[svn-commits] bebuild: tag 11.0.0-beta2 r373257 - /tags/11.0.0-beta2/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Sep 20 14:21:31 CDT 2012
Author: bebuild
Date: Thu Sep 20 14:21:27 2012
New Revision: 373257
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=373257
Log:
Importing files for 11.0.0-beta2 release.
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tags/11.0.0-beta2/.lastclean (with props)
tags/11.0.0-beta2/.version (with props)
tags/11.0.0-beta2/ChangeLog (with props)
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+2012-09-20 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.0.0-beta2 Released.
+
+2012-09-20 18:59 +0000 [r373235-373240] Matthew Jordan <mjordan at digium.com>
+
+ * configs/extensions.conf.sample, CHANGES, apps/app_queue.c:
+ app_queue: Support an 'agent available' hint Sets INUSE when no
+ free agents, NOT_INUSE when an agent is free. modifes
+ handle_statechange() scan members loop to scan for a free agent
+ and updates the Queue:queuename_avial devstate. Previously exited
+ early if the member was found in the queue. Now Exits later when
+ both a member was found, and a free agent was found. alecdavis
+ (license 585) Reported by: Alec Davis Tested by: alecdavis
+ Review: https://reviewboard.asterisk.org/r/2121/ ~~~~ Support all
+ ways a member can be available for 'agent available' hints Alec's
+ patch in r373188 added the ability to subscribe to a hint for
+ when Queue members are available. This patch modifies the check
+ that determines when a Queue member is available by refactoring
+ the availability checks in num_available_members into a shared
+ function is_member_available. This should now handle the
+ ringinuse option, as well as device state values other than
+ AST_DEVICE_NOT_INUSE.
+
+ * res/res_rtp_asterisk.c, /: When processing RFC 2833 DTMF,
+ accomodate increasing timestamps in End events While endpoints
+ should not be changing the source timestamp between DTMF event
+ packets, the fact is there exists those endpoints that do exactly
+ that. To work around this, we absorb timestamps within the
+ expected re-transmit period. Note that this period only affects
+ End of Event packets, so it should not prevent the detection of
+ new DTMF digits that happen to arrive right on top of each other.
+ (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson
+ Tested by: mjordan, Vladimir Mikhelson Review:
+ https://reviewboard.asterisk.org/r/2124 ........ Merged revisions
+ 373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373237 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * configs/extensions.conf.sample, CHANGES, apps/app_queue.c: Add
+ queue monitoring hints This patch adds support for hints on a
+ queue. Hints can be added using the nomenclature 'Queue:name',
+ where name is the name of the queue being monitored. This nifty
+ feature was done by Alec Davis. Review:
+ https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis
+ Tested by: alecdavis patches: review1619.diff2 by alecdavis
+ (license 585)
+
+2012-09-20 18:18 +0000 [r373229] Joshua Colp <jcolp at digium.com>
+
+ * channels/sip/include/sip.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, channels/chan_sip.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ configs/sip.conf.sample, include/asterisk/rtp_engine.h: Add
+ support for DTLS-SRTP to res_rtp_asterisk and chan_sip. As
+ mentioned on the review for this, WebRTC has moved towards
+ choosing DTLS-SRTP as the mechanism for key exchange for SRTP.
