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2020-09-08*: use MASTER_SITE_PERL_CPANwiz2-4/+4
2020-09-06p5-Device-Modem: update to 1.59.wiz2-8/+7
1.59 Mon Jun 15 08:17:54 CEST 2020 - Merged pull request #6 from ghciv6/fix_multi_instance_log fixed log handling with multi instances and typo in close(). Thanks to @ghciv6 ! 1.58 - Updated test suite a bit. - Added the tests to the manifest. - Got rid of indirect object syntax. - Moved test.pl to the actual test suite. - Updated $VERSION declarations according to: http://www.dagolden.com/index.php/369/version-numbers-should-be-boring/ - Added some extra tests (xt/author, xt/release). - Fixed some spelling.
2020-08-31*: bump PKGREVISION for perl-5.32.wiz19-38/+38
2020-08-18*: revbump for libsndfileleot1-2/+2
2020-08-18synce-libsynce: fix buildtnn2-1/+22
2020-08-17*: revbump after fontconfig bl3 changes (libuuid removal)leot13-24/+26
2020-08-13asterisk16: Update to 16.12.0ryoon4-35/+53
Changelog: Bugs fixed in this release: ----------------------------------- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco Kroon) [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2 (Reported by Jared Smith) [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an INVITE. (Reported by Frederic LE FOLL) [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed environment. (Reported by Università di Bologna - CESIA VoIP) [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf (Reported by Marin Odrljin) [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill Kervaski) [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching (Reported by Joshua C. Colp) [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua C. Colp) [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale lastcall time) (Reported by Walter Doekes) [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported by Walter Doekes) [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes) [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id parameter (Reported by sungtae kim) [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) [ASTERISK-28940] - /channels/create doesn't get any parameters from the body (Reported by sungtae kim) [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by Walter Doekes) [ASTERISK-28900] - res_fax: Double frame free when gateway in use with off-nominal format usage (Reported by Gregory Massel) [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by Alexander Traud) [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by nappsoft) [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) Improvements made in this release: ----------------------------------- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set (Reported by sungtae kim) [ASTERISK-28958] - Continue reading string when ping received by websocket (Reported by Nickolay V. Shmyrev) [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin Harwell) [ASTERISK-28949] - res_http_websocket: Add masking to websocket client (Reported by Moises Silva) [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by Kevin Harwell)
2020-07-30kermit: add a more detailed patch commentgutteridge2-3/+7
2020-07-30kermit: fix compilation on Linux with glibc >= 2.28gutteridge2-6/+29
Fix taken from the upstream project's 9.0.305 Alpha.01 release, noted to be a temporary workaround. (Separately, from how I read the change log, there has been no stable 9.0 release since 9.0.302.) Tested on Debian 9.13 (which has an older version of glibc which wouldn't reproduce the issue) and Fedora 31 & 32. (This issue was reported on pkgsrc-users back in July 2019 by Pierre Dupond, and I'd provided a workaround for it in that email chain, but I'd never actually committed anything to pkgsrc.)
