[svn-commits] bebuild: tag 1.8.9.0-rc1 r349394 - /tags/1.8.9.0-rc1/ChangeLog
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Dec 29 13:19:22 CST 2011
Author: bebuild
Date: Thu Dec 29 13:19:18 2011
New Revision: 349394
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=349394
Log:
Add change log for 1.8.9.0
Added:
tags/1.8.9.0-rc1/ChangeLog (with props)
Added: tags/1.8.9.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.9.0-rc1/ChangeLog?view=auto&rev=349394
==============================================================================
--- tags/1.8.9.0-rc1/ChangeLog (added)
+++ tags/1.8.9.0-rc1/ChangeLog Thu Dec 29 13:19:18 2011
@@ -1,0 +1,36241 @@
+2011-12-29 15:13 +0000 [r349339] Matthew Jordan <mjordan at digium.com>
+
+ * main/rtp_engine.c: Handle AST_CONTROL_UPDATE_RTP_PEER frames in
+ local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER
+ frames in the local bridge loop causes the loop to exit
+ prematurely. This causes a variety of negative side effects,
+ depending on when the loop exits. This patch handles the frame by
+ essentially swallowing the frame in the local loop, as the
+ current channel drivers expect the RTP bridge to handle the
+ frame, and, in the case of the local bridge loop, no additional
+ action is necessary. (issue ASTERISK-19040) (issue
+ ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340)
+ (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/
+
+2011-12-28 21:30 +0000 [r349289] Sean Bright <sean at malleable.com>
+
+ * main/audiohook.c: Use ast_audiohook_write_list_empty to determine
+ if our lists are empty instead of duplicating that logic.
+
+2011-12-27 20:48 +0000 [r349194] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_musiconhold.c, res/res_timing_pthread.c,
+ include/asterisk/module.h, res/res_timing_dahdi.c,
+ res/res_timing_timerfd.c: Fix timing source dependency issues
+ with MOH Prior to this patch, res_musiconhold existed at the same
+ module priority level as the timing sources that it depends on.
+ This would cause a problem when music on hold was reloaded, as
+ the timing source could be changed after res_musiconhold was
+ processed. This patch adds a new module priority level,
+ AST_MODPRI_TIMING, that the various timing modules are now loaded
+ at. This now occurs before loading other resource modules, such
+ that the timing source is guaranteed to be set prior to resolving
+ the timing source dependencies. (closes issue ASTERISK-17474)
+ Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf,
+ Wes Van Tlghem, elguero, Thomas Arimont Patches:
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff
+ uploaded by elguero (License #5026)
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff
+ uploaded by elguero (License #5026)
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by
+ elguero (License #5026) Review:
+ https://reviewboard.asterisk.org/r/1578/
+
+2011-12-27 17:09 +0000 [r349144] Sean Bright <sean at malleable.com>
+
+ * main/audiohook.c: Once an audiohook is attached to a channel, we
+ continue to transcode all of the frames, even after all of the
+ hooks are detached. This patch short-cicuits us out before we
+ transcode unnecessarily.
+
+2011-12-23 17:25 +0000 [r349044] Sean Bright <sean at malleable.com>
+
+ * apps/app_chanspy.c: In ChanSpy, don't create audiohooks that will
+ never be used. When ChanSpy is initialized it creates and
+ attaches 3 audiohooks: 1) Read audio off of the channel that we
+ are spying on 2) Write audio to the channel that we are spying on
+ 3) Write audio to the channel that is bridged to the channel that
+ we are spying on. The first is always necessary, but the others
+ are used only when specific options are passed to the ChanSpy
+ application (B, d, w, and W to be specific). When those flags are
+ not passed, neither of those audiohooks are ever sent frames, but
+ we still try to process the hooks for each voice frame that we
+ recieve on the channel. So in short - only create and attach
+ audiohooks that we actually need.
+
+2011-12-23 15:24 +0000 [r348992] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_dial.c: Fix missing doc tags found while fixing
+ ASTERISK-18689 Add missing <variable></variable> tags in app_dial
+ documentation.
