
>    and instructions via early media and only answering the channel
>    when it is time for the caller to record their message. This
>    behavior can be activated by passing the new 'e' option to
>    VoiceMail.
> 
> ------------------------------------------------------------------------------
> --- Functionality changes from Asterisk 18.0.0 to Asterisk 18.1.0 ------------
> ------------------------------------------------------------------------------
> 
> Core
> ------------------
>  * Added debug logging categories that allow a user to output debug
>    information based on a specified category. This lets the user
>    limit, and filter debug output to data relevant to a particular
>    context, or topic. For instance the following categories are
>    now available for debug logging purposes:
> 
>    dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet
> 
>    These debug categories can be enable/disable via an Asterisk
>    CLI command:
> 
>      core set debug category <category>[:<sublevel>] [category[:<sublevel] ...]
>      core set debug category off [<category> [<category>] ...]
> 
>    If no sub-level is associated all debug statements for a given
>    category are output. If a sub-level is given then only those
>    statements assigned a value at or below the associated sub-level
>    are output.
> 
> app_confbridge
> ------------------
>  * app_confbridge now has the ability to force the estimated bitrate
>    on an SFU bridge.  To use it, set a bridge profile's remb_behavior
>    to "force" and set remb_estimated_bitrate to a rate in bits per
>    second.  The remb_estimated_bitrate parameter is ignored if
>    remb_behavior is something other than "force".
> 
> ------------------------------------------------------------------------------
> --- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
> ------------------------------------------------------------------------------
> 
> chan_pjsip
> ------------------
>  * The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a
>    warning, and returns unsuccessful if it's used on a channel
>    prior to answering.
> 
> logger
> ------------------
>  * Added a new log formatter called "plain" that always prints
>    file, function and line number if available (even for verbose
>    messages) and never prints color control characters.  Most
>    suitable for file output but can be used for other channels as
>    well.
> 
>    You use it in logger.conf like so:
>    debug => [plain]debug
>    console => [plain]error,warning,debug,notice,pjsip_history
>    messages => [plain]warning,error,verbose
> 
> ------------------------------------------------------------------------------
> --- New functionality introduced in Asterisk 18.0.0 --------------------------
> ------------------------------------------------------------------------------
> 
> Core
> ------------------
>  * The Streams API becomes the home for the core ACN capabilities.
>    These include...
> 
>     * Parsing and formatting of codec negotation preferences.
>     * Resolving pending streams and topologies with those configured
>       using configured preferences.
>     * Utility functions for creating string representations of
>       streams, topologies, and negotiation preferences.
> 
>    For codec negotiation preferences:
>     * Added ast_stream_codec_prefs_parse() which takes a string
>       representation of codec negotiation preferences, which may
>       come from a pjsip endpoint for example, and populates a
>       ast_stream_codec_negotiation_prefs structure.
>     * Added ast_stream_codec_prefs_to_str() which does the reverse.
>     * Added many functions to parse individual parameter name
>       and value strings to their respectrive enum values, and the
>       reverse.
> 
>    For streams:
>     * Added ast_stream_create_resolved() which takes a "live" stream
>       and resolves it with a configured stream and the negotiation
>       preferences to create a new stream.
>     * Added ast_stream_to_str() which create a string representation
>       of a stream suitable for debug or display purposes.
> 
>    For topology:
>     * Added ast_stream_topology_create_resolved() which takes a
>       "live" topology and resolves it, stream by stream, with a
>       configured topology stream and the negotiation preferences
>       to create a new topology.
>     * Added ast_stream_topology_to_str() which create a string
>       representation of a topology suitable for debug or display
>       purposes.
>     * Renamed ast_format_caps_from_topology() to
>       ast_stream_topology_get_formats() to be more consistent with
>       the existing ast_stream_get_formats().
> 
>    Additional changes:
>     * A new function ast_format_cap_append_names() appends the
>       results to the ast_str buffer instead of replacing buffer
>       contents.
