[svn-commits] lmadsen: tag 1.4.26-rc1 r197689 - /tags/1.4.26-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu May 28 12:13:54 CDT 2009
Author: lmadsen
Date: Thu May 28 12:13:50 2009
New Revision: 197689
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=197689
Log:
Importing files for 1.4.26-rc1 release.
Added:
tags/1.4.26-rc1/.lastclean (with props)
tags/1.4.26-rc1/.version (with props)
tags/1.4.26-rc1/ChangeLog (with props)
Added: tags/1.4.26-rc1/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.4.26-rc1/.lastclean?view=auto&rev=197689
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--- tags/1.4.26-rc1/ChangeLog (added)
+++ tags/1.4.26-rc1/ChangeLog Thu May 28 12:13:50 2009
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+2009-05-28 15:51 +0000 [r197620] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: 'iax show peer blah' now outputs whether or
+ not peer 'blah' is in trunk mode or not.
+
+2009-05-28 15:27 +0000 [r197588] Mark Michelson <mmichelson at digium.com>
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Allow
+ for media to arrive from an alternate source when responding to a
+ reinvite with 491. When we receive a SIP reinvite, it is possible
+ that we may not be able to process the reinvite immediately since
+ we have also sent a reinvite out ourselves. The problem is that
+ whoever sent us the reinvite may have also sent a reinvite out to
+ another party, and that reinvite may have succeeded. As a result,
+ even though we are not going to accept the reinvite we just
+ received, it is important for us to not have problems if we
+ suddenly start receiving RTP from a new source. The fix for this
+ is to grab the media source information from the SDP of the
+ reinvite that we receive. This information is passed to the RTP
+ layer so that it will know about the alternate source for media.
+ Review: https://reviewboard.asterisk.org/r/252
+
+2009-05-28 15:21 +0000 [r197562] Eliel C. Sardanons <eliels at gmail.com>
+
+ * channels/chan_sip.c: Use the address we already know when
+ reloading a peer with nat=yes. If we already have an address for
+ a peer, and we are reloading the sip configuration, try to use
+ that address to contact the peer, instead of getting it from the
+ Contact. (closes issue #15194) Reported by: ibc Patches:
+ sip.patch uploaded by eliel (license 64) Tested by: manwe
+
+2009-05-28 14:49 +0000 [r197537] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_chanspy.c, include/asterisk/audiohook.h,
+ main/audiohook.c: Add flags to chanspy audiohook so that audio
+ stays in sync. There are two flags being added to the chanspy
+ audiohook here. One is the pre-existing
+ AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
+ the read and write slinfactories on the audiohook do not skew
+ beyond a certain tolerance. In addition, there is a new audiohook
+ flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
+ we do not allow for a slinfactory to build up a substantial
+ amount of audio before flushing it. For this particular issue,
+ this means that the person spying on the call will hear the
+ conversations in real time with very little delay in the audio.
+ (closes issue #13745) Reported by: geoffs Patches: 13745.patch
+ uploaded by mmichelson (license 60) Tested by: snblitz
+
+2009-05-28 13:44 +0000 [r197466] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix a bug where the flag indicating the
+ presence of rport would get overwritten by the nat setting. The
+ presence of rport is now stored as a separate flag. Once the
+ dialog is setup and authenticated (or it passes through
+ unauthenticated) the proper nat flag is set. (closes issue
+ #13823) Reported by: dimas
+
+2009-05-27 20:12 +0000 [r197264] Sean Bright <sean.bright at gmail.com>
+
+ * Makefile: Use bash explicitly when calling
+ build_tools/mkpkgconfig from the Makefile. Since we use bashisms
+ in build_tools/mkpkgconfig, we should call on bash explicitly
+ when running from the Makefile, otherwise we get errors during a
+ 'make install.' (closes issue #15209) Reported by: seandarcy
+
+2009-05-27 20:07 +0000 [r197259] Olle Johansson <oej at edvina.net>
+
+ * doc/asterisk-conf.txt: Typo fix
+
+2009-05-27 19:09 +0000 [r197194] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_cut.c: Use a different determinator on whether to
+ print the delimiter, since leading fields may be blank. (closes
+ issue #15208) Reported by: ramonpeek Patch by me, though inspired
+ in part by a patch from ramonpeek
+
+2009-05-27 16:49 +0000 [r197124] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, include/asterisk/channel.h: Fix broken attended
+ transfers The bridge was terminating immediately after the
+ attended transfer was completed. The problem was because upon
+ reentering ast_channel_bridge nexteventts was checked to see if
+ it was set and if so could possibly return AST_BRIDGE_COMPLETE.