+
+2011-12-23 02:09 +0000 [r348940] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/pbx.h, main/pbx.c, channels/chan_sip.c: Fix
+ extension state callback references in chan_sip. Chan_sip gives a
+ dialog reference to the extension state callback and assumes that
+ when ast_extension_state_del() returns, the callback cannot
+ happen anymore. Chan_sip then reduces the dialog reference count
+ associated with the callback. Recent changes (ASTERISK-17760)
+ have resulted in the potential for the callback to happen after
+ ast_extension_state_del() has returned. For chan_sip, this could
+ be very bad because the dialog pointer could have already been
+ destroyed. * Added ast_extension_state_add_destroy() so chan_sip
+ can account for the sip_pvt reference given to the extension
+ state callback when the extension state callback is deleted. *
+ Fix pbx.c awkward statecbs handling in
+ ast_extension_state_add_destroy() and handle_statechange() now
+ that the struct ast_state_cb has a destructor to call. * Ensure
+ that ast_extension_state_add_destroy() will never return -1 or 0
+ for a successful registration. * Fixed pbx.c statecbs_cmp() to
+ compare the correct information. The passed in value to compare
+ is a change_cb function pointer not an object pointer. * Make
+ pbx.c ast_merge_contexts_and_delete() not perform callbacks with
+ AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
+ deadlocking when those locks are held during the callback. *
+ Removed unused lock declaration for the pbx.c store_hints list.
+ (closes issue ASTERISK-18844) Reported by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1635/
+
+2011-12-22 22:31 +0000 [r348888] Matthew Jordan <mjordan at digium.com>
+
+ * cel/cel_pgsql.c: Fix for memory leaks / cleanup in cel_pgsql
+ There were a number of issues in cel_pgsql's pgsql_log method: *
+ If either sql or sql2 could not be allocated, the method would
+ return while the pgsql_lock was still locked * If the execution
+ of the log statement succeeded, the sql and sql2 structs were
+ never free'd * Reconnection successes were logged as ERRORs. In
+ general, the severity of several logging statements was reduced
+ (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/
+
+2011-12-22 18:38 +0000 [r348833] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/frame.h: Allow packetization vaules > 127
+ According to the RTP packetization documentation, and the maximum
+ values listed in AST_FORMAT_LIST, we should support values > that
+ the signed char array that ast_codec_pref makes available to
+ store the value. All places in the code treat the framing field
+ as though it were an int array instaead of a char array anyway,
+ so this just fixes the type of the array. (closes issue
+ ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/
+
+2011-12-20 23:08 +0000 [r348735] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS number
+ if it is blank. Some ISDN switches complain or block the call if
+ the RDNIS number is empty. * Made chan_iax2 not save a RDNIS
+ number into the ast_channel if the string is blank. This is what
+ other channel drivers do. (closes issue ASTERISK-17152) Reported
+ by: rmudgett
+
+2011-12-19 21:31 +0000 [r348647] Richard Mudgett <rmudgett at digium.com>
+
+ * configure, configure.ac: Fix crashes on other platforms caused by
+ interference from Darwin weak symbol support. Support weak
+ symbols on a platform specific basis. The Mac OS X (Darwin)
+ support must be isolated from the other platforms because it has
+ caused other platforms to crash. Several other platforms
+ including Linux have GCC versions that define the weak attribute.
+ However, this attribute is only setup for use in the code by
+ Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
+ Review: https://reviewboard.asterisk.org/r/1617/
+
+2011-12-18 18:27 +0000 [r348516] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configs/sip.conf.sample, /: Correct two flaws in sip.conf.sample
+ related to AST-2011-013. * The sample file listed *two* values
+ for the 'nat' option as being the default. Only 'force_rport' is
+ the default. * The warning about having differing 'nat' settings
+ confusingly referred to both peers and users. ........ Merged
+ revisions 348515 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2011-12-16 23:51 +0000 [r348310-348464] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, main/features.c: Clean-up on isle five for
+ __ast_request_and_dial() and ast_call_forward(). * Add locking
+ when a channel inherits variables and datastores in
+ __ast_request_and_dial() and ast_call_forward(). Note: The
+ involved channels are not active so there was minimal potential
+ for problems. * Remove calls to ast_set_callerid() in
+ __ast_request_and_dial() and ast_call_forward() because the set
+ information is for the wrong direction. * Don't use C++ keywords
+ for variable names in ast_call_forward(). * Run the redirecting
+ interception macro if defined when forwarding a call in
+ ast_call_forward(). Note: Currently will never execute because
+ the only callers that supply a calling channel supply a hungup or
+ zombie channel. * Make feature_request_and_dial() put the
+ transferee into autoservice when it calls ast_call_forward() in
+ case a redirection interception macro is run. Note: Currently
+ will never happen because the caller channel (Party B) is always
+ hungup at this time. * Make feature_request_and_dial() ignore the
+ AST_CONTROL_PROCEEDING frame to silence a log message.