> 
> app_bridgeaddchan
> ------------------
>  * The BridgeAdd application now behaves more like the Bridge
>    application.  The application now sets the BRIDGERESULT channel
>    variable to indicate what happened when the channel resumes in
>    dialplan.  This is instead of hanging up the channel on failure
>    conditions.
> 
> res_pjsip
> ------------------
>  * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
>    have been added to res_pjsip endpoints that specify the preferred
>    order of codecs to use between those received/sent in an SDP
>    offer and those set in the endpoint configuration.
> 
> ------------------------------------------------------------------------------
> --- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
> ------------------------------------------------------------------------------
> 
> AMI
> ------------------
>  * You can now specify an optional 'Content-Type' as an argument
>    for the Asterisk SendText manager action.
> 
> ARI
> ------------------
>  * A new parameter 'inhibitConnectedLineUpdates' is now available
>    in the 'bridges.addChannel' call. This prevents the identity of
>    the newly connected channel from being presented to other bridge
>    members.
> 
> ARI Channels
> ------------------
>  * The Channel resource has a new sub-resource "externalMedia".
>    This allows an application to create a channel for the sole
>    purpose of exchanging media with an external server.  Once
>    created, this channel could be placed into a bridge with existing
>    channels to allow the external server to inject audio into the
>    bridge or receive audio from the bridge.  See
>    https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
>    for more information.
> 
> Core
> ------------------
>  * H.265/HEVC is now a supported video codec and it can be used by
>    specifying "h265" in the allow line.  Please note however, that
>    handling of the additional SDP parameters described in RFC 7798
>    section 7.2 is not yet supported.
> 
> Features
> ------------------
>  * Adds support for AudioSocket, a very simple bidirectional audio
>    streaming protocol. There are both channel and application
>    interfaces.
> 
>    A description of the protocol can be found on the referenced
>    wiki page. A short talk about the reasons and implementation
>    can be found on YouTube at the link provided.
> 
>    ARI support has also been added via the existing "externalMedia"
>    ARI functionality. The UUID is specified using the arbitrary
>    "data" field.
> 
>    Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
>    YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI
> 
> Messaging
> ------------------
>  * In order to reduce the amount of AMI and ARI events generated,
>    the global "Message/ast_msg_queue" channel can be set to suppress
>    it's normal channel housekeeping events such as "Newexten",
>    "VarSet", etc. This can greatly reduce load on the manager and
>    ARI applications when the Digium Phone Module for Asterisk is
>    in use.  To enable, set "hide_messaging_ami_events" in asterisk.conf
>    to "yes"  In Asterisk versions <18, the default is "no" preserving
>    existing behavior.  Beginning with Asterisk 18, the option will
>    default to "yes".
> 
> STIR/SHAKEN
> ------------------
>  * STIR/SHAKEN support has been added to Asterisk. Configuration
>    is done in stir_shaken.conf. There is a sample configuration
>    file to help you get started
>    (asterisk/configs/samples/stir_shaken.conf.sample).  Once that's
>    set up, you can enable STIR/SHAKEN on any endpoint by setting
>    stir_shaken to yes on the endpoint configuration object. This
>    will add an Identity header on outgoing INVITEs, and check for
>    an Identity header on incoming INVITEs. This option has been
>    added to Alembic as well.
> 
>    The information received on an incoming INVITE can be checked
>    using the STIR_SHAKEN dialplan function. There are two variations:
> 
>    STIR_SHAKEN(count)
>    STIR_SHAKEN(0, verify_result)
> 
>    The first variation will tell you how many STIR/SHAKEN results
>    are on the channel. The second fetches information for a specific
>    result. The first parameter is the index, followed by what
>    information you want to retrieve.  The available options are
>    'verify_result', 'identity', and 'attestation'.
> 
> app_chanisavail
> ------------------
>  * The ChanIsAvail application now tolerates empty positions in
>    the supplied device list.  Dialplan can now be simplified by
>    not having to check for empty positions in the device list.
> 
> app_confbridge
> ------------------
>  * A new bridge profile option, maximum_sample_rate, has been added
>    which sets a maximum sample rate that the bridge will be mixed
>    at. This allows the bridge to move below the maximum sample rate
>    as needed but caps it at the maximum.
