--- /dev/null
+2011-12-19 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.6.2.22 Released
+
+2011-12-18 18:25 +0000 [r348515] Kevin P. Fleming <kpfleming@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 'yes' is the
+ default.
+
+ * The warning about having differing 'nat' settings
+ confusingly referred to both peers and users.
+
+2011-12-08 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.6.2.21 Released.
+
+ * AST-2011-013, AST-2011-014
+
+2011-12-08 21:03 +0000 [r347659] Leif Madsen <lmadsen@digium.com>
+
+ * /: Update svn:externals to use menuselect from 1.6.2.20 and not
+ later. This change is required because when making security
+ releases, if you pull from menuselect/trunk you'll get changes
+ meant for later versions of Asterisk.
+
+2011-12-08 16:17 +0000 [r347530] Terry Wilson <twilson@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)
+
+2011-11-21 20:33 +0000 [r345800-345827] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't set the nat default twice. Cleaning up
+ a small 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
+
+2011-08-05 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.20 Released.
+
+2011-08-01 21:19 +0000 [r330490-330505] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c: fixes reference leak pointed out by rmudgett in
+ https://reviewboard.asterisk.org/r/1337/
+
+ * main/features.c: Asterisk 18103 - Fix reload crash caused by
+ destroying default parking lot Default parking lot was being
+ destroyed in reload and was not being rebuilt properly. This
+ patch keeps features.c reload from destroying the default parking
+ lot in 1.6.2. Bug was caused by a hasty backport which didn't
+ test reload enough times to catch the problem. (closes issue
+ ASTERISK-18103) Reported by: 808blogger Review:
+ https://reviewboard.asterisk.org/r/1337/
+
+2011-07-08 22:26 +0000 [r327255] Jason Parker <jparker@digium.com>
+
+ * cdr, formats, codecs/gsm/src, funcs, bridges, codecs/lpc10,
+ main/db1-ast/btree, codecs/g722, main, main/db1-ast/recno, res,
+ pbx, res/ael, channels, main/stdtime, codecs, agi, utils,
+ main/db1-ast/hash, apps, main/db1-ast/db, main/db1-ast/mpool: Add
+ .o files to svn:ignore property, since it's only ignored if
+ locally configured to do so.
+
+2011-06-28 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.19 Released.
+
+2011-06-28 20:06 +0000 [r325277] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 325275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r325275 | twilson | 2011-06-28 15:03:19 -0500 (Tue, 28 Jun 2011)
+ | 2 lines Don't leak SIP username information ........
+
+2011-06-23 18:21 +0000 [r324643] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Addresses AST-2011-008, memory corruption
+ and remote crash in SIP driver. AST-2011-008
+
+2011-06-23 18:18 +0000 [r324634] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /, main/features.c: Merged revisions 324627
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011)
+ | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver
+ Thanks to twilson for identifying the issue and providing the
+ patches. AST-2011-010 ........
+
+2011-06-21 16:10 +0000 [r324306] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c: ConfBridge does not handle hangup properly
+ When playing back a prompt to a channel, confbridge neglects to
+ check for hangup events causing lockup condititions for hangups
+ that occur before actually joining the conference. This change
+ ensures that the user is removed from the conference in the event
+ of a premature hangup. Review:
+ https://reviewboard.asterisk.org/r/1277/
+
+2011-06-15 18:13 +0000 [r323733] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 323732 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011)
+ | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a
+ recent DTMF change. This patch makes sure that dynamic features
+ are also checked when deciding whether or not to pass DTMF
+ through or store it for interpreting. (closes issue
+ ASTERISK-17914) Reported by: vrban ........
+
+2011-06-15 15:22 +0000 [r323579] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /: Merged revisions 323559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun
+ 2011) | 25 lines Resolve a segfault/bus error when we try to map
+ memory that falls on a page boundary. The fix for ASTERISK-15359
+ was incorrect in that it added 1 to the length of the mmap'd
+ region. The problem with this is that reading/writing to that
+ extra byte outside of the bounds of the underlying fd causes a
+ bus error. The real issue is that we are working with both a FILE
+ * and the raw fd underneath it and not synchronizing between
+ them. The code that was removed in ASTERISK-15359 was correct,
+ but we weren't flushing the FILE * before mapping the fd. Looking
+ at the manager code in 1.4 reveals that the FILE * in 'struct
+ mansession' is never used except to create a temporary file that
+ we immediately fdopen. This means we just need to write a 0 byte
+ to the fd and everything will just work. The other branches
+ require a call to fflush() which, while not a guaranteed fix,
+ should reduce the likelihood of a crash. This all makes sense in
+ my head. (closes issue ASTERISK-16460) Reported by:
+ Ravelomanantsoa Hoby (hoby) Patches:
+ issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license
+ #5060) ........
