[asterisk-commits] lmadsen: tag 1.8.1-rc1 r295161 - /tags/1.8.1-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Nov 16 12:49:40 CST 2010
Author: lmadsen
Date: Tue Nov 16 12:49:36 2010
New Revision: 295161
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=295161
Log:
Importing files for 1.8.1-rc1 release.
Added:
tags/1.8.1-rc1/.lastclean (with props)
tags/1.8.1-rc1/.version (with props)
tags/1.8.1-rc1/ChangeLog (with props)
Added: tags/1.8.1-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.1-rc1/.lastclean?view=auto&rev=295161
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--- tags/1.8.1-rc1/ChangeLog (added)
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+2010-11-16 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.1-rc1 Released.
+
+2010-11-15 18:30 +0000 [r294989-295078] Tilghman Lesher <tlesher at digium.com>
+
+ * tests/test_expr.c (added), /: Merged revisions 295062 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295062 | tilghman | 2010-11-15 12:24:02 -0600
+ (Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15
+ Nov 2010) | 2 lines Create test verifying results of expression
+ parser ........ ................
+
+ * funcs/func_curl.c, /: Merged revisions 294988 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010)
+ | 8 lines It is possible to crash Asterisk by feeding the curl
+ engine invalid data. (closes issue #18161) Reported by: wdoekes
+ Patches: 20101029__issue18161.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman ........
+
+2010-11-12 21:14 +0000 [r294905-294911] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 294910 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12
+ Nov 2010) | 4 lines Return correct error code if lock path fails.
+ The recent changes to open_mailbox actually caused it to be
+ fixed, but let's be consistent. Reported by alecdavis in
+ asterisk-dev. ........
+
+ * apps/app_voicemail.c, /: Merged revisions 294904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600
+ (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010)
+ | 16 lines Fix regression causing abort in voicemail after
+ opening a mailbox with no mesgs. In order to be more safe, some
+ error handling code was changed to respect more error conditions
+ including the potential memory allocation failure for deleted and
+ heard message tracking introduced in 293004. However,
+ last_message_index returns -1 for zero messages (perhaps as
+ expected) and was triggering the stricter error checking. Because
+ last_message_index is only called directly in one place, just
+ return 0 from open_mailbox (for file based storage) when no
+ messages are detected unless a real error has occurred. (closes
+ issue #18240) Reported by: leobrown Patches:
+ bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
+ Tested by: pabelanger ........ ................
+
+2010-11-12 02:45 +0000 [r294823] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, /: Merged revisions
+ 294822 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600
+ (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010)
+ | 11 lines Asterisk is getting a "No D-channels available!"
+ warning message every 4 seconds. Asterisk is just whining too
+ much with this message: "No D-channels available! Using Primary
+ channel XXX as D-channel anyway!". Filtered the message so it
+ only comes out once if there is no D channel available without an
+ intervening D channel available period. (closes issue #17270)
+ Reported by: jmls ........ ................