> 
>  * A new option, "text_messaging", has been added to the user
>    profile which allows control over whether text messaging is
>    enabled or disabled for a user. If enabled (the default) text
>    messages will be sent to the user. If disabled no text messages
>    will be sent to the user.
> 
> app_dial
> ------------------
>  * The Dial application now tolerates empty positions in the supplied
>    destination list.  Dialplan can now be simplified by not having
>    to check for empty positions in the destination list.  If there
>    are no endpoints to dial then DIALSTATUS is set to CHANUNAVAIL.
> 
> app_mixmonitor
> ------------------
>  * An option 'S' has been added to MixMonitor. If used in combination
>    with the r() and/or t() options, if a frame is available to
>    write to one of those files but not the other, a frame of silence
>    if written to the file that does not have an audio frame. This
>    should prevent the two files from "drifting" when mixed after
>    the fact.
> 
>  * If the 'filename' argument to MixMonitor() ended with '.wav49,'
>    Asterisk would silently convert the extension to '.WAV' when
>    opening the file for writing. This caused the MIXMONITOR_FILENAME
>    variable to reference the wrong file. The MIXMONITOR_FILENAME
>    variable will now reflect the name of the file that Asterisk
>    actually used instead of the filename that was passed to the
>    application.
> 
> app_page
> ------------------
>  * The Page application now tolerates empty positions in the supplied
>    destination list.  Dialplan can now be simplified by not having
>    to check for empty positions in the destination list.
> 
> app_voicemail
> ------------------
>  * A feature was added in Asterisk 13.27.0 and 16.4.0 that removed
>    lock files from the Asterisk voicemail directory on startup.
>    Some users that store their voicemails on network storage devices
>    experienced slow startup times due to the relative expense of
>    traversing the voicemail directory structure looking for orphaned
>    lock files. This feature has now been removed.
> 
>    Users who require the lock files to be removed at startup should
>    modify their startup scripts to do so before starting the asterisk
>    process.
> 
> chan_pjsip
> ------------------
>  * A new dialplan function, PJSIP_MOH_PASSTRHOUGH, has been added
>    to chan_pjsip. This allows the behaviour of the moh_passthrough
>    endpoint option to be read or changed in the dialplan. This
>    allows control on a per-call basis.
> 
> chan_rtp
> ------------------
>  * The UnicastRTP channel driver provided by chan_rtp now accepts
>    "<hostname>:<port>" as an alternative to "<ip_address>:<port>"
>    in the destination.  The first AAAA (preferred) or A record
>    resolved will be used as the destination.  The lookup is
>    synchronous so beware of possible dialplan delays if you specify
>    a hostname.
> 
> func_curl
> ------------------
>  * A new parameter, httpheader, has been added to CURLOPT function.
>    This parameter allows to set custom http headers for subsequent
>    calls of CURL function.  Any setting of headers will replace
>    the default curl headers (e.g. "Content-type:
>    application/x-www-form-urlencoded")
> 
>  * A new option, followlocation, can now be enabled with the
>    CURLOPT() dialplan function. Setting this will instruct cURL to
>    follow 3xx redirects, which it does not by default.
> 
> func_jitterbuffer
> ------------------
>  * The JITTERBUFFER dialplan function now has an option to enable
>    video synchronization support. When enabled and used with a
>    compatible channel driver (chan_sip, chan_pjsip) the video is
>    buffered according to the size of the audio jitterbuffer and is
>    synchronized to the audio.
> 
> func_volume
> ------------------
>  * Accept decimal number as argument.
> 
> http
> ------------------
>  * You can now disable the /httpstatus page served by Asterisk's
>    built-in HTTP server by setting 'enable_status' to 'no' in
>    http.conf.
> 
> minmemfree
> ------------------
>  * The 'minmemfree' configuration option now counts memory allocated
>    to the filesystem cache as "free" because it is memory that is
>    available to the process.
> 
> res_ari_channels
> ------------------
>  * When creating a channel in ARI using the create call
>    you can now specify dialplan variables to be set as part of the
>    same operation.