2020-07-21py-esptool: updated to 2.8adam3-22/+24
Version 2.8 Features esptool.py image_info now prints a summary of segment memory types (IRAM, DRAM, etc) based on the address range. esptool.py write_flash will warn if it looks like a bootloader binary is built for ESP32-S2 or another newer chip (support for flashing ESP32-S2 will be added in a future version.) Bug Fixes Removed ESP8266 SDK & ESP-IDF dependencies when building the flasher stub binaries. Previously the SDKs were used to include some register address macros, only. This removes any uncertainty about whether the flasher stub binary is a derived work of either SDK. The flasher stub binary itself is the same as the binary in v2.7. Fixed minor issues running esptool automated tests on macOS. Minor flake8 fixes including compatibility with newer flake8 versions. ESP32 Only Features Support detection of new ESP32 silicon revisions New esptool.py elf2image --min-rev X option allows creating a .bin file which only supports a minimum ESP32 silicon revision. Bugfixes Fix burning custom MAC with espefuse.py when 3/4 Coding Scheme is set
2020-06-12asterisk16: Update to 16.11.0ryoon2-20/+19
Changelog: Bugs fixed in this release: ----------------------------------- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body (Reported by sungtae kim) [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by nappsoft) [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim) [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when transport=transport-udp6 (Reported by Peter Sokolov) [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not configured. (Reported by Alexander Traud) [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec (Reported by nappsoft) [ASTERISK-28870] - streams: One memory leak and one issue cloning streams (Reported by George Joseph) [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call (Reported by lvl) [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output (Reported by Etienne Lessard) [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling (Reported by Shlomi Gutman) [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua C. Colp) [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud) Improvements made in this release: ----------------------------------- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by Joshua C. Colp) [ASTERISK-28896] - ari: Add support for specifying variables on channel create (Reported by Joshua C. Colp) [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported by Guido Falsi) [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms (Reported by Guido Falsi) [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi) [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio (Reported by Peter Turczak)
2020-06-11efax-gtk: Update to 3.2.15ryoon2-8/+7
Changelog: Version 3.2.15 (3rd June 2020) -------------- Fix build for gcc-10 (efax/efaxlib.h, efax/efaxlib.c, efax/Makefile.am, efax/Makefile.in). Version 3.2.14 (6th March 2020) -------------- Remove X11 specific code to allow the program to run better against wayland compositors (acinclude.m4, configure.ac; dialogs.cpp, helpfile.cpp, logger.cpp, main.cpp, mainwindow.cpp, prog_defs.h; src/Makefile.am). Fix label layout in settings dialog (settings.cpp). Apply SO_REUSEADDR option when constructing sockets (socket_server.cpp). Deal with strict aliasing warning (efax/efaxos.c).
2020-06-02Revbump for icuadam14-28/+28
2020-05-31comms/asterisk16: remove unknow configure optionrillig1-3/+1
2020-05-22revbump after updating security/nettleadam5-10/+10
2020-05-21comms/asterisk15: remove unknown configure option --with-ltdlrillig1-3/+1
This option has been removed in 2018, see ChangeLog.
2020-05-21(comms/obexapp) Build fix: Remove obexapp.1 obexapp.h from SUBST_FILES.pathsmef1-2/+2
2020-05-20mark packages that fail with -Werror=char-subscriptsrillig2-2/+16
These packages are susceptible to bugs when confronted with non-ASCII characters. See https://gcc.gnu.org/bugzilla/show_bug.cgi?id=94182. It takes some time to analyze and fix these individually, therefore they are only marked as "needs work".
2020-05-17g/c references to openjdk7tnn1-2/+2
2020-05-14Fix compare of pointer and NUL constant. Allow newer libtiff. Bumpjoerg4-5/+28
revision.