+
+ * main/channel.c: Fix cut and past error in ast_call_forward().
+ (issue ASTERISK-18836)
+
+ * include/asterisk/cdr.h, apps/app_followme.c, apps/app_queue.c,
+ res/res_monitor.c, main/channel.c, main/pbx.c,
+ apps/app_authenticate.c, funcs/func_cdr.c, main/features.c: Fix
+ crash during CDR update. The ast_cdr_setcid() and
+ ast_cdr_update() were shown in ASTERISK-18836 to be called by
+ different threads for the same channel. The channel driver thread
+ and the PBX thread running dialplan. * Add lock protection around
+ CDR API calls that access an ast_channel pointer. (closes issue
+ ASTERISK-18836) Reported by: gpluser Review:
+ https://reviewboard.asterisk.org/r/1628/
+
+ * apps/app_parkandannounce.c: Fix ParkAndAnnounce to pass the
+ CallerID to the announcing channel. ParkAndAnnounce tried to pass
+ the CallerID to the announcing channel but the ID was wiped out
+ by the channel masquerade done when parking the call. * Save the
+ CallerID before parking the channel to pass it to the announcing
+ channel. * Fixed a minor memory leak in ParkAndAnnounce. *
+ Updated some ParkAndAnnounce log messages.
+
+2011-12-14 22:01 +0000 [r348212] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_fax.c: Don't clear LOCALSTATIONID before sending or
+ receiving. The user may set that variable. ASTERISK-18921
+
+2011-12-14 20:34 +0000 [r348154-348157] Jonathan Rose <jrose at digium.com>
+
+ * configs/features.conf.sample: Fix accidental use of tabs instead
+ of spaces from previous features.conf.sample change
+
+ * configs/features.conf.sample: Document PARKINGSLOT variable in
+ features.conf.sample (issue ASTERISK-16239)
+
+2011-12-13 23:00 +0000 [r348101] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_followme.c, bridges/bridge_builtin_features.c: Fix
+ FollowMe CallerID on outgoing calls. The addition of the
+ Connected Line support changed how CallerID is passed to outgoing
+ calls. The FollowMe application was not updated to pass CallerID
+ to the outgoing calls. * Fix FollowMe CallerID on outgoing calls.
+ * Restructured findmeexec() to fix several memory leaks and
+ eliminate some duplicated code. * Made check the return value of
+ create_followme_number(). Putting a NULL into the numbers list is
+ bad if create_followme_number() fails. * Fixed a couple uses of
+ ast_strdupa() inside loops. * The changes to
+ bridge_builtin_features.c fix a similar CallerID issue with the
+ bridging API attended and blind transfers. (Not used at this
+ time.) (closes issue ASTERISK-17557) Reported by: hamlet505a
+ Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1612/
+
+2011-12-13 15:16 +0000 [r348048] Stefan Schmidt <sst at sil.at>
+
+ * channels/chan_sip.c: Fix possible misshandling of an incoming SIP
+ response as a peer poke response. Also make sure peer has even
+ qualify enabled when handle a peer poke response. (closes issue
+ ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
+ UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
+ by: David Vossel
+
+2011-12-12 19:22 +0000 [r347995] Terry Wilson <twilson at digium.com>
+
+ * res/res_srtp.c: Add a separate buffer for SRTCP packets The
+ function ast_srtp_protect used a common buffer for both SRTP and
+ SRTCP packets. Since this function can be called from multiple
+ threads for the same SRTP session (scheduler for SRTCP and
+ channel for SRTP) it was possible for the packets to become
+ corrupted as the buffer was used by both threads simultaneously.
+ This patch adds a separate buffer for SRTCP packets to avoid the
+ problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
+ Collins)
+
+2011-12-09 01:19 +0000 [r347811] Richard Mudgett <rmudgett at digium.com>
+
+ * main/pbx.c: Fix some parsing issues in
+ add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
+ potential sign extension issue. * Fix infinite loop in
+ add_exten_to_pattern_tree() handling of character set escape
+ handling. * Added buffer overflow checks in
+ add_exten_to_pattern_tree() character set collection. * Made
+ ignore empty character sets. * Added escape character handling to
+ end-of-range character in character sets. This has a slight
+ change in behavior if the end-of-range character is an escape
+ character. You must now escape it. * Fix potential sign extension
+ issue when expanding character set ranges. * Made remove
+ duplicated characters from character sets. The duplicate
+ characters lower extension matching priority and prevent
+ duplicate extension detection. * Fix escape character handling
+ when the escape character is trying to escape the end-of-string.