+ (closes issue #15183) Reported by: andrebarbosa Tested by:
+ andrebarbosa, tootai, loloski
+
+2009-05-27 13:54 +0000 [r197024] Sean Bright <sean.bright at gmail.com>
+
+ * apps/app_queue.c: Fix handling of the 'state_interface' option of
+ the 'queue add member' CLI command. This change relates to
+ r184980, which was a backport of the state interface changes to
+ app_queue from trunk. trunk and all of the 1.6.x branches are not
+ affected. 'queue add member' allows for specifying an interface
+ to use for device state when adding a queue member via CLI, but
+ the validation code was not properly updated to reflect this
+ optional argument. (closes issue #15198) Reported by: loloski
+ Patches: 05272009_app_queue.diff uploaded by seanbright (license
+ 71) Tested by: loloski
+
+2009-05-26 18:14 +0000 [r196826] Russell Bryant <russell at digium.com>
+
+ * res/res_convert.c: Resolve a file handle leak. The frames here
+ should have always been freed. However, out of luck, there was
+ never any memory leaked. However, after file streams became
+ reference counted, this code would leak the file stream for the
+ file being read. (closes issue #15181) Reported by: jkroon
+
+2009-05-26 13:06 +0000 [r196657] Joshua Colp <jcolp at digium.com>
+
+ * contrib/scripts/safe_asterisk: Remove some bash specific stuff
+ from safe_asterisk. (closes issue #10812) Reported by: paravoid
+ Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license
+ 46)
+
+2009-05-22 13:54 +0000 [r196116] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_misdn.c: Fix a bug where using immediate with mISDN
+ caused a cause code of 16 to get sent back instead of 1 if the
+ 's' extension did not exist. (closes issue #12286) Reported by:
+ lmamane
+
+2009-05-21 19:04 +0000 [r195991] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Sign problem calculating timestamp for iax
+ frame leads to no audio on the receiving peer. There are rare
+ cases in which a frame's delivery timestamp is slightly less than
+ the iax2_pvt's offset. This causes the pvt's timestamp to be a
+ small negative number, but since the timestamp value is unsigned
+ it looks like a huge positive number. This patch checks for this
+ negative case and sets the ms to zero. A similar check is already
+ done right below this one in the 'else' statement. (closes issue
+ #15032) Reported by: guillecabeza Patches:
+ chan_iax2.c.patch_timestamp uploaded by guillecabeza (license
+ 380) Tested by: guillecabeza (closes issue #14216) Reported by:
+ Andrey Sofronov
+
+2009-05-21 15:25 +0000 [r195881] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/cdr.c, res/res_features.c, include/asterisk/cdr.h: This
+ commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and
+ AST_CDR_FLAG_LOCKED from being updated in certain cases. This is
+ accomplished by adding two functions to update the answer time
+ and disposition of calls that checks for the proper lock flags.
+ These functions are used in the ast_bridge_call() function so
+ that ForkCDR(A) calls are respected. This patch also modifies the
+ way ast_bridge_call() chooses the cdr record to base the
+ bridged_cdr on. Previously the first unlocked cdr record would be
+ chosen, now instead the first cdr record is chosen and forked cdr
+ records are moved to the bridge_cdr. This allows the original cdr
+ record and any forked cdr records to be properly updated with
+ answer and end times. (closes issue #13797) Reported by: sh0t
+ Tested by: sh0t (closes issue #14744) Reported by: deepesh
+
+2009-05-20 17:30 +0000 [r195635-195688] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: Fix some code that wrongly assumed a pointer
+ would always be non-NULL when dealing with CDRs after a bridge.