2020-05-07repair build break, apply -Wno-error=incompatible-pointer-typesplunky2-4/+10
2020-05-06revbump after boost updateadam5-8/+10
2020-05-05asterisk14: updated to 14.7.8adam3-14/+39
asterisk 14.7.8: * AST-2018-009: Fix crash processing websocket HTTP Upgrade requests The HTTP request processing in res_http_websocket allocates additional space on the stack for various headers received during an Upgrade request. An attacker could send a specially crafted request that causes this code to overflow the stack, resulting in a crash. * No longer allocate memory from the stack in a loop to parse the header values. NOTE: There is a slight API change when using the passed in strings as is. We now require the passed in strings to no longer have leading or trailing whitespace. This isn't a problem as the only callers have already done this before passing the strings to the affected function. asterisk 14.7.7: * AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses. When endpoint specific ACL rules block a SIP request they respond with a 403 forbidden. However, if an endpoint is not identified then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints. * Made endpoint specific ACL rules now respond with a 401 unauthorized which is the same as if an endpoint were not identified. The fix is accomplished by replacing the found endpoint with the artificial endpoint which always fails authentication. asterisk 14.7.6: * AST-2018-003: Crash with an invalid SDP fmtp attribute pjproject's fmtp retrieval function failed to catch invalid fmtp attributes. Because of this Asterisk would crash if given an SDP with an invalid fmtp attribute. When retrieving the format this patch now makes sure the fmtp attribute is available. If not available it now returns an error status. * AST-2018-002: Crash with an invalid SDP media format description pjproject's media format parsing algorithm failed to catch invalid values. Because of this Asterisk would crash if given an SDP with a invalid media format description. When parsing the media format description this patch now properly parses the value and returns an error status if it can't successfully parse/convert the value. * AST-2018-005: res_pjsip_transport_management: Move to core Since res_pjsip_transport_management provides several attack mitigation features, its functionality moved to res_pjsip and this module has been removed. This way the features will always be available if res_pjsip is loaded. * AST-2018-005: Fix tdata leaks when calling pjsip_endpt_send_response(2) pjsip_distributor: authenticate() creates a tdata and uses it to send a challenge or failure response. When pjsip_endpt_send_response2() succeeds, it automatically decrements the tdata ref count but when it fails, it doesn't. Since we weren't checking for a return status, we weren't decrementing the count ourselves on error and were therefore leaking tdatas. res_pjsip_session: session_reinvite_on_rx_request wasn't decrementing the ref count if an error happened while sending a 491 response. pre_session_setup wasn't decrementing the ref count if while sending an error after a pjsip_inv_verify_request failure. res_pjsip: ast_sip_send_response wasn't decrementing the ref count on error. * AST-2018-005: Add a check for NULL tdata in ast_sip_failover_request It was discovered that there are some corner cases where a pjsip tsx might have no last_tx so calling ast_sip_failover_request with a NULL last_tx as its tdata would cause a crash. * AST-2018-004: Restrict the number of Accept headers in a SUBSCRIBE. When receiving a SUBSCRIBE request the Accept headers from it are stored locally. This operation has a fixed limit of 32 Accept headers but this limit was not enforced. As a result it was possible for memory outside of the allocated space to get written to resulting in a crash. This change enforces the limit so only 32 Accept headers are processed.
2020-05-05srtp: updated to 2.3.0adam17-457/+75
libsrtp 2.3.0 Major changes in this release are a fuzzer for libsrtp, NSS as optional crypto back end and cmake support for building. For more details and a complete list of changes please see the CHANGES file. libsrtp 2.2.0 First release in the 2.2 series. The major change with this release is that the all the code has been reformatted to be consistent and this consistency can be enforced with the include .clang-format file. This resulted in a lot of none functional changes but was considered worth it to simplify maintenance in the future. There are numerous other minor fixes, see the CHANGES file for more details. libsrtp 2.1.0 First release in the 2.1 series. libsrtp 2.0.0 Initial libsrtp 2.0 release.
