head 1.6; access; symbols pkgsrc-2025Q2:1.5.0.66 pkgsrc-2025Q2-base:1.5 pkgsrc-2025Q1:1.5.0.64 pkgsrc-2025Q1-base:1.5 pkgsrc-2024Q4:1.5.0.62 pkgsrc-2024Q4-base:1.5 pkgsrc-2024Q3:1.5.0.60 pkgsrc-2024Q3-base:1.5 pkgsrc-2024Q2:1.5.0.58 pkgsrc-2024Q2-base:1.5 pkgsrc-2024Q1:1.5.0.56 pkgsrc-2024Q1-base:1.5 pkgsrc-2023Q4:1.5.0.54 pkgsrc-2023Q4-base:1.5 pkgsrc-2023Q3:1.5.0.52 pkgsrc-2023Q3-base:1.5 pkgsrc-2023Q2:1.5.0.50 pkgsrc-2023Q2-base:1.5 pkgsrc-2023Q1:1.5.0.48 pkgsrc-2023Q1-base:1.5 pkgsrc-2022Q4:1.5.0.46 pkgsrc-2022Q4-base:1.5 pkgsrc-2022Q3:1.5.0.44 pkgsrc-2022Q3-base:1.5 pkgsrc-2022Q2:1.5.0.42 pkgsrc-2022Q2-base:1.5 pkgsrc-2022Q1:1.5.0.40 pkgsrc-2022Q1-base:1.5 pkgsrc-2021Q4:1.5.0.38 pkgsrc-2021Q4-base:1.5 pkgsrc-2021Q3:1.5.0.36 pkgsrc-2021Q3-base:1.5 pkgsrc-2021Q2:1.5.0.34 pkgsrc-2021Q2-base:1.5 pkgsrc-2021Q1:1.5.0.32 pkgsrc-2021Q1-base:1.5 pkgsrc-2020Q4:1.5.0.30 pkgsrc-2020Q4-base:1.5 pkgsrc-2020Q3:1.5.0.28 pkgsrc-2020Q3-base:1.5 pkgsrc-2020Q2:1.5.0.24 pkgsrc-2020Q2-base:1.5 pkgsrc-2020Q1:1.5.0.4 pkgsrc-2020Q1-base:1.5 pkgsrc-2019Q4:1.5.0.26 pkgsrc-2019Q4-base:1.5 pkgsrc-2019Q3:1.5.0.22 pkgsrc-2019Q3-base:1.5 pkgsrc-2019Q2:1.5.0.20 pkgsrc-2019Q2-base:1.5 pkgsrc-2019Q1:1.5.0.18 pkgsrc-2019Q1-base:1.5 pkgsrc-2018Q4:1.5.0.16 pkgsrc-2018Q4-base:1.5 pkgsrc-2018Q3:1.5.0.14 pkgsrc-2018Q3-base:1.5 pkgsrc-2018Q2:1.5.0.12 pkgsrc-2018Q2-base:1.5 pkgsrc-2018Q1:1.5.0.10 pkgsrc-2018Q1-base:1.5 pkgsrc-2017Q4:1.5.0.8 pkgsrc-2017Q4-base:1.5 pkgsrc-2017Q3:1.5.0.6 pkgsrc-2017Q3-base:1.5 pkgsrc-2017Q2:1.5.0.2 pkgsrc-2017Q2-base:1.5 pkgsrc-2017Q1:1.4.0.6 pkgsrc-2017Q1-base:1.4 pkgsrc-2016Q4:1.4.0.4 pkgsrc-2016Q4-base:1.4 pkgsrc-2016Q3:1.4.0.2 pkgsrc-2016Q3-base:1.4 pkgsrc-2016Q2:1.3.0.2 pkgsrc-2016Q2-base:1.3 pkgsrc-2016Q1:1.1.1.1.0.4 pkgsrc-2016Q1-base:1.1.1.1 pkgsrc-2015Q4:1.1.1.1.0.2 pkgsrc-2015Q4-base:1.1.1.1 pkgsrc-base:1.1.1.1 TNF:1.1.1; locks; strict; comment @# @; 1.6 date 2025.07.09.07.16.57; author jnemeth; state dead; branches; next 1.5; commitid kLQuP0CZAP9SB22G; 1.5 date 2017.05.13.22.39.13; author jnemeth; state Exp; branches; next 1.4; commitid soIBzYcfbdo4rhRz; 1.4 date 2016.07.24.06.35.50; author jnemeth; state Exp; branches; next 1.3; commitid TkvH9ALlFgiYnxfz; 1.3 date 2016.06.09.04.41.49; author jnemeth; state Exp; branches; next 1.2; commitid l7iS7v2ifXPgiK9z; 1.2 date 2016.05.06.07.41.06; author jnemeth; state Exp; branches; next 1.1; commitid Af0yz0jKv0qeso5z; 1.1 date 2015.12.05.23.29.10; author jnemeth; state Exp; branches 1.1.1.1; next ; commitid HrxiuMojJLTuCOLy; 1.1.1.1 date 2015.12.05.23.29.10; author jnemeth; state Exp; branches; next ; commitid HrxiuMojJLTuCOLy; desc @@ 1.6 log @remove seriously outdated Asterisk 13 as per pkgsrc-users e-mail on May 19th @ text @$NetBSD: patch-main_Makefile,v 1.5 2017/05/13 22:39:13 jnemeth Exp $ --- main/Makefile.orig 2016-07-21 14:54:02.000000000 +0000 +++ main/Makefile @@@@ -96,6 +96,10 @@@@ ifeq ($(OSARCH),OpenBSD) AST_LIBS+=-lcrypto -lpthread -lm $(EDITLINE_LIB) endif +ifeq ($(OSARCH),dragonfly) + AST_LIBS+=-lpthread +endif + ifeq ($(OSARCH),SunOS) AST_LIBS+=-lpthread -ldl -lrt -lnsl -lsocket -lresolv ASTSSL_LIBS+=-L/opt/ssl/lib -L/usr/local/ssl/lib @@@@ -337,10 +341,10 @@@@ bininstall: $(INSTALL) -m 755 $(MAIN_TGT) "$(DESTDIR)$(ASTSBINDIR)/" ifeq ($(AST_ASTERISKSSL),yes) ifeq ($(findstring darwin,$(OSARCH)),) # not Darwin - $(INSTALL) -m 755 $(ASTSSL_LIB).$(ASTSSL_SO_VERSION) "$(DESTDIR)$(ASTLIBDIR)/" - $(LN) -sf $(ASTSSL_LIB).$(ASTSSL_SO_VERSION) "$(DESTDIR)$(ASTLIBDIR)/$(ASTSSL_LIB)" + $(INSTALL) -m 755 $(ASTSSL_LIB).$(ASTSSL_SO_VERSION) "$(DESTDIR)$(PREFIX)/lib/" + $(LN) -sf $(ASTSSL_LIB).$(ASTSSL_SO_VERSION) "$(DESTDIR)$(PREFIX)/lib/$(ASTSSL_LIB)" else # Darwin - $(INSTALL) -m 755 $(ASTSSL_LIB) "$(DESTDIR)$(ASTLIBDIR)/" + $(INSTALL) -m 755 $(ASTSSL_LIB) "$(DESTDIR)$(PREFIX)/lib/" endif endif ifeq ($(PJPROJECT_BUNDLED),yes) @ 1.5 log @Update to Asterisk 13.15.0. This is mostly a bug fix release with a few minor enhancements. 13.14.1 was released to fix AST-2017-001. ----- 13.15.0 The Asterisk Development Team would like to announce the release of Asterisk 13.15.0. The release of Asterisk 13.15.