> 
> res_musiconhold
> ------------------
>  * This fix allows a realtime moh class to be unregistered from
>    the command line. This is useful when the contents of a directory
>    referenced by a realtime moh class have changed.  The realtime
>    moh class is then reloaded on the next request and uses the new
>    directory contents.
> 
>  * A new mode - playlist - has been added to res_musiconhold. This
>    mode allows the user to specify the files (or URLs) to play
>    explicitly by putting them directly in musiconhold.conf.
> 
> res_pjsip
> ------------------
>  * Added a new PJSIP system setting called disable_rport.
>    Default is no to keep support working as before.
> 
>    If it is false (default) it adds the 'rport' parameter in the
>    outgoing request message.  If it is true it does not add the
>    'rport' parameter in the outgoing request message.
> 
>    This is a system option, but working as a global option.
> 
> res_pjsip_endpoint_identifier_ip
> ------------------
>  * In 'type = identify' sections, the addresses specified for the
>    'match' clause can now include a port number. For IP addresses,
>    the port is provided by including a colon after the address,
>    followed by the desired port number. If supplied, the netmask
>    should follow the port number. To specify a port for IPv6
>    addresses, the address itself must be enclosed in brackets to
>    be parsed correctly.
> 
> res_pjsip_logger
> ------------------
>  * The PJSIP packet logger now has the following CLI commands:
> 
>    pjsip set logger pcap <filename>
> 
>    When used this will create a pcap file containing the incoming
>    and outgoing SIP packets, in unencrypted form.
> 
>    pjsip set logger console <on / off>
> 
>    This allows you to toggle logging to console on and off.
> 
>    pjsip set logger host <IP/subnet mask> add
> 
>    This allows you to add an additional IP address or subnet mask
>    to logging, allowing you to log multiple instead of just a single
>    IP address or all traffic.
> 
>    The normal "pjsip set logger host" CLI command has also been
>    expanded to allow subnet masks as well.
> 
> res_pjsip_session
> ------------------
>  * When placing an outgoing call to a PJSIP endpoint the intent
>    of any requested formats will now be respected. If only an audio
>    format is requested (such as ulaw) but the underlying endpoint
>    does not support the format the resulting SDP will still only
>    contain an audio stream, and not any additional streams such as
>    video.
> 
>  * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
>    have been added to res_pjsip endpoints that specify the preferred
>    order of codecs to use between those received/sent in an SDP
>    offer and those set in the endpoint configuration.
> 
> res_rtp_asterisk
> ------------------
>  * This change include a new cli command 'rtp show settings'
> 
>    The command display by general settings of rtp configuration.
>    For this point is added the fields: rtpstart, rtpend, dtmftimeout,
>    rtpchecksum, strictrtp, learning_min_sequential and icesupport.
> 
>  * The blacklist mechanism in res_rtp_asterisk for ICE and STUN
>    was converted to an ACL mechanism.
> 
>    As such six new options are now available:
> 
>    ice_deny
>    ice_permit
>    ice_acl
>    stun_deny
>    stun_permit
>    stun_acl
> 
>    These options have their obvious meanings as used elsewhere.
> 
>    Backwards compatibility was maintained by adding {stun,ice}_blacklist
>    as aliases for {stun,ice}_deny.
> 
> res_sorcery_memory_cache
> ------------------
>  * The SorceryMemoryCacheExpireObject AMI action and CLI
>    command allow expiring of a specific object within the sorcery
>    memory cache. This is done by removing the object from the cache
>    with the expectation that the cache will then re-populate the
>    object when it is next needed.
> 
>    For full backend caching this does not occur. The cache won't
>    repopulate until an entire refresh is done resulting in the
>    possibility that objects are missing until that time.
> 
>    The AMI action and CLI command will now not allow expiring of
>    an object if the cache is configured as a full backend cache.
>    Instead you must use either the SorceryMemoryCacheExpire or
>    SorceryMemoryCachePopulate AMI actions or their associated CLI
>    commands.
> 
> taskprocessor.c
> ------------------
>  * Added two new CLI commands to reset stats for taskprocessors.