+ (closes issue #15079) Reported by: barryf
+
+ * apps/app_meetme.c: Fix a bug where the MeetMe option 'D' did not
+ actually prompt for the pin. (closes issue #15050) Reported by:
+ pmhaddad
+
+2009-05-19 20:12 +0000 [r195520] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Ensure thread keys are initialized before
+ attempting to access them. (closes issue #14889) Reported by:
+ jaroth Patches: app_voicemail.c.patch uploaded by msirota
+ (license 758) Tested by: msirota, BlargMaN
+
+2009-05-19 14:41 +0000 [r195448] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix a bug where direct RTP setup would
+ partially occur even when disabled if the calling channel was
+ answered. (issue #13545) Reported by: davidw (issue #14244)
+ Reported by: mbnwa
+
+2009-05-18 20:24 +0000 [r195366] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, res/res_smdi.c: Add a similar dependency on
+ SMDI for voicemail as already exists for ADSI. (closes issue
+ #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt
+ uploaded by tilghman (license 14)
+ 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman
+ (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by
+ tilghman (license 14)
+
+2009-05-18 15:51 +0000 [r195095-195206] Joshua Colp <jcolp at digium.com>
+
+ * main/frame.c: Fix a typo which caused loss of audio when using
+ G729 in some scenarios with a smoother present. (closes issue
+ #15105) Reported by: bamby Patches: process-vad-correctly.diff
+ uploaded by bamby (license 430)
+
+ * main/rtp.c, channels/chan_sip.c: Fix a bug where the codecs of
+ the called party leg were not properly sent back to the caller
+ call leg when reinvited. (closes issue #13569) Reported by:
+ bkw918
+
+2009-05-18 12:57 +0000 [r195020] Russell Bryant <russell at digium.com>
+
+ * main/manager.c: Don't try to unlock a bogus channel. (closes
+ issue #15144) Reported by: cristiandimache
+
+2009-05-15 22:43 +0000 [r194873] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: IAX2 REGAUTH loop IAX was not sending
+ REGREJ to terminate invalid registrations. Instead it sent
+ another REGAUTH if the authentication challenge failed. This
+ caused a loop of REGREQ and REGAUTH frames. (Related to Security
+ fix AST-2009-001) (closes issue #14867) Reported by: aragon
+ Tested by: dvossel (closes issue #14717) Reported by: mobeck
+ Patches: regauth_loop_update_patch.diff uploaded by dvossel
+ (license 671) Tested by: dvossel
+
+2009-05-15 18:43 +0000 [r194764] Russell Bryant <russell at digium.com>
+
+ * configs/logger.conf.sample: Fix some spelling fail.
+
+2009-05-15 15:40 +0000 [r194557-194685] David Vossel <dvossel at digium.com>
+
+ * channels/iax2-parser.c, channels/chan_iax2.c: Update to previous
+ IAX2 "Ghost" Channels patch. Fixed some comments made on
+ reviewboard for the previous patch. (issue #14207)
+
+ * channels/iax2-parser.c, channels/iax2-parser.h,
+ channels/chan_iax2.c: IAX2 "Ghost" Channels There is a bug
+ tracker issue where people are reporting "Ghost" channels in
+ their 'iax2 show channels' output. The confusion is caused by
+ channels being listed as "(NONE)" with format "unknown". These
+ are not channels of coarse. They are usually just pending
+ registration or poke requests, but it is confusing output. To
+ help make sense of this I have added two columns to 'iax2 show
+ channels'. One shows the first message which started the
+ transaction, and the second shows the last message sent by either
+ side of the call. This helps diagnose why the entry exists and
+ why it may not go away. (closes issue #14207) Reported by:
+ clive18 Review: https://reviewboard.asterisk.org/r/246/
+
+2009-05-14 22:23 +0000 [r194509] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /: Update URL to Reviewboard
+
+2009-05-14 22:17 +0000 [r194484] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix a race condition where a reinvite could
+ trigger a 482 response. The loop detection/spiral detection code
+ in chan_sip used the owner channel's state as a criterion for
+ determining if the incoming INVITE is a looped request. The
+ problem with this is that the INVITE-handling code happens in a
+ different thread than the thread that marks the owner channel as
+ being up. As a result, if a reinvite were to come in very
+ quickly, say from another Asterisk on the same LAN, it was
+ possible for the reinvite to arrive before the owner channel had
+ been set to the up state. This patch corrects the problem by
+ using the invitestate of the sip_pvt instead, since that can be
+ guaranteed to be set correctly by the time the reinvite arrives.