2020-05-03remserial: link network libs on SunOSwiedi1-1/+3
2020-05-01asterisk16: Update to 16.10.0ryoon4-29/+29
Changelog: 16.10.0: New Features made in this release: ----------------------------------- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on Linux (Reported by Matt Addison) Bugs fixed in this release: ----------------------------------- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C. Colp) [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters (Reported by sungtae kim) [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio (Reported by Alexey Vasilyev) [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander Traud) [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for a user (Reported by Joshua C. Colp) [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by Alexander Traud) [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK (Reported by nappsoft) [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets (Reported by Joshua Roys) [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK (Reported by nappsoft) [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol) [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option (Reported by Patrick Wakano) [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed only. (Reported by Alexander Traud) [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. (Reported by Alexander Traud) [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported by Alexander Traud) [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet (Reported by Kevin Harwell) [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect. (Reported by Alexander Traud) [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers. (Reported by Alexander Traud) [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by Alexander Traud) [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield, channame from dialplan (Reported by Sébastien Duthil) [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction. (Reported by Alexander Traud) [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with clang. (Reported by Alexander Traud) [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang. (Reported by Alexander Traud) [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR (Reported by Jason Hord) [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported by Kevin Harwell) [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load' (Reported by Alexander Traud) [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load' (Reported by Alexander Traud) [ASTERISK-28790] - Crash during conference call using confbridge and video (Reported by Pascal Cadotte Michaud) [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is running as root (Reported by Jaco Kroon) [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number (Reported by Jaco Kroon) [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream (Reported by Joshua C. Colp) [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by sungtae kim) [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have reflected state (Reported by Joshua C. Colp) [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but not used (Reported by Torrey Searle) [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when there is a SSRC change (Reported by Paulo Vicentini) [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold) [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject files (Reported by Jared Smith) [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string (Reported by George Joseph) Improvements made in this release: ----------------------------------- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi) [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis - Prescom) [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't (Reported by Joshua Elson) [ASTERISK-28782] - Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add video (Reported by Joshua C. Colp) 16.9.0: Bugs fixed in this release: ----------------------------------- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding() (Reported by lvl) [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper) [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp) [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph) [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup (Reported by Kevin Harwell) [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN (Reported by Ross Beer) [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes (Reported by Joshua C. Colp) [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should return 503 (Reported by Walter Doekes) [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues (Reported by EDV O-TON) [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by Sébastien Duthil) [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer) [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be changed (Reported by Alex) [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer) [ASTERISK-28679] - stasis application is destroyed after its creation (Reported by Francois Blackburn) [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected (Reported by Peter Sokolov) Improvements made in this release: ----------------------------------- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh) [ASTERISK-28733] - stream: Add support for adding/removing streams during SFU/calls (Reported by Joshua C. Colp) [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor (Reported by xrobau) [ASTERISK-28726] - install_prereq script uses the interactive mode when installing aptitude (Reported by Sylvain Afchain) 16.8.0: New Features made in this release: ----------------------------------- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything (Reported by candrews) [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on source port (Reported by Sean Bright) Bugs fixed in this release: ----------------------------------- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported by Francois Blackburn) [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer) [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer) [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by Maciej Michno) [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40 (Reported by Andrew Siplas) [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output (Reported by Sean Bright) [ASTERISK-24484] - Update documentation for statsd module - usage requirements unclear (Reported by Dan Jenkins) [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM (Reported by Kevin Flyn) [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan (Reported by Frank Matano) [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used (Reported by Stas Kobzar) [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present instead of just one (Reported by AvayaXAsterisk) [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to return error and hangup (Reported by Corey Farrell) [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M. Lee) [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve XML version number (Reported by Bryan Nelson) [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens) [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) [ASTERISK-28562] - SIP WSS message not processed until next frame arrives (Reported by Robert Sutton) [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax (Reported by Richard Kenner) [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by Boris P. Korzun) [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP (Reported by Jean Aunis - Prescom) [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order mark is present (Reported by Robin Leffmann) [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by Pascal Cadotte Michaud) [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph) [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft) [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config option (Reported by Kevin Harwell) [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR. (Reported by Frederic LE FOLL) [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation (Reported by Pascal Cadotte Michaud) [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G) [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections (Reported by Jaco Kroon) [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response Contact (Reported by Ross Beer) [ASTERISK-28625] - Playback of local files impacted by large media cache (Reported by Kevin Reeves) Improvements made in this release: ----------------------------------- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp)
2020-04-26comms/pilot-link: fix unknown configure optionsrillig1-5/+2
2020-04-12Recursive revision bump after textproc/icu updateadam10-20/+20
2020-04-09comms/kermit: forgot to commit distinfo.rhialto1-3/+5
2020-04-08comms/kermit: this patch should be added too.rhialto1-0/+31
2020-04-08comms/kermit: Adapt patches to openssl 1.1.1e.rhialto4-13/+560
Parts are inspired by the FreeBSD port. I could not easily find a telnetd with SSL support so I did not really test it. Without SSL/TLS, it disconnects from NetBSD's telnetd if telnetd is run with "-a valid" ("Authentication failed: No authentication method available"); but "telnetd -a none" works.
