summaryrefslogtreecommitdiff
path: root/comms
AgeCommit message (Collapse)AuthorFilesLines
2017-04-22Revbump after icu updateadam14-28/+28
2017-04-18Updated minicom to 2.7.1.wiz2-10/+9
New for version 2.7.1: - CVE-2017-7467: Fix an out of bounds data access that can lead to remote code execution. This issue was found by Solar Designer of Openwall during a security audit of the Virtuozzo 7 product, which contains derived downstream code in its prl-vzvncserver component. The corresponding Virtuozzo 7 fix is: https://src.openvz.org/projects/OVZ/repos/prl-vzvncserver/commits/6d95404e75b98f36b1cc85ee23df99dcf06ca13f Openwall would like to thank the Virtuozzo company for funding the effort.
2017-04-13Update DeforaOS Phone to version 0.5.1khorben3-11/+11
This release brings: - parameter database for mobile data access - additional USSD codes for T-Mobile (Germany) - build fixes
2017-04-04Updated py-colorama to 0.3.7.wiz2-7/+7
0.3.7 * Fix issue #84: check if stream has 'closed' attribute before testing it * Fix issue #74: objects might become None at exit 0.3.6 * Fix issue #81: fix ValueError when a closed stream was used 0.3.5 * Bumping version to re-upload a wheel distribution 0.3.4 * Fix issue #47 and #80 - stream redirection now strips ANSI codes on Linux * Fix issue #53 - strip readline markers * Fix issue #32 - assign orig_stdout and orig_stderr when initialising * Fix issue #57 - Fore.RESET did not reset style of LIGHT_EX colors. Fixed by Andy Neff * Fix issue #51 - add context manager syntax. Thanks to Matt Olsen. * Fix issue #48 - colorama didn't work on Windows when environment variable 'TERM' was set. * Fix issue #54 - fix pylint errors in client code. * Changes to readme and other improvements by Marc Abramowitz and Zearin 0.3.3 * Fix Google Code issue #13 - support changing the console title with OSC escape sequence * Fix Google Code issue #16 - Add support for Windows xterm emulators * Fix Google Code issue #30 - implement \033[nK (clear line) * Fix Google Code issue #49 - no need to adjust for scroll when new position is already relative (CSI n A\B\C\D) * Fix Google Code issue #55 - erase_data fails on Python 3.x * Fix Google Code issue #46 - win32.COORD definition missing * Implement \033[0J and \033[1J (clear screen options) * Fix default ANSI parameters * Fix position after \033[2J (clear screen) * Add command shortcuts: colorama.Cursor, colorama.ansi.set_title, colorama.ansi.clear_line, colorama.ansi.clear_screen * Fix issue #22 - Importing fails for python3 on Windows * Thanks to John Szakmeister for adding support for light colors * Thanks to Charles Merriam for adding documentation to demos
2017-03-31Recursive bump for gpgme update which removed a support library.wiz1-2/+2
2017-02-21Add an upper API version restriction.cherry1-2/+3
The current only user of this buildlink file is asterisk-chan-dongle (which is yet to be committed). With further users, comms/asterisk may need to find a version specific directory as newer versions are imported.
2017-02-17Don't define accept4 locally on new enough NetBSD current.joerg2-1/+22
2017-02-17Add missing includes.joerg3-9/+23
2017-02-12Recursive revbump from fonts/harfbuzzryoon15-30/+30
2017-02-10Add buildlink support.cherry1-0/+12
This will aid subsequent module builds
2017-02-10Um, need bsd.prefs.mk before testing ${OPSYS}.he1-3/+4
2017-02-10Don't enable the inet6 option on the various BSDs, since their stackhe2-4/+10
require separate inet6 and inet sockets, and conserver as of 8.2.1 doesn't do that. Bump PKGREVISION.
2017-02-06Recursive bump for harfbuzz's new graphite2 dependency.wiz15-32/+30
2017-01-19Convert all occurrences (353 by my count) ofagc5-19/+19
MASTER_SITES= site1 \ site2 style continuation lines to be simple repeated MASTER_SITES+= site1 MASTER_SITES+= site2 lines. As previewed on tech-pkg. With thanks to rillig for fixing pkglint accordingly.
