[asterisk-commits] bebuild: tag 1.8.11.0 r360767 - in /tags/1.8.11.0: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Mar 29 10:25:28 CDT 2012
Author: bebuild
Date: Thu Mar 29 10:25:23 2012
New Revision: 360767
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=360767
Log:
Importing files for 1.8.11.0 release.
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tags/1.8.11.0/.version (with props)
tags/1.8.11.0/ChangeLog (with props)
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+2012-03-29 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.11.0 Released.
+
+2012-02-27 23:34 +0000 [r357093] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c: Fix callerid of Originated calls. Thanks to Matt
+ Riddell for tracking this down. (closes issue ASTERISK-19385)
+ Reported by: ornix
+
+2012-02-27 16:03 +0000 [r356963] Terry Wilson <twilson at digium.com>
+
+ * main/features.c: Copy CDR variables when set during a bridge This
+ patch makes sure amaflags, accountcode, and userfield get copied
+ to the bridge CDR when set during a bridge (like via a custom
+ feature). (closes issue ASTERISK-16990) Review:
+ https://reviewboard.asterisk.org/r/1721/
+
+2012-02-27 15:14 +0000 [r356917] Jonathan Rose <jrose at digium.com>
+
+ * res/res_odbc.c: Remove possible segfaults from res_odbc by adding
+ locks around usage of odbc handle (closes issue ASTERISK-19011)
+ Reported by: Walter Doekes Patches:
+ issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch
+ uploaded by Walter Doekes (license 5674) review:
+ https://reviewboard.asterisk.org/r/1719/ review:
+ https://reviewboard.asterisk.org/r/1622/
+
+2012-02-25 17:21 +0000 [r356797] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c: Fix crash in app_voicemail during
+ close_mailbox In r354890, a memory leak in app_voicemail was
+ fixed by properly disposing of the allocated heard/deleted
+ pointers. However, there are situations, particularly when no
+ messages are found in a folder, where these pointers are not
+ allocated and not NULL. In that case, an invalid free would be
+ attempted, which could crash app_voicemail. As there are a number
+ of code paths where this could occur, this patch uses the number
+ of messages detected in the folder before it attempts to free the
+ pointers. This resolves the crash detected in the Asterisk Test
+ Suite's check_voicemail_nominal test.
+
+2012-02-24 18:23 +0000 [r356677] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/tcptls.h, channels/sip/include/sip.h,
+ channels/chan_sip.c: Fix worker thread resource leak in SIP
+ TCP/TLS. The SIP TCP/TLS worker threads were created joinable but
+ noone could join them if they died on their own. * Fix the SIP
+ TCP/TLS worker threads to not be created joinable. *
+ _sip_tcp_helper_thread() only needs one parameter since the pvt
+ parameter is only passed in as NULL and never used. (closes issue
+ ASTERISK-19203) Reported by: Steve Davies Review:
+ https://reviewboard.asterisk.org/r/1714/
+
+2012-02-24 17:41 +0000 [r356604-356650] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_srtp.c: Remove srtp_shutdown from res_srtp The patch for
+ ASTERISK-19253 included properly shutting down the libsrtp
+ library in the case of module unload. Unfortunately, not all
+ distributions have the srtp_shutdown call. As such, this patch
+ removes calling srtp_shutdown.
+
+ * include/asterisk/rtp_engine.h, res/res_srtp.c,
+ channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h,
+ main/rtp_engine.c: Allow SRTP policies to be reloaded Currently,
+ when using res_srtp, once the SRTP policy has been added to the
+ current session the policy is locked into place. Any attempt to
+ replace an existing policy, which would be needed if the remote
+ endpoint negotiated a new cryptographic key, is instead rejected
+ in res_srtp. This happens in particular in transfer scenarios,
+ where the endpoint that Asterisk is communicating with changes
+ but uses the same RTP session. This patch modifies res_srtp to
+ allow remote and local policies to be reloaded in the underlying
+ SRTP library. From the perspective of users of the SRTP API, the
+ only change is that the adding of remote and local policies are
+ now added in a single method call, whereas they previously were
+ added separately. This was changed to account for the differences
+ in handling remote and local policies in libsrtp. Review:
+ https://reviewboard.asterisk.org/r/1741/ (closes issue
+ ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
+ Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt
+ Jordan (license 6283) (with some small modifications for this
+ check-in)
+
+2012-02-23 19:49 +0000 [r356521] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c, main/features.c: Fix blind transfer parking
+ issues if the dialed extension is not recognized as a parking
+ extension. Custom parking extensions may not be coded such that
+ the first and only extension priority is the Park application.