+ We could have continued processing characters after the end of
+ the exten string. We could have added the previous character to
+ the pattern matching tree incorrectly. (closes issue
+ ASTERISK-18909) Reported by: Luke-Jr
+
+2011-12-08 21:28 +0000 [r347718] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: Fix regression when using tcpenable=no and
+ tlsenable=yes. The tlsenable settings are tucked away in
+ main/tcptls.c, so I missed them when resolving ASTERISK-18837.
+ This should resolve the test suite breakage of the sip tls tests.
+ Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
+ Jordan
+
+2011-12-08 17:50 +0000 [r347595] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c: Mark channel running the h exten with the
+ soft-hangup flag. When a bridge is broken, ast_bridge_call()
+ might execute the h exten on the calling channel. However, that
+ channel may not have been the channel that broke the bridge by
+ hanging up. The channel executing the h exten must be in a hung
+ up state so things like AGI run in the correct mode. * Make sure
+ ast_bridge_call() marks the channel it is executing the h exten
+ on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
+ to match the pbx.c main dialplan execution loop when it executes
+ the h exten.) (closes issue ASTERISK-18811) Reported by: David
+ Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: David Hajek, rmudgett
+
+2011-12-08 16:19 +0000 [r347531] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Don't crash on INFO automon request with
+ no channel AST-2011-014. When automon was enabled in
+ features.conf, it was possible to crash Asterisk by sending an
+ INFO request if no channel had been created yet. (closes issue
+ ASTERISK-18805) ........ Merged revisions 347530 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2011-12-07 21:36 +0000 [r347438] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c: Update AMI Getvar and Setvar documentation about
+ supplying a channel name. (closes issue ASTERISK-18958) Reported
+ by: Red Patches: jira_asterisk_18958_v1.8.patch (license #5621)
+ patch uploaded by rmudgett
+
+2011-12-07 20:23 +0000 [r347369] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_meetme.c: Fix: Meetme recording variables from realtime
+ DB use null entries over channel variables Meetme would attempt
+ to substitute the realtime values of RECORDING_FILE and
+ RECORDING_FORMAT from the meetme db entry instead of using the
+ channel variable set for those variables in spite of those
+ database entries being NULL or even lacking a column to represent
+ them. (closes issue ASTERISK-18873) Reported by: Byron Clark
+ Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
+ 6157)
+
+2011-12-06 23:47 +0000 [r347292] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
+ signals case insensitive. (closes issue ASTERISK-18924) Reported
+ by: Kevin Taylor
+
+2011-12-06 21:44 +0000 [r347239] Jonathan Rose <jrose at digium.com>
+
+ * main/pbx.c: Documents CHANNEL(musicclass) taking priority over
+ m([x]) in waitExten If waitExten specifies a music class to use
+ with its music on hold option, it will use CHANNEL(musicclass)
+ instead if that channel variable has been set on the initiating
+ channel. This documents that behavior in the waitExten app so
+ that this can be known without checking the documentation of the
+ code in function local_ast_moh_start. (closes issue
+ ASTERISK-18804)
+
+2011-12-06 19:39 +0000 [r347111-347166] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: Don't allow transport=tcp when tcpenable=no.