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release: *New Features made in this release:* ----------------------------------- - [ASTERISK-26878 ] - func_channel: Add ability to get the callid so dialplan has access to it. (Reported by Richard Mudgett) - [ASTERISK-26863 ] - res_pjsip: Add endpoint identification scheme based on a configured SIP header/value (Reported by Matt Jordan) - [ASTERISK-17428 ] - [patch] Allow "Comedian Mail" branding to be removed (Reported by John Covert) *Bugs fixed in this release:* ----------------------------------- - [ASTERISK-26851 ] - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) - [ASTERISK-26897 ] - chan_sip: Security vulnerability with client code header (Reported by Alex Villacs Lasso) - [ASTERISK-26916 ] - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) - [ASTERISK-26705 ] - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) - [ASTERISK-26850 ] - res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets (Reported by Max Norba) - [ASTERISK-26484 ] - res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument. (Reported by Vinod Dharashive) - [ASTERISK-26776 ] - res_pjsip_pubsub: Crash when generating xpidf content (Reported by Andrew Green) - [ASTERISK-26880 ] - Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled (Reported by Kirsty Tyerman) - [ASTERISK-26862 ] - app_queue: Queue stops calling members with local interface after forwarding in previous call (Reported by Robert Mordec) - [ASTERISK-26732 ] - res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome (Reported by Dan Jenkins) - [ASTERISK-26879 ] - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) - [ASTERISK-26867 ] - autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade). (Reported by Krzysztof Trempala) - [ASTERISK-26869 ] - res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension (Reported by Torrey Searle) - [ASTERISK-26668 ] - core: Malformed pattern matching extension (various factors) results in crash (Reported by xrobau) - [ASTERISK-26865 ] - chan_iax2: Reload of iax peer results in loss of host address/port (Reported by Richard Begg) - [ASTERISK-26872 ] - Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal) (Reported by Matt Jordan) - [ASTERISK-26717 ] - Document the fact that Asterisk HEP support only works with the PJSIP channel driver (Reported by Olivier Krief) - [ASTERISK-26643 ] - Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk (Reported by Roman Bedros) - [ASTERISK-25237 ] - stasis_cache.c:845 caching_topic_exec: - misleading ERROR message (Reported by Smirnov Aleksey) - [ASTERISK-26857 ] - chan_pjsip: Dialplan function race condition (Reported by Joshua Colp) - [ASTERISK-26841 ] - chan_sip: Call not cancelled after receiving a 422 response (Reported by Jean Aunis - Prescom) - [ASTERISK-26822 ] - pjsip/cli_commands: pjsip show channelstats shows wrong codec (Reported by Kevin Harwell) - [ASTERISK-26685 ] - res_pjsip: Crash when using IPv6 and Transport ws,wss (Reported by Michael Balen) - [ASTERISK-24562 ] - app_voicemail: Cannot set fromstring on a per-mailbox basis (Reported by Mark Scholten) - [ASTERISK-26598 ] - Saynumber is trying to get "and" from "digits/" subfolder (Reported by Jonathan Harris) - [ASTERISK-17067 ] - Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) - [ASTERISK-26796 ] - res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' (Reported by Jrgen H) - [ASTERISK-25628 ] - res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging (Reported by Dmitry Wagin) - [ASTERISK-26825 ] - pjsip.conf.sample: user_agent: still refers to branch 12 (Reported by Tzafrir Cohen) - [ASTERISK-26823 ] - PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist (Reported by Mark Michelson) - [ASTERISK-26623 ] - res_pjsip: Crash when calling PJSIPShowEndpoint (Reported by Jrgen H) - [ASTERISK-26808 ] - res_pjsip_outbound_registration doesn't know about network change events (Reported by George Joseph) - [ASTERISK-26313 ] - chan_sip : Asterisk restart seems to be required for changing encryption option (Reported by benasse) - [ASTERISK-26781 ] - bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio (Reported by Sean Bright) - [ASTERISK-26782 ] - res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication (Reported by Peter Sokolov) - [ASTERISK-26812 ] - [patch] Fix download_externals To Allow The Use Of curl Or wget (Reported by Michael L. Young) - [ASTERISK-18271 ] - Pattern matching with res_config_mysql extensions does not behave as expected (Reported by Charlie Smurthwaite) - [ASTERISK-26669 ] - PJSIP Segfault 13.13.1 (Bundled PJSIP) (Reported by Nic Colledge) - [ASTERISK-18731 ] - [patch] DUNDi weight parameter not processed correctly (Reported by Peter Racz) - [ASTERISK-26580 ] - [patch] Error during LDAP modify action when user unregisters (Reported by Nicholas John Koch) - [ASTERISK-26799 ] - res_pjsip: Using an auth object for inbound and outbound authentication fails. (Reported by Richard Mudgett) - [ASTERISK-26738 ] - Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c (Reported by Michael Maier) - [ASTERISK-25893 ] - Function vmauthenticate accesses uninitialized memory (Reported by Filip Jenicek) - [ASTERISK-26802 ] - [patch] Integrity Check Of PJSIP Download Fails (Reported by Michael L. Young) - [ASTERISK-15858 ] - [patch] Fix query with double backslash in string literals and stop log warnings (Reported by Humberto Figuera) - [ASTERISK-26057 ] - res_config_sqlite3 uses incorrect query - unnecessary escape (Reported by Stepan) - [ASTERISK-23457 ] - SQlite3: Realtime queue loading fails after PRAGMA query result (Reported by Scott Griepentrog) - [ASTERISK-26794 ] - http: Crash on Reload Only in ast_tcptls_server_start (Reported by Joshua Elson) - [ASTERISK-26714 ] - Phone default have not ringing on ARM (Reported by Igor Goncharovsky) - [ASTERISK-26696 ] - pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh (Reported by Zach R) - [ASTERISK-26756 ] - res_pjsip_mwi: Asterisk does not terminate MWI subscription (Reported by Carl Fortin) - [ASTERISK-26109 ] - Asterisk fails building with OpenSSL 1.1.0 (Reported by Tzafrir Cohen) - [ASTERISK-26723 ] - VoiceMailPlayMsg not playing messages via realtime (Reported by Ryan Rittgarn) - [ASTERISK-18286 ] - [patch] 'Silence' is truncated in Record() (Reported by var) - [ASTERISK-26248 ] - chan_pjsip: Error when calling PJSIP client with domain specified (Reported by Norbert Varga) - [ASTERISK-26788 ] - core: Protect flags during ast_waitfor (Reported by Joshua Colp) - [ASTERISK-26115 ] - pbx: AMI Originate ignore "failed" extension on call failure (Reported by Nasir Iqbal) - [ASTERISK-26785 ] - configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample (Reported by Torrey Searle) - [ASTERISK-26772 ] - Crash in srv.c on startup with pjsip (Reported by nappsoft) - [ASTERISK-26770 ] - res_stasis_device_state: Duplicate subscriptions when multiple received at same time (Reported by Joshua Colp) *Improvements made in this release:* ----------------------------------- - [ASTERISK-26864 ] - res_pjsip_session: Add support for overlap dialling (Reported by Richard Begg) - [ASTERISK-26846 ] - chan_sip: Add rtcp-mux support (Reported by Sean Bright) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.15.0 *Thank you for your continued support of Asterisk!* ----- 13.14.0 The Asterisk Development Team has announced the release of Asterisk 13.14.0. The release of Asterisk 13.14.0 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-26630 - Make logging PJPROJECT messages a bit easier (Reported by Richard Mudgett) Bugs fixed in this release: ----------------------------------- * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported by nappsoft) * ASTERISK-26704 - res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'. (Reported by Anthony Messina) * ASTERISK-21094 - MixMonitorMute mutes through stream if already slinear (e.g. Originate) (Reported by David Woolley) * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be hung up via ARI (Reported by Tom Pawelek) * ASTERISK-26632 - core: Possibility of a frame "imbalance" leading to stuck channels. (Reported by Mark Michelson) * ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor) (Reported by Dmitriy) * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect (Reported by Michael Maier) * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support for SRV (Reported by Joshua Colp) * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett) * ASTERISK-26740 - voicemail API test: uses varlibdir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact Headers Enabled (Reported by JoshE) * ASTERISK-26621 - app_queue: Queue application does not ring members with Local interface (Reported by Jonas Kellens) * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with MWI wasn't registered (Reported by Michael Kuron) * ASTERISK-25494 - build: GCC 5.1.x catches some new const, array bounds and missing paren issues (Reported by George Joseph) * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton) * ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose) * ASTERISK-26653 - pjproject_bundled doesn't verify already downloaded tarballs (Reported by George Joseph) * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed, setting up new calls (Reported by Walter