2017-01-18Add two patches so that this at least semi-works when the inet6he6-8/+100
option is used: * Use correct sockaddr length when doing getnameinfo() for inet6, so we avoid an early return with "permanent failure" from getnameinfo() * Use temp variables for walking the address lists so that we avoid trying freeaddrinfo(NULL) and getting SEGV This still isn't fully baked and backward compatible: with the inet6 option turned on, on NetBSD the conserver process only opens an inet6 server socket and no longer serves an inet socket (a Linuxism, I suspect), making it troublesome to interoperate with older versions of conserver or installations on hosts without IPv6 connectivity. PKGREVISION bumped.
2017-01-01Revbump after boost updateadam5-8/+10
2017-01-01Add python-3.6 to incompatible versions.wiz2-4/+4
2016-12-12Revert "Specify readline requirement on 30 packages"wiz1-2/+1
Many of these definitely do not depend on readline. So there must be a different underlying problem, and that should be tracked down instead of papering over it.
2016-12-11Update to Asterisk 11.25.1: this fixes AST-2016-009.jnemeth2-12/+11
Asterisk Project Security Advisory - ASTERISK-2016-009 Product Asterisk Summary Nature of Advisory Authentication Bypass Susceptibility Remote unauthenticated sessions Severity Minor Exploits Known No Reported On October 3, 2016 Reported By Walter Doekes Posted On Last Updated On December 8, 2016 Advisory Contact Mmichelson AT digium DOT com CVE Name Description The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue l does not affect you. Resolution chan_sip has been patched to only treat spaces and horizontal tabs as whitespace following a header name. This allows for Asterisk and authenticating proxies to view requests the same way Affected Versions Product Release Series Asterisk Open Source 11.x All Releases Asterisk Open Source 13.x All Releases Asterisk Open Source 14.x All Releases Certified Asterisk 13.8 All Releases Corrected In Product Release Asterisk Open Source 11.25.1, 13.13.1, 14.2.1 Certified Asterisk 11.6-cert16, 13.8-cert4 Patches SVN URL Revision Links Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/ASTERISK-2016-009.pdf and http://downloads.digium.com/pub/security/ASTERISK-2016-009.html Revision History Date Editor Revisions Made November 28, 2016 Mark Michelson Initial writeup Asterisk Project Security Advisory - ASTERISK-2016-009 Copyright (c) 2016 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form.
2016-12-09Update comms/py-gammu to py-gammu-2.7leot2-7/+7
Changes: 2.7 === * Needs Gammu >= 1.37.90 due to API changes. 2.6 === * Fixed error when creating new contact. * Fixed possible testsuite errors.
2016-12-09Update comms/gammu to gammu-1.37.91leot4-15/+14
Changes: 20161023 - 1.37.91 [!] * Changed version of the shared library. [-] * Improved support for ZTE MF100. [-] * Ignore unsolicited +CLCC: reply. [-] * Correctly report when some SMSD SQL backend is not compiled in. [-] * Fix build of MySQL backend on Linux. 20161018 - 1.37.90 [-] * Improved support Huawei K3770. [!] * API changes in some parameter types. [-] * Fixed various Windows compilation issues. [-] * Fixed several resource leaks. [-] * Create outbox SMS atomically in FILES backend. [!] * Removed getlocation command as we no longer fit into their usage policy. [-] * Fixed call diverts on TP-LINK MA260. [+] * Initial support for Oracle database. [!] * Removed unused daemons, pbk and pbk_groups tables from the SMSD schema. [+] * SMSD outbox entries now can have priority set in the database. [+] * Added SIM IMSI to the SMSD status table. [+] * Added CheckNetwork directive. [+] * SMSD attempts to power on radio if disabled. [-] * Fixed processing of AT unsolicited responses in some cases. [-] * Fixed parsing USSD responses from some devices. 20160816 - 1.37.4 [-] * Improved support for Huawei E3131. [-] * Fixed SMS support for MULTIBAND 900E. [-] * Fixed SMS created in text mode. 20160524 - 1.37.3 [-] * Improved support for Huawei E398. [-] * Improved support for Huawei/Vodafone K4505. [-] * Fixed possible crash if SMSD used in library. [-] * Improved support for Huawei E180. 20160413 - 1.37.2 [-] * Fixed compilation of SMSD. 20160413 - 1.37.1 [-] * Properly report errors in HEX encoded strings from SMSD SQL backends. [-] * Configurable SMSD table names. [-] * Improved support for Huawei E303. [-] * Improved support for Vodafone K4511. [-] * Improved support for Telit M2M modules.