+ When tcpenable=no, sending to transport=tcp hosts was still
+ allowed. Resolving the source address wasn't possible and yielded
+ the string "(null)" in SIP messages. Fixed that and a couple of
+ not-so-correct log messages. (closes issue ASTERISK-18837)
+ Reported by: Andreas Topp Review:
+ https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan
+
+ * apps/app_voicemail.c: Add regression tests for issue
+ ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
+ Reviewed by: Matt Jordan
+
+ * apps/app_voicemail.c: Move setting of voicemail zonetag and
+ locale up a bit. The voicemail [general] zonetag and locale
+ variables weren't loaded until after the mailboxes were
+ initialized. This caused the settings to be unset for those
+ mailboxes until a reload was performed. (closes issue
+ ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570
+ Reviewed by: Matt Jordan
+
+2011-12-06 17:05 +0000 [r347058] Matthew Jordan <mjordan at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: Fixed crash from
+ orphaned MWI subscriptions in chan_sip This patch resolves the
+ issue where MWI subscriptions are orphaned by subsequent SIP
+ SUBSCRIBE messages. When a peer is removed, either by pruning
+ realtime SIP peers or by unloading / loading chan_sip, the MWI
+ subscriptions that were orphaned would still be on the event
+ engine list of valid subscriptions but have a pointer to a peer
+ that no longer was valid. When an MWI event would occur, this
+ would cause a seg fault. (closes issue ASTERISK-18663) Reported
+ by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches:
+ blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
+ Review: https://reviewboard.asterisk.org/r/1610/
+
+2011-12-05 17:39 +0000 [r347006] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Restore call progress code for analog
+ ports. Extracting sig_analog from chan_dahdi lost call progress
+ detection functionality. * Fix analog ports from considering a
+ call answered immediately after dialing has completed if the
+ callprogress option is enabled. (closes issue ASTERISK-18841)
+ Reported by: Richard Miller Patches: chan_dahdi.diff (license
+ #5685) patch uploaded by Richard Miller (Modified by me)
+ sig_analog.c.diff (license #5685) patch uploaded by Richard
+ Miller (Modified by me) sig_analog.h.diff (license #5685) patch
+ uploaded by Richard Miller
+
+2011-12-05 14:56 +0000 [r346954] Jonathan Rose <jrose at digium.com>
+
+ * main/pbx.c: Resolve duplicate label used in multiple priorities
+ for the same extension. Prior to this patch, if labels with the
+ same name were used for different priorities in the same
+ extension, the new label would be accepted, but it would be
+ unusable since attempts to reach that label would just go to the
+ first one. Now pbx.c detects this, generates a warning in logs,
+ and culls the label before adding it to the dialplan. (closes
+ issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
+ pbx.c.patch uploaded by Kenneth Shumard (License 5077)
+
+2011-12-05 14:45 +0000 [r346951] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_jabber.exports.in: Fix chan_jingle/gtalk load regression
+ introduced in r346087 Add missing symbol exports for
+ ast_aji_client_destroy and ast_aji_buddy_destroy for usage
+ outside res_jabber. Testing of these changes focused on
+ res_jabber itself, so this problem was missed. Reported-by:
+ Michael Spiceland
+
+2011-12-04 09:57 +0000 [r346899] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
+ domain ACL bypass. The code that allowed admins to create users
+ with domain-only uri's had stopped to work in 1.8 because of the
+ reqresp parser rewrites. This is fixed now: if you have a
+ [mydomain.com] sip user, you can register with useraddr
+ sip:mydomain.com. Note that in that case -- if you're using
+ domain ACLs (a configured domain list) -- mydomain.com must be in
+ the allow list as well. Reviewboard r1606 shows a list of
+ registration combinations and which SIP response codes are
+ returned. Review: https://reviewboard.asterisk.org/r/1533/
+ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
+ issue ASTERISK-18741)
+
+2011-12-02 16:19 +0000 [r346762] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, channels/chan_h323.c: process null frame
+ pointer returned by ast_rtp_instance_read correctly (closes issue
+ ASTERISK-16697) Reported by: under Patches: segfault.diff
+ (License #5871) patch uploaded by under
+
+2011-12-01 21:11 +0000 [r346700] Richard Mudgett <rmudgett at digium.com>
+
+ * configs/res_stun_monitor.conf.sample, include/asterisk/stun.h,
+ main/stun.c, res/res_stun_monitor.c: Re-resolve the STUN address
+ if a STUN poll fails for res_stun_monitor. The STUN socket must
+ remain open between polls or the external address seen by the
+ STUN server is likely to change. However, if the STUN request
+ poll fails then the STUN server address needs to be re-resolved
+ and the STUN socket needs to be closed and reopened. * Re-resolve
+ the STUN server address and create a new socket if the STUN
+ request poll fails. * Fix ast_stun_request() return value
+ consistency. * Fix ast_stun_request() to check the received
+ packet for expected message type and transaction ID. * Fix
+ ast_stun_request() to read packets until timeout or an associated
+ response packet is found. The stun_purge_socket() hack is no
+ longer required. * Reduce ast_stun_request() error messages to
+ debug output. * No longer pass in the destination address to
+ ast_stun_request() if the socket is already bound or connected to
+ the destination. (closes issue ASTERISK-18327) Reported by:
+ Wolfram Joost Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1595/
+
+2011-12-01 20:36 +0000 [r346564-346697] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
+ ringing. 183 Ringing isn't even a thing. 183 is actually a
+ session progress message. (closes issue ASTERISK-18925) Reported
+ by: Sebastian Denz Tested by: jrose Patches:
+ asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
+ Denz (License #6139)
+
+ * include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c:
+ r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) |
+ 18 lines Cleaning up chan_sip/tcptls file descriptor closing.
