pkgsrc-Changes-HG archive
[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index][Old Index]
[pkgsrc/trunk]: pkgsrc/comms/asterisk14 update Asterisk to 14.7.5 -- this is ...
details: https://anonhg.NetBSD.org/pkgsrc/rev/1e02f961988d
branches: trunk
changeset: 374483:1e02f961988d
user: jnemeth <jnemeth%pkgsrc.org@localhost>
date: Wed Jan 24 05:51:40 2018 +0000
description:
update Asterisk to 14.7.5 -- this is a bug fix and security update,
it fixes AST-2017-005, AST-2017-006, AST-2017-006, AST-2017-008,
AST-2017-009, AST-2017-010, AST-2017-011, AST-2017-012, AST-2017-013,
and AST-2017-014. Note that several of these are related to PJSIP
which pkgsrc doesn't use.
----- 14.7.5 -----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.18.5, 14.7.5,
15.1.5 and 13.18-cert2.
The following security vulnerabilities were resolved in these versions:
* AST-2017-014: Crash in PJSIP resource when missing a contact header
A select set of SIP messages create a dialog in Asterisk. Those SIP messages
must contain a contact header. For those messages, if the header was not
present and using the PJSIP channel driver, it would cause Asterisk to crash.
The severity of this vulnerability is somewhat mitigated if authentication is
enabled. If authentication is enabled a user would have to first be authorized
before reaching the crash point.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
Thank you for your continued support of Asterisk!
----- 14.7.4 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert9, 13.18.4,
14.7.4 and 15.1.4.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-012: Remote Crash Vulnerability in RTCP Stack
If a compound RTCP packet is received containing more than
one report (for example a Receiver Report and a Sender
Report) the RTCP stack will incorrectly store report
information outside of allocated memory potentially causing
a crash.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-012.html
http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
Thank you for your continued support of Asterisk!
----- 14.7.3 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert8, 13.18.3,
14.7.3 and 15.1.3.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
If the chan_skinny (AKA SCCP protocol) channel driver is
flooded with certain requests it can cause the asterisk
process to use excessive amounts of virtual memory
eventually causing asterisk to stop processing requests of
any kind.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
Thank you for your continued support of Asterisk!
----- 14.7.2 -----
The Asterisk Development Team would like to announce the release
of Asterisk 14.7.2.
The release of Asterisk 14.7.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.7.2
Thank you for your continued support of Asterisk!
----- 14.7.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 14.7.0.
The release of Asterisk 14.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Improvements made in this release:
-----------------------------------
* ASTERISK-27278 - [patch] chan_sip: Provide access to read the
full SIP Request-URI from INVITE
(Reported by David J. Pryke)
* ASTERISK-27255 - alembic: Add support for Microsoft SQL
server
(Reported by Florian Floimair)
* ASTERISK-27253 - [patch] libsrtp-2.1.x support
(Reported by Alexander Traud)
* ASTERISK-27220 - Enable CHANNEL function to get from and to
tag from SIP Headers
(Reported by Andre Nazario)
* ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif
(Reported by Andrey)
* ASTERISK-27173 - Support for GMIME 3.0
(Reported by Tzafrir Cohen)
* ASTERISK-27092 - [patch] app_queue: Add Priority to AMI
QueueStatus
(Reported by Niklas Larsson)
* ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
chan_pjsip
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-27047 - res_pjsip: user=phone added to Anonymous
caller-id when it shouldn't be.
(Reported by dtryba)
* ASTERISK-27270 - cdr_mysql: various crashes at second module
reload if cdr_mysql.conf is configured
(Reported by Tzafrir Cohen)
* ASTERISK-25266 - Application Originate returns SUCCESS to
ORIGINATE_STATUS upon failure to originate
(Reported by Allen Ford)
* ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing
unavailable endpoints
(Reported by Richard Mudgett)
* ASTERISK-27305 - res_ari: Memory leaks in ARI when using
Content-Type: application/json
(Reported by David Hajek)
* ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
* ASTERISK-27324 - [patch] Dual-Stack server cannot be used as
IPv4 client via TCP/TLS
(Reported by Alexander Traud)
* ASTERISK-27317 - vector: multiple evaluation of elem in
AST_VECTOR_ADD_SORTED.