2016-12-04Recursive revbump from textproc/icu 58.1ryoon14-25/+28
2016-12-04Specify readline requirement on 30 packagesmarino1-1/+2
Solves: /usr/libexec/binutils225/elf/ld.gold: error: cannot find -lreadline The missing specification is obvious on DragonFly because there's no publically accessible version of readline in base.
2016-12-03Correct the if statement to AND, not OR.sevan1-2/+2
Unbreak builds on FreeBSD & DragonFly BSD
2016-12-03Add dfu-util.sevan1-1/+2
2016-12-03Import dfu-util 0.9sevan4-0/+40
ok wiedi
2016-11-27Update to Asterisk 14.2.0: this is mostly a bugfix release with some minorjnemeth4-16/+22
improvements. pkgsrc change: adapt to new res_resolver_unbound module. The Asterisk Development Team has announced the release of Asterisk 14.2.0. The release of Asterisk 14.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Improvements made in this release: ----------------------------------- * ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan) * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton) Bugs fixed in this release: ----------------------------------- * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11) * ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett) * ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett) * ASTERISK-26556 - manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes (Reported by Michelle Dupuis) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph) * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason) * ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp) * ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit IPv6 transport configured (Reported by Joshua Colp) * ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli) * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov) * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan) * ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph) * ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp) * ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used (Reported by Frankie Chin) * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud) * ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c (Reported by Matt Krokosz) * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone) * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev) * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp) * ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard) * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav) * ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell) * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour) * ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters) * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp) * ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton) * ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel) * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud) * ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session (Reported by Alexei Gradinari) * ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients (Reported by Andrew Nagy) * ASTERISK-26444 - 'features show' command in CLI does not return prompt. (Reported by John Kiniston) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen) * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini) * ASTERISK-26439 - chan_rtp: Crash when originating (Reported by Kayode) * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud) * ASTERISK-26618 - build: Backport addition of librt check to configure.ac (Reported by Kevin Harwell) New Features made in this release: ----------------------------------- * ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge (Reported by Matt Jordan) * ASTERISK-26492 - ARI: Add ability to specify channel variables on websocket events (Reported by Mark Michelson) * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events (Reported by Matt Jordan) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0 Thank you for your continued support of Asterisk!
2016-11-27Update to Asterisk 13.13.0: this is mainly a bug fix release with somejnemeth3-15/+15
minor improvements. The Asterisk Development Team has announced the release of Asterisk 13.13.0. The release of Asterisk 13.13.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge (Reported by Matt Jordan) * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11) * ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett) * ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett) * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph) * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason) * ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp) * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov) * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan) * ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph) * ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli) * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud) * ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp) * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp) * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone) * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev) * ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard) * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav) * ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell) * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour) * ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters) * ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton) * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp) * ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel) * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud) * ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients (Reported by Andrew Nagy) * ASTERISK-26444 - 'features show' command in CLI does not return prompt. (Reported by John Kiniston) * ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session (Reported by Alexei Gradinari) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen) * ASTERISK-26439 - chan_rtp: Crash when originating (Reported by Kayode) * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini) * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud) * ASTERISK-26618 - build: Backport addition of librt check to configure.ac (Reported by Kevin Harwell) Improvements made in this release: ----------------------------------- * ASTERISK-25063 - [patch]add X.509 subject alternative name support to Asterisk TLS support (Reported by Maciej Szmigiero) * ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan) * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0 Thank you for your continued support of Asterisk!