Doekes) * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp line (Reported by Jørgen H) * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all aors (Reported by George Joseph) * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" (Reported by Juris Breicis) * ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems without IPv6 (Reported by Guido Falsi) * ASTERISK-24330 - Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@@@GLIBC_2.2' (Reported by Tzafrir Cohen) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Reported by Alexei Gradinari) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) Improvements made in this release: ----------------------------------- * ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions (Reported by Rusty Newton) * ASTERISK-26527 - Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav) * ASTERISK-26624 - res_calendar_caldav: Add support for gmail (Reported by Eduardo Scudeller Libardi) * ASTERISK-26562 - app_controlplayback: Transmit Silence on ControlPlayback pause (Reported by Mikheili Dautashvili) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.14.0 Thank you for your continued support of Asterisk! ----- @ text @d1 1 a1 1 $NetBSD: patch-main_Makefile,v 1.4 2016/07/24 06:35:50 jnemeth Exp $ @ 1.4 log @Update to Asterisk 13.10.0: this is mainly a bug fix release. The Asterisk Development Team has announced the release of Asterisk 13.10.0. The release of Asterisk 13.10.0 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-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port", "call_id" to contacts (Reported by Alexei Gradinari) * ASTERISK-25994 - [patch]res_pjsip: module load priority (Reported by Alexei Gradinari) * ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported by Alexei Gradinari) * ASTERISK-25835 - Authentication using 'Username' field from Digest (Reported by Ross Beer) * ASTERISK-25930 - PJSIP: disable multi domain to improve realtime performace (Reported by Alexei Gradinari) Bugs fixed in this release: ----------------------------------- * ASTERISK-26160 - pjsip: Updated->Reachable during qualify (Reported by Matt Jordan) * ASTERISK-26177 - func_odbc: Database handle is kept when it should be released (Reported by Leandro Dardini) * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request due to server timeout (Reported by Ross Beer) * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details (Reported by Corey Farrell) * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported by George Joseph) * ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim generates a compile error (Reported by George Joseph) * ASTERISK-26128 - Alembic scripts are failing (Reported by Mark Michelson) * ASTERISK-26139 - test_res_pjsip_scheduler: Compile failure if pjproject isn't installed in a system location (Reported by George Joseph) * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by Alexander Traud) * ASTERISK-26127 - res_pjsip_session: Crash due to race condition between res_pjsip_session unload and timer (Reported by Joshua Colp) * ASTERISK-26083 - ARI: Announcer channels staying around after playback to a bridge is finished (Reported by Per Jensen) * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported by Alexander Traud) * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '>' (Reported by Vasil Kolev) * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by Alexander Traud) * ASTERISK-25262 - Memory leak when a caller channel does multiple dials and CEL is enabled (Reported by Etienne Lessard) * ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels (Reported by Niklas Larsson) * ASTERISK-26096 - res_hep: Crash when configuration file is missing (Reported by Niklas Larsson) * ASTERISK-26089 - Invalid security events during boot using PJSIP Realtime (Reported by Scott Griepentrog) * ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by Ross Beer) * ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B. Davis) * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock) * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported by Alexander Traud) * ASTERISK-26070 - ari/channels: Creating a local channel without an originator adds all audio formats to it's capabilities (Reported by George Joseph) * ASTERISK-26078 - core: Memory leak in logging (Reported by Etienne Lessard) * ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered properly (Reported by Ross Beer) * ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work - documentation needs