+ This patch attempts to eliminate various possible instances of
+ undefined behavior caused by invoking close/fclose in situations
+ where fclose may have already been issued on a
+ tcptls_session_instance and/or closing file descriptors that
+ don't have a valid index for fd (-1). Thanks for more than a
+ little help from wdoekes. (closes issue ASTERISK-18700) Reported
+ by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane
+ Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas
+ Review: https://reviewboard.asterisk.org/r/1576/
+
+2011-11-30 19:36 +0000 [r346472] Leif Madsen <lmadsen at digium.com>
+
+ * configs/queues.conf.sample: Update queues.conf.sample
+ documentation. Update the documentation surrounding the use of
+ MONITOR_EXEC to make it more clear that it can be used for both
+ Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
+ Reported by: David Woolley Patches:
+ issue18817_mixmonitor_queues_doc.diff by Michael L. Young
+ (License #5026)
+
+2011-11-28 14:30 +0000 [r346292] Stefan Schmidt <sst at sil.at>
+
+ * res/res_rtp_asterisk.c: Fix regression that 'rtp/rtcp set debup
+ ip' only works when also a port was specified. (closes issue
+ ASTERISK-18693) Reported by: Davide Dal Fra Review:
+ https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
+ Doekes
+
+2011-11-23 22:52 +0000 [r346239] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/acl.h,
+ channels/chan_skinny.c, channels/chan_h323.c, main/acl.c: Fix
+ calls to ast_get_ip() not initializing the address family.
+
+2011-11-23 20:15 +0000 [r346144-346147] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
+ function. In r116240, get_msg_text() got an extra parameter to
+ fix the unwanted addition of trailing newlines to SIP MESSAGE
+ bodies. This caused all linefeeds to be trimmed, which isn't
+ right either. This is a stop-gap; the right fix is to return the
+ original SIP request body. Review:
+ https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan
+
+ * include/asterisk/strings.h: Fix ast_str_truncate signedness
+ warning and documentation. Review:
+ https://reviewboard.asterisk.org/r/1594
+
+2011-11-23 17:12 +0000 [r346086] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_gtalk.c, res/res_jabber.c, channels/chan_jingle.c,
+ include/asterisk/jabber.h: Fix res_jabber resource leaks This
+ should fix almost all resource leaks in res_jabber that involve
+ ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where
+ ast_aji_get_client would sometimes bump an object's refcount and
+ sometimes not. Review: https://reviewboard.asterisk.org/r/1553
+
+2011-11-23 16:09 +0000 [r346030] Terry Wilson <twilson at digium.com>
+
+ * res/res_musiconhold.c: Resume playing existing hold music for
+ cached realtime MOH As a result of the fix for ASTERISK-18039,
+ realtime caching MOH no longer properly resumes playing back a
+ file between different holds in the same call. This is because
+ scanning for new files causes the existing file array to be
+ emptied and we were just comparing that the saved pointer to the
+ filename matched the pointer to the filename in a particular
+ position in the array. An easy fix is to save the filename
+ instead of a pointer to it and then do a strcmp instead of
+ comparing the addresses. (closes issue ASTERISK-18912) Review:
+ https://reviewboard.asterisk.org/r/1596/
+
+2011-11-22 22:55 +0000 [r345976] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/dnsmgr.h, main/dnsmgr.c: Fix dnsmgr entries to
+ ask for the same address family each time. The dnsmgr refresh
+ would always get the first address found regardless of the
+ original address family requested. So if you asked for only IPv4
+ addresses originally, you might get an IPv6 address on refresh. *
+ Saved the original address family requested by
+ ast_dnsmgr_lookup() to be used when the address is refreshed.