+
+2010-11-11 22:17 +0000 [r294740-294745] Russell Bryant <russell at digium.com>
+
+ * doc/CCSS_architecture.pdf (removed): Remove CCSS architecture
+ PDF. It has been moved to:
+ https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture
+
+ * doc/digium-mib.txt (removed), doc/followme.txt (removed),
+ doc/building_queues.txt (removed), doc/timing.txt (removed),
+ doc/advice_of_charge.txt (removed), doc/unistim.txt (removed),
+ doc/video_console.txt (removed), doc/macroexclusive.txt
+ (removed), doc/google-soc2009-ideas.txt (removed), doc/README.txt
+ (added), doc/callfiles.txt (removed), doc/externalivr.txt
+ (removed), doc/codec-64bit.txt (removed),
+ build_tools/prep_tarball, doc/video.txt (removed), doc/jingle.txt
+ (removed), doc/modules.txt (removed), doc/manager_1_1.txt
+ (removed), doc/PEERING (removed), doc/snmp.txt (removed),
+ doc/siptls.txt (removed), doc/HOWTO_collect_debug_information.txt
+ (removed), doc/ldap.txt (removed), doc/sip-retransmit.txt
+ (removed), doc/distributed_devstate.txt (removed),
+ doc/voicemail_odbc_postgresql.txt (removed), doc/tex (removed),
+ doc/queue.txt (removed), doc/jabber.txt (removed),
+ doc/chan_sip-perf-testing.txt (removed), Makefile,
+ doc/asterisk-mib.txt (removed), doc/database_transactions.txt
+ (removed), doc/smdi.txt (removed), doc/janitor-projects.txt
+ (removed), doc/hoard.txt (removed), doc/res_config_sqlite.txt
+ (removed), doc/osp.txt (removed), doc/speechrec.txt (removed),
+ doc/sms.txt (removed), doc/distributed_devstate-XMPP.txt
+ (removed), doc/valgrind.txt (removed), doc/realtimetext.txt
+ (removed), doc/cli.txt (removed), doc/rtp-packetization.txt
+ (removed), doc/datastores.txt (removed), doc/CODING-GUIDELINES
+ (removed), doc/ss7.txt (removed), doc/backtrace.txt (removed),
+ doc/India-CID.txt (removed): Remove most of the contents of the
+ doc dir in favor of the wiki content. This merge does the
+ following things: * Removes most of the contents from the doc/
+ directory in favor of the wiki - http://wiki.asterisk.org/ *
+ Updates the build_tools/prep_tarball script to know how to export
+ the contents of the wiki in both PDF and plain text formats so
+ that the documentation is still included in Asterisk release
+ tarballs.
+
+2010-11-11 21:58 +0000 [r294640-294734] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294733 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600
+ (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
+ | 18 lines Fix problem with qualify option packets for realtime
+ peers never stopping. The option packets not only never stopped,
+ but if a realtime peer was not in the peer list multiple options
+ dialogs could accumulate over time. This scenario has the
+ potential to progress to the point of saturating a link just from
+ options packets. The fix was to ensure that the poke scheduler
+ checks to see if a peer is in the peer list before continuing to
+ poke. The reason a peer must be in the peer list to be able to
+ properly manage an options dialog is because otherwise the call
+ pointer is lost when the peer is regenerated from the database,
+ which is how existing qualify dialogs are detected. (closes issue
+ #16382) (closes issue #17779) Reported by: lftsy Patches:
+ bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
+ zerohalo ........ ................
+
+ * main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
+ revisions 294639 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294639 | jpeeler | 2010-11-11 13:31:00 -0600
+ (Thu, 11 Nov 2010) | 53 lines Merged revisions 294384 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
+ | 47 lines Fix a deadlock in device state change processing.
+ Copied from some notes from the original author (Russell):
+ Deadlock scenario: Thread 1: device state change thread Holds -
+ rdlock on contexts Holds - hints lock Waiting on channels
+ container lock Thread 2: SIP monitor thread Holds the "iflock"
+ Holds a sip_pvt lock Holds channel container lock Waiting for a
+ channel lock Thread 3: A channel thread (chan_local in this case)
+ Holds 2 channel locks acquired within app_dial Holds a 3rd
+ channel lock it got inside of chan_local Holds a local_pvt lock
+ Waiting on a rdlock of the contexts lock A bunch of other threads
+ waiting on a wrlock of the contexts lock To address this
+ deadlock, some locking order rules must be put in place and
+ enforced. Existing relevant rules: 1) channel lock before a pvt
+ lock 2) contexts lock before hints lock 3) channels container
+ before a channel What's missing is some enforcement of the order
+ when you involve more than any two. To fix this problem, I put in
+ some code that ensures that (at least in the code paths involved
+ in this bug) the locks in (3) come before the locks in (2). To
+ change the operation of thread 1 to comply, I converted the
+ storage of hints to an astobj2 container. This allows processing
+ of hints without holding the hints container lock. So, in the
+ code path that led to thread 1's state, it no longer holds either
+ the contexts or hints lock while it attempts to lock the channels
+ container. (closes issue #18165) Reported by: antonio ABE-2583
+ ........ ................