(Reported by Corey Farrell)
* ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
ast_strings_match
(Reported by Corey Farrell)
* ASTERISK-27284 - Status of RFC 3323 and PJSIP
(Reported by dtryba)
* ASTERISK-27296 - [patch] False positive busy checks when
icalendar's recurrence-id mechanism is involved
(Reported by Beno?t Dereck-Tricot)
* ASTERISK-27216 - app_queue: does its
check-makeannouncement-logic twice each head-caller-loop
(Reported by Stefan Engstr?m)
* ASTERISK-27298 - Problem with expires on pjsip /
outbound-publish
(Reported by Cyrille Demaret)
* ASTERISK-27295 - Contact is improperly translated after
d178f497
(Reported by Sean Bright)
* ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833
(SSRC Changes)
(Reported by Ross Beer)
* ASTERISK-27289 - A codeblock that maintains a bug,but maybe
the codeblock will never run
(Reported by Huangyx)
* ASTERISK-27283 - Realtime config fail with PostgreSQL version
before 9.1
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-27257 - bridge_native_rtp: half-way direct media
when using early bridging
(Reported by Jean Aunis - Prescom)
* ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer -
Possible PJSIP Vulnerability
(Reported by Ross Beer)
* ASTERISK-26606 - tcptls: Incorrect OpenSSL function call
leads to misleading error report
(Reported by Bob Ham)
* ASTERISK-16898 - SRTP unprotect: authentication failure when
RTP sequence number switches from 65535 -> 0
(Reported by Marcello Ceschia)
* ASTERISK-27274 - RTCP needs better packet validation to
resist port scans.
(Reported by Richard Mudgett)
* ASTERISK-27252 - RTP: One way audio with direct media and
strictrtp=yes.
(Reported by Richard Mudgett)
* ASTERISK-25524 - module reload res_calendar.so does not
reload everything in calendar.conf
(Reported by Jesper)
* ASTERISK-24588 - res_calendar does not process CalDAV from
Owncloud [fix included]
(Reported by Stefan Gofferje)
* ASTERISK-25523 - res_calendar: Warning about invalid channel
value (for notification) occurs even when event has no
notification configured.
(Reported by Jesper)
* ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and
wireshark disagree
(Reported by Tzafrir Cohen)
* ASTERISK-27248 - [patch]external_media_address and
external_signaling_address don't always honor localnet
(Reported by Walter Doekes)
* ASTERISK-27217 - chan_sip: Asterisk crashing when
subscription doesn't get set
(Reported by Bryan Walters)
* ASTERISK-24066 - res_smdi: convert to astobj2
(Reported by Corey Farrell)
* ASTERISK-17540 - SDP origin attribute modified when issuing
re-INVITE because of directmedia=yes
(Reported by saghul)
* ASTERISK-27254 - alembic: prune_on_boot fix erroneous
(Reported by Florian Floimair)
* ASTERISK-27232 - When in queue on g722 with interruptions,
music on hold can get stuck and no longer play
(Reported by Jens T.)
* ASTERISK-27024 - nat/external_media settings ignored in 14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-26879 - PJSIP external_media_address ignored if no
local_net options are provided
(Reported by Matt Jordan)
* ASTERISK-27165 - CDR: CDR(start,u) function won't work in
cdr_custom config
(Reported by Jacek Konieczny)
* ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at
channel_internal_api.c:478 during T.38 Fax Receive
(Reported by Ross Beer)
* ASTERISK-27225 - Crash when freeing dtls_cfg->cafile
(Reported by Richard Kenner)
* ASTERISK-27177 - ooh323c: misleading indentation in
addons/ooh323c/src/ooSocket.c
(Reported by Tzafrir Cohen)
* ASTERISK-27241 - libc segfault upon entry into app_directory
(Reported by David Moore)
* ASTERISK-27152 - Sending a "tel" uri in a From or To header
in an unauthenticated message causes asterisk to crash
(Reported by Ross Beer)
* ASTERISK-27103 - core: ast_safe_system command injection
possible.
(Reported by Corey Farrell)
* ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
with strict RTP enabled
(Reported by Joshua Colp)
* ASTERISK-26994 - Confbridge: CBAnn channels intermittently
become stuck when caller hangs up before recording name
(Reported by James Terhune)
* ASTERISK-20858 - app_minivm fails to clean up mkstemp files
(Reported by Walter Doekes)
* ASTERISK-16777 - several filename bugs in Record()
application
(Reported by klaus3000)
* ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
is used
(Reported by Torrey Searle)
* ASTERISK-27168 - alembic: PJSIP scripts are missing column
dtls_fingerprint in ps_endpoints table
(Reported by Florian Floimair)
* ASTERISK-19103 - When using realtime queues, function
QUEUE_MEMBER_LIST() will return an error if no other
app/function has loaded the queues first. This problem does not
exist if queues.conf is used.