>    You can reset stats for a single, specific taskprocessor ('core
>    reset taskprocessor <taskprocessor>'), or you can reset all
>    taskprocessors ('core reset taskprocessors'). These commands
>    will reset the counter for the number of tasks processed as well
>    as the max queue size.
> 
>  * Added "like" support for 'core show taskprocessors'. Now you
>    can specify a specific set of taskprocessors (or just one) by
>    adding the keyword "like" to the above command, followed by your
>    search criteria.
> 
> Status:
> 
> Vendor Tag:   TNF
> Release Tags: pkgsrc-base
>               
> C pkgsrc/comms/asterisk18/DESCR
> C pkgsrc/comms/asterisk18/Makefile
> C pkgsrc/comms/asterisk18/PLIST
> C pkgsrc/comms/asterisk18/distinfo
> C pkgsrc/comms/asterisk18/options.mk
> C pkgsrc/comms/asterisk18/files/asterisk.sh
> N pkgsrc/comms/asterisk18/files/smf/manifest.xml
> N pkgsrc/comms/asterisk18/patches/patch-funcs_func__channel.c
> N pkgsrc/comms/asterisk18/patches/patch-funcs_func__env.c
> N pkgsrc/comms/asterisk18/patches/patch-funcs_func__pjsip__endpoint.c
> N pkgsrc/comms/asterisk18/patches/patch-include_asterisk_autoconfig.h.in
> N pkgsrc/comms/asterisk18/patches/patch-main_acl.c
> N pkgsrc/comms/asterisk18/patches/patch-main_app.c
> N pkgsrc/comms/asterisk18/patches/patch-main_ast__expr2.c
> N pkgsrc/comms/asterisk18/patches/patch-main_astmm.c
> N pkgsrc/comms/asterisk18/patches/patch-main_callerid.c
> N pkgsrc/comms/asterisk18/patches/patch-Makefile
> N pkgsrc/comms/asterisk18/patches/patch-channels_pjsip_dialplan__functions.c
> N pkgsrc/comms/asterisk18/patches/patch-funcs_func__cdr.c
> N pkgsrc/comms/asterisk18/patches/patch-funcs_func__strings.c
> N pkgsrc/comms/asterisk18/patches/patch-main_Makefile
> N pkgsrc/comms/asterisk18/patches/patch-main_asterisk.c
> N pkgsrc/comms/asterisk18/patches/patch-main_bridge__basic.c
> N pkgsrc/comms/asterisk18/patches/patch-main_cdr.c
> N pkgsrc/comms/asterisk18/patches/patch-main_conversions.c
> N pkgsrc/comms/asterisk18/patches/patch-main_cel.c
> N pkgsrc/comms/asterisk18/patches/patch-main_dns__naptr.c
> N pkgsrc/comms/asterisk18/patches/patch-main_enum.c
> N pkgsrc/comms/asterisk18/patches/patch-main_features.c
> N pkgsrc/comms/asterisk18/patches/patch-main_http.c
> N pkgsrc/comms/asterisk18/patches/patch-main_indications.c
> N pkgsrc/comms/asterisk18/patches/patch-apps_app__voicemail.c
> N pkgsrc/comms/asterisk18/patches/patch-contrib_scripts_vmail.cgi
> N pkgsrc/comms/asterisk18/patches/patch-include_asterisk_lock.h
> N pkgsrc/comms/asterisk18/patches/patch-apps_app__adsiprog.c
> N pkgsrc/comms/asterisk18/patches/patch-main_manager.c
> N pkgsrc/comms/asterisk18/patches/patch-main_pbx.c
> N pkgsrc/comms/asterisk18/patches/patch-main_pbx__builtins.c
> N pkgsrc/comms/asterisk18/patches/patch-main_pbx__timing.c
> N pkgsrc/comms/asterisk18/patches/patch-main_sched.c
> N pkgsrc/comms/asterisk18/patches/patch-main_test.c