+ Since there is a switch statement further in the INVITE-handling
+ code, the AST_STATE_RINGING state also checks the invitestate of
+ the sip_pvt in case we should actually be treating the channel as
+ if it were up already. (closes issue #12215) Reported by: jpyle
+ Patches: 12215_confirmed.patch uploaded by mmichelson (license
+ 60) Tested by: lmadsen
+
+2009-05-13 19:41 +0000 [r194356] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Remove an extraneous unlocking operation from
+ ast_channel_free. In the case that we could not remove the
+ desired channel from the list of channels, there was an extra
+ call to unlock the channel list. Since we unlock the list later
+ on in the function anyway, this results in the list being
+ unlocked twice yet only being locked once. (closes issue #15098)
+ Reported by: tim_ringenbach Patches: remove_extra_unlock.diff
+ uploaded by tim (license 540)
+
+2009-05-13 16:18 +0000 [r194322] Doug Bailey <dbailey at digium.com>
+
+ * main/pbx.c: Pull in a piece of murf's 88166 patch that makes it
+ safe to call pbx_substitute_variables_helper_full with a
+ non-zero'd buffer
+
+2009-05-21 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.25
+
+2009-05-13 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.25-rc1
+
+2009-05-13 13:38 +0000 [r194208] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Fix RFC2833 issues with DTMF getting duplicated and
+ with duration wrapping over. (closes issue #14815) Reported by:
+ geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88)
+ Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue
+ #14460) Reported by: moliveras Tested by: moliveras
+
+2009-05-13 00:52 +0000 [r194137] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c: Fix logic for how to proceed with a single digit
+ extension. (closes issue #15091) Reported by: andrew Patches:
+ 20090512__issue15091.diff.txt uploaded by tilghman (license 14)
+ Tested by: andrew
+
+2009-05-12 22:15 +0000 [r194028] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: This change modifies app_queue to properly
+ generate CDR records in failure situations. This involves setting
+ a proper cdr disposition coresponding to the given failure
+ condition and ensuring the proper information is stored in the
+ cdr record. (closes issue #13691) Reported by: dferrer Tested by:
+ mnicholson (closes issue #13637) Reported by: atis Tested by:
+ atis
+
+2009-05-12 20:39 +0000 [r193955] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Avoid initializing routines if the
+ authentication fails. Fixes a crash (RR) issue. (closes issue
+ #14508) Reported by: tiziano Patches:
+ 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license
+ 377)
+
+2009-05-12 18:18 +0000 [r193880] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Set the invitestate to INV_CANCELLED only if
+ we are actually sending a SIP CANCEL. The problem was that the
+ hangup code was setting the invitestate too early. The result of
+ this was that we would always send a CANCEL request, even if it
+ was not an appropriate time to do so (e.g. we have not yet
+ received a provisional response for our INVITE). Note that this
+ same fix had been applied to trunk and the 1.6.X branches
+ starting with revision 155467. This is why you will see this
+ revision being blocked from those places. AST-216
+
+2009-05-11 22:48 +0000 [r193755] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Move 300 bytes around on the stack, to make
+ more room for an extension buffer. This allows more concurrent
+ extensions to be copied for a single voicemail, without creating
+ a possibility of upsetting existing users, where a dialplan could
+ run out of stack space where it had run fine before.
+ Alternatively, we could have allocated off the heap, but that is
+ a larger change and would have increased the chance for
+ instability introduced by this change. This is really solved
+ starting in 1.6.0.11, as the use of an ast_str buffer allows an
+ unlimited number of extensions (up to available memory). We
+ additionally create a new warning message when the buffer length
+ is exceeded, permitting administrators to see an issue after the
+ fact, whereas previously the list was silently truncated. (closes
+ issue #14739) Reported by: p_lindheimer Patches:
+ 20090417__bug14739.diff.txt uploaded by tilghman (license 14)
+ Tested by: p_lindheimer
+
+2009-05-11 19:09 +0000 [r193613] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Sent wrong message to clear a call we
+ started if the other end has not responed yet. In the state
+ MISDN_CALLING (i.e. SETUP was sent but no answer has arrived
+ yet), it is not allowed to clear the call with RELEASE_COMPLETE.
+ It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only
+ allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a,
+ 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer.
+ JIRA ABE-1862
+
+2009-05-11 17:35 +0000 [r193544] Leif Madsen <lmadsen at digium.com>
+
+ * funcs/func_channel.c: Document CHANNEL(transfercapability) in CLI
+ documentation. (issue #15073) Reported by: pkempgen Patches:
+ 20090511__issue15073.diff.txt uploaded by tilghman (license 14)
+
+2009-05-08 21:01 +0000 [r193391] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c: Set the proper disposition on originated calls.