(Reported by Jim Van Meggelen)
* ASTERISK-21241 - When using voicemail as announce only
(maxmsg=0), the star dtmf to enter the voicemail is not honored
(Reported by Eelco Brolman)
* ASTERISK-27204 - [patch] app_queue: Wrong queue stat
calculation
(Reported by sungtae kim)
* ASTERISK-27207 - XMPP OAuth not working due to inverted
logic
(Reported by Michael Kuron)
* ASTERISK-27174 - res_calendar_icalendar: Recurring events not
being loaded from Google calendar using ical
(Reported by Mark Thompson)
* ASTERISK-27202 - If wget is not installed and "or" is not
available, external components (excluding pjsip) are not
installed
(Reported by Se?n C. McCord)
* ASTERISK-27147 - Either asterisk or pjproject isn't re-using
tcp connections (again)
(Reported by George Joseph)
* ASTERISK-27193 - IPv6 receive address in message doesn't
include brackets
(Reported by Scott Griepentrog)
* ASTERISK-26745 - Asymmetric codecs when
asymmetric_rtp_codec=no
(Reported by Jesse Ross)
* ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics
are not available when native bridge is used
(Reported by Torrey Searle)
* ASTERISK-27110 - RTP session is not fully destroyed on
channel hangup
(Reported by Matt Jordan)
* ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
* ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces
around the status element in XML
(Reported by Abraham Liebsch)
* ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
devmode enabled.
(Reported by Corey Farrell)
* ASTERISK-27130 - Applications ARI: Unsubscribe action for
deviceStates does not remove old subscriptions properly
(Reported by Sergej Kasumovic)
* ASTERISK-25810 - say.c calls for sounds in the subdir
"digits" that don't exist (in Core). SayUnixTime or other Say...
apps will fail out when they call these sounds.
(Reported by Nicolas Riendeau)
* ASTERISK-27142 - sounds: Conflict between files in
asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
(Reported by Corey Farrell)
* ASTERISK-27133 - res_rtp_asterisk: RTCP does not use ICE when
RTCP-MUX in use
(Reported by Joshua Colp)
* ASTERISK-27123 - confbridge: Name recordings are left on
filesystem
(Reported by Sergej Kasumovic)
* ASTERISK-27122 - chan_iax2: On reload MWI taskprocessors keep
adding up
(Reported by Sergej Kasumovic)
* ASTERISK-26807 - sounds: New 3-D Binaural audio features
require new sound prompts
(Reported by Rusty Newton)
* ASTERISK-25816 - French conf-adminmenu, conf-usermenu prompts
differ in content from the English files
(Reported by Benoit Duverger)
* ASTERISK-26274 - Resolve open sounds issues and then create a
new sounds release (1.5.1? or 1.6?)
(Reported by Rusty Newton)
* ASTERISK-27128 - [patch]res_stasis_snoop: When recording a
snoop channel (using ARI) where no media is being received, no
recording happens when theres no media
(Reported by Dan Jenkins)
* ASTERISK-27124 - app_playback.c:say_date_generic use
timezonename parameter
(Reported by Holger Hans Peter Freyther)
* ASTERISK-27127 - configs: Erroneous load directive in sample
configuration results in "Error loading module
'res_pjsip_multihomed.so'"
(Reported by HZMI8gkCvPpom0tM)
* ASTERISK-27105 - [patch]core: when setting 'maxfiles' in
asterisk.conf, a message is printed, even in rasterisk -x
(Reported by Tzafrir Cohen)
* ASTERISK-27036 - res_pjsip: Asterisk crashes when an
extension tries to use PJSIP trunk with from_user containing '@'
(Reported by Maxim Vasilev)
* ASTERISK-27023 - res_rtp_asterisk: Deadlock when TURN session
in use
(Reported by Jatin Jain)
* ASTERISK-27093 - ODBC deadlocks when app_directory tries to
play back non-existent voicemail greeting
(Reported by James Terhune)
New Features made in this release:
-----------------------------------
* ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action
(Reported by Thomas Sevestre)
* ASTERISK-27117 - core: Add support for timelen parsing to
ast_parse_arg and ACO.
(Reported by Corey Farrell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.7.0
Thank you for your continued support of Asterisk!
----- 14.6.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 14.6.0.
The release of Asterisk 14.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27108 - Crash using 'data get' CLI command
(Reported by Sean Bright)
* ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add
only really different domain with TLS.
(Reported by Alexander Traud)
* ASTERISK-27100 - channel: ast_waitfordigit_full fails to
clear flag in an error branch.