+ This commit adds support for this but makes it available for
+ normal SIP clients as well. Testing has been done to ensure that
+ this introduces no regressions with existing behavior and also
+ that it functions as expected. Review:
+ https://reviewboard.asterisk.org/r/2113/
+
+2012-09-20 17:15 +0000 [r373220] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/features.h, main/channel.c,
+ apps/app_directed_pickup.c, funcs/func_channel.c,
+ main/features.c, include/asterisk/channel.h: Named call pickup
+ groups. Fixes, missing functionality, and improvements. *
+ ASTERISK-20383 Missing named call pickup group features:
+ CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
+ CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() -
+ Needs to also select from named pickup groups. * ASTERISK-20384
+ Using the pickupexten, the pickup channel selection could fail
+ even though there was a call it could have picked up. In a call
+ pickup race when there are multiple calls to pickup and two
+ extensions try to pickup a call, it is conceivable that the loser
+ will not pick up any call even though it could have picked up the
+ next oldest matching call. Regression because of the named call
+ pickup group feature. * See ASTERISK-20386 for the implementation
+ improvements. These are the changes in channel.c and channel.h. *
+ Fixed some locking issues in CHANNEL(). (closes issue
+ ASTERISK-20383) Reported by: rmudgett (closes issue
+ ASTERISK-20384) Reported by: rmudgett (closes issue
+ ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2112/
+
+2012-09-20 13:00 +0000 [r373211] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Correct handling of unknown SDP stream types
+ When the patch to handle arbitrary SDP stream arrangements went
+ into Asterisk, it also included an ability to transparently
+ decline unknown stream types. The scanf calls used were not
+ checked properly causing this part of the functionality to be
+ broken. (closes issue ASTERISK-20203)
+
+2012-09-18 20:14 +0000 [r373133] Sean Bright <sean at malleable.com>
+
+ * main/manager.c, /: Don't crash when passing a NULL message to
+ __astman_get_header. Before this commit, __astman_get_header
+ would blindly dereference the passed in 'struct message *' to
+ traverse the header list. There are cases, however, such as
+ '*CLI> sip qualify peer foo' where the message pointer is NULL,
+ so we need to check for that. ........ Merged revisions 373131
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373132 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-18 15:47 +0000 [r373119] dlee <dlee at localhost>:
+
+ * Makefile, include/asterisk/utils.h, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
+ -fnested-functions compile flag, if needed. In order to use
+ nested functions on some versions of GCC (e.g. GCC on OS X), the
+ -fnested-functions flag must be passed to the compiler. This
+ patch adds detection logic to ./configure to add the flag if
+ necessary. It also adds a comment to utils.h as to why the nested
+ function needs a prototype. (closes issue ASTERISK-20399)
+ Reported by: David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2102/
+
+2012-09-15 00:27 +0000 [r373107] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_ss7.c, /: Made companding law for SS7 calls only
+ determined by SS7 signaling type. For SS7, the companding law for
+ a call was chosen inconsistently depending upon ss7type (ITU vs
+ ANSI) and the DAHDI companding default (T1 vs E1). For incoming
+ calls, the companding law was determined by ss7type. For outgoing
+ calls, the companding law was determined by the DAHDI default.
+ With the wrong combination you would get A-law/u-law conflicts.
+ An A-law/u-law conflict sounds like bad static on the line. SS7
+ ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
+ noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
+ with T1 line: ok * Fix the companding law used to be determined
+ by the SS7 signaling type only. ........ Merged revisions 373090
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373101 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-14 19:50 +0000 [r373079] Matthew Jordan <mjordan at digium.com>
+
+ * main/tcptls.c, /, channels/chan_sip.c, main/libasteriskssl.c:
+ Resolve memory leaks in TLS initialization and TLS client
+ connections This patch resolves two sources of memory leaks when
+ using TLS in Asterisk: 1) It removes improper initialization (and
+ multiple re-initializations) of portions of the SSL library.
+ Asterisk calls SSL_library_init and SSL_load_error_strings during
+ SSL initialization; collectively this obviates the need for
+ calling any of the following during initialization or client
+ connection handling: * ERR_load_crypto_strings (handled by
+ SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
+ SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
+ SSL_library_init) 2) Failure to completely clean up all memory
+ allocated by Asterisk and by the SSL library for TLS clients.
+ This included not freeing the SSL_CTX object in the SIP channel
+ driver, as well as not clearing the error stack when the TLS
+ client exited. Note that these memory leaks were found by Thomas
+ Arimont, and this patch was essentially written by him with some
+ minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
+ Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
+ Arimont (license 5525) Review:
+ https://reviewboard.asterisk.org/r/2105 ........ Merged revisions
+ 373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373062 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-13 20:04 +0000 [r373029-373047] dlee <dlee at localhost>:
+
+ * main/Makefile: Fixed make clean when configured
+ --disable-asteriskssl
+
+ * main/channel.c, /, include/asterisk/channel.h: Fix timeouts for
+ ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass
+ its timeout to ast_waitfor_nandfds, expecting it to decrement the
+ timeout by however many milliseconds were waited. This is a
+ problem if it consistently waits less than 1ms. The timeout will
+ never be decremented, and we wait... FOREVER! This patch makes
+ ast_waitfordigit_full manage the timeout itself. It maintains the
+ previously undocumented behavior that negative timeouts wait
+ forever. (closes issue ASTERISK-20375) Reported by: Mark
+ Michelson Tested by: Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/2109/ ........ Merged
+ revisions 373024 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373025 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-12 20:53 +0000 [r372995] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_motif.c: Skip any non-content information when
+ looking for and handling content. This fixes a bug with Jitsi and
+ conference calling. Jitsi implements XEP-0298 which places some
+ conference-info information in the session-initiate request which
+ chan_motif did not expect to occur.