+ (closes issue #14167) Reported by: jpt Patches:
+ call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
+ Tested by: dlotina, rmartinez, mnicholson
+
+2009-05-08 14:51 +0000 [r193262] David Vossel <dvossel at digium.com>
+
+ * channels/misdn_config.c: "misdn show config" segfaults asterisk,
+ if no MSN lists (closes issue #14976) Reported by: alecdavis
+ Patches: misdn_config.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis, FabienToune
+
+2009-05-08 14:03 +0000 [r193193] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configs/logger.conf.sample, main/logger.c: Make absolute paths
+ for logger channels work properly (Note: This is not a new
+ feature, it was previously undocumented and broken.) The Asterisk
+ logger has a feature to support absolute pathnames for logger
+ channels, but the code implementing the feature was broken. This
+ has been fixed, and the absolute path feature is now documented
+ in the sample logger.conf.
+
+2009-05-07 23:41 +0000 [r193119] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c: Fix Background within a Macro for FreePBX. If the
+ single digit DTMF is an extension in the specified context, then
+ go there and signal no DTMF. Otherwise, we should exit with that
+ DTMF. If we're in Macro, we'll exit and seek that DTMF as the
+ beginning of an extension in the Macro's calling context. If
+ we're not in Macro, then we'll simply seek that extension in the
+ calling context. Previously, someone complained about the
+ behavior as it related to the interior of a Gosub routine, and
+ the fix (#14011) inadvertently broke FreePBX (#14940). This
+ change should fix both of these situations, but with the possible
+ incompatibility that if a single digit extension does not exist
+ (but a longer extension COULD have matched), it would have
+ previously gone immediately to the "i" extension, but will now
+ need to wait for a timeout. (closes issue #14940) Reported by:
+ p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
+ tilghman (license 14) Tested by: p_lindheimer
+
+2009-05-07 22:17 +0000 [r193050] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Give a more helpful message when an
+ incoming call's dialed extension does not match. Added the dialed
+ extension and context to the chan_misdn messages warning that the
+ dialed number cannot be matched in the dialplan.
+
+2009-05-07 16:29 +0000 [r192932] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Eliminate repetition of fullcontact during
+ reconstruction. If the fullcontact field appears in both the
+ sippeers and the sipregs table, then during reconstruction of the
+ field, it will otherwise be doubled. (closes issue #14754)
+ Reported by: Alexei Gradinari Patches:
+ 20090506__bug14754.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen
+
+2009-05-06 22:15 +0000 [r192858] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_features.c: Make ParkedCall application stop execution of
+ the dialplan after hang up Just changed park_exec to always
+ return non-zero. I really wasn't entirely sure at first if this
+ was a bug. Decided it was since it would be surprising when not
+ using ParkedCall in the dialplan to hang up and have dialplan
+ execution continue. (closes issue #14555) Reported by:
+ francesco_r
+
+2009-05-06 13:30 +0000 [r192633] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Update some old logic to stop both begin and
+ end DTMF frames from reaching the core if rfc2833 is not enabled.
+ (closes issue #15036) Reported by: dimas Patches: v1-15036.patch
+ uploaded by dimas (license 88)
+
+2009-05-05 19:56 +0000 [r192524] Sean Bright <sean.bright at gmail.com>
+
+ * static-http/astman.js: Fix Javascript error when using astman.js
+ in Internet Explorer. Internet Explorer (tested with 7.0) does
+ not like trailing commas on constructs like object initializers,
+ so get rid of them to avoid some errors. (closes issue #15026)
+ Reported by: rajnishgiri Patches: bug15026.patch uploaded by
+ seanbright (license 71) Tested by: seanbright
+
+2009-05-05 18:22 +0000 [r192429-192454] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: Fix an incorrect assumption that certain
+ values on the channel will always exist when they may not. The
+ CDR code involved with bridges wrongly assumed that the currently
+ executing application and data values will always exist. It is
+ possible for this to be false when call forwarding is involved.
+ (closes issue #14984) Reported by: gincantalupo
+
+ * apps/app_followme.c: Fix a bug where the followme application
+ would continue trying numbers after the caller hung up. (closes
+ issue #13624) Reported by: sgenyuk
+
+2009-05-04 22:37 +0000 [r192213] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: global mohinterpret setting is ignored
+ mohinterpret and mohsuggest global variables were not copied over
+ during build_users and build_peers. (closes issue #14728)
+ Reported by: dimas Patches: v1-14728.patch uploaded by dimas
+ (license 88) Tested by: dimas, dvossel
+
+2009-05-02 18:48 +0000 [r191628-191778] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Fix a bug which resulted from the Hebrew
+ voicemail commit. This fixes a case where a certain message could
+ get played twice. (closes issue #13155) Reported by:
+ greenfieldtech Patches: app_voicemail.c.multi-lang-patch uploaded
+ by greenfieldtech (license 369) Tested by: greenfieldtech
+
+ * apps/app_chanspy.c: Kevin has informed me that thi sort of thing
+ is not necessary.