(Reported by Corey Farrell)
* ASTERISK-27090 - PJSIP: Deadlock using TCP transport
(Reported by Richard Mudgett)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)
* ASTERISK-27065 - call hangup after leaving app_queue
(Reported by Marek Cervenka)
* ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code()
(Reported by Ross Beer)
* ASTERISK-24052 - app_voicemail reloads result in leaked IMAP sockets.
(Reported by Louis Jocelyn Paquet)
* ASTERISK-27074 - core_local: local channel data not being
properly unref'ed and unlocked
(Reported by Kevin Harwell)
* ASTERISK-27075 - bridge: stuck channel(s) after failed
attended transfer
(Reported by Kevin Harwell)
* ASTERISK-27060 - Comment typo format_g729.c
(Reported by Matthew Fredrickson)
* ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
execution and application unregistration
(Reported by Frederic LE FOLL)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
sorcery.c
(Reported by Ryan Smith)
* ASTERISK-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by J?rgen H)
* ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for
RTCP component
(Reported by Michael Walton)
* ASTERISK-26923 - bridging: T.38 request is lost when channels
are added to bridge
(Reported by Torrey Searle)
* ASTERISK-27053 - res_pjsip_refer/session: Calls dropped
during transfer
(Reported by Kevin Harwell)
* ASTERISK-27052 - Asterisk build process fails with flag
--with-pjproject-bundled with curl download command and slow
network
(Reported by alex)
* ASTERISK-27039 - chan_pjsip: Device state is idle when
channel from endpoint is in early media
(Reported by Joshua Colp)
* ASTERISK-26996 - chan_pjsip: Flipping between codecs
(Reported by Michael Maier)
* ASTERISK-26281 - chan_pjsip would send INVITE to
'Unreachable' endpoints
(Reported by Jacek Konieczny)
* ASTERISK-26973 - bridge: Crash when freeing frame and
snooping
(Reported by Michel R. Vaillancourt)
* ASTERISK-19291 - Background in realtime
(Reported by Andrew Nowrot)
* ASTERISK-27025 - channel / meetme: Fix missing parentheses
(Reported by Joshua Colp)
* ASTERISK-27021 - GET /recordings/stored returns 500 Internal
Server Error
(Reported by Tim Morgan)
* ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets
in wrong byte order on Intel platform when using slin codec
(Reported by Frankie Chin)
* ASTERISK-23951 - Asterisk attempts and fails to build
format_mp3 even if mp3lib was not downloaded
(Reported by Tzafrir Cohen)
* ASTERISK-25294 - srtp's crypto_get_random deprecated
(Reported by Tzafrir Cohen)
* ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't
describe BEEP argument
(Reported by Rusty Newton)
* ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set
variable" command without args
(Reported by Antoine Pitrou)
* ASTERISK-25662 - Malformed AGI 520 Usage response
(Reported by Tony Mountifield)
* ASTERISK-27008 - res_format_attr_h264: SDP parse fails if
fmtp optional parameters have a space
(Reported by John Harris)
* ASTERISK-26399 - app_queue: Agent not called when caller is
parked
(Reported by wushumasters)
* ASTERISK-26400 - app_queue: Queue member stops being called
after AMI "Redirect" action for queues with wrapuptime
(Reported by Etienne Lessard)
* ASTERISK-26715 - app_queue: Member will not receive any new
calls after doing a transfer if wrapuptime = greater than 0 and
using Local channel
(Reported by David Brillert)
* ASTERISK-26975 - app_queue: Non-zero wrapup time can cause
agents not to receive queue calls after transfer queue call
(Reported by Lorne Gaetz)
* ASTERISK-27012 - app_confbridge: ConfBridge sometimes does
not play user name recording while leaving
(Reported by Robert Mordec)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
(Reported by Javier Riveros)
* ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
completion failure/delay if client offers rtcp-mux as
negotiable
(Reported by Stefan Engstr?m)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
Improvements made in this release:
-----------------------------------
* ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
block PJSIP taskprocessor on startup
(Reported by Alexei Gradinari)
* ASTERISK-27043 - Core/BuildSystem: Add defines to fix build
with LibreSSL
(Reported by Guido Falsi)
* ASTERISK-27042 - Unpatched asterisk sources fail to build on
FreeBSD due to missing crypt.h file
(Reported by Guido Falsi)
* ASTERISK-26419 - audiohooks: Remove redundant codec
translations when using audiohooks
(Reported by Michael Walton)
* ASTERISK-26976 - libsrtp-2.x.x support
(Reported by Alex)
* ASTERISK-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0
Thank you for your continued support of Asterisk!