clarification for when read/write is possible (Reported by Private Name) * ASTERISK-25777 - data race in threadpool (Reported by Badalian Vyacheslav) * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files (Reported by Tzafrir Cohen) * ASTERISK-26029 - parking: ast_parking_park_call should return parking_space instead of parking_exten (Reported by Diederik de Groot) * ASTERISK-25938 - res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero. (Reported by Edwin Vandamme) * ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final response (Reported by Javier Riveros ) * ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown fields (Reported by Joshua Colp) * ASTERISK-24986 - keepalive INFO packages ignored by asterisk (Reported by Ilya Trikoz) * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early nosignal packet (Reported by George Joseph) * ASTERISK-26030 - call cut because of double Session-Expires header in re-invite after proxy authentication is required (Reported by George Joseph) * ASTERISK-25964 - Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight (Reported by Matt Jordan) * ASTERISK-26005 - res_pjsip: Multiple SIP messages are combined into 1 TCP packet (Reported by Ross Beer) * ASTERISK-25352 - res_hep_rtcp correlation_id is different then res_hep (Reported by Kevin Scott Adams) * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported by Tzafrir Cohen) * ASTERISK-26007 - res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon) * ASTERISK-25990 - PJSIP TLS registration should respect client_uri scheme when generating Contact URI (Reported by Sebastian Damm) * ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use source port in nonce verification (Reported by Mark Michelson) * ASTERISK-25993 - pjproject: Allow bundling to not require everything it does (Reported by Joshua Colp) * ASTERISK-25956 - Compilation error in conditionally compiled code in config_options.c (Reported by Chris Trobridge) * ASTERISK-25998 - file: Crash when using nativeformats (Reported by Joshua Colp) * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime (Reported by Ross Beer) * ASTERISK-25968 - pjproject_bundled: Configure and make need to be re-tested (Reported by George Joseph) * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is in the process of being recorded during reload (Reported by John Campbell) * ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by Dmitriy Serov) * ASTERISK-25963 - func_odbc requires reconnect checks for stale connections (Reported by Ross Beer) * ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crash when running test (Reported by Joshua Colp) * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive sometimes two calls (Reported by nik600) * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself (Reported by Jonathan R. Rose) * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) New Features made in this release: ----------------------------------- * ASTERISK-25904 - PJSIP: add contact.updated event (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-13.10.0 Thank you for your continued support of Asterisk! @ text @d1 1 a1 1 $NetBSD: patch-main_Makefile,v 1.3 2016/06/09 04:41:49 jnemeth Exp $ a29 11 @@@@ -365,9 +369,7 @@@@ endif ifneq ($(ASTPJ_LIB).$(ASTPJ_SO_VERSION),.) rm -f "$(DESTDIR)$(ASTLIBDIR)/$(ASTPJ_LIB).$(ASTPJ_SO_VERSION)" endif -ifneq ($(LDCONFIG),) - $(LDCONFIG) $(LDCONFIG_FLAGS) "$(DESTDIR)$(ASTLIBDIR)/" -endif + rm -f "$(DESTDIR)$(PREFIX)/lib/$(ASTSSL_LIB).$(ASTSSL_SO_VERSION)" clean:: rm -f asterisk libasteriskssl.o @ 1.3 log @Upgrade to Asterisk 13.9.1: this is a bugfix release. Note that since the package doesn't support PJSIP (yet), all reference to PJSIP bugs are not applicable. ----- 13.9.1 The Asterisk Development Team has announced the release of Asterisk 13.9.1. The release of Asterisk 13.9.1 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-26007 - res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.1 Thank you for your continued support of Asterisk! ----- 13.9.0 The Asterisk Development Team has announced the release of Asterisk 13.9.0. The release of Asterisk 13.9.0 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-25963 - func_odbc requires reconnect checks for stale connections (Reported by Ross Beer) * ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by Dmitriy Serov) * ASTERISK-25938 - res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero. (Reported by Edwin Vandamme) * ASTERISK-25927 - Removed