+
+2012-09-12 18:23 +0000 [r372984] Jonathan Rose <jrose at digium.com>
+
+ * res/res_xmpp.c: res_xmpp: Fix a segfault caused by bodyless
+ messages (closes issue ASTERISK-20361) Reported by: Noah
+ Engelberth Review: https://reviewboard.asterisk.org/r/2108/
+
+2012-09-12 15:19 +0000 [r372937] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Add channel name to a warning to make
+ debugging easier. The "autodestruct with owner in place" message
+ is typically indicative of a channel reference leak. Printing out
+ the name of the channel in the message may be helpful when trying
+ to debug the issue. ........ Merged revisions 372932 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372933 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-12 14:18 +0000 [r372930] dlee <dlee at localhost>:
+
+ * main/Makefile: Fixed r372696 when configured
+ --disable-asteriskssl; properly install libasteriskssl.dylib on
+ OS X. I didn't realize that libasteriskssl.c was still compiled,
+ even when you disable asteriskssl; it simple gets statically
+ linked into asterisk.
+
+2012-09-11 22:32 +0000 [r372917] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_local.c, /: chan_local: Switch from using a random
+ 4 digit hex identifier to unique id Changes chan_local channels
+ to use an 8 digit hex identifier generated atomically and
+ sequentially in order to eliminate the chance of having multiple
+ channels with the same name during high call volume situations.
+ (issue ASTERISK-20318) Reported by: Dan Cropp Review:
+ https://reviewboard.asterisk.org/r/2104/ ........ Merged
+ revisions 372902 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372916 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-11 21:15 +0000 [r372886-372888] Mark Michelson <mmichelson at digium.com>
+
+ * main/asterisk.c, /, include/asterisk/_private.h, main/message.c:
+ Fix inability to shutdown gracefully due to an unending channel
+ reference. message.c makes use of a special message queue channel
+ that exists in thread storage. This channel never goes away due
+ to the fact that the taskprocessor used by message.c does not get
+ shut down, meaning that it never ends the thread that stores the
+ channel. This patch fixes the problem by shutting down the
+ taskprocessor when Asterisk is shut down. In addition, the thread
+ storage has a destructor that will release the channel reference
+ when the taskprocessor is destroyed. (closes issue AST-937)
+ Reported by Jason Parker Patches: AST-937.patch uploaded by Mark
+ Michelson (License #5049) Tested by Jason Parker ........ Merged
+ revisions 372885 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/features.c: Fix bad channel application data reference.
+ When channels get bridged due to an AMI bridge action or a DTMF
+ attended transfer, the two channels that get bridged have their
+ application data pointing to the other channel's name. This means
+ that if one channel is hung up but the other moves on, it means
+ that the channel that moves on will have its application data
+ pointing at freed memory. (issue ASTERISK-20335) Reported by:
+ aragon ........ Merged revisions 372840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372841 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-11 17:16 +0000 [r372864] dlee <dlee at localhost>:
+
+ * Makefile, /: Corrects the astsbindir setting when installing the
+ sample asterisk.conf. (closes issue ASTERISK-20406) ........
+ Merged revisions 372863 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-10 20:59 +0000 [r372795-372806] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_iax2.c: Ensure iax2 debug output is displayed
+ when expected When IAX2 debug was changed from iax_showframe to
+ iax_outputframe, some instances were missed (or added afterward).
+ This was causing debug output to not be displayed when expected.
+ (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
+ John Covert ........ Merged revisions 372804 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372805 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_jingle.c, include/asterisk/doxygen/architecture.h,
+ main/devicestate.c, channels/chan_gtalk.c, res/res_jabber.c:
+ Deprecate chan_gtalk, chan_jingle, and res_jabber chan_gtalk,
+ chan_jingle, and res_jabber are now deprecated in favor of using
+ chan_motif and res_xmpp. They are a feature-equivalent
+ replacement and are written to be more easily maintainable.