2016-11-27Update to Asterisk 11.25.0: this is a bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 11.25.0. The release of Asterisk 11.25.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen) * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini) * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0 Thank you for your continued support of Asterisk!
2016-11-24Update doxygen-depend version to 1.8.12 (or add new BUILD_DEPENDS+)mef1-2/+2
2016-11-24Adjust PLIST for doxygen update 1.8.11 to 1.8.12, PKGREVISION++.mef2-5/+5
2016-11-11Update to Asterisk 14.1.2: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 14.1.2. The release of Asterisk 14.1.2 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.2 Thank you for your continued support of Asterisk!
2016-11-11Update the Asterisk 13.12.2: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 13.12.2. The release of Asterisk 13.12.2 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.2 Thank you for your continued support of Asterisk!
2016-10-29Update to Asterisk 13.12.1: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 13.12.1. The release of Asterisk 13.12.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.1 Thank you for your continued support of Asterisk!
2016-10-28Update to Asterisk 14.1.1: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 14.1.1. The release of Asterisk 14.1.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.1 Thank you for your continued support of Asterisk!
2016-10-28Update to Asterisk 11.24.1: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 11.24.1. The release of Asterisk 11.24.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1 Thank you for your continued support of Asterisk!
2016-10-27Update to Asterisk 14.1.0: this is mostly a bug fix release.jnemeth5-33/+20
The Asterisk Development Team has announced the release of Asterisk 14.1.0. The release of Asterisk 14.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-26277 - Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-26477 - pjproject: SEGV during SSL operations (Reported by George Joseph) * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. (Reported by effie mouzeli) * ASTERISK-26416 - pjproject-bundled: configure fails to check for all required utilities (Reported by Corey Farrell) * ASTERISK-26466 - core: Be forgiving on external callerid that may be flawed so we don't drop events (Reported by Richard Mudgett) * ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported by Carlos Chavez) * ASTERISK-26446 - app_dial: There's no way to override the hangupcause on unanswered channels (Reported by George Joseph) * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in the console or verbose when starting (Reported by Dan Jenkins) * ASTERISK-24311 - Populating database via Alembic fails when using same database for multiple schema sets (Reported by Dafi Ni) * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud) * ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell) * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek) * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows do_monitor lock (Reported by Barry Flanagan) * ASTERISK-26397 - manager: PresenceState action crashes Asterisk 14 (Reported by Andrew Nagy) * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by Joshua Colp) * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen) * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled (Reported by Anthony Messina) * ASTERISK-26391 - Consoles do not display verbose logger messages even when requested. (Reported by Marcelo Terres) * ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen) * ASTERISK-26365 - rtp: Offer with multiple payloads for same codec is incorrectly handled (Reported by Joshua Colp) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten for connectionful protocols (Reported by Joshua Colp) * ASTERISK-26367 - rtp: Timestamps broken when video frame is across multiple RTP packets (Reported by Joshua Colp) * ASTERISK-26375 - res_pjsip_transport_management: Log message states seconds, but time value is milliseconds (Reported by Joshua Colp) * ASTERISK-26364 - res_pjsip: Don't assume a request will have target addresses (Reported by Joshua Colp) * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett) * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes unexpected callerid (Reported by Kevin Harwell) * ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov) * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard) * ASTERISK-26264 - res_pjsip: Crash when applying ACL from non-existent endpoint (Reported by nappsoft) * ASTERISK-26341 - ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list (Reported by Matt Jordan) * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. (Reported by Etienne Lessard) * ASTERISK-26331 - Crash on "core show channeltype Surrogate" in ast_format_cap_get_names (Reported by CGI.NET) * ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft) * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock) * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard) * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard) * ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian 6 (Reported by George Joseph) * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud) * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard) * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert) * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett) * ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen) * ASTERISK-22374 - Finish mapping the sip.conf parameters to res_sip.conf parameters (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. (Reported by Ali Ghavidel) * ASTERISK-25472 - Swagger scripts are