diffstat:
comms/asterisk14/Makefile | 11 ++--
comms/asterisk14/PLIST | 18 ++++---
comms/asterisk14/distinfo | 20 ++++----
comms/asterisk14/patches/patch-apps_app__queue.c | 57 +++++++++++++----------
4 files changed, 58 insertions(+), 48 deletions(-)
diffs (248 lines):
diff -r 42efee77e3e0 -r 1e02f961988d comms/asterisk14/Makefile
--- a/comms/asterisk14/Makefile Wed Jan 24 00:46:30 2018 +0000
+++ b/comms/asterisk14/Makefile Wed Jan 24 05:51:40 2018 +0000
@@ -1,12 +1,11 @@
-# $NetBSD: Makefile,v 1.17 2018/01/01 21:18:17 adam Exp $
+# $NetBSD: Makefile,v 1.18 2018/01/24 05:51:40 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
-DISTNAME= asterisk-14.5.0
+DISTNAME= asterisk-14.7.5
#PKGREVISION= 6
-PKGREVISION= 4
CATEGORIES= comms net audio
MASTER_SITES= http://downloads.asterisk.org/pub/telephony/asterisk/
MASTER_SITES+= http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
@@ -142,7 +141,7 @@
.include "options.mk"
# check sounds/Makefile for current version when upgrading package
-DISTFILES+= asterisk-extra-sounds-en-gsm-1.5.tar.gz
+DISTFILES+= asterisk-extra-sounds-en-gsm-1.5.1.tar.gz
# Override default paths in config files
SUBST_CLASSES+= configs
@@ -252,9 +251,9 @@
post-install:
# check sounds directory for current versions when upgrading package
- ${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.5.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
+ ${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.6.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
${TAR} xzf ${WRKSRC}/sounds/asterisk-moh-opsound-wav-2.03.tar.gz -C ${DESTDIR}${ASTDATADIR}/moh
- ${TAR} xzf ${DISTDIR}/${DIST_SUBDIR}/asterisk-extra-sounds-en-gsm-1.5.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
+ ${TAR} xzf ${DISTDIR}/${DIST_SUBDIR}/asterisk-extra-sounds-en-gsm-1.5.1.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
${INSTALL_DATA} ${WRKSRC}/BUGS ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
${INSTALL_DATA} ${WRKSRC}/CHANGES ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
${INSTALL_DATA} ${WRKSRC}/COPYING ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
diff -r 42efee77e3e0 -r 1e02f961988d comms/asterisk14/PLIST
--- a/comms/asterisk14/PLIST Wed Jan 24 00:46:30 2018 +0000
+++ b/comms/asterisk14/PLIST Wed Jan 24 05:51:40 2018 +0000
@@ -1,4 +1,4 @@
-@comment $NetBSD: PLIST,v 1.4 2017/06/21 13:33:47 jnemeth Exp $
+@comment $NetBSD: PLIST,v 1.5 2018/01/24 05:51:40 jnemeth Exp $
include/asterisk.h
include/asterisk/_private.h
include/asterisk/abstract_jb.h
@@ -517,15 +517,15 @@
libdata/asterisk/scripts/ast_logescalator
libdata/asterisk/scripts/ast_loggrabber
libdata/asterisk/scripts/refcounter.py
-libdata/asterisk/sounds/en/.asterisk-core-sounds-en-gsm-1.5
+libdata/asterisk/sounds/en/.asterisk-core-sounds-en-gsm-1.6
libdata/asterisk/sounds/en/1-for-am-2-for-pm.gsm
libdata/asterisk/sounds/en/1-yes-2-no.gsm
-libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.5
-libdata/asterisk/sounds/en/CHANGES-asterisk-extra-en-1.5
-libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.5
-libdata/asterisk/sounds/en/CREDITS-asterisk-extra-en-1.5
-libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.5
-libdata/asterisk/sounds/en/LICENSE-asterisk-extra-en-1.5
+libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.6
+libdata/asterisk/sounds/en/CHANGES-asterisk-extra-en-1.5.1
+libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.6
+libdata/asterisk/sounds/en/CREDITS-asterisk-extra-en-1.5.1
+libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.6
+libdata/asterisk/sounds/en/LICENSE-asterisk-extra-en-1.5.1
libdata/asterisk/sounds/en/OfficeSpace.gsm
libdata/asterisk/sounds/en/Randulo-allison.gsm
libdata/asterisk/sounds/en/SIP_Test_Failure.gsm
@@ -791,6 +791,8 @@
libdata/asterisk/sounds/en/confbridge-begin-glorious-b.gsm
libdata/asterisk/sounds/en/confbridge-begin-glorious-c.gsm
libdata/asterisk/sounds/en/confbridge-begin-leader.gsm
+libdata/asterisk/sounds/en/confbridge-binaural-off.gsm
+libdata/asterisk/sounds/en/confbridge-binaural-on.gsm
libdata/asterisk/sounds/en/confbridge-conf-begin.gsm
libdata/asterisk/sounds/en/confbridge-conf-end.gsm
libdata/asterisk/sounds/en/confbridge-dec-list-vol-in.gsm
diff -r 42efee77e3e0 -r 1e02f961988d comms/asterisk14/distinfo
--- a/comms/asterisk14/distinfo Wed Jan 24 00:46:30 2018 +0000
+++ b/comms/asterisk14/distinfo Wed Jan 24 05:51:40 2018 +0000
@@ -1,18 +1,18 @@
-$NetBSD: distinfo,v 1.6 2017/06/21 13:33:47 jnemeth Exp $
+$NetBSD: distinfo,v 1.7 2018/01/24 05:51:40 jnemeth Exp $
-SHA1 (asterisk-14.5.0/asterisk-14.5.0.tar.gz) = 9cba1c356293db67bcc3685bb8f1f9fd21e321f0
-RMD160 (asterisk-14.5.0/asterisk-14.5.0.tar.gz) = 59b19305f1c64d55a91ec25d893a02e99c48fdfe
-SHA512 (asterisk-14.5.0/asterisk-14.5.0.tar.gz) = 04dbea932900ecd3218629b2f19d20ad544cd7c02014fb4bd659e638e4a068ba179e6a4400bed788316fd337102ed8290c95823304567f378f9626361fd18c5e
-Size (asterisk-14.5.0/asterisk-14.5.0.tar.gz) = 40730634 bytes
-SHA1 (asterisk-14.5.0/asterisk-extra-sounds-en-gsm-1.5.tar.gz) = 831ae6442e23cbef1e7d1c84798778ad0b0524d1
-RMD160 (asterisk-14.5.0/asterisk-extra-sounds-en-gsm-1.5.tar.gz) = d52df795201c53fc4cd7d99ed41516e312f6f0f3
-SHA512 (asterisk-14.5.0/asterisk-extra-sounds-en-gsm-1.5.tar.gz) = c7d3c3fd2c854e6776801312d34bf69bbed78a443c16121637f508c5275f18b1d415cbb6e4f6f8c5aa3769cbbfa1a11485b9972053777f3ac39256c2c81729f1
-Size (asterisk-14.5.0/asterisk-extra-sounds-en-gsm-1.5.tar.gz) = 4256538 bytes
+SHA1 (asterisk-14.7.5/asterisk-14.7.5.tar.gz) = b378be5598e76f2385298bab346bf489796cefa7
+RMD160 (asterisk-14.7.5/asterisk-14.7.5.tar.gz) = 1cd1ac72c758bebe54d7fdefe2a7fd59640d7863
+SHA512 (asterisk-14.7.5/asterisk-14.7.5.tar.gz) = e6ac50d116528aeb2d2f0ac05ce2d3f5c037b87926fffa0d958d34f02957f13c8a01894c40d7a20ad52d3f3b929f3521a7969e19f485f19bef1d53e8d5390c81
+Size (asterisk-14.7.5/asterisk-14.7.5.tar.gz) = 40819648 bytes
+SHA1 (asterisk-14.7.5/asterisk-extra-sounds-en-gsm-1.5.1.tar.gz) = 8bd05d42d45454b642f1d2e598e00e2189747846
+RMD160 (asterisk-14.7.5/asterisk-extra-sounds-en-gsm-1.5.1.tar.gz) = 2320f0c9b884c1d7e80003668fbae03cf4495842
+SHA512 (asterisk-14.7.5/asterisk-extra-sounds-en-gsm-1.5.1.tar.gz) = 6da96ecf9fb2051fd7efc1c5f9b346f6ec7b31d06b7008e0612c869984a3212141ec981132ddd55215339e04c6c27b48d8b3737bd1fa974bffd628a0505212b4
+Size (asterisk-14.7.5/asterisk-extra-sounds-en-gsm-1.5.1.tar.gz) = 4254022 bytes
SHA1 (patch-Makefile) = 8e6c47cabfc2dffcfd8c5a5d2eb0c76e864a5519