+
+2010-11-10 23:26 +0000 [r294569-294605] Tilghman Lesher <tlesher at digium.com>
+
+ * pbx/pbx_spool.c: Fixing the Mac OS X build (bamboo warning)
+
+ * pbx/pbx_spool.c: Properly queue files with inotify(7). (closes
+ issue #18089) Reported by: abelbeck Patches:
+ 20101021__issue18089.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-11-10 14:14 +0000 [r294501-294535] Russell Bryant <russell at digium.com>
+
+ * UPGRADE.txt, res/ais/clm.c, res/ais/evt.c: Tweak a couple of CLI
+ commands back to their original form. The "module" in this case
+ is two parts, so there are two words before the verb of the CLI
+ command.
+
+ * main/devicestate.c, /: Merged revisions 294500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010)
+ | 7 lines Improve a debug message to be more readable and
+ consistent. (closes issue #18282) Reported by: klaus3000 Patches:
+ ast_devstate2str-patch.txt uploaded by klaus3000 (license 65)
+ ........
+
+2010-11-09 22:46 +0000 [r294466] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c: Allow ast_do_masquerade() failure to be reported
+ again.
+
+2010-11-09 20:33 +0000 [r294430] Tilghman Lesher <tlesher at digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 294429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010)
+ | 8 lines Detect GMime properly on systems where gmime flags and
+ libs are configured with pkg-config. (closes issue #16155)
+ Reported by: jcollie Patches: 20100917__issue16155.diff.txt
+ uploaded by tilghman (license 14) Tested by: tilghman ........
+
+2010-11-09 16:55 +0000 [r294349] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/channel.h, channels/sig_pri.c, main/channel.c,
+ channels/chan_misdn.c, channels/sig_analog.c: Analog lines do not
+ transfer CONNECTED LINE or execute the interception macros. Add
+ connected line update for sig_analog transfers and simplify the
+ corresponding sig_pri and chan_misdn transfer code. Note that if
+ you create a three-way call in sig_analog before transferring the
+ call, the distinction of the caller/callee interception macros
+ make little sense. The interception macro writer needs to be
+ prepared for either caller/callee macro to be executed. The
+ current implementation swaps which caller/callee interception
+ macro is executed after a three-way call is created. Review:
+ https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA
+ SWP-2372
+
+2010-11-08 22:32 +0000 [r294278-294313] Jeff Peeler <jpeeler at digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 294312 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08
+ Nov 2010) | 1 line add missing unlock not present in 294277
+ ........
+
+ * include/asterisk/timing.h, main/timing.c, main/channel.c, /,
+ res/res_timing_timerfd.c: Merged revisions 294277 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08
+ Nov 2010) | 16 lines Fix playback failure when using IAX with the
+ timerfd module. To fix this issue the alert pipe will now be used
+ when the timerfd module is in use. There appeared to be a race
+ that was not solved by adding locking in the timerfd module, but
+ needed to be there anyway. The race was between the timer being
+ put in non-continuous mode in ast_read on the channel thread and
+ the IAX frame scheduler queuing a frame which would enable
+ continuous mode before the non-continuous mode event was read.
+ This race for now is simply avoided. (closes issue #18110)
+ Reported by: tpanton Tested by: tpanton I put tested by tpanton
+ because it was tested on his hardware. Thanks for the remote
+ access to debug this issue! ........
+
+2010-11-08 20:56 +0000 [r294243] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov
+ 2010) | 8 lines Go off hold when we get an empty reinvite telling
+ us to. (closes issue 0014448) Reported by: frawd (closes issue
+ #17878) Reported by: frawd ........
+
+2010-11-08 19:56 +0000 [r294207] Terry Wilson <twilson at digium.com>
+
+ * configs/calendar.conf.sample, res/res_calendar.c: Set a default
+ waittime, and make sure to convert it to milliseconds
+
+2010-11-08 17:16 +0000 [r294125] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: valgrind reported references to freed
+ memory during a mISDN hangup collision. Bad things have been
+ happening in chan_misdn because the chan_misdn channel private
+ struct chan_list is not protected from reentrancy. Hangup
+ collisions have be causing read and write accesses to freed
+ memory. Converted chan_misdn struct chan_list to an ao2 object
+ for its reference counting feature. ********** Removed an
+ impediment to converting chan_list to an ao2 object. The use of
+ the other_ch member in chan_list is shaky at best. It is set if
+ the incoming and outgoing call legs are mISDN. The use of the
+ other_ch member goes against the Asterisk architecture and can
+ even cause problems. 1) It is used to disable echo cancellation.