not replacing format variable in file brief (Reported by Corey Farrell) * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it (Reported by József Dudás) * ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite tests fail (Reported by Richard Mudgett) * ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. (Reported by Alexander Traud) * ASTERISK-25492 - ARI: Path parameters are case sensitive (Reported by Joshua Colp) * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via chan_pjsip (Reported by Ross Beer) * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell) * ASTERISK-26246 - Security: Privilege escalation by AMI adding dialplan extensions. (Reported by Richard Mudgett) * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell) * ASTERISK-26241 - res_pjsip: When using compact headers, rpid and pai are incorrectly generated (Reported by George Joseph) * ASTERISK-25797 - app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension (Reported by Etienne Lessard) * ASTERISK-26239 - res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname (Reported by George Joseph) * ASTERISK-26238 - res_pjsip: Empty global default_from_user causes crash (Reported by Joshua Colp) * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP 'identify_by' enum (Reported by Joshua Colp) * ASTERISK-26253 - sdp_srtp: libsrtp now a required dependency, shouldn't be (Reported by Ben Merrills) * ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade + indicate (Reported by Ross Beer) * ASTERISK-26183 - alembic: error when using sqlalchemy version 1.1.0b2 (Reported by Kevin Harwell) * ASTERISK-26283 - res_resolver_unbound: fails configure on older Ubuntu and CentOS (Reported by George Joseph) * ASTERISK-26280 - DNS lookups can block channel media paths (Reported by Mark Michelson) * ASTERISK-26278 - asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM. (Reported by Corey Farrell) * ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c (Reported by Richard Mudgett) * ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell) * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex for get all (Reported by Dmitry) * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets with IP6 (Reported by Alexander Traud) * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy) * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden) * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard Mudgett) Improvements made in this release: ----------------------------------- * ASTERISK-26409 - codec_opus: Update Asterisk to support the l translation codec. (Reported by Kevin Harwell) * ASTERISK-26289 - Announcer channels in ConfBridges cause inefficiencies (Reported by Mark Michelson) * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used (Reported by Alexei Gradinari) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.0 Thank you for your continued support of Asterisk!
2016-10-27Update to Asterisk 13.12.0: this is mostly a bug fix release.jnemeth4-20/+20
The Asterisk Development Team has announced the release of Asterisk 13.12.0. The release of Asterisk 13.12.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-26277 - Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-26477 - pjproject: SEGV during SSL operations (Reported by George Joseph) * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. (Reported by effie mouzeli) * ASTERISK-26416 - pjproject-bundled: configure fails to check for all required utilities (Reported by Corey Farrell) * ASTERISK-26466 - core: Be forgiving on external callerid that may be flawed so we don't drop events (Reported by Richard Mudgett) * ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported by Carlos Chavez) * ASTERISK-26446 - app_dial: There's no way to override the hangupcause on unanswered channels (Reported by George Joseph) * ASTERISK-24311 - Populating database via Alembic fails when using same database for multiple schema sets (Reported by Dafi Ni) * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud) * ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell) * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek) * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows do_monitor lock (Reported by Barry Flanagan) * ASTERISK-26397 - manager: PresenceState action crashes Asterisk 14 (Reported by Andrew Nagy) * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by Joshua Colp) * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen) * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled (Reported by Anthony Messina) * ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen) * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten for connectionful protocols (Reported by Joshua Colp) * ASTERISK-26367 - rtp: Timestamps broken when video frame is across multiple RTP packets (Reported by Joshua Colp) * ASTERISK-26375 - res_pjsip_transport_management: Log message states seconds, but time value is milliseconds (Reported by Joshua Colp) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett) * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes unexpected callerid (Reported by Kevin Harwell) * ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov) * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard) * ASTERISK-26264 - res_pjsip: Crash when applying ACL from non-existent endpoint (Reported by nappsoft) * ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen) * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. (Reported by Etienne Lessard) * ASTERISK-26331 - Crash on "core show channeltype Surrogate" in ast_format_cap_get_names (Reported by CGI.NET) * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock) * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard) * ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian 6 (Reported by George Joseph) * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud) * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard) * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard) * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert) * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett) * ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft) * ASTERISK-22374 - Finish mapping the sip.conf parameters to res_sip.conf parameters (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-25472 - Swagger scripts are not replacing format variable in file brief (Reported by Corey Farrell) * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. (Reported by Ali Ghavidel) * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it (Reported by József Dudás) * ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. (Reported by Alexander Traud) * ASTERISK-25492 - ARI: Path parameters are case sensitive (Reported by Joshua Colp) * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell) * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via chan_pjsip (Reported by Ross Beer) * ASTERISK-26246 - Security: Privilege escalation by AMI adding dialplan extensions. (Reported by Richard Mudgett) * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell) * ASTERISK-26241 - res_pjsip: When using compact headers, rpid and pai are incorrectly generated (Reported by George Joseph) * ASTERISK-26239 - res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname (Reported by George Joseph) * ASTERISK-26238 - res_pjsip: Empty global default_from_user causes crash (Reported by Joshua Colp) * ASTERISK-25797 - app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension (Reported by Etienne Lessard) * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP 'identify_by' enum (Reported by Joshua Colp) * ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade + indicate (Reported by Ross Beer) * ASTERISK-26183 - alembic: error when using sqlalchemy version 1.1.0b2 (Reported by Kevin Harwell) * ASTERISK-26280 - DNS lookups can block channel media paths (Reported by Mark Michelson) * ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c (Reported by Richard Mudgett) * ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell) * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex for get all (Reported by Dmitry) * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets with IP6 (Reported by Alexander Traud) * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy) * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden) Improvements made in this release: ----------------------------------- * ASTERISK-26409 - codec_opus: Update Asterisk to support the translation codec. (Reported by Kevin Harwell) * ASTERISK-26289 - Announcer channels in ConfBridges cause inefficiencies (Reported by Mark Michelson) * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used (Reported by Alexei Gradinari) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.0 Thank you for your continued support of Asterisk!
2016-10-26Update to Asterisk 11.24.0: this is a bug fix release.jnemeth3-12/+49
The Asterisk Development Team has announced the release of Asterisk 11.24.0. The release of Asterisk 11.24.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud) * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek) * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett) * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard) * ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen) * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock) * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard) * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard) * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud) * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard) * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert) * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett) * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have multiple native formats (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change) (Reported by Krzysztof Trempala) * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell) * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell) * ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell) * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy) * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard Mudgett) * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and attended transfer handling (Reported by Ben Smithurst) * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional code. (Reported by Corey Farrell) * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even on lost packets. (Reported by Alexander Traud) * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files (Reported by Tzafrir Cohen) * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by Richard Miller) * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported by Corey Farrell) * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance (Reported by Edwin Vandamme) * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by Alexander Traud) * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when building menuselect (Reported by Jeffrey Walton) * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and bugs (Reported by Alexei Gradinari) * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp) * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph) Improvements made in this release: ----------------------------------- * ASTERISK-26220 - Add support for noreturn function attributes. (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-11.24.0 Thank you for your continued support of Asterisk!