+ (closes issue ASTERISK-20298) Review:
+ https://reviewboard.asterisk.org/r/2082/ Reported-by: Leif Madsen
+
+2012-09-10 19:19 +0000 [r372777] dlee <dlee at localhost>:
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Eliminate "type-punned
+ pointer" build warning. Removes "res_rtp_asterisk.c:706: warning:
+ dereferencing type-punned pointer will break strict-aliasing
+ rules" warning from the build on 32-bit platforms. The problem is
+ that 'size' was referenced aliased to both (pj_size_t *) and
+ (pj_ssize_t *). Now just make a copy of size that is the right
+ type so there isn't any pointer aliasing happening. It also adds
+ comments and asserts regarding what looks like an inappropriate
+ use of pj_sock_sendto, but is actually totally fine. (closes
+ issue ASTERISK-20368) Reported by: Shaun Ruffel Tested by:
+ Michael L. Young Patches:
+ 0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch
+ uploaded by Shaun Ruffel (license 5417) slightly modified by
+ David M. Lee.
+
+2012-09-10 18:50 +0000 [r372768] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Document that 'p' option will
+ continue in dialplan. (closes issue AST-991) Reported by John
+ Bigelow ........ Merged revisions 372765 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372767 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-10 18:37 +0000 [r372766] Kinsey Moore <kmoore at digium.com>
+
+ * /: Recorded merge of revisions 372764 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Warn on
+ CLI when UDPTL init fails This adds a CLI warning when a SDP
+ offer is rejected due to UDPTL initialization failure.
+ Previously, there was no indication of the reason for offer
+ rejection in this case. (closes issue ASTERISK-20357)
+ Reported-by: Francesco Usseglio Gaudi ........ Merged revisions
+ 372763 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-10 17:33 +0000 [r372754] Jonathan Rose <jrose at digium.com>
+
+ * main/channel.c, /: Masquerade: Retain parkinglot settings made by
+ CHANNEL function. Prior to this patch, the user would have a
+ parkinglot set on a channel that was parked and when the channel
+ was retrieved, any attempt by that channel to park would simply
+ use the default. This patch makes parkinglot values set in this
+ way be retained through the masquerade. (closes issue AST-990)
+ Reported by: Nick Huskinson Patches:
+ masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
+ (license 6182) ........ Merged revisions 372736 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372737 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-09 01:25 +0000 [r372711] Matthew Jordan <mjordan at digium.com>
+
+ * channels/sip/sdp_crypto.c, /: Only re-create an SRTP session when
+ needed In r356604, SRTP handling was fixed to accomodate multiple
+ crypto keys in an SDP offer and the ability to re-create an SRTP
+ session when the crypto keys changed. In certain circumstances -
+ most notably when a phone is put on hold after having been
+ bridged for a significant amount of time - the act of re-creating
+ the SRTP session causes problems for certain models of phones.
+ The patch committed in r356604 always re-created the SRTP session
+ regardless of whether or not the cryptographic keys changed.
+ Since this is technically not necessary, this patch modifies the
+ behavior to only re-create the SRTP session if Asterisk detects
+ that the remote key has changed. This allows models of phones
+ that do not handle the SRTP session changing to continue to work,
+ while also providing the behavior needed for those phones that do
+ re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported
+ by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
+ https://reviewboard.asterisk.org/r/2099 ........ Merged revisions
+ 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372710 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-08 05:51 +0000 [r372696] dlee <dlee at localhost>:
+
+ * /, main/Makefile: Recorded merge of revisions 372695 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add
+ OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c. Without
+ this flag, those files will compile with the system installed
+ OpenSSL headers (if they exist). This is a real bummer if a
+ different path was specified using --with-ssl= (closes issue
+ ASTERISK-20392) ........ Merged revisions 372682 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-07 23:07 +0000 [r372622-372657] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup().
+ (closes issue ASTERISK-20349) Reported by: Brent Eagles ........