+ This could be bad if the call is forked and the winning call leg
+ is not mISDN or the winning call leg is not the last mISDN
+ channel called by the fork. The other_ch would become a dangling
+ pointer. 2) It is used when the far end is alerting to hear the
+ far end's inband audio instead of Asterisk's generated ringback
+ tone. This is bad if the call is forked. You would only hear the
+ last forked mISDN channel and it may not be ringing yet. The
+ other_ch would become a dangling pointer if the call is later
+ transferred. ********** JIRA SWP-2423 JIRA ABE-2614
+
+2010-11-05 22:03 +0000 [r294084] Brett Bryant <bbryant at digium.com>
+
+ * channels/chan_sip.c: Fixed deadlock avoidance issues while
+ locking channel when adding the Max-Forwards header to a request.
+ (closes issue #17949) (closes issue #18200) Reported by: bwg
+ Review: https://reviewboard.asterisk.org/r/997/
+
+2010-11-05 16:05 +0000 [r294047-294049] Terry Wilson <twilson at digium.com>
+
+ * contrib/scripts/ast_tls_cert: Corret spelling and example
+
+ * contrib/scripts/ast_tls_cert: Tell people to use the correct
+ common name for clients as well
+
+2010-11-05 00:07 +0000 [r293970] Shaun Ruffell <sruffell at digium.com>
+
+ * codecs/codec_dahdi.c, /: Merged revisions 293969 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500
+ (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010)
+ | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when
+ receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically
+ commit 9034) added the capability for the wctc4xxp to return more
+ than a single packet of data in response to a read. However, when
+ decoding packets, codec_dahdi was still assuming that the default
+ number of samples was in each read. In other words, each packet
+ your provider sent you, regardless of size, would result in 20 ms
+ of decoded data (30 ms if decoding G723). If your provider was
+ sending 60 ms packets then codec_dahdi would end up stripping 40
+ ms of data from each transcoded frame resulting in "choppy"
+ audio. This would only affect systems where G729 packets are
+ arriving in sizes greater than 20ms or G723 packets arriving in
+ sizes greater than 30ms. DAHDI-744. ........ ................
+
+2010-11-04 21:39 +0000 [r293924] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Fixes ringback tone on sip semi-attended
+ transfer. ABE-2168
+
+2010-11-04 13:27 +0000 [r293887] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_sip.c: Do not output port in IPaddress for AMI
+ sippeers. (closes issue #18248) Reported by: orn Patches:
+ ami_sippeers.patch uploaded by pabelanger (license 224) Tested
+ by: orn
+
+2010-11-03 18:35 +0000 [r293807] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293806 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500
+ (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010)
+ | 20 lines Party A in an analog 3-way call would continue to hear
+ ringback after party C answers. All parties are analog FXS ports.
+ 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to
+ bring C into 3-way call before C answers. (A and B hear ringback)
+ 4) C answers 5) A continues to hear ringback during the 3-way
+ call. (All parties can hear each other.) * Fixed use of wrong
+ variable in dahdi_bridge() that stopped ringback on the wrong
+ subchannel. * Made several debug messages have more information.
+ A similar issue happens if B and C are SIP channels. B continues
+ to hear ringback. For some reason this only affects v1.8 and
+ trunk. * Don't start ringback on the real and 3-way subchannels
+ when creating the 3-way conference. Removing this code is benign
+ on v1.6.2 and earlier. ........ ................
+
+2010-11-03 18:05 +0000 [r293803] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/rtp_engine.h, main/rtp_engine.c,
+ channels/chan_sip.c: Avoid valgrind warnings for
+ ast_rtp_instance_get_xxx_address The documentation for
+ ast_rtp_instance_get_(local/remote)_address stated that they
+ returned 0 for success and -1 on failure. Instead, they returned
+ 0 if the address structure passed in was already equivalent to
+ the address instance local/remote address or 1 otherwise. 90% of
+ the calls to these functions completely ignored the return
+ address and passed in an uninitialized struct, which would make
+ valgrind complain even though the operation was technically safe.