2016-10-25add and enable asterisk14jnemeth1-1/+2
2016-10-25 Initial import of Asterisk 14. It has been tested to compilejnemeth58-0/+6653
and run, but not a lot of functional testing. This does not have the new PJSIP, which will be coming in a followup commit. This also does not have the patches for compiling with Clang. For upgrading instructions, please see: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+14 ----- 14.0.0 ----- The Asterisk Development Team is pleased to announce the release of Asterisk 14.0.0. Asterisk 14 is the next major release series of Asterisk. It is a Standard Support release, similar to Asterisk 12. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions For important information regarding upgrading to Asterisk 14, please see the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+14 A short list of new features includes: * A complete overhaul of the core DNS support in Asterisk, including implementing full NAPTR and SRV support in the PJSIP stack via the libunbound library. * The ability to publish extension state to a SIP Subscription server, such as Kamailio. This includes the ability to automatically generate a hint in the dialplan based on device state changes using the new autohint setting. * Playback of media from a remote HTTP server via a URI is now supported by all dialplan applications and AGI. Media retrieved using a URI is cached in a media cache and re-used when possible. * When using ARI to manipulate media on a resource, a list of media resources can now be supplied. The media resources will be played back sequentially in the order that they are provided. * Channels created via ARI can now be created and handed off to Stasis for external control prior to performing the outbound dial. This enables applications to set additional state on the channel prior to dialing, as well as enabling certain early media scenarios. And much more! More information about the new features can be found on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Documentation A full list of all new features can also be found in the CHANGES file: https://github.com/asterisk/asterisk/blob/14/CHANGES For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.0.0 Thank you for your continued support of Asterisk! ----- 14.0.1 ----- The Asterisk Development Team has announced the release of Asterisk 14.0.1. The release of Asterisk 14.0.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Improvements made in this release: ----------------------------------- * ASTERISK-26409 - codec_opus: Update Asterisk to support the translation codec. (Reported by Kevin Harwell) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.1 Thank you for your continued support of Asterisk! ----- 14.0.2 ----- The Asterisk Development Team has announced the release of Asterisk 14.0.2. The release of Asterisk 14.0.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in the console or verbose when starting (Reported by Dan Jenkins) * ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell) * ASTERISK-26425 - download_externals: ignore xmlstarlet return code for optional element (Reported by Kevin Harwell) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.2 Thank you for your continued support of Asterisk!
2016-10-09Recursive bump for all users of pgsql now that the default is 95.wiz2-4/+4
2016-10-07Revbump post boost updateadam4-6/+8
2016-09-26srtp: do not conflict with builtin hmac in netbsd-7.99.x, use another namemaya3-1/+85
(local_hmac). Fixes build on NetBSD. Patch by Sérgio de Almeida Lenzi
2016-09-23Update to Asterisk 11.23.1: this is a security fix release to fixjnemeth5-54/+154
AST-2016-007. Note that on Oct. 25th, this branch of Asterisk will switch to security fixes, and one year later it will read end-of-life. pkgsrc changes: - don't use gethostbyname_r on NetBSD - eliminate conflict with new hmac(1) function on NetBSd ----- AST-2016-007 The overlap dialing feature in chan_sip allows chan_sip to report to a device that the number that has been dialed is incomplete and more digits are required. If this functionality is used with a device that has performed username/password authentication RTP resources are leaked. This occurs because the code fails to release the old RTP resources before allocating new ones in this scenario. If all resources are used then RTP port exhaustion will occur and no RTP sessions are able to be set up.