+
+ * apps/app_chanspy.c: Move static buffers to outside for loops in
+ app_chanspy. Similar to seanbright's commit 191422, this moves
+ some static buffers to be defined outside of for loops since it
+ is undefined if memory will be re-used or if the stack will grow
+ with each iteration of the loop.
+
+2009-05-01 20:00 +0000 [r191559] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: SIP Response 410 maps to cause code 22 (or
+ 23), not 1. (closes issue #14993) Reported by: BigJimmy Patches:
+ causepatch uploaded by BigJimmy (license 371)
+
+2009-05-01 17:40 +0000 [r191488] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c: Fix DTMF not being sent to other side after a
+ partial feature match This fixes a regression from commit 176701.
+ The issue was that ast_generic_bridge never exited after the
+ feature digit timeout had elapsed, which prevented the queued
+ DTMF from being sent to the other side. This issue was reported
+ to me directly.
+
+2009-05-01 15:42 +0000 [r191422] Sean Bright <sean.bright at gmail.com>
+
+ * apps/app_queue.c: Move the defintion of the a couple arrays out
+ of loops. According to Kevin, it is unspecified as to whether a
+ variable defined inside a block is allocated once by the compiler
+ or for each pass through the block (loops being the only
+ interesting case), so just define these before we get into our
+ loop to be sure.
+
+2009-04-29 23:10 +0000 [r191220] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/h323/ast_h323.cxx, channels/chan_h323.c: Allow H.323 to
+ compile with FDLEAK checking enabled.
+
+2009-04-29 18:07 +0000 [r191096] David Brooks <dbrooks at digium.com>
+
+ * pbx/pbx_config.c: Patch to fix tab-completion crash on "remove
+ extension" This patch simply removes some old code back before
+ Asterisk used editline. This fixes the crash that occurred when
+ tab-completing "remove extension". (closes issue #14689) Reported
+ by: isaacgal
+
+2009-04-29 15:23 +0000 [r191041] Sean Bright <sean.bright at gmail.com>
+
+ * apps/app_queue.c: Fix a crash in app_queue with very long member
+ lists. A user reported via #asterisk that with very long lists of
+ members, a crash occurs in ast_strdupa, so just use a single
+ buffer and ast_copy_string instead of stack allocating copys of
+ each interface name.
+
+2009-04-27 19:29 +0000 [r190721] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in: Fix 'inconsistent
+ line endings' when autoconf 2.63 is used Attempt to make
+ configure script regeneration 'safe' using autoconf 2.63, which
+ embeds a bare CR into the script, thus making Subversion complain
+ about inconsistent line endings This commit changes the MIME type
+ of the configure script to be 'binary' thus making Subversion no
+ longer inspect line endings, and as a bonus 'svn diff' will no
+ longer try to generate diff output for it, which is not generally
+ useful anyway.
+
+2009-04-27 19:03 +0000 [r190661-190662] Russell Bryant <russell at digium.com>
+
+ * res/res_smdi.c: Fix a typo from 190661.
+
+ * res/res_smdi.c: Resolve a crash in res_smdi when used with
+ chan_dahdi. When chan_dahdi goes to get an SMDI message, it
+ provides no search criteria. It just grabs the next message that
+ arrives. This code was written with the SMDI dialplan functions
+ in mind, since that is now the preferred method of using SMDI.
+ However, this broke support of it being used from chan_dahdi.
+ (closes AST-212)
+
+2009-04-23 21:07 +0000 [r190356] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Remove a bogus ast_channel_unlock().
+
+2009-04-23 19:13 +0000 [r190286] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_local.c: Fix a bug in chan_local glare hangup
+ detection. If both sides of a Local channel were hung up at
+ around the same time it was possible for one thread to destroy
+ the local private structure and have the other thread immediately
+ try to remove the already freed structure from the local channel
+ list.
+
+2009-04-23 10:07 +0000 [r190187] Olle Johansson <oej at edvina.net>
+
+ * include/asterisk/lock.h: unistd.h is required for usleep() on
+ Darwin. It will not hurt to include it always on other platforms
+ either.