SHA1 (patch-addons_chan__ooh323.c) = 9cba619ced6a4449604faebeac33d91a23519c48
SHA1 (patch-apps_app__dumpchan.c) = 127ac02bdc180ad2334cd095aa6e646feb6fba10
SHA1 (patch-apps_app__followme.c) = c6a5790b5e9b34d07dbfdd66a58e2854c8c72695
-SHA1 (patch-apps_app__queue.c) = ac673bf85469a26d72317b05ffaaa63f15e96987
+SHA1 (patch-apps_app__queue.c) = a2378ed32b1ad68bf6473f30a57e23326ef59802
SHA1 (patch-apps_app__sms.c) = ae81daf6ccf8c8fdf2251dba305e137bb9ab6b05
SHA1 (patch-apps_app__voicemail.c) = ee46ffd64a15ef79fc568edd3d5eb68cd86865f7
SHA1 (patch-build__tools_mkpkgconfig) = 7fab8fcf46d9f8a3b98455674fec6307ec472b23
diff -r 42efee77e3e0 -r 1e02f961988d comms/asterisk14/patches/patch-apps_app__queue.c
--- a/comms/asterisk14/patches/patch-apps_app__queue.c Wed Jan 24 00:46:30 2018 +0000
+++ b/comms/asterisk14/patches/patch-apps_app__queue.c Wed Jan 24 05:51:40 2018 +0000
@@ -1,8 +1,8 @@
-$NetBSD: patch-apps_app__queue.c,v 1.2 2017/06/21 13:33:48 jnemeth Exp $
+$NetBSD: patch-apps_app__queue.c,v 1.3 2018/01/24 05:51:40 jnemeth Exp $
---- apps/app_queue.c.orig 2017-05-30 17:50:46.000000000 +0000
+--- apps/app_queue.c.orig 2017-12-22 22:26:07.000000000 +0000
+++ apps/app_queue.c
-@@ -5447,7 +5447,7 @@ static int wait_our_turn(struct queue_en
+@@ -5439,7 +5439,7 @@ static int wait_our_turn(struct queue_en
if ((status = get_member_status(qe->parent, qe->max_penalty, qe->min_penalty, qe->parent->leavewhenempty, 0))) {
*reason = QUEUE_LEAVEEMPTY;
@@ -11,29 +11,29 @@
res = -1;
qe->handled = -1;
break;
-@@ -6824,8 +6824,8 @@ static int try_calling(struct queue_ent
+@@ -6819,8 +6819,8 @@ static int try_calling(struct queue_ent
/* if setinterfacevar is defined, make member variables available to the channel */
/* use pbx_builtin_setvar to set a load of variables with one call */
- if (qe->parent->setinterfacevar) {
-- snprintf(interfacevar, sizeof(interfacevar), "MEMBERINTERFACE=%s,MEMBERNAME=%s,MEMBERCALLS=%d,MEMBERLASTCALL=%ld,MEMBERPENALTY=%d,MEMBERDYNAMIC=%d,MEMBERREALTIME=%d",
+ if (qe->parent->setinterfacevar && interfacevar) {
+- ast_str_set(&interfacevar, 0, "MEMBERINTERFACE=%s,MEMBERNAME=%s,MEMBERCALLS=%d,MEMBERLASTCALL=%ld,MEMBERPENALTY=%d,MEMBERDYNAMIC=%d,MEMBERREALTIME=%d",
- member->interface, member->membername, member->calls, (long)member->lastcall, member->penalty, member->dynamic, member->realtime);
-+ snprintf(interfacevar, sizeof(interfacevar), "MEMBERINTERFACE=%s,MEMBERNAME=%s,MEMBERCALLS=%d,MEMBERLASTCALL=%jd,MEMBERPENALTY=%d,MEMBERDYNAMIC=%d,MEMBERREALTIME=%d",
++ ast_str_set(&interfacevar, 0, "MEMBERINTERFACE=%s,MEMBERNAME=%s,MEMBERCALLS=%d,MEMBERLASTCALL=%jd,MEMBERPENALTY=%d,MEMBERDYNAMIC=%d,MEMBERREALTIME=%d",
+ member->interface, member->membername, member->calls, (intmax_t)member->lastcall, member->penalty, member->dynamic, member->realtime);
- pbx_builtin_setvar_multiple(qe->chan, interfacevar);
- pbx_builtin_setvar_multiple(peer, interfacevar);
+ pbx_builtin_setvar_multiple(qe->chan, ast_str_buffer(interfacevar));
+ pbx_builtin_setvar_multiple(peer, ast_str_buffer(interfacevar));
}
-@@ -6833,8 +6833,8 @@ static int try_calling(struct queue_ent
+@@ -6828,8 +6828,8 @@ static int try_calling(struct queue_ent
/* if setqueueentryvar is defined, make queue entry (i.e. the caller) variables available to the channel */