+ These custom parking extensions will not be recognized as parking
+ extensions. When a call is blind transferred to an extension that
+ is not recognized as a parking extension, the normal blind
+ transfer code causes the transferred channel to start executing
+ dialplan. Calls that get parked in this manner do not know the
+ original channel name that parked the call so the original parker
+ could never be called back if the parked call is not retrieved
+ before the timeout time. The parking space is also announced to
+ the call being parked as a side effect of not knowing the
+ original parking channel. * Fix handling of BLINDTRANSFER channel
+ variable for call parking. * Fixed SIP blind transfer using the
+ wrong dialplan context variable to check for the parking
+ extension. (closes issue ASTERISK-19322) Reported by: aragon
+ Tested by: rmudgett, jparker Review:
+ https://reviewboard.asterisk.org/r/1730/ JIRA AST-766
+
+2012-02-23 15:37 +0000 [r356475] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix ACK routing for non-2xx responses. When
+ we send an ACK for a 2xx response to an INVITE, we are supposed
+ to use the learned route set. However, when we receive a non-2xx
+ final response to an INVITE, we are supposed to send the ACK to
+ the same place we initially sent the INVITE. We had been doing
+ this up until the changes went in that would build a route set
+ from provisional responses. That introduced a regression where we
+ would use the learned route set under all circumstances. With
+ this change, we now will set the destination of our ACK based on
+ the invitestate. If it is INV_COMPLETED then that means that we
+ have received a non-2xx final response (INV_TERMINATED indicates
+ a 2xx response was received). If it is INV_CANCELLED, then that
+ means the call is being canceled, which means that we should be
+ ACKing a 487 response. The other change introduced here is
+ setting the invitestate to INV_CONFIRMED when we send an ACK
+ *after* the reqprep instead of before. This way, we can tell in
+ reqprep more easily what the invitestate is prior to sending the
+ ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
+ patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
+ (license #5049) (with some slight modifications prior to commit)
+
+2012-02-23 03:36 +0000 [r356335-356430] Paul Belanger <pabelanger at digium.com>
+
+ * addons/chan_ooh323.c: Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2)
+
+ * apps/app_rpt.c: Missed one strsep() function
+
+ * apps/app_rpt.c: Add back strsep() function for previous commit
+
+2012-02-22 21:08 +0000 [r356291] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/calendar.h, main/loader.c, res/res_calendar.c:
+ Track module use count for res_calendar If the res_calendar
+ module was followed immediately by one of the calendar tech
+ modules and "core stop gracefully" was run, Asterisk would crash.
+ This patch adds use count tracking for res_calendar so that it is
+ unloaded after the tech modules when shutting down gracefully. It
+ is now not possible to unload all the of the calendar modules via
+ "module unload res_calednar.so", but it is still possible to
+ unload them all via "module unload -h res_calendar.so". Review:
+ https://reviewboard.asterisk.org/r/1752/
+
+2012-02-22 20:20 +0000 [r356290] Paul Belanger <pabelanger at digium.com>
+
+ * apps/app_rpt.c: Fix -Werror=unused-but-set-variable compiler
+ error (gcc 4.6.2) Review:
+ https://reviewboard.asterisk.org/r/1763/
+
+2012-02-22 14:50 +0000 [r356214] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: Fix potential buffer overrun and memory leak
+ when executing "sip show peers" The "sip show peers" command uses
+ a fix sized array to sort the current peers in the peers
+ ao2_container. The size of the array is based on the current
+ number of peers in the container. However, once the size of the
+ array is determined, the number of peers in the container can
+ change, as the peers container is not locked. This could cause a
+ buffer overrun when populating the array, if peers were added to
+ the container after the array was created. Additionally, a memory
+ leak of the allocated array would occur if a user caused the
+ _show_peers method to return CLI_SHOWUSAGE. We now create a
+ snapshot of the current peers using an ao2_callback with the
+ OBJ_MULTIPLE flag. This size of the array is set to the number of
+ peers that the iterator will iterate over; hence, if peers are
+ added or removed from the peers container it will not affect the
+ execution of the "sip show peers" command. Review:
+ https://reviewboard.asterisk.org/r/1738/ (closes issue
+ ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas
+ Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey
+ Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan
+ (license 6283)
+
+2012-02-21 11:16 +0000 [r355901-356107] Sean Bright <sean at malleable.com>
+
+ * channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
+ make sense when an IP is passed in.