> N pkgsrc/comms/asterisk18/patches/patch-main_utils.c
> N pkgsrc/comms/asterisk18/patches/patch-menuselect_menuselect.c
> N pkgsrc/comms/asterisk18/patches/patch-include_asterisk_sha1.h
> N pkgsrc/comms/asterisk18/patches/patch-apps_app__queue.c
> N pkgsrc/comms/asterisk18/patches/patch-apps_app__sms.c
> N pkgsrc/comms/asterisk18/patches/patch-cdr_cdr__pgsql.c
> N pkgsrc/comms/asterisk18/patches/patch-cel_cel__pgsql.c
> N pkgsrc/comms/asterisk18/patches/patch-res_res__hep__pjsip.c
> N pkgsrc/comms/asterisk18/patches/patch-res_res__limit.c
> N pkgsrc/comms/asterisk18/patches/patch-sounds_Makefile
> N pkgsrc/comms/asterisk18/patches/patch-tests_test__locale.c
> N pkgsrc/comms/asterisk18/patches/patch-tests_test__voicemail__api.c
> N pkgsrc/comms/asterisk18/patches/patch-utils_Makefile
> N pkgsrc/comms/asterisk18/patches/patch-utils_db1-ast_include_db.h
> N pkgsrc/comms/asterisk18/patches/patch-channels_chan__sip.c
> N pkgsrc/comms/asterisk18/patches/patch-channels_chan__pjsip.c
> N pkgsrc/comms/asterisk18/patches/patch-channels_pjsip_cli__commands.c
> N pkgsrc/comms/asterisk18/patches/patch-configure.ac
> N pkgsrc/comms/asterisk18/patches/patch-configure
> N pkgsrc/comms/asterisk18/patches/patch-funcs_func__pjsip__aor.c
> N pkgsrc/comms/asterisk18/patches/patch-main_ast__expr2.y
> N pkgsrc/comms/asterisk18/patches/patch-addons_chan__ooh323.c
> N pkgsrc/comms/asterisk18/patches/patch-main_cli.c
> N pkgsrc/comms/asterisk18/patches/patch-main_logger.c
> N pkgsrc/comms/asterisk18/patches/patch-res_res__calendar__caldav.c
> N pkgsrc/comms/asterisk18/patches/patch-res_res__musiconhold.c
> N pkgsrc/comms/asterisk18/patches/patch-utils_extconf.c
> N pkgsrc/comms/asterisk18/patches/patch-apps_app__chanspy.c
> N pkgsrc/comms/asterisk18/patches/patch-apps_app__directory.c
> N pkgsrc/comms/asterisk18/patches/patch-apps_app__dumpchan.c
> N pkgsrc/comms/asterisk18/patches/patch-res_res__xmpp.c
> N pkgsrc/comms/asterisk18/patches/patch-utils_smsq.c
> N pkgsrc/comms/asterisk18/patches/patch-include_asterisk_strings.h
> N pkgsrc/comms/asterisk18/patches/patch-apps_app__followme.c
> N pkgsrc/comms/asterisk18/patches/patch-apps_app__minivm.c
> N pkgsrc/comms/asterisk18/patches/patch-build__tools_mkpkgconfig
> N pkgsrc/comms/asterisk18/patches/patch-main_tdd.c
> N pkgsrc/comms/asterisk18/patches/patch-funcs_func__pjsip__contact.c
> N pkgsrc/comms/asterisk18/patches/patch-main_stdtime_localtime.c
> N pkgsrc/comms/asterisk18/patches/patch-pbx_pbx__config.c
> N pkgsrc/comms/asterisk18/patches/patch-pbx_pbx__dundi.c
> N pkgsrc/comms/asterisk18/patches/patch-res_ael_pval.c
> N pkgsrc/comms/asterisk18/patches/patch-res_res__calendar.c
> N pkgsrc/comms/asterisk18/patches/patch-res_res__calendar__icalendar.c
> N pkgsrc/comms/asterisk18/patches/patch-res_res__pjproject.c
> 
> 6 conflicts created by this import.
> Use the following command to help the merge:
> 
>       cvs checkout -jTNF:yesterday -jTNF pkgsrc/comms/asterisk18
>