/* use pbx_builtin_setvar to set a load of variables with one call */
- if (qe->parent->setqueueentryvar) {
-- snprintf(interfacevar, sizeof(interfacevar), "QEHOLDTIME=%ld,QEORIGINALPOS=%d",
+ if (qe->parent->setqueueentryvar && interfacevar) {
+- ast_str_set(&interfacevar, 0, "QEHOLDTIME=%ld,QEORIGINALPOS=%d",
- (long) (time(NULL) - qe->start), qe->opos);
-+ snprintf(interfacevar, sizeof(interfacevar), "QEHOLDTIME=%jd,QEORIGINALPOS=%d",
++ ast_str_set(&interfacevar, 0, "QEHOLDTIME=%jd,QEORIGINALPOS=%d",
+ (intmax_t) (time(NULL) - qe->start), qe->opos);
- pbx_builtin_setvar_multiple(qe->chan, interfacevar);
- pbx_builtin_setvar_multiple(peer, interfacevar);
+ pbx_builtin_setvar_multiple(qe->chan, ast_str_buffer(interfacevar));
+ pbx_builtin_setvar_multiple(peer, ast_str_buffer(interfacevar));
}
-@@ -8063,8 +8063,8 @@ static int queue_exec(struct ast_channel
+@@ -8043,8 +8043,8 @@ static int queue_exec(struct ast_channel
}
}
@@ -44,7 +44,7 @@
qe.chan = chan;
qe.prio = prio;
-@@ -8114,8 +8114,8 @@ check_turns:
+@@ -8094,8 +8094,8 @@ check_turns:
record_abandoned(&qe);
reason = QUEUE_TIMEOUT;
res = 0;
@@ -55,7 +55,7 @@
break;
}
-@@ -8160,7 +8160,7 @@ check_turns:
+@@ -8142,7 +8142,7 @@ check_turns:
if ((status = get_member_status(qe.parent, qe.max_penalty, qe.min_penalty, qe.parent->leavewhenempty, 0))) {
record_abandoned(&qe);
reason = QUEUE_LEAVEEMPTY;
@@ -64,7 +64,7 @@
res = 0;
break;
}
-@@ -8183,7 +8183,7 @@ check_turns:
+@@ -8165,7 +8165,7 @@ check_turns:
record_abandoned(&qe);
reason = QUEUE_TIMEOUT;
res = 0;
@@ -73,7 +73,7 @@
break;
}
-@@ -8211,8 +8211,8 @@ stop:
+@@ -8193,8 +8193,8 @@ stop:
if (!qe.handled) {
record_abandoned(&qe);
ast_queue_log(args.queuename, ast_channel_uniqueid(chan), "NONE", "ABANDON",
@@ -84,7 +84,7 @@
res = -1;
} else if (qcontinue) {
reason = QUEUE_CONTINUE;
-@@ -8220,7 +8220,7 @@ stop:
+@@ -8205,7 +8205,7 @@ stop:
}
} else if (qe.valid_digits) {
ast_queue_log(args.queuename, ast_channel_uniqueid(chan), "NONE", "EXITWITHKEY",
@@ -93,7 +93,7 @@
}
}
-@@ -9465,9 +9465,9 @@ static char *__queues_show(struct manses
+@@ -9445,9 +9445,9 @@ static char *__queues_show(struct manses
do_print(s, fd, " Callers: ");
for (qe = q->head; qe; qe = qe->next) {
@@ -106,12 +106,21 @@
do_print(s, fd, ast_str_buffer(out));
}
}
-@@ -9837,7 +9837,7 @@ static int manager_queues_status(struct
+@@ -9817,7 +9817,7 @@ static int manager_queues_status(struct
"CallerIDName: %s\r\n"
"ConnectedLineNum: %s\r\n"
"ConnectedLineName: %s\r\n"
- "Wait: %ld\r\n"
+ "Wait: %jd\r\n"
+ "Priority: %d\r\n"
"%s"
"\r\n",
- q->name, pos++, ast_channel_name(qe->chan), ast_channel_uniqueid(qe->chan),
+@@ -9826,7 +9826,7 @@ static int manager_queues_status(struct
+ S_COR(ast_channel_caller(qe->chan)->id.name.valid, ast_channel_caller(qe->chan)->id.name.str, "unknown"),
+ S_COR(ast_channel_connected(qe->chan)->id.number.valid, ast_channel_connected(qe->chan)->id.number.str, "unknown"),
+ S_COR(ast_channel_connected(qe->chan)->id.name.valid, ast_channel_connected(qe->chan)->id.name.str, "unknown"),
+- (long) (now - qe->start), qe->prio, idText);
++ (intmax_t) (now - qe->start), qe->prio, idText);
+ ++q_items;
+ }
+ }
Home |
Main Index |
Thread Index |
Old Index