+
+ * channels/chan_iax2.c: Remove spurious warning when
+ 'qualifyfreqnotok' is set successfully. (closes issue
+ ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright
+ Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John
+ Covert (license 5512)
+
+ * channels/chan_dahdi.c: This was a LOG_NOTICE, so roll it back.
+
+ * channels/chan_dahdi.c: Change some debug messages from LOG_DEBUG
+ to ast_debug.
+
+ * channels/chan_iax2.c: Add some boilerplate documentation for
+ IAXVAR and IAXPEER.
+
+ * channels/chan_iax2.c: Set the length of the ast_sockaddr, so that
+ we can set it's port later. Without this, the call to
+ ast_sockaddr_set_port a few lines later is a noop.
+
+2012-02-18 07:55 +0000 [r355850] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/sig_pri.c, channels/sig_ss7.c, channels/sig_pri.h,
+ channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_ss7.h,
+ channels/sig_analog.h: push 'outgoing' flag from sig_XXX up to
+ chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept
+ in sync, particulary FXS ast_hangup didn't clear the 'outgoing'
+ flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
+ Now provides a callback for all the low level sig_XXX modules.
+ (issue ASTERISK-19316) alecdavis (license 585) Reported by:
+ Jeremy Pepper Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1747/
+
+2012-02-18 03:59 +0000 [r355839] Paul Belanger <pabelanger at digium.com>
+
+ * res/res_pktccops.c: Fix -Werror=unused-but-set-variable compiler
+ error (gcc 4.6.2)
+
+2012-02-17 22:01 +0000 [r355746-355793] Sean Bright <sean at malleable.com>
+
+ * channels/chan_iax2.c, configs/iax.conf.sample: Don't allow
+ trunkfreq to be greater than 1000ms.
+
+ * channels/chan_iax2.c: Pass the correct value to
+ ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
+ variable to determine how often to send trunk packets, but this
+ value is in milliseconds while ast_timer_set_rate() expects the
+ rate argument to be ticks per second. So we divide 1000 by
+ trunkfreq and pass that in instead. With a default of 20ms, this
+ change makes IAX2 send trunk packets every 20ms instead of every
+ 50ms. Tracked down by myself and Bob Wienholt.
+
+2012-02-17 18:57 +0000 [r355721-355732] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix regressions with regards to route-set
+ creation on early dialogs. This fixes two main issues: 1.
+ Asterisk would send a CANCEL to the route created by the
+ provisional response instead of using the same destination it did
+ in the initial INVITE. 2. If a new route set arrives in a 200 OK
+ than was in the 1XX response (perfectly possible if our outbound
+ INVITE gets forked), then the route set in the 200 OK needs to
+ overwrite the route set in the 1XX response. (closes issue
+ ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten
+ Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson
+ (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt
+ (license 6034) Review: https://reviewboard.asterisk.org/r/1749
+
+ * main/translate.c: Revert change to translate.c as it has caused
+ an infinite loop to occur in circumstances.
+
+2012-02-17 00:25 +0000 [r355700-355711] Paul Belanger <pabelanger at digium.com>
+
+ * addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/perutil.c:
+ Revert 355700 and 355701
+
+ * addons/ooh323c/src/ooSocket.c: Missed a variable
+
+ * addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/perutil.c:
+ Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
+
+2012-02-16 20:01 +0000 [r355622] Sean Bright <sean at malleable.com>
+
+ * main/audiohook.c: Revert a change to audio_audiohook_write_list
+ that had no affect. When I made this change initially, I was
+ under the false impression that the audiohooks structure remained
+ on the channel after all of the hooks had been detached. This is
+ not the case, ast ast_read takes care of removing the audiohooks
+ structure if the lists are empty.