+
+2009-04-22 21:35 +0000 [r190092] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Detect availability of
+ pthread_rwlock_timedwrlock() before using it. (closes issue
+ #14930) Reported by: tilghman Patches:
+ 20090420__bug14930.diff.txt uploaded by tilghman (license 14)
+ Tested by: mvanbaak, tilghman
+
+2009-04-22 19:20 +0000 [r189991] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/h323/ast_h323.cxx, channels/chan_h323.c,
+ channels/h323/chan_h323.h: Make chan_h323 respect packetization
+ settings Previously, packetization settings were ignored and now
+ they are not. A new config option 'autoframing' has been added to
+ mirror the way chan_sip handles it. Turning on the autoframing
+ option (available both as a global option or per peer) overrides
+ the local settings with the remote packetization settings.
+ Testing was performed with varying packetization levels with the
+ following codecs: ulaw, alaw, gsm, and g729. (closes issue
+ #12415) Reported by: pj Patches:
+ 2009012200_h323packetization.diff.txt uploaded by mvanbaak
+ (license 7), modified by me
+
+2009-04-22 14:29 +0000 [r189849] Michiel van Baak <michiel at vanbaak.info>
+
+ * contrib/scripts/get_ilbc_source.sh: replace sed with tr to remove
+ \r from downloaded file On some systems, sed does not recognize
+ \r in the pattern the way it was used here. Use tr instead
+ because this works the same across systems. (closes issue #14936)
+ Reported by: leobrown Patches: 2009042201_14936.diff.txt uploaded
+ by mvanbaak (license 7) Tested by: leobrown, mvanbaak
+
+2009-04-21 15:52 +0000 [r189601-189664] Doug Bailey <dbailey at digium.com>
+
+ * utils/muted.c: Remove daemon call on systems that do not support
+ forking.
+
+ * main/config.c, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, configure.ac: Add check in configure
+ script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h This
+ allows config.c to compile when linked against uclibc that does
+ not support these parameters
+
+2009-04-20 22:02 +0000 [r189537] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_odbc.c, funcs/func_strings.c: Add a workaround for
+ func_odbc/ARRAY() for problems that occur with certain special
+ characters. In certain cases, due to the way Set() works in 1.4,
+ values may not get set properly. This is a workaround for 1.4
+ only that corrects for these issues, without making func_odbc
+ more difficult to use properly. (closes issue #14614) Reported
+ by: wdoekes Patches: 20090309__bug14614__2.diff.txt uploaded by
+ tilghman (license 14)
+ double_set_unescape_workaround_for_func_odbc.osso-and-tilghman-1.diff
+ uploaded by wdoekes (license 717) Tested by: wdoekes, tilghman
+
+2009-04-20 21:10 +0000 [r189463-189465] Terry Wilson <twilson at digium.com>
+
+ * apps/app_dial.c: Update CDR appropriately when
+ AST_CAUSE_NO_ANSWER is set
+
+ * apps/app_dial.c: Don't treat a NOANSWER like a CHANUNAVAIL
+
+2009-04-20 20:58 +0000 [r189462] Sean Bright <sean.bright at gmail.com>
+
+ * pbx/ael/ael.tab.c, pbx/ael/ael.y: Properly handle @s within hints
+ in AEL. AEL was not handling the case of a device hint containing
+ an @ symbol, which caused parking hints (e.g.
+ hint(park:exten at context)) to error out the parser. This patch
+ makes AEL treat the @ the same way it treats colon and ampersand
+ now, meaning the characters are included in verbatim. (closes
+ issue #14941) Reported by: bpgoldsb Patches: bug14941.patch
+ uploaded by seanbright (license 71) Tested by: bpgoldsb
+
+2009-04-20 19:10 +0000 [r189391] Doug Bailey <dbailey at digium.com>
+
+ * main/manager.c, main/db1-ast/recno/rec_open.c,
+ channels/chan_iax2.c: Clean up problem with manager
+ implementation of mmap where it was not testing against
+ MAP_FAILED response. Got rid of shadowed variable used in
+ processign the mmap results. Change test of mmap results to
+ compare against MAP_FAILED
+
+2009-04-20 14:04 +0000 [r189277] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Move the check for chan->fdno == -1 to after the
+ zombie/hangup check. Many users were finding that their hung up
+ channels were staying up and causing 100% CPU usage. (issue
+ #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
+ uploaded by mmichelson (license 60) Tested by: falves11, bamby
+
+2009-04-18 01:27 +0000 [r189203] David Vossel <dvossel at digium.com>
+
+ * channels/chan_agent.c: Fixed autologoff in agents.conf not
+ working when agent logs in via AgentLogin app An agent logs in by
+ calling an extension that calls the AgentLogin app. In
+ agents.conf ackcall=always is set, so when they get a call they
+ have the choice to either acknowledge it or ignore it.