option "registertrying" is still documented in sip.conf.sample (Reported by Etienne Lessard) * ASTERISK-25947 - Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object. (Reported by Richard Mudgett) * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis fails to get app name (Reported by John Bigelow) * ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed ConnectedLine information (Reported by George Joseph) * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP thread (Reported by Joshua Colp) * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events not raised (Reported by Joshua Colp) * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers table unless rt (Reported by Jaco Kroon) * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of app_queue.c (Reported by Sébastien Couture) * ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George Joseph) * ASTERISK-25707 - Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions (Reported by George Joseph) * ASTERISK-25123 - Bracketed IPv6 Contact header parameter unparsable with Asterisk/PJSIP (Reported by Anthony Messina) * ASTERISK-25874 - app_voicemail: Stack buffer overflow in test_voicemail_notify_endl (Reported by Badalian Vyacheslav) * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set (Reported by Jaco Kroon) * ASTERISK-25885 - res_pjsip: Race condition between adding contact and automatic expiration (Reported by Joshua Colp) * ASTERISK-25910 - pjproject: Via headers are not parsed when "received" contains an IPv6 address (Reported by George Joseph) * ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails (Reported by Harley Peters) * ASTERISK-25894 - [patch] webrtc video broken due to missing marker bits in RTP streams (Reported by Jacek Konieczny) * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk (Reported by Robert McGilvray) * ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj (Reported by Hans van Eijsden) * ASTERISK-25882 - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2) (Reported by Richard Mudgett) * ASTERISK-25867 - [patch] Video delay on app_echo (Reported by Jacek Konieczny) * ASTERISK-24605 - res_parking option parkeddynamic does not work with the core Features 'parkcall' (DTMF initiated parking) (Reported by Philip Correia) * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime (Reported by Ross Beer) * ASTERISK-24596 - Unclear how to use Park application with res_parking 'parkeddynamic' enabled. Documentation? (Reported by Philip Correia) * ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs (Reported by Taylor Hawkes) * ASTERISK-25825 - Crashes during shutdown when running CLI commands (Reported by Mark Michelson) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton) * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to data corruption (Reported by Gianluca Merlo) Improvements made in this release: ----------------------------------- * ASTERISK-25865 - Message-Account Missing From PJSIP MWI (Reported by Ross Beer) * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad de Wet) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0 Thank you for your continued support of Asterisk! @ text @d1 1 a1 1 $NetBSD: patch-main_Makefile,v 1.2 2016/05/06 07:41:06 jnemeth Exp $ d3 1 a3 1 --- main/Makefile.orig 2016-05-13 17:41:41.000000000 +0000 d16 1 a16 14 @@@@ -224,11 +228,7 @@@@ endif $(ASTSSL_LIB): $(ASTSSL_LIB).$(ASTSSL_SO_VERSION) $(ECHO_PREFIX) echo " [LN] $< -> $@@" - $(CMD_PREFIX) if [ -x "$(LDCONFIG)" ] ; then \ - $(LDCONFIG) $(LDCONFIG_FLAGS) . 2>/dev/null ;\ - else \ - $(LN) -sf $< $@@ ;\ - fi + $(CMD_PREFIX) $(LN) -sf $< $@@ else # Darwin ASTSSL_LIB:=libasteriskssl.dylib @@@@ -345,10 +345,10 @@@@ bininstall: d30 1 a30 1 @@@@ -373,9 +373,7 @@@@ endif @ 1.2 log @Update to Asterisk 13.8.2: this is mainly a bug fix release. It also contains fixes for AST-2016-004 and AST-2016-005. However, those issues only affected the pjsip module. Since Asterisk in pkgsrc doesn't (yet) use pjsip, it wasn't affected. ----- 13.8.2 The Asterisk Development Team has announced the release of Asterisk 13.8.2. The release of Asterisk 13.8.2 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-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events not raised (Reported by Joshua Colp) * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP thread (Reported by Joshua Colp) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.2 Thank you for your continued support of Asterisk! ----- 13.8.0 The Asterisk Development