+
+2012-02-16 19:26 +0000 [r355574-355608] Richard Mudgett <rmudgett at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_c_declare_check.m4 (added), configure.ac,
+ formats/format_ogg_vorbis.c: Fix compile problem when old version
+ of libvorbisfile v1.1.2 is used. The principle difference between
+ libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition
+ of the predefined callbacks OV_CALLBACKS_xxx in
+ vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the
+ configure script to detect if libvorbisfile.h declares
+ OV_CALLBACKS_NOCLOSE. * Copied the declaration of
+ OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile.
+ (closes issue ASTERISK-19370) Reported by: Jonn Taylor
+
+ * res/res_monitor.c: Fix AMI Monitor action without File header
+ converting channel name into filename. * Fix potential Solaris
+ crash if Monitor application has a urlbase and no fname_base
+ option.
+
+2012-02-15 19:26 +0000 [r355448-355529] Sean Bright <sean at malleable.com>
+
+ * channels/chan_iax2.c: When IAX2 debugging is enabled, make sure
+ to log 'apathetic' messages too.
+
+ * channels/chan_iax2.c: Only use maxtrunkcall and maxnontrunkcall
+ in chan_iax2 if IAX_OLD_FIND is specified. These variables are
+ only accessed from the IAX_OLD_FIND path, so there is no reason
+ to keep them updated otherwise.
+
+ * channels/chan_iax2.c: Use TRUNK_CALL_START as originally
+ intended. Back in r646, TRUNK_CALL_START was added and defined as
+ 0x4000. That same value was also hard-coded in one part of the
+ IAX2 code instead of using the #define. TRUNK_CALL_START has
+ changed over the years (for dealing with LOW_MEMORY), but the
+ hard-coded usage was never updated to match. This patch fixes
+ that.
+
+2012-02-14 19:10 +0000 [r355319-355365] Richard Mudgett <rmudgett at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ formats/format_ogg_vorbis.c: Fix voicemail problems when using
+ ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
+ format because it did not implement the seek and tell format
+ callbacks among other problems. Since we were already using the
+ libvorbis and libvorbisenc libraries we can use libvorbisfile as
+ it is also part of the vorbis library package. * Made use the
+ libvorbisfile to handle the ogg/vorbis file stream. The
+ format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
+ (closes issue ASTERISK-16926) Reported by: sque Patches:
+ ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
+ by sque
+
+ * cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock in
+ cel_sqlite_custom reload. (closes issue ASTERISK-19356) Reported
+ by: Alex Villacis Lasso Patches:
+ asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
+ (license #5617) patch uploaded by Alex Villacis Lasso Review:
+ https://reviewboard.asterisk.org/r/1740/
+
+2012-02-14 16:26 +0000 [r355268] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Properly invert the return of a strncmp
+ call. This was causing identification that should have been made
+ private to be public. (closes issue AST-814) reported by Patrick
+ Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
+ (license 5430)
+
+2012-02-14 15:50 +0000 [r355228] Jason Parker <jparker at digium.com>
+
+ * configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3 CDRs
+ by default in sample configs.
+
+2012-02-14 13:33 +0000 [r355182] Sean Bright <sean at malleable.com>
+
+ * channels/chan_iax2.c: Clear the high order bit from the
+ destination call number before sending. send_apathetic_reply
+ takes the incoming frame's source call number as the destination
+ call number for the outgoing frame. If the incoming frame was a
+ full frame, then the high order bit of the source call number is
+ set and will be interpreted as a retransmit when sent back out as
+ the destination call number.