+ This patch fixes the documentation and converts the
+ get_xxx_address functions to void since all they really do is
+ copy the address and cannot fail. Additionally two new functions
+ (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created
+ for the 3 times where the return value was actually checked. The
+ get_and_cmp_local_address function is currently unused, but
+ exists for the sake of symmetry. The only functional change as a
+ result of this change is that we will not do an
+ ast_sockaddr_cmp() on (mostly uninitialized) addresses before
+ doing the ast_sockaddr_copy() in the get_*_address functions. So,
+ even though it is an API change, it shouldn't have a noticeable
+ change in behavior. Review:
+ https://reviewboard.asterisk.org/r/995/
+
+2010-11-02 23:09 +0000 [r293724] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500
+ (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
+ | 8 lines Add enabled/disabled information for rtautoclear sip
+ show settings output. When setting to zero/"no", the numeric
+ default was shown making it not obvious the disabled setting was
+ respected. (closes issue #18123) Reported by: zerohalo ........
+ ................
+
+2010-11-02 21:29 +0000 [r293648] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293647 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500
+ (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010)
+ | 6 lines Make warning message have more useful information in
+ it. Change "Unable to get index, and nullok is not asserted" to
+ "Unable to get index for '<channel-name>' on channel <number>
+ (<function>(), line <number>)". ........ ................
+
+2010-11-02 20:45 +0000 [r293611] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * main/manager.c: If manager and tls are disabled, do not display
+ TCP/TLS Bindaddress.
+
+2010-11-01 17:29 +0000 [r293530] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Analog 3-way call would not connect all
+ parties if one was using sig_pri. Also the "dahdi show channel"
+ would not show the correct 3-way call status. * Synchronized the
+ inthreeway flag between chan_dahdi and sig_analog. * Fixed a
+ my_set_linear_mode() sign error and made take an analog sub
+ channel enum.
+
+2010-11-01 16:09 +0000 [r293496] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_iax2.c: Use ast_sockaddr_from_sin function not
+ memcpy This resolves some IAX2 registration issue report on the
+ asterisk-users mailing list. (closes issue #18202) Reported by:
+ pabelanger Patches: update_registry.patch.v2 uploaded by
+ pabelanger (license 224) Tested by: pabelanger, Nic Colledge
+ (mailing list) Review: https://reviewboard.asterisk.org/r/993
+
+2010-11-01 14:58 +0000 [r293493] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Only offer codecs both sides support for
+ directmedia When using directmedia, Asterisk needs to limit the
+ codecs offered to just the ones that both sides recognize,
+ otherwise they may end up sending audio that the other side
+ doesn't understand. (closes issue #17403) Reported by: one47
+ Patches: sip_codecs_simplified4 uploaded by one47 (license 23)
+ Tested by: one47, falves11 Review:
+ https://reviewboard.asterisk.org/r/967/
+
+2010-10-30 01:53 +0000 [r293341-293418] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293417 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500
+ (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some more code that serves no purpose.
+ ........ ................
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500
+ (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some code that serves no purpose.
+ ........ ................
+
+2010-10-29 21:48 +0000 [r293305] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: Modify sip_setoption to not complain about
+ unknown options. This now behaves just like the other setoption
+ callbacks. For the curious the offending option for the reporter
+ was AST_OPTION_CHANNEL_WRITE which was getting passed due to a
+ fix for chan_local in 286189. (closes issue #17985) Reported by:
+ globalnetinc
+
+2010-10-28 20:00 +0000 [r293197] Tilghman Lesher <tlesher at digium.com>
+
+ * res/ael/ael.tab.h, main/ast_expr2.c, /, main/ast_expr2.h,
+ res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c: Merged
+ revisions 293195-293196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500
+ (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........ ................ r293196 | tilghman | 2010-10-28
+ 14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions
+ 293194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........ ................