+ autologoff=10 is set as well, so if the agent ignores the call
+ over 10sec one may assume that the agent should be logged out
+ (and in this case hungup on as well), but this was not happening.
+ (closes issue #14091) Reported by: evandro Patches:
+ autologoff.diff uploaded by dvossel (license 671) Review:
+ http://reviewboard.digium.com/r/225/
+
+2009-04-17 21:27 +0000 [r189134] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_lib.c: Modifed/added some debug messages.
+ JIRA ABE-1835
+
+2009-04-17 15:43 +0000 [r189009] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c: Make Busy() application set the CDR disposition to
+ BUSY. (closes issue #14306) Reported by: cristiandimache
+
+2009-04-17 14:41 +0000 [r188937-188946] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix a bug where a value used to create the
+ channel name was bogus. This commit fixes the scenario where an
+ incoming call is authenticated using a peer entry. Previously the
+ channel name was created using either the username setting from
+ the sip.conf entry or the IP address that the call came from. Now
+ the channel name will be created using the peer name itself. This
+ commit will not change the way the channel name is generated for
+ users or friends. (closes issue #14256) Reported by: Nick_Lewis
+ Patches: chan_sip.c-chname.patch uploaded by Nick (license 657)
+ Tested by: Nick_Lewis, file
+
+ * channels/chan_dahdi.c: Fix a situation where the DAHDI channel
+ private structure lock was not unlocked when it should have been.
+ (issue AST-210)
+
+2009-04-16 21:41 +0000 [r188835] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Only update realtime, if global option
+ rtupdate != false (closes issue #14885) Reported by: deepesh
+ Patches: 20090413__bug14885.diff.txt uploaded by tilghman
+ (license 14) Tested by: deepesh
+
+2009-04-16 21:37 +0000 [r188833] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Only disable mISDN DSP if Asterisk DSP is
+ enabled. Leave jitter setting alone. JIRA ABE-1835
+
+2009-04-16 21:02 +0000 [r188773] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Umask should not be exported into global
+ namespace. (closes issue #14912) Reported by: jcapp
+
+2009-04-15 22:08 +0000 [r188646] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c: National prefix inserted even when caller
+ ID not available When the caller ID is restricted, the expected
+ behavior is for the caller id to be blank. In chan_dahdi, the
+ national prefix is placed onto the callers number even if its
+ restricted (empty) causing the caller id to be the national
+ prefix rather than blank. (closes issue #13207) Reported by:
+ shawkris Patches: national_prefix.diff uploaded by dvossel
+ (license 671) Review: http://reviewboard.digium.com/r/220/
+
+2009-04-15 20:04 +0000 [r188582] Mark Michelson <mmichelson at digium.com>
+
+ * main/file.c: Update ast_readvideo_callback to match
+ ast_readaudio_callback. This fixes potential refcount errors that
+ may occur on ast_filestreams. AST-208
+
+2009-04-14 15:02 +0000 [r188287] David Vossel <dvossel at digium.com>
+
+ * main/audiohook.c: audio_audiohook_write_list() does not correctly
+ update sample size after ast_translate.
+ audio_audiohook_write_list() does not take into account that the
+ sample size may change after translation depending on if the
+ original frame is is 8khz or 16khz. While no 16kz codecs are
+ supported in 1.4 at the moment, this will save headaches in the
+ future if they ever are. the sample size is now updated after
+ translating to reflect this possibility. Thanks to jcolp and
+ mmichelson for helping me work this out. (issue AST-197)
+
+2009-04-13 23:04 +0000 [r188149] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_odbc.c: If fileconfig limit exceeds our maximum, then set
+ the limit to the maximum. (Closes issue #14888) Reported by:
+ falves11
+
+2009-04-10 22:16 +0000 [r187962] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/Makefile: Fix module embedding for chan_h323. Include
+ libchanh323.a in the modules.link file so that all the symbols
+ can be resolved at link time. (closes issue #11966) Reported by:
+ dome
+
+2009-04-10 19:26 +0000 [r187865] Russell Bryant <russell at digium.com>
+
+ * channels/chan_dahdi.c: Support "signaling" in addition to
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