+
+2012-02-14 09:41 +0000 [r355136] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: call manager_event only if there is not
+ null channel structure (Closes issue ASTERISK-19298) Reported by:
+ robinfood Patches: issue19298.patch uploaded by may213 (License
+ #5415)
+
+2012-02-13 22:02 +0000 [r355056] Richard Mudgett <rmudgett at digium.com>
+
+ * pbx/pbx_spool.c: Fix occasional incorrectly delayed call-file
+ execution. Since the dir timestamp is available at one second
+ resolution, we cannot know if it was updated within the same
+ second after we scanned it. Therefore, we will force another scan
+ if the dir was just modified. * Changed to force another scan if
+ the directory was just modified. (closes issue ASTERISK-19081)
+ Reported by: Knut Bakke Review:
+ https://reviewboard.asterisk.org/r/1688/
+
+2012-02-13 19:49 +0000 [r355009] Joshua Colp <jcolp at digium.com>
+
+ * pbx/pbx_config.c: Only allow one 'dialplan reload' to execute at
+ a time as otherwise they would share the same common local
+ context list. (closes issue AST-758)
+
+2012-02-13 17:22 +0000 [r354953] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_config_pgsql.c, configs/extconfig.conf.sample: Fix
+ reconnecting to pgsql database after connection loss. There can
+ only be one database connection in res_config_pgsql just like
+ res_config_sqlite. If the connection is lost, the connection may
+ not get reestablished to the same database if the res_pgsql.conf
+ and extconfig.conf files are inconsistent. * Made only use the
+ configured database from res_pgsql.conf. * Fixed potential buffer
+ overwrite of last[] in config_pgsql(). (closes issue
+ ASTERISK-16982) Reported by: german aracil boned Review:
+ https://reviewboard.asterisk.org/r/1731/
+
+2012-02-10 21:45 +0000 [r354889] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c: Fix a voicemail memory leak with
+ heard/deleted messages. open_mailbox() was changed quite a long
+ time ago to allocate this memory. close_mailbox() should have
+ been changed to be responsible for freeing it.
+
+2012-02-10 18:03 +0000 [r354835] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c: Fix AMI Redirect ExtraChannel not redirecting to
+ the same exten and context. The astman_get_header() never returns
+ NULL so the check by the code for NULL would never fail. (closes
+ issue ASTERISK-16974) Reported by: Nuno Borges Patches:
+ 0018325.patch (license #6116) patch uploaded by Nuno Borges
+ (modified)
+
+2012-02-09 22:01 +0000 [r354749] Terry Wilson <twilson at digium.com>
+
+ * funcs/func_cdr.c: Note that CDRs are immutable once a bridge is
+ torn down CDRs cannot be modified after a bridge is torn down,
+ (e.g. after Dial() returns) even though the CDR() function may be
+ called. Since modifying the CDR code to change this behavior
+ could very easily break all kinds of things, this patch just
+ documents this limitation. (closes issues ASTERISK-16923) Review:
+ https://reviewboard.asterisk.org/r/1720/
+
+2012-02-09 20:49 +0000 [r354655-354702] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Fix parsing of SIP headers where compact and
+ non-compact headers are mixed Change parsing of SIP headers so
+ that compactness of the header no longer influences which header
+ will be chosen. Previously, a non-compact header would be chosen
+ instead of a preceeding compact-form header. (closes issue
+ ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/
+
+ * main/config.c: Make the config parser remove escaping backslashes
+ The config parser in Asterisk does not currently remove a
+ backslash that is used to escape a semicolon which would
+ otherwise be interpreted as the start of a comment. The change
+ here causes that backslash to be removed, but does not create a
+ real escape system in the config parser. The biggest complication
+ with a real escape system would be breaking existing configs
+ everywhere (parsing \\ as \ and breaking on escaped non-semicolon
+ characters) even though it would be the "right" way to do things.
+ (closes issue ASTERISK-17121) Review:
+ https://reviewboard.asterisk.org/r/1724/
+
+2012-02-09 18:58 +0000 [r354594-354640] Mark Michelson <mmichelson at digium.com>
+
+ * main/translate.c: Remove outdated comment.