+ Merged revisions 372655 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372656 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, funcs/func_math.c: Remove annoying unconditional debug message
+ from INC/DEC functions. (closes issue AST-1001) Reported by:
+ Guenther Kelleter ........ Merged revisions 372628 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372629 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_queue.c: Fix exception path typo in app_queue.c
+ try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy
+ Pepper Patches: fix-local-channel-locking.patch (license #6350)
+ patch uploaded by Jeremy Pepper ........ Merged revisions 372624
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 372625 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_voicemail.c, /: Fix VoicemailUserEntry event headers
+ ServerEmail and MailCommand reported values. The AMI action
+ VoicemailUsersList VoicemailUserEntry event headers ServerEmail
+ and MailCommand did not report the global values if they were not
+ overridden. The VoicemailUserEntry event header ServerEmail was
+ not populated with the global value if the voicemail user did not
+ override it. The VoicemailUserEntry event header MailCommand was
+ never populated with a value. * Removed unused struct ast_vm_user
+ member mailcmd[]. (closes issue AST-973) Reported by: John
+ Bigelow Tested by: rmudgett ........ Merged revisions 372620 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372621 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-07 21:04 +0000 [r372609-372611] dlee <dlee at localhost>:
+
+ * res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin,
+ res/pjproject/third_party/bin, res/pjproject/third_party/gsm/lib,
+ res/pjproject/lib, res/pjproject/pjlib/lib,
+ res/pjproject/third_party/gsm/bin, res/pjproject/pjnath/lib,
+ res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/lib,
+ res/pjproject/pjsip/bin, res/pjproject/pjsip-apps/bin,
+ res/pjproject/pjmedia/lib, res/pjproject/third_party/lib,
+ codecs/ilbc: svn:ignore cleanup. * pjproject bin and lib
+ directories should pretty much ignore everything * Ignore *.o in
+ codecs/ilbc
+
+ * res/Makefile: Fix parallel make for res_asterisk_rtp. Fixes a
+ build regression introduced in r369517 "Add support for
+ ICE/STUN/TURN in res_rtp_asterisk and chan_sip." [1]. [1]
+ http://svnview.digium.com/svn/asterisk?view=revision&revision=369517
+ When compiling asterisk in parallel like: $ make -j 10 It's
+ possible to get errors like the following:
+ .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing
+ separator. Stop. make[4]: *** [depend] Error 2 make[3]: *** [dep]
+ Error 1 make[2]: ***
+ [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a]
+ Error 2 make[3]: warning: jobserver unavailable: using -j1. Add
+ `+' to parent make rule. This is because the build system is
+ trying to build each of the libraries in pjproject in parallel.
+ Now the build will build pjproject in a single job and link the
+ results into res_asterisk_rtp. Parallel builds, on one test
+ system, saves ~1.5 minutes from a default Asterisk build: Single
+ job: $ git clean -fdx >/dev/null && time ( ./configure >/dev/null
+ 2>&1 && make >/dev/null 2>&1 ) real 2m34.529s user 1m41.810s sys
+ 0m15.970s Parallel make: $ git clean -fdx >/dev/null && time (
+ ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 ) real
+ 1m2.353s user 2m39.120s sys 0m18.850s (closes issue
+ ASTERISK-20362) Reported by: Shaun Ruffel Patches:
+ 0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch
+ uploaded by Shaun Ruffel (License #5417)
+
+2012-09-07 02:26 +0000 [r372531-372583] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_minivm.c: Free ast_str objects when temp file fails
+ to be created in MiniVM The previous commit (r372554) was from a
+ patch that was written before r366880, which ensured that ast_str
+ objects allocated in the sendmail routine were free'd in off
+ nominal paths. This commit frees the string objects in the off
+ nominal path introduced in r372554. (issue ASTERISK-17133)
+ Reported by: Tzafrir Cohen ........ Merged revisions 372581 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372582 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_minivm.c: Fix file descriptor leak and pointer scope
+ issue in MiniVM when sending mail When MiniVM sends an e-mail and
+ it has the volgain option set, it will spawn sox in a separate
+ process to handle the manipulation of the sound file. In doing
+ so, it creates a temporary file. There are two problems here: 1)
+ The file descriptor returned from mkstemp is leaked 2) The
+ finalfilename character pointer points to a buffer that loses
+ scope once volgain processing is finished. Note that in r316265,
+ Russell fixed some gcc warnings by using the return value of the
+ mkstemp call. A warning was placed in minivm that the file
+ descriptor was going to be leaked. This patch reverts that
+ change, as it handles the leak and 'uses' the file descriptor
+ returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
+ Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
+ Cohen (license #5035) ........ Merged revisions 372554 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372555 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_queue.c: Update QueueMemberStatus event documentation to
+ include member status values The Status: header in a