2020-03-22asterisk16: fix L§inux packaging issuestnn4-9/+17
2020-03-22comms/py-serial: fix nonexistent files in SUBST blockrillig1-24/+5
2020-03-22asterisk16: configure asks for -ledit. Comply.tnn1-1/+2
2020-03-22asterisk-sounds-native: removetnn4-764/+0
This package has been redundant since asterisk 1.4. Source/explanation: https://www.voip-info.org/asterisk-native-sounds/
2020-03-22asterisk-sounds-native: removetnn2-3/+3
This package has been redundant since asterisk 1.4. Source/explanation: https://www.voip-info.org/asterisk-native-sounds/
2020-03-20*: Convert broken sourceforge HOMEPAGEs back to httpnia4-8/+8
2020-03-20asterisk-sounds-native: adjust workaround for missing x-bit on directoriestnn1-2/+2
2020-03-15lirc: add missing includetnn2-2/+16
2020-03-10librsvg: update bl3.mk to remove libcroco in rust casewiz13-26/+26
recursive bump for the dependency change
2020-03-08*: recursive bump for libffiwiz28-48/+56
2020-03-06comms: Remove libopensync-plugin-evolution2nia8-113/+1
Fails to build against current evolution and upstream site appears to be dead (parking page)?
2020-02-18Fix YYDEBUG usage.joerg2-3/+29
2020-02-15Add comms/remserial 1.4manu6-1/+116
The remserial program acts as a communications bridge between a TCP/IP network port and a Linux device such as a serial port. Any character-oriented Linux /dev device will work. The program can also use pseudo-ttys as the device. A pseudo-tty is like a serial port in that it has a /dev entry that can be opened by a program that expects a serial port device, except that instead of belonging to a physical serial device, the data can be intercepted by another program. The remserial program uses this to connect a network port to the "master" (programming) side of the pseudo-tty allowing the device driver (slave) side to be used by some program expecting a serial port. See example 3 below for details. The program can operate as a server accepting network connections from other machines, or as a client, connecting to remote machine that is running the remserial program or some other program that accepts a raw network connection. The network connection passes data as-is, there is no control protocol over the network socket. Multiple copies of the program can run on the same computer at the same time assuming each is using a different network port and device.
2020-02-15Make sure power is enabled on startup. Useful for D-Link DWM-157manu4-4/+28
Submitted upstream as https://github.com/gammu/gammu/pull/516
2020-01-27asterisk-sounds-native: pkglint cleanupwiz2-6/+5
2020-01-27comms/asterisk16: Check for clang correctlygdt1-2/+2
(This is a simple pkglint autofix, testing for clang being in PKGSRC_COMPILER, rather than equal to, avoiding failure with ccache/distcc.)
2020-01-27comms/asterisk-sounds-native: Add EOL-ish caution to DESCRgdt1-0/+3
2020-01-26all: migrate homepages from http to httpsrillig9-18/+18
pkglint -r --network --only "migrate" As a side-effect of migrating the homepages, pkglint also fixed a few indentations in unrelated lines. These and the new homepages have been checked manually.
2020-01-26all: migrate some SourceForge homepage URLs back from https to httprillig6-12/+12
https://mail-index.netbsd.org/pkgsrc-changes/2020/01/18/msg205146.html In the above commit, the homepage URLs were migrated from http to https, assuming that SourceForge would use the same host names for both http and https connections. This assumption was wrong. Their documentation at https://sourceforge.net/p/forge/documentation/Custom%20VHOSTs/ states that the https URLs use the domain sourceforge.io instead. To make the homepages from the above commit reachable again, pkglint has been extended to check for reachable homepages. This check is only enabled when the --network command line option is given. Each of the homepages that referred to https://$project.sourceforge.net before was migrated to https://$project.sourceforge.io (27), and if that was not reachable, to the fallback URL http://$project.sourceforge.net (163).
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-24 04:27:52 +0000