Team has announced the release of Asterisk 13.8.0. The release of Asterisk 13.8.0 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-24919 - res_pjsip_config_wizard: Ability to write contents to file (Reported by Ray Crumrine) * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel Journo) * ASTERISK-25480 - [patch]Add field PauseReason on QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena) Bugs fixed in this release: ----------------------------------- * ASTERISK-25849 - chan_pjsip: transfers with direct media sometimes drops audio (Reported by Kevin Harwell) * ASTERISK-25113 - install_prereq in Debian 8 without "standard system utilities" (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so (Reported by Sergio Medina Toledo) * ASTERISK-25023 - Deadlock in chan_sip in update_provisional_keepalive (Reported by Arnd Schmitter) * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local channel (Reported by Filip Frank) * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when separating multiple AORs (Reported by Mateusz Kowalski) * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into Stasis application. (Reported by Javier Riveros ) * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean Bright) * ASTERISK-25582 - Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt Jordan) * ASTERISK-25811 - Unable to delete object from sorcery cache (Reported by Ross Beer) * ASTERISK-25800 - [patch] Calculate talktime when is first call answered (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to PJSIP requirement (Reported by Gergely Dömsödi) * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub (Reported by Jacques Peacock) * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing OutboundSubscriptionDetail ami action (Reported by Kevin Harwell) * ASTERISK-25721 - [patch] res_phoneprov: memory leak and heap-use-after-free (Reported by Badalian Vyacheslav) * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes returns garbage (Reported by Etienne Lessard) * ASTERISK-25751 - res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp) * ASTERISK-25606 - Core dump when using transports in sorcery (Reported by Martin Moučka) * ASTERISK-20987 - non-admin users, who join muted conference are not being muted (Reported by hristo) * ASTERISK-25737 - res_pjsip_outbound_registration: line option not in Alembic (Reported by Joshua Colp) * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause ast_frdup crash (Reported by Walter Doekes) * ASTERISK-25742 - Secondary IFP Packets can result in accessing uninitialized pointers and a crash (Reported by Torrey Searle) * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST Vulnerability - Investigate vulnerability of HTTP server (Reported by Alex A. Welzl) * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with non-default timert1 (Reported by Alexander Traud) * ASTERISK-25702 - PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by Nic Colledge) * ASTERISK-25730 - build: make uninstall after make distclean tries to remove root (Reported by George Joseph) * ASTERISK-25725 - core: Incorrect XML documentation may result in weird behavior (Reported by Joshua Colp) * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in sip_sipredirect (Reported by Badalian Vyacheslav) * ASTERISK-25709 - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Reported by Mark Michelson) * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported by Badalian Vyacheslav) * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build script (Reported by Joshua Colp) * ASTERISK-25712 - Second call to already-on-call phone and Asterisk sends "Ready" (Reported by Richard Mudgett) * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow (Reported by Badalian Vyacheslav) * ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report incorrect values (Reported by Gianluca Merlo) * ASTERISK-25611 - core: threadpool thread_timeout_thrash unit test sporadically failing (Reported by Joshua Colp) * ASTERISK-24097 - Documentation - CHANNEL function help text missing 'linkedid' argument (Reported by Steven T. Wheeler) * ASTERISK-25700 - main/config: Clean config maps on shutdown. (Reported by Corey Farrell) * ASTERISK-25696 - bridge_basic: don't cache xferfailsound during a transfer (Reported by Kevin Harwell) * ASTERISK-25697 - bridge_basic: don't play an attended transfer fail sound after target hangs up (Reported by Kevin Harwell) * ASTERISK-25683 - res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG (Reported by yaron nahum) * ASTERISK-25686 - PJSIP: qualify_timeout is a double, database schema is an integer (Reported by Marcelo Terres) * ASTERISK-25690 - Hanging up when executing connected line sub does not cause hangup (Reported by Joshua Colp) * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh reload' cause a crash (Reported by Sean Bright) * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed (Reported by Olivier Krief) * ASTERISK-25637 - Multi homed server using wrong IP (Reported by Daniel Journo) * ASTERISK-25394 - pbx: Incorrect device and presence state when changing hint details (Reported by Joshua Colp) * ASTERISK-25640 - pbx: Deadlock on features reload and state change hint. (Reported by Krzysztof Trempala) * ASTERISK-25681 - devicestate: Engine thread is not shut down (Reported by Corey Farrell) * ASTERISK-25680 - manager: manager_channelvars is not cleaned at shutdown (Reported by Corey Farrell) * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by Corey Farrell) * ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by Daniel Journo) * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported by Corey Farrell) * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey Farrell) * ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by Mark Michelson) * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference (Reported by Corey Farrell) * ASTERISK-25647 - bug of cel_radius.c: wrong point of ADD_VENDOR_CODE (Reported by Aaron An) * ASTERISK-25317 - asterisk sends too many stun requests (Reported by Stefan Engström) * ASTERISK-25137 - endpoint stasis messages are delivered twice (Reported by Vitezslav Novy) * ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are sent for every status change (Reported by George Joseph) * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel (Reported by Dmitry Melekhov) * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade Brandon) * ASTERISK-25442 - using realtime (mysql) queue members are never updated in wait_our_turn function (app_queue.c) (Reported by Carlos Oliva) * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend caching (Reported by Joshua Colp) * ASTERISK-25601 - json: Audit reference usage and thread safety (Reported by Joshua Colp) * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by sungtae kim) Improvements made in this release: ----------------------------------- * ASTERISK-25495 - [patch] Prevent old-update packages on repository Debian systems (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps (Reported by Andrew Nagy) * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for Anonymous (Reported by Anthony Messina) * ASTERISK-24813 - asterisk.c: #if statement in listener() confuses code folding editors (Reported by Corey Farrell) * ASTERISK-25767 - [patch] Add check to configure for sanitizes (Reported by Badalian Vyacheslav) * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the core set (Reported by Rusty Newton) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0 Thank you for your continued support of Asterisk! @ text @d1 1 a1 1 $NetBSD: patch-main_Makefile,v 1.1.1.1 2015/12/05 23:29:10 jnemeth Exp $ d3 1 a3 1 --- main/Makefile.orig 2016-04-20 10:46:18.000000000 +0000 d20 6 a25 5 -ifneq ($(LDCONFIG),) - $(CMD_PREFIX) $(LDCONFIG) $(LDCONFIG_FLAGS) . 2>/dev/null -else $(CMD_PREFIX) $(LN) -sf $< $@@ -endif @ 1.1 log @Initial revision @ text @d1 1 a1 1 $NetBSD: patch-main_Makefile,v 1.2 2013/05/05 01:32:35 jnemeth Exp $ d3 1 a3 1 --- main/Makefile.orig 2015-10-09 21:48:48.000000000 +0000 d5 1 a5 1 @@@@ -93,6 +93,10 @@@@ ifeq ($(OSARCH),OpenBSD) d16 4 a19 4 @@@@ -218,9 +222,6 @@@@ ifeq ($(GNU_LD),1) endif $(ECHO_PREFIX) echo " [LD] $^ -> $@@" $(CMD_PREFIX) $(CC) $(STATIC_BUILD) -o $@@ $(CC_LDFLAGS_SO) $^ $(CC_LIBS) d21 3 a23 1 - $(LDCONFIG) $(LDCONFIG_FLAGS) . d26 3 a28 3 $(ASTSSL_LIB): $(ASTSSL_LIB).$(ASTSSL_SO_VERSION) $(LN) -sf $< $@@ @@@@ -260,13 +261,13 @@@@ bininstall: d37 2 a38 1 $(INSTALL) -m 755 $(ASTSSL_LIB) "$(DESTDIR)$(ASTLIBDIR)/" a39 3 ifneq ($(LDCONFIG),) - $(LDCONFIG) $(LDCONFIG_FLAGS) "$(DESTDIR)$(ASTLIBDIR)/" + $(LDCONFIG) $(LDCONFIG_FLAGS) "$(DESTDIR)$(PREFIX)/lib/" d41 4 a45 6 $(LN) -sf asterisk "$(DESTDIR)$(ASTSBINDIR)/rasterisk" @@@@ -274,10 +275,7 @@@@ endif binuninstall: rm -f "$(DESTDIR)$(ASTSBINDIR)/$(MAIN_TGT)" rm -f "$(DESTDIR)$(ASTSBINDIR)/rasterisk" - rm -f "$(DESTDIR)$(ASTLIBDIR)/$(ASTSSL_LIB).$(ASTSSL_SO_VERSION)" @ 1.1.1.1 log @`cat ../comment` @ text @@