+ QueueMemberStatus event (and other QueueMember* events) is the
+ numeric value of the device state corresponding to that Queue
+ Member. As those values are not exactly obvious, listing them in
+ the documentation is useful. Matt Riddell reported this
+ indirectly through the wiki page. (closes issue ASTERISK-20243)
+ Reported by: Matt Riddell
+
+2012-09-06 22:12 +0000 [r372523] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/sig_pri.c: Fix loss of MOH on an ISDN channel when
+ parking a call for the second time. Using the AMI redirect action
+ to take an ISDN call out of a parking lot causes the MOH state to
+ get confused. The redirect action does not take the call off of
+ hold. When the call is subsequently parked again, the call no
+ longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on
+ repeated AST_CONTROL_HOLD frames if it is already in a state
+ where it is supposed to be sending MOH. The MOH may have been
+ stopped by other means. (Such as killing the generator.) This
+ simple fix is done rather than making the AMI redirect action
+ post an AST_CONTROL_UNHOLD unconditionally when it redirects a
+ channel and thus potentially breaking something with an
+ unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches:
+ jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by
+ rmudgett ........ Merged revisions 372521 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 372522 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 21:42 +0000 [r372519] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_queue.c: Ensure listed queues are not offered for
+ completion When using tab-completion for the list of queues on
+ "queue reset stats" or "queue reload
+ {all|members|parameters|rules}", the tab-completion listing for
+ further queues erroneously listed queues that had already been
+ added to the list. The tab-completion listing now only displays
+ queues that are not already in the list. (closes issue AST-963)
+ Reported-by: John Bigelow ........ Merged revisions 372517 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372518 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 18:55 +0000 [r372500] dsessions <dsessions at localhost>:
+
+ * channels/chan_sip.c, configs/res_ldap.conf.sample: LDAP Realtime
+ Peers Cannot Register Prior to 1.8, it was not necessary for an
+ explicit "type" to be set for an asterisk LDAP realtime peer. Now
+ the routine find_peer actually checks the type field during
+ registration and fails to find the peer if it is not set. The
+ attached patches make the realtime type equal whatever type is
+ being searched for if the type is 0 upon return from routine
+ build_peer. (closes issue ASTERISK-17222) Reported by: John
+ Covert Patch by: David Vossel Tested by: Darren Sessions Review:
+ https://reviewboard.asterisk.org/r/2095/
+
+2012-09-06 15:56 +0000 [r372473] Jonathan Rose <jrose at digium.com>
+
+ * /, UPGRADE-1.8.txt: chan_sip: Note change in behavior to how
+ directmediapermit/deny ACL works r366547 introduced a change to
+ the directmedia ACL for chan_sip which modified the behavior
+ significantly. Prior to the patch, this option would bridge peers
+ with directmedia if a peer's IP address matched its own
+ directmedia ACL. After that patch, the peer would check the
+ bridged peer's ACL instead. This change has been present since
+ 1.8.14.0. That patched failed to document the change in
+ Upgrade.txt, so this patch adds mention of that change to
+ UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)
+ ........ Merged revisions 372471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372472 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 14:30 +0000 [r372446] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_queue.c: Ensure "rules" is tab-completable for "queue
+ show" Previously, tabbing at the end of "queue show" produced a
+ list of available queues about which information could be shown,
+ but did not include an alternative command, "rules", to access
+ information about queue rules. The "rules" item should now be
+ shown in the list of tab-completable items. (closes issue
+ AST-958) Reported-by: John Bigelow ........ Merged revisions
+ 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372445 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 02:50 +0000 [r372392-372419] Matthew Jordan <mjordan at digium.com>
+
+ * /, pbx/pbx_dundi.c: Fix DUNDi message routing bug when
+ neighboring peer is unreachable Consider a scenario where DUNDi
+ peer PBX1 has two peers that are its neighbors, PBX2 and PBX3,
+ and where PBX2 and PBX3 are also neighbors. If the connection is
+ temporarily broken between PBX1 and PBX3, PBX1 should not include
+ PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER
+ message, as it cannot send messages to PBX3. If it does, PBX2
+ will assume that PBX3 already received the message and fail to
+ forward the message on to PBX3 itself. This patch fixes this by
+ only including peers in a DPDISCOVER message that are reachable
+ by the sending node. This includes all peers with an empty
+ address (00:00:00:00:00:00) and that are have been reached by a
+ qualify message. This patch also prevents attempting to qualify a
+ dynamic peer with an empty address until that peer registers.
+ (closes issue ASTERISK-19309) Reported by: Peter Racz patches:
+ dundi_routing.patch uploaded by Peter Racz (license 6290) The
+ patch uploaded by Peter was modified slightly for this commit.