+
+ * main/translate.c: Fix translation path choices. This change makes
+ it so computational cost is not taken into account when deciding
+ if a multistep path is better than a single-step path. This means
+ that the only time a multistep path will be chosen is if no
+ single-step path exists. This ensures a better quality
+ translation even if it turns out to be slightly slower. (closes
+ issue ASTERISK-16821) reported by Andrew Lindh Review:
+ https://reviewboard.asterisk.org/r/1715
+
+2012-02-09 17:07 +0000 [r354547] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: Clean-up of minor formatting issues in
+ r354542/3/4 rmudgett pointed out some formatting issues in the
+ check-in for ASTERISK-19290. This cleans those up. Review:
+ https://reviewboards.asterisk.org/r/1722/
+
+2012-02-09 16:56 +0000 [r354545] Mark Michelson <mmichelson at digium.com>
+
+ * CHANGES, res/res_fax.c: Adding reload support to res_fax.so
+ (closes issue ASTERISK-16712) reported by Frank DiGennaro Review:
+ https://reviewboard.asterisk.org/r/1713
+
+2012-02-09 16:30 +0000 [r354542] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
+ codes In ASTERISK-18924, SIP INFO DTMF handlingw as changed to
+ account for both lowercase alphatbetic DTMF events, as well as
+ uppercase alphabetic DTMF events. When this occurred, the
+ comparison of the character buffer containing the event code was
+ changed such that the buffer was first compared again '0' and '9'
+ to determine if it was numeric. Unfortunately, since the first
+ character in the buffer will typically be '1' in the case of
+ non-numeric event codes (10-16), this caused those codes to be
+ converted to a DTMF event of '1'. This patch fixes that, and
+ cleans up handling of both application/dtmf-relay and
+ application/dtmf content types. Review:
+ https://reviewboard.asterisk.org/r/1722/ (closes issue
+ ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan
+
+2012-02-09 02:52 +0000 [r354495] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce. Well,
+ thats embarrasing. I forgot to initialize the caller_id storage.
+ (closes issue ASTERISK-19311) Reported by: tootai Tested by:
+ rmudgett
+
+2012-02-09 02:23 +0000 [r354492] Russell Bryant <russell at russellbryant.com>
+
+ * main/channel.c: Remove some unnecessary locking from
+ ast_hangup(). This patch removes some unnecessary locking of the
+ channels container in ast_hangup(). The reason this came up is
+ that this lock can very quickly block the entire system. If any
+ of the channel cleanup code decides to block, it causes a problem
+ for the whole system. For example, when audiohooks get destroyed,
+ if that blocks for a while waiting on the mixmonitor thread to
+ exit because it's busy blocking on some I/O, it causes a problem
+ for many other threads in the meantime. Review:
+ https://reviewboard.asterisk.org/r/1712/
+
+2012-02-07 20:53 +0000 [r354348] Terry Wilson <twilson at digium.com>
+
+ * contrib/realtime/postgresql/realtime.sql, channels/chan_sip.c:
+ Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
+ instead of "" 2. Don't set ipaddr or port to the string "(null)"
+ when they are empty 3. Add missing required fields, set default
+ for lastms to 0, and modify the length of the ipaddr field to 45
+ in the Postgresql realtime.sql file. (closes issue
+ ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/
+
+2012-02-07 15:04 +0000 [r354263] Jonathan Rose <jrose at digium.com>
+
+ * cdr/cdr_pgsql.c: Fix column duplication bug in module reload for
+ cdr_pgsql. Prior to this patch, attempts to reload cdr_pgsql.so
+ would cause the column list to keep its current data and then add
+ a second copy during the reload. This would cause attempts to log
+ the CDR to the database to fail. This patch also cleans up some
+ unnecessary null checks for ast_free and deals with a few
+ potential locking problems. (closes issue ASTERISK-19216)
+ Reported by: Jacek Konieczny Review:
+ https://reviewboard.asterisk.org/r/1711/
+
+2012-02-06 22:58 +0000 [r354116-354216] Richard Mudgett <rmudgett at digium.com>
+
+ * pbx/pbx_config.c: Improved documentation of CLI "dialplan add
+ extension" command. * Documented dialplan add extension
+ <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
+ of command without the app-data value. There are many
+ applications that do no need any parameters so it is silly to
+ require that field for all commands. * Fixed a couple
+ ast_malloc/ast_free mismatches with ast_add_extension2() calls.