2016-09-23Update to Asterisk 13.11.2: this is mainly a bug fix releasejnemeth6-47/+158
including two security issues: AST-2016-006 and AST-2016-007. Note that AST-2016-006 only affected setups using PJSIP, which pkgsrc Asterisk does not. pkgsrc changes: - don't use gethostbyname_r on NetBSD - eliminte conflict with new hmac(1) function on NetBSD ----- AST-2016-006 Asterisk can be crashed remotely by sending an ACK to it from an endpoint username that Asterisk does not recognize. Most SIP request types result in an "artificial" endpoint being looked up, but ACKs bypass this lookup. The resulting NULL pointer results in a crash when attempting to determine if ACLs should be applied. This issue was introduced in the Asterisk 13.10 release and only affects that release. This issue only affects users using the PJSIP stack with Asterisk. Those users that use chan_sip are unaffected. ----- AST-2016-007 The overlap dialing feature in chan_sip allows chan_sip to report to a device that the number that has been dialed is incomplete and more digits are required. If this functionality is used with a device that has performed username/password authentication RTP resources are leaked. This occurs because the code fails to release the old RTP resources before allocating new ones in this scenario. If all resources are used then RTP port exhaustion will occur and no RTP sessions are able to be set up. ----- 13.11.2 The Asterisk Development Team has announced the release of Asterisk 13.11.2. The release of Asterisk 13.11.2 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.2 Thank you for your continued support of Asterisk! ----- 13.11.0 The Asterisk Development Team has announced the release of Asterisk 13.11.0. The release of Asterisk 13.11.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by Alexei Gradinari) Bugs fixed in this release: ----------------------------------- * ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft) * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard) * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden) * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard Mudgett) * ASTERISK-26227 - sqlalchemy error due to long identifier name (Reported by Mark Michelson) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26214 - Allow arbitrary time for fax detection to end on a channel (Reported by Richard Mudgett) * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and attended transfer handling (Reported by Ben Smithurst) * ASTERISK-26216 - res_fax: Deadlock when detect fax while channel executing Playback (Reported by Richard Mudgett) * ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and DTD in docs. (Reported by Alexander Traud) * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional code. (Reported by Corey Farrell) * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even on lost packets. (Reported by Alexander Traud) * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files (Reported by Tzafrir Cohen) * ASTERISK-26200 - [patch] res_pjsip_mwi: improve realtime performance - remove unneeded check on endpoint's contacts. (Reported by Alexei Gradinari) * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by Richard Miller) * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported by Corey Farrell) * ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb (Reported by Corey Farrell) * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance (Reported by Edwin Vandamme) * ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for ast_threadpool_serializer_group (Reported by Corey Farrell) * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by Alexander Traud) * ASTERISK-26160 - pjsip: Updated->Reachable during qualify (Reported by Matt Jordan) * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when building menuselect (Reported by Jeffrey Walton) * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and bugs (Reported by Alexei Gradinari) * ASTERISK-26177 - func_odbc: Database handle is kept when it should be released (Reported by Leandro Dardini) * ASTERISK-26184 - chan_sip: Reference leaks in error paths. (Reported by Corey Farrell) * ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged during duplicate replacement (Reported by Corey Farrell) * ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for reuse (Reported by Scott Griepentrog) * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp) * ASTERISK-26172 - res_sorcery_realtime: fix bug when successful sql UPDATE is treated as failed if there is no affected rows. (Reported by Alexei Gradinari) * ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered (Reported by Dmitriy Serov) * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request due to server timeout (Reported by Ross Beer) * ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by Alexei Gradinari) * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph) * ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13 (Reported by Daniel Denson) * ASTERISK-26326 - Crash when dialing MulticastRTP channel (Reported by George Joseph) Improvements made in this release: ----------------------------------- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by Corey Farrell) * ASTERISK-22131 - Update the make dependencies script to pull, build, and install the correct pjproject (Reported by Matt Jordan) * ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip (Reported by JoshE) * ASTERISK-26159 - res_hep: enabled by default and information sent to default address (Reported by Ross Beer) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.0 Thank you for your continued support of Asterisk!
2016-09-08Use PKGMANDIR.jperkin2-4/+4
2016-08-03Revbump after graphics/gd updateadam14-25/+28
2016-07-24Update to 0.8wen2-8/+7
No upstream changelog found.
2016-07-24Update to 1.61wen2-8/+7
Upstream changes: 1.61 Tue Jun 21 21:05:12 CEST 2016 - Fixed RT#115491, remove the use of the encodings pragma, now deprecated. - Plenty of style, test and functionality fixes contributed by Joel Maslak and Paul Cochrane, as part of the CPAN PR Challenge. Awesome job, thanks! - Amended the main module documentation to make it clear this module is in maintenance mode and hasn't seen any major development work in years.
generated by cgit v1.2.3 (git 2.39.1) at 2024-05-08 04:15:48 +0000