+ ........ Merged revisions 372417 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372418 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_followme.c: Allow configured numbers for FollowMe to
+ be greater than 90 characters When parsing a 'number' defined in
+ followme.conf, FollowMe previously parsed the number in the
+ configuration file into a buffer with a length of 90 characters.
+ This can artificially limit some parallel dial scenarios. This
+ patch allows for numbers of any length to be defined in the
+ configuration file. Note that Clod Patry originally wrote a patch
+ to fix this problem and received a Ship It! on the JIRA issue.
+ The patch originally expanded the buffer to 256 characters.
+ Instead, the patch being committed duplicates the string in the
+ config file on the stack before parsing it for consumption by the
+ application. (closes issue ASTERISK-16879) Reported by: Clod
+ Patry Tested by: mjordan patches: followme_no_limit.diff uploaded
+ by Clod Patry (license #5138) Slightly modified for this commit.
+ ........ Merged revisions 372390 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372391 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 19:43 +0000 [r372373] Richard Mudgett <rmudgett at digium.com>
+
+ * main/dsp.c, /: Fix compile error. ........ Merged revisions
+ 372372 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 19:24 +0000 [r372365] Kinsey Moore <kmoore at digium.com>
+
+ * main/manager.c, /: Correct documentation for ModuleLoad AMI
+ action The documentation incorrectly listed 'rtp' as a reloadable
+ subsystem and left out many other reloadable subsystems. It is
+ now also documented that subsystems may only be reloaded, not
+ loaded or unloaded. (closes issue AST-977) Reported-by: John
+ Bigelow ........ Merged revisions 372354 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372358 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 18:46 +0000 [r372342] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/dsp.c, /: dsp.c: in ast_mf_detect_init incorrectly sets
+ goertzel samples to 160, should be MF_GSIZE Related
+ https://reviewboard.asterisk.org/r/2097/ ........ Merged
+ revisions 372339 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372341 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 18:36 +0000 [r372340] Kinsey Moore <kmoore at digium.com>
+
+ * main/pbx.c, /: Ensure counts generated in
+ manager_show_dialplan_helper are correct When
+ manager_show_dialplan_helper was written, the counter increment
+ for the total number of contexts was placed with the extensions
+ increment instead of in the enclosing loop. This function should
+ now generate correct context counts. (closes issue AST-970)
+ Reported-by: John Bigelow ........ Merged revisions 372337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372338 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 17:35 +0000 [r372327-372328] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_rtp_asterisk.c: Fix coding guidelines issue with a recent
+ commit.
+
+ * res/res_rtp_asterisk.c: Fix RTP/RTCP read error message
+ confusion. The RTP/RTCP read error message can report "fail:
+ success" when the read failure is because of an ICE failure. *
+ Changed __rtp_recvfrom() to generate a PJ ICE message when ICE
+ fails. * Changed RTP/RTCP read error message to indicate an
+ unspecified error when errno is zero. (closes issue
+ ASTERISK-20288) Reported by: Joern Krebs Patches:
+ jira_asterisk_20288_err_msg.patch (license #5621) patch uploaded
+ by rmudgett (modified)
+
+2012-09-05 16:04 +0000 [r372311] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c,
+ include/asterisk/rtp_engine.h: Re-fix sending unnegotiated
+ payloads during a P2P RTP bridge. The previous fix still would
+ look in the static_RTP_PT table, which is inappropriate since we
+ specifically want to find a codec that has been negotiated.
+ (closes issue ASTERISK-20296) reported by NITESH BANSAL Patches:
+ codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
+
+2012-09-05 13:47 +0000 [r372289] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c, /: Fix memory leaks in app_voicemail when
+ using IMAP storage or realtime config This patch fixes two memory
+ leaks: 1. When find_user is called with NULL as its first
+ parameter, the voicemail user returned is allocated on the heap.
+ The inboxcount2 function uses find_user in such a fashion when
+ counting new messages, and fails to free the resulting voicemail
+ user object. 2. When populate_defaults is called on a voicemail
+ user, it wipes whatever flags have been set on the object by
+ copying over the global flags object. If the VM_ALLOCED flag was
+ ste on the voicemail user prior to doing so, that flag is
+ removed. This leaks the voicemail user when free_user is later
+ called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
+ patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
+ Patch slightly modified for this commit. Review:
+ https://reviewboard.asterisk.org/r/2096 ........ Merged revisions
+ 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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