+
+2011-11-22 20:29 +0000 [r345923] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * include/asterisk/logger.h: Clarify why the AST_LOG_* macros exist
+ next to the LOG_* macros. (issue ASTERISK-17973)
+
+2011-11-21 21:03 +0000 [r345828-345829] Terry Wilson <twilson at digium.com>
+
+ * CHANGES: Change nat=yes to nat=force_rport in CHANGES Fix a small
+ documentation merge issue ASTERISK-18862
+
+ * configs/sip.conf.sample, CHANGES, /, channels/chan_sip.c: Default
+ to nat=yes; warn when nat in general and peer differ It is
+ possible to enumerate SIP usernames when the general and
+ user/peer nat settings differ in whether to respond to the port a
+ request is sent from or the port listed for responses in the Via
+ header. In 1.4 and 1.6.2, this would mean if one setting was
+ nat=yes or nat=route and the other was either nat=no or
+ nat=never. In 1.8 and 10, this would mean when one was
+ nat=force_rport and the other was nat=no. In order to address
+ this problem, it was decided to switch the default behavior to
+ nat=yes/force_rport as it is the most commonly used option and to
+ strongly discourage setting nat per-peer/user when at all
+ possible. For more discussion of the issue, please see:
+ http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
+ (closes issue ASTERISK-18862) Review:
+ https://reviewboard.asterisk.org/r/1591/ ........ Merged
+ revisions 345776 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged
+ revisions 345800 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2011-11-19 15:08 +0000 [r345682] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * main/db.c: Update the documentation to better clarify how the
+ existing commands work. Review:
+ https://reviewboard.asterisk.org/r/1593/
+
+2011-11-17 17:06 +0000 [r345546] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Remove dead code since pri_grab() can never
+ fail. Dead code makes programmers sick. I am sick of looking at
+ it.
+
+2011-11-17 17:04 +0000 [r345545] Jason Parker <jparker at digium.com>
+
+ * apps/app_confbridge.c: Fix documentation of 's' option. The menu
+ key is #, not *. Reported by p3nguin on #asterisk.
+
+2011-11-16 14:42 +0000 [r345487] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_voicemail.c: Guarantee messages go into the right
+ folders with multiple recipients Before, using the U flag in
+ Voicemail with multiple recipients would put urgent messages in
+ the INBOX folder for all users past the first thanks to a bug
+ with the message copying function. This would also cause messages
+ to fail to be sent if the INBOX directory hadn't been created for
+ that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt
+ Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1589/
+
+2011-11-15 20:09 +0000 [r345219-345431] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_agi.c: Make FastAGI HANGUP show up in AGI debug output. *
+ Change from using send() to ast_agi_send() so the HANGUP shows up
+ in the AGI debug output. (closes issue ASTERISK-18723) Reported
+ by: James Van Vleet Patches: jira_asterisk_18723_v1.8.patch
+ (license #5621) patch uploaded by rmudgett
+
+ * channels/sig_pri.c: Fix typo in sig_pri using wrong structure
+ name. It is fortunate that the typo does not alter generated code
+ since the e->restart.channel and e->ring.channel members are in
+ the same position. (closes issue ASTERISK-18868) Reported by:
+ zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by
+ zvision
+
+ * apps/app_queue.c: Make queue log indicate if ADDMEMBER is paused
+ for AMI and realtime. * Add parameter to queue log ADDMEMBER to
+ indicate if the member is paused. (closes issue ASTERISK-18645)
+ Reported by: garlew Patches: paused.diff (License #5337) patch
+ uploaded by garlew Tested by: rmudgett, garlew Review:
+ https://reviewboard.asterisk.org/r/1469/
+
+ * UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/chan_sip.c: Restore SIP DTMF overlap dialing method. The
+ recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
+ working correctly removed a long standing ability to do overlap
+ dialing using DTMF in the early media phase of a call. See
+ ASTERISK-18702 it has a very good description of the issue. I
+ started with Pavel Troller's chan_sip.diff patch on issue
+ ASTERISK-18702. * Added 'dtmf' enum value to sip.conf
+ allowoverlap config option. The new option value causes the
+ Incomplte application to not send anything with chan_sip so the
+ caller can supply more digits via DTMF. * Renames
+ SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
+ since that is what it really means. * Fixed get_destination()
+ inconsistency with the pickup extension matching. * Fixed
+ initialization of PAGE3 of global_flags in reload_config().