+
+2011-06-10 19:15 +0000 [r323039] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Unlock the sip channel during fax detection
+ like chan_dahdi does to prevent a deadlock with
+ ast_autoservice_stop. (closes issue ASTERISK-17798) tested by
+ mnicholson
+
+2011-06-09 15:37 +0000 [r322668-322699] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: unlock pvt when we drop voice frames
+ received in early media when in t.38 mode
+
+ * channels/chan_sip.c: fix for previous commit
+
+ * /, channels/chan_sip.c: Merged revisions 322646 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r322646 | mnicholson | 2011-06-09 10:10:30 -0500 (Thu, 09 Jun
+ 2011) | 5 lines don't drop any voice frames when checking for
+ T.38 during early media (closes issue ASTERISK-17705) Review:
+ https://reviewboard.asterisk.org/r/1186/ patch by oej reported by
+ oej ........
+
+2011-05-27 08:24 +0000 [r321210] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/features.c: Fix *8 directed pickup locks system during
+ pickupsound play out move playout from sip_pickup_thread to
+ bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2
+ threads trying to write audio to same channel. In addition fixes
+ choppy audio beep in issue 19177. (issue #18654) (issue #19177)
+ Reported by: Docent Patches: review1232-1.6.2.diff.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1232/
+
+2011-05-23 16:15 +0000 [r320506-320562] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c: Adds missing part to the ast_tcptls_server_start
+ fails second attempt to bind patch. (closes issue #19289)
+ Reported by: wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls_2.patch uploaded by
+ wdoekes (license 717)
+
+ * apps/app_chanspy.c: Fixes chanspy enforced mode lacking a
+ channel_unlock. (closes issue #19348) Reported by: wdoekes
+ Patches: issue19348_chanspy_missing_channel_unlock.patch uploaded
+ by wdoekes (license 717)
+
+2011-05-22 23:25 +0000 [r320444] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/res_odbc.c: Don't crash when the connection fails. (closes
+ issue #19250) Reported by: seadweller Patches:
+ 20110514__issue19250.diff.txt uploaded by tilghman (license 14)
+ Tested by: seadweller, sum
+
+2011-05-20 21:24 +0000 [r320271] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c: Fixes issue with ast_tcptls_server_start failing
+ on second attempt to bind. (closes issue #19289) Reported by:
+ wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls.patch uploaded by
+ wdoekes (license 717)
+
+2011-05-20 20:44 +0000 [r320236] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 320235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011)
+ | 13 lines The meetme CLI command completion leaves conferences
+ mutex locked. When issuing a meetme kick CLI command and an
+ invalid (non-existent) conference number is specified, pressing
+ Tab leaves the conferences mutex locked and, therefore, all
+ conferences deadlock. Add missing unlock. (closes issue #19336)
+ Reported by: zvision Patches: app_meetme.diff uploaded by zvision
+ (license 798) ........
+
+2011-05-20 18:45 +0000 [r320179] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: This commit modifies the way polling is done
+ on TLS sockets. Because of the buffering the TLS layer does,
+ polling is unreliable. If poll is called while there is data
+ waiting to be read in the TLS layer but not at the network layer,
+ the messaging processing engine will not proceed until something
+ else writes data to the socket, which may not occur. This change
+ modifies the logic around TLS sockets to only poll after a failed
+ read on a non-blocking socket. This way we know that there is no
+ data waiting to be read from the buffering layer. (closes issue
+ #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson
+
+2011-05-18 23:11 +0000 [r319528-319653] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 319652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011)
+ | 8 lines Make sure everyone gets an unhold when a transfer
+ succeeds Some phones, like the Snom phones, send a hold to the
+ transfer target after before sending the REFER. We need to make
+ sure that we unhold the parties that are being connected after
+ the masquerade. If Local channels with the /nm option are used
+ when dialing the parties, hold music would still be playing on
+ the transfer target, even after being connected with the
+ transferee. ........
+
+ * apps/app_dial.c, /: Merged revisions 319527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011)
+ | 10 lines Fix app_dial ring groups Revert part of r315643. We
+ need to remove the datastore here as well. The code in bridging
+ code will catch anything that app_dial might miss. (closes issue
+ #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff
+ uploaded by elguero (license 37) ........
