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authorjnemeth <jnemeth>2016-09-23 17:50:19 +0000
committerjnemeth <jnemeth>2016-09-23 17:50:19 +0000
commitef32b2d15a5edb5cdebc7818efcede5ceaec82ca (patch)
tree734cfe283a3324387af116868f08eaf427d9a006 /comms/asterisk13/PLIST
parent091ef44cd950828a5153d4b41922d128dd9c6f49 (diff)
downloadpkgsrc-ef32b2d15a5edb5cdebc7818efcede5ceaec82ca.tar.gz
Update to Asterisk 13.11.2: this is mainly a bug fix release
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!
Diffstat (limited to 'comms/asterisk13/PLIST')
-rw-r--r--comms/asterisk13/PLIST6
1 files changed, 4 insertions, 2 deletions
diff --git a/comms/asterisk13/PLIST b/comms/asterisk13/PLIST
index 6f48feb3d8a..2f217405764 100644
--- a/comms/asterisk13/PLIST
+++ b/comms/asterisk13/PLIST
@@ -1,4 +1,4 @@
-@comment $NetBSD: PLIST,v 1.6 2016/07/24 06:35:50 jnemeth Exp $
+@comment $NetBSD: PLIST,v 1.7 2016/09/23 17:50:19 jnemeth Exp $
include/asterisk.h
include/asterisk/_private.h
include/asterisk/abstract_jb.h
@@ -298,8 +298,8 @@ lib/asterisk/modules/chan_bridge_media.so
lib/asterisk/modules/chan_iax2.so
${PLIST.mgcp}lib/asterisk/modules/chan_mgcp.so
${PLIST.jabber}lib/asterisk/modules/chan_motif.so
-lib/asterisk/modules/chan_rtp.so
lib/asterisk/modules/chan_oss.so
+lib/asterisk/modules/chan_rtp.so
lib/asterisk/modules/chan_sip.so
lib/asterisk/modules/chan_skinny.so
lib/asterisk/modules/chan_unistim.so
@@ -419,6 +419,8 @@ lib/asterisk/modules/res_format_attr_h263.so
lib/asterisk/modules/res_format_attr_h264.so
lib/asterisk/modules/res_format_attr_opus.so
lib/asterisk/modules/res_format_attr_silk.so
+lib/asterisk/modules/res_format_attr_siren14.so
+lib/asterisk/modules/res_format_attr_siren7.so
lib/asterisk/modules/res_format_attr_vp8.so
lib/asterisk/modules/res_hep.so
lib/asterisk/modules/res_hep_rtcp.so