+ (closes issue ASTERISK-18702) Reported by: Pavel Troller Review:
+ https://reviewboard.asterisk.org/r/1517/ Review:
+ https://reviewboard.asterisk.org/r/1582/
+
+ * main/pbx.c: Fix Progress spelling error in main/pbx.c. (closes
+ issue ASTERISK-18857) Reported by: David M Patches:
+ mainpbx-trivial.patch (License #6326) patch uploaded by David M
+
+2011-11-14 19:05 +0000 [r345163] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Don't read past end of input when calling write()
+ int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames *
+ sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when
+ new_frames == 1. Otherwise we start reading into adjacent
+ variables declared on the stack. The read end discards what is
+ read, so the values don't matter but it's not a good idea to read
+ past where we want even though new_frames is almost always 1 and
+ should never be large. This patch is basically taken out of
+ kpfleming's eventfd branch, as he mentioned that he remembered
+ fixing it there when I talked to him about this issue. Review:
+ https://reviewboard.asterisk.org/r/1583/
+
+2011-11-14 19:00 +0000 [r345160] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/sip/include/reqresp_parser.h: Update reqresp_parser
+ parse_uri doxygen comments. The issue mentioned in the bug report
+ had been fixed recently by twilson. The reporter included this
+ documentation fix. (closes issue ASTERISK-18572) Reported by:
+ Richard Miller Patch by: Richard Miller (modified)
+
+2011-11-14 15:08 +0000 [r345063] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Ensure that a null vmexten does not cause a
+ segfault When sip_send_mwi_to_peer was modified recently to avoid
+ deadlocks, vmexten was not expected to be null. This change
+ handles that situation to avoid a segfault.
+
+2011-11-14 15:00 +0000 [r345062] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_voicemail.c: Moves voicemail setup password entry to the
+ end of the setup process. This change was made because
+ forcegreeting and forcename settings in voicemail could be
+ circumvented by hanging up after entering a password, because the
+ only way voicemail currently observes whether a mailbox is new or
+ not is by checking to see if the password is the same as the
+ mailbox number or not. (closes issue ASTERISK-18282) Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/
+
+2011-11-12 16:05 +0000 [r344965] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * channels/chan_misdn.c: mISDN Round Robin break when no channel is
+ available Prevent channels been parsed repetitively.
+
+2011-11-12 00:24 +0000 [r344899] Terry Wilson <twilson at digium.com>
+
+ * res/res_musiconhold.c: Don't forget to rescan MOH files for
+ cached realtime classes Realtime MOH class caching was
+ implemented because without it, you would build a completely new
+ MOH class and would start the music over at the beginning each
+ time hold was pressed in a conversation. Unfortunately, this
+ broke re-scanning for file changes for realtime MOH classes. This
+ patch corrects that issue. (closes issue ASTERISK-18039) Review:
+ https://reviewboard.asterisk.org/r/1579/
+
+2011-11-11 21:54 +0000 [r344835-344843] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * main/utils.c, include/asterisk/stringfields.h,
+ include/asterisk/utils.h: Use __alignof__ instead of sizeof for
+ stringfield length storage. Kevin P Fleming suggested that
+ r343157 should use __alignof__ instead of sizeof. For most
+ systems this won't be an issue, but better fix it now while it's
+ still fresh. Review: https://reviewboard.asterisk.org/r/1573
+
+ * channels/sip/reqresp_parser.c: Remove unneeded if(params) checks
+ in reqresp_parser. Nick Lewis added them in
+ https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent
+ reason. There is no way that params could become NULL in that
+ piece of code, so I removed these excess checks again.
+
+ * main/manager.c: Fix bad quoting of multiline mxml opaque_data
+ that caused invalid xml. The opaque_data was added and enclosed
+ in single quotes, assuming it would be only a single line. The
+ rest of the lines were appended after the closing quote. (closes
+ issue ASTERISK-18852) Reported by: peep_ on IRC Review:
+ https://reviewboard.asterisk.org/r/1577
+
+2011-11-11 20:42 +0000 [r344823] Matthew Jordan <mjordan at digium.com>
+
+ * main/file.c: Video format was treated as audio when removed from
+ the file playback scheduler This patch fixes the format type
+ check in ast_closestream and filestream_destructor. Previously a
+ comparison operator was used, but since audio formats are no
+ longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats
+ that have a value greater than the video formats), a bitwise AND
+ operation is used instead. Duplicated code was also moved to
+ filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo
+ Bedrij Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1580/
+
+2011-11-11 20:10 +0000 [r344769] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Fix regression introduced by SDP fixups If
+ capability is adjusted when switching to UDPTL during fax
+ transmission, fax teardown fails. Make sure capability is only
+ touched if RTP is active. This regression was introduced in
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