+ (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
+ by: rmudgett
+
+ * main/features.c: Add missing headers to AMI UnParkedCall event to
+ uniquely identify the call. The AMI UnParkedCall event was
+ missing the Parkinglot and Uniqueid headers that the AMI
+ ParkedCall event contains. (closes issue ASTERISK-19240) Reported
+ by: Michael Yara
+
+2012-02-03 21:24 +0000 [r353999] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due
+ to r335976 Bad locking order was added to chan_agent to prevent
+ segfaults from having no locking in a patch by irroot. This patch
+ addresses the bad locking order by releasing locks before getting
+ the right locking order to stop deadlocks from occuring when
+ doing multiple interactions with agents. (closes issue
+ ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
+ https://reviewboard.asterisk.org/r/1708/
+
+2012-02-02 22:26 +0000 [r353915] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Ensure entering T.38 passthrough does not
+ cause an infinite loop After R340970 Asterisk was still polling
+ the RTCP file descriptor after RTCP is shut down and removed. If
+ the descriptor happened to have data ready when the removal
+ occured then Asterisk would go into an infinite loop trying to
+ read data that it can never actually access. This change disables
+ the audio RTCP file descriptor for the duration of the T.38
+ transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
+ Vrban
+
+2012-02-02 20:01 +0000 [r353867] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
+ Restore the 'w' modifier support for ISDN spans.
+ Dial(DAHDI/g0/1234w888) This feature also causes the sending
+ complete ie to be sent for switch types that do not automatically
+ send the ie. (EuroISDN/ETSI) The main difference between dialing
+ Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
+ sending of the sending complete ie. (closes issue ASTERISK-19176)
+ Reported by: rmudgett Tested by: rmudgett
+
+2012-02-02 18:31 +0000 [r353818] Jonathan Rose <jrose at digium.com>
+
+ * funcs/func_curl.c: Backports some documentation for func_curl
+ from 10 to 1.8 For some reason this function was completely
+ undocumented in 1.8. I copied the 10 docs over to 1.8 and removed
+ references to an enumerator that was added in the Asterisk 10
+ version of func_curl. That was the only change I noted. (closes
+ issue ASTERISK-19186) Reported by: Olivier Krief
+
+2012-02-02 16:58 +0000 [r353770] Mark Michelson <mmichelson at digium.com>
+
+ * UPGRADE.txt, configs/manager.conf.sample,
+ include/asterisk/manager.h, configs/http.conf.sample,
+ main/manager.c, main/http.c: Fix TLS port binding behavior as
+ well as reload behavior: * Removes references to tlsbindport from
+ http.conf.sample and manager.conf.sample * Properly bind to port
+ specified in tlsbindaddr, using the default port if specified. *
+ On a reload, properly close socket if the service has been
+ disabled. A note has been added to UPGRADE.txt to indicate how
+ ports must be set for TLS. (closes issue ASTERISK-16959) reported
+ by Olaf Holthausen (closes issue ASTERISK-19201) reported by
+ Chris Mylonas (closes issue ASTERISK-19204) reported by Chris
+ Mylonas Review: https://reviewboard.asterisk.org/r/1709
+
+2012-02-02 16:57 +0000 [r353720-353769] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: Fix sip show peers port output, align
+ columns, and fix ami port output. A previous patch I committed
+ from ASTERISK-16930 unexpectedly changed some output for the AMI
+ action "sippeers" which this patch changes back. Also, this
+ aligns the output for the cli command "sip show peers" and fixes
+ another issue that patch introduced by using
+ ast_sockaddr_stringify calls multiple times without immediately
+ using the pointer. I also went ahead and did a little janitorial
+ work to clean up whitespace in _sip_show_peers. (issue
+ ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
+ Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
+ Walter Doekes (license 5674)
+
+ * channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers for
+ various functions in chan_sip There are a number of cleaner
+ looking wrappers for ast_sockaddr_stringify_fmt available which
+ are slightly more readable than using a direct call to
+ ast_sockaddr_stringify_fmt. This patch switches a number of those
+ calls in chan_sip to use those wrappers and is generally
+ harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
+ Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
+ Michael L. Young (license 5026)
+
+2012-02-01 15:50 +0000 [r353598] Sean Bright <sean at malleable.com>
+
+ * include/asterisk/audiohook.h: Resolve an overlap in the
+ ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
+ AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
+ unintended side effects. This patch moves
+ AST_AUDIOHOOK_TRIGGER_WRITE, and updates
+ AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