+
+2011-05-16 18:00 +0000 [r319202] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Unlink a peer from peers_by_ip when expiring
+ a registration Review: https://reviewboard.asterisk.org/r/1218/
+
+2011-05-16 15:56 +0000 [r319144] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes issue with peer ref-counting during
+ handle_request_subscribe. (closes issue #19293) Reported by:
+ irroot
+
+2011-05-16 15:51 +0000 [r319141] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Make sure tcptls_session exists before
+ dereferencing it. (closes issue #19192) Reported by: stknob
+ Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by
+ Chainsaw (license 723) Tested by: vois, Chainsaw
+
+2011-05-13 01:14 +0000 [r318636-318735] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/features.h, /, channels/chan_sip.c,
+ apps/app_directed_pickup.c, main/features.c: Merged revisions
+ 318734 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r318734 | rmudgett | 2011-05-12 20:09:40 -0500
+ (Thu, 12 May 2011) | 43 lines Merged revisions 318671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 * The
+ applicable fixes for v1.4 are the SIP deadlock and the in
+ progress masquerade check for multiple parties trying to pickup
+ the same call. issue18654_v1.4.patch uploaded by rmudgett
+ (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch
+ uploaded by rmudgett (license 664) ........ r318671 | alecdavis |
+ 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix
+ directed group pickup feature code *8 with pickupsounds enabled
+ Since 1.6.2, the new pickupsound and pickupfailsound in
+ features.conf cause many issues. 1).
+ chan_sip:handle_request_invite() shouldn't be playing out the
+ fail/success audio, as it has 'netlock' locked. 2). dialplan
+ applications for directed_pickups shouldn't beep. 3). feature
+ code for directed pickup should beep on success/failure if
+ configured. Created a sip_pickup() thread to handle the pickup
+ and playout the audio, spawned from handle_request_invite. Moved
+ app_directed:pickup_do() to features:ast_do_pickup(). Functions
+ below, all now use the new ast_do_pickup() app_directed_pickup.c:
+ pickup_by_channel() pickup_by_exten() pickup_by_mark()
+ pickup_by_part() features.c: ast_pickup_call() (closes issue
+ #18654) Reported by: Docent Patches:
+ ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license
+ 585) Tested by: lmadsen, francesco_r, amilcar, isis242,
+ alecdavis, irroot, rymkus, loloski, rmudgett Review:
+ https://reviewboard.asterisk.org/r/1185/ ........
+ ................
+
+ * channels/chan_sip.c: Merged revision 222981 from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 Similar
+ deadlock possible when running the Pickup application internally.
+ ------------------------------------------------------------------------
+ r222981 | dvossel | 2009-10-08 17:04:41 -0500 (Thu, 08 Oct 2009)
+ | 13 lines Deadlock between ast_cel_report_event and
+ ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner
+ channel while only the pvt lock is held. Since pbx_exec calls
+ ast_cel_report_event which attempts to lock the channel, invalid
+ locking order occurs. Channels should be locked before pvt's.
+ (closes issue #15512) Reported by: lmsteffan Patches:
+ ast_cel_deadlock_15512.diff uploaded by dvossel (license 671)
+
+2011-05-11 17:15 +0000 [r318548] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Clean up several chan_sip reference leaks
+ Several situations in the code could lead to peers or sip_pvt
+ references being leaked. This would cause RTP ports to never be
+ destroyed (leading to exhaustion of all available RTP ports) and
+ memory leaks. The original patch for this issue from rgagnon was
+ the result of an obscene amount of testing and hard work, for
+ which I am very grateful. I did some cleanup and added a few
+ additional refcount fixes that I found. (closes issue #17255)
+ Reported by: kvveltho Patches:
+ tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by
+ rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes,
+ loloski Review: https://reviewboard.asterisk.org/r/1101/ Review:
+ https://reviewboard.asterisk.org/r/1207/
+
+2011-05-09 20:04 +0000 [r318331] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't offer video to directmedia callee
+ unless caller offered it as well Make sure that when directmedia
+ is enabled, that video is not offered to the callee even if it
+ supports it. p->vrtp will not exist since the caller didn't offer
+ video. (closes issue #19195) Reported by: one47 Patches:
+ sip_cant_add_video_rtp uploaded by one47 (license 23)
+
+2011-05-09 16:51 +0000 [r318230] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes cases where sip_set_rtp_peer can
+ return too early during media path reset. (closes issue #19225)
+ Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by
+ one47 (license 23)
+
+2011-05-06 19:34 +0000 [r317859] Matthew Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_lua.c: pbx_lua autoservice fixes Don't start an
+ autoservice in pbx_lua if pbx_lua already started one and don't
+ stop one if we didn't start one. Also start and stop the
+ autoservice when transferring control from and to the pbx.
+
+2011-05-06 18:03 +0000 [r317720] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 317719 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317719 | rmudgett | 2011-05-06 12:59:05 -0500 (Fri, 06 May 2011)
+ | 11 lines Regression after r297603 (Improve handling of REGISTER
+ requests with multiple contact headers.