+
+2010-10-28 16:11 +0000 [r293159] Jeff Peeler <jpeeler at digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 293158 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28
+ Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically
+ when you're using characters above \x7f or invalid character
+ escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
+ Patches: issue18060_func_strings_filter_infinite_loop.patch
+ uploaded by wdoekes (license 717) Tested by: wdoekes ........
+
+2010-10-26 18:49 +0000 [r293119] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 293118 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500
+ (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010)
+ | 29 lines Fix inprocess_container in voicemail to correctly
+ restrict max messages. The comparison function logic was off, so
+ the number of sessions for a given mailbox were not being
+ incremented properly. This problem caused the maximum number of
+ messages per folder to not be respected when simultaneously
+ leaving multiple voicemails just below the threshold. These
+ problems should be fixed by the above, but just in case: Fixed
+ resequence_mailbox to rely on the actual number of detected
+ number of files in a directory rather than just assuming only 10
+ messages more than the maximum had been left. Also if more
+ messages than the maximum are deleted they are actually removed
+ now. The second purpose of this commit should have been separated
+ out probably, but is related to the above. Again, if the number
+ of messages in a given voicemail folder exceeds the maximum set
+ limit make sure to allocate enough space for the deleted and
+ heard index tracking array. A few random fixes: There was a
+ forgotten decrement of the inprocess count in imap_store_file.
+ When using IMAP storage, do not look in the directory where file
+ based storage messages may still reside and influence the message
+ count. Ensure to use only the first format in sendmail. ABE-2516
+ ........ ................
+
+2010-10-26 16:32 +0000 [r293046-293081] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: No need to define the struct if there are no
+ users.
+
+ * channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Allow the DAHDI driver to compile, even with a
+ sufficiently older version of libpri. Fixes our Bamboo builds.
+
+2010-10-25 21:15 +0000 [r292906-292969] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/sig_pri.c: Several more defines that need to be altered
+ for compiling against an older version of libpri
+
+ * channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Allow the DAHDI driver to compile, even with a
+ sufficiently older version of libpri. Fixes our Bamboo builds.
+
+2010-10-25 19:07 +0000 [r292868] David Vossel <dvossel at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 292867 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500
+ (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010)
+ | 27 lines This patch turns chan_local pvts into astobj2 objects.
+ chan_local does some dangerous things involving deadlock
+ avoidance. tech_pvt functions like hangup and queue_frame are
+ provided with a locked channel upon entry. Those functions are
+ completely safe as long as you don't attempt to give up that
+ channel lock, but that is impossible to guarantee due to the
+ required deadlock avoidance necessary to lock both the tech_pvt
+ and both channels involved. In the past, we have tried to account
+ for this by doing things like setting a "glare" flag that
+ indicates what function should destroy the pvt. This was used in
+ local_hangup and local_queue_frame to decided who should destroy
+ the pvt if they collided in separate threads. I have removed the
+ need to do this by converting all chan_local tech_pvts to
+ astobj2. This means we can ref a pvt before deadlock avoidance
+ and not have to worry about that pvt possibly getting destroyed
+ under us. It also cleans up where we destroy the tech_pvt. The
+ only unlink from the tech_pvt container occurs in local_hangup
+ now, which is where it should occur. Since there still may be
+ thread collisions on some functions like local_hangup after
+ deadlock avoidance, I have added some checks to detect those
+ collisions and exit appropriately. I think this patch is going to
+ solve quite a bit of weirdness we have had with local channels in
+ the past. ........ ................
+
+2010-10-22 22:35 +0000 [r292794-292825] Terry Wilson <twilson at digium.com>
+
+ * contrib/scripts/ast_tls_cert: Don't create directories without at
+ least o+x Also, making files that you are going to modify
+ read-only is dumb.
+
+ * contrib/scripts/ast_tls_cert: Make files readable only by the
+ owner
+
+2010-10-22 21:28 +0000 [r292787] Leif Madsen <lmadsen at digium.com>
+
+ * configs/res_ldap.conf.sample, contrib/scripts/asterisk.ldif, /,
+ channels/chan_sip.c: Merged revisions 292786 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010)
+ | 13 lines Update the LDIF file for LDAP. The LDIF file
+ asterisk.ldif was quite a bit out of date from the
+ asterisk.ldap-schema file, so I've now updated that to be in
+ sync. The asterisk.ldif file being out of sync was a problem on
+ my systems where I was doing an ldapadd to import the schema into
+ the LDAP database, and the existing file would cause problems and
+ ERROR messages when registering. Additional documention has been
+ added based on feedback in the issue I'm closing. (closes issue
+ #13861) Reported by: scramatte Patches: ldap-update.txt uploaded
+ by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
+ rgenthner ........
+
+2010-10-22 17:09 +0000 [r292741] Mark Michelson <mmichelson at digium.com>
+
+ * tests/test_event.c: Prevent multiple runs of event_sub_test from
+ producing false failure results. The array of test subscriptions
+ was declared "static," meaning that the data.count field would
+ retain its value between runs of the test. After the first test
+ run, this would result in false reports of test failures. I chose
+ to just remove the "static" keyword from the structure since it's
+ not a huge deal to construct this structure during each run of
+ the test. Another alternative would have been to zero out the
+ data.count fields of each test subscription instead.
+
+2010-10-22 16:49 +0000 [r292740] Terry Wilson <twilson at digium.com>
+
+ * contrib/scripts/ast_tls_cert (added): Add TLS cert helper script
+ This script is useful for quickly generating self-signed CA,
+ server, and client certificates for use with Asterisk. It is
+ still recommended to obtain certificates from a recognized
+ Certificate Authority and to develop an understanding how SSL
+ certificates work. Real security is hard work. OPTIONS: -h Show
+ this message -m Type of cert "client" or "server". Defaults to
+ server. -f Config filename (openssl config file format) -c CA
+ cert filename (creates new CA cert/key as ca.crt/ca.key if not
+ passed) -k CA key filename -C Common name (cert field) For a
+ server cert, this should be the same address that clients attempt
+ to connect to. Usually this will be the Fully Qualified Domain
+ Name, but might be the IP of the server. For a CA or client cert,
+ it is merely informational. Make sure your certs have unique
+ common names. -O Org name (cert field) An informational string
+ (company name) -o Output filename base (defaults to asterisk) -d
+ Output directory (defaults to the current directory) Example: To
+ create a CA and a server (pbx.mycompany.com) cert with output in
+ /tmp: ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp
+ This will create a CA cert and key as well as asterisk.pem and
+ the the two files that it is made from: asterisk.crt and
+ asterisk.key. Copy asterisk.pem and ca.crt somewhere (like
+ /etc/asterisk) and set tlscertfile=/etc/asterisk.pem and
+ tlscafile=/etc/ca.crt. Since this is a self-signed key, many
+ devices will require you to import the ca.crt file as a trusted
+ cert. To create a client cert using the CA cert created by the
+ example above: ast_tls_cert -m client -c /tmp/ca.crt -k
+ /tmp/ca.key -C "Joe User" -O \ "My Company" -d /tmp -o joe_user
+ This will create client.crt/key/pem in /tmp. Use this if your
+ device supports a client certificate. Make sure that you have the
+ ca.crt file set up as a tlscafile in the necessary Asterisk
+ configs. Make backups of all .key files in case you need them
+ later.
+
+2010-10-22 15:47 +0000 [r292704] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, main/channel.c, channels/chan_misdn.c:
+ Connected line is not updated when chan_dahdi/sig_pri or
+ chan_misdn transfers a call. When a call is transfered by ECT or
+ implicitly by disconnect in sig_pri or implicitly by disconnect
+ in chan_misdn, the connected line information is not exchanged.
+ The connected line interception macros also need to be executed
+ if defined. The CALLER interception macro is executed for the
+ held call. The CALLEE interception macro is executed for the
+ active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches:
+ abe_2589_c3bier.patch uploaded by rmudgett (license 664)
+ abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/958/
+
+2010-10-21 22:09 +0000 [r292667] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/misdn/ie.c: Compile correctly on Linux
+ (asterisk/localtime.h depends upon asterisk/autoconfig.h loading
+ first).
+
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