+ This will affect existing modules that use these flags, so be
+ sure to recompile as necessary. (closes issue ASTERISK-19246)
+ Reported by: feyfre
+
+2012-02-01 15:02 +0000 [r353550] Matthew Jordan <mjordan at digium.com>
+
+ * contrib/init.d/etc_default_asterisk: Added clarification for the
+ VERBOSITY setting to etc_default_asterisk Clarified that using
+ the VERBOSITY setting in etc_default_asterisk is the same as
+ using the -v command line switch, which causes Asterisk to launch
+ in console mode. (closes issue ASTERISK-17030) Reported by: Jonas
+
+2012-01-31 23:41 +0000 [r353502] Terry Wilson <twilson at digium.com>
+
+ * res/res_calendar.c: Allow res_calendar to be unloaded The
+ calendaring tech modules depend on res_calendar and initially
+ res_calendar just bumped the use count so that it couldn't be
+ unloaded. res_calendar can potentially create many threads and
+ I've seen issues where the Asterisk shutdown has failed where it
+ looked like these threads could be the culprit. This patch adds
+ unload support for res_calendar. Unloading res_calendar will also
+ unload the dependant tech modules as well. (closes issue
+ ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
+
+2012-01-31 16:51 +0000 [r353454] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/channel.h, main/manager.c: Fix memory leak in
+ error paths for action_originate(). * Fix memory leak of vars in
+ error paths for action_originate(). * Moved struct
+ fast_originate_helper tech and data members to stringfields. *
+ Simplified ActionID header handling for fast_originate(). * Added
+ doxygen note to ast_request() and ast_call() and the associated
+ channel callbacks that the data/addr parameters should be treated
+ as const char *. Review: https://reviewboard.asterisk.org/r/1690/
+
+2012-01-30 23:17 +0000 [r353371] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/dnsmgr.h, main/dnsmgr.c, channels/chan_sip.c:
+ Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
+ currently takes a pointer to an ast_sockaddr and updates it
+ anytime an address resolves to something different. There are a
+ couple of issues with this. First, the ast_sockaddr is usually
+ the address of an ast_sockaddr inside a refcounted struct and we
+ never bump the refcount of those structs when using dnsmgr. This
+ makes it possible that a refresh could happen after the
+ destructor for that object is called (despite ast_dnsmgr_release
+ being called in that destructor). Second, the module using dnsmgr
+ cannot be aware of an address changing without polling for it in
+ the code. If an action needs to be taken on address update (like
+ re-linking a SIP peer in the peers_by_ip table), then polling for
+ this change negates many of the benefits of having dnsmgr in the
+ first place. This patch adds a function to the dnsmgr API that
+ calls an update callback instead of blindly updating the address
+ itself. It also moves calls to ast_dnsmgr_release outside of the
+ destructor functions and into cleanup functions that are called
+ when we no longer need the objects and increments the refcount of
+ the objects using dnsmgr since those objects are stored on the
+ ast_dnsmgr_entry struct. A helper function for returning the
+ proper default SIP port (non-tls vs tls) is also added and used.
+ This patch also incorporates changes from a patch posted by Timo
+ Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
+ ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/
+
+2012-01-30 22:40 +0000 [r353320-353368] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_sip.c: prevent debug messsges displaying -ve Cseq
+ numbers. Missed in R353320
+
+ * channels/sip/include/sip.h, channels/sip/include/dialog.h,
+ channels/chan_sip.c: RFC3261 Section 8.1.1.5. The sequence number
+ value MUST be expressible as a 32-bit unsigned integer * fix: use
+ %u instead of %d when dealing with CSeq numbers - to remove
+ possibility of -ve numbers. * fix: change all uses of seqno and
+ friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
+ Summary of CSeq numbers. An initial CSeq number must be less than
+ 2^31 A CSeq number can increase in value up to 2^32-1 An
+ incrementing CSeq number must not wrap around to 0. Tested with
+ Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
+ Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1699/
+
+2012-03-05 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.10.0 Released.
+
+2012-03-01 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.10.0-rc4 Released.
+
+ * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
+ a port of 0.
+
+ In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which
+ changed the behavior of ast_find_ourip such that port number was
+ wiped out. This caused the port in internip (which is used for
+ Contact and Call-ID on NOTIFYs) to be 0. This change causes
+ ast_find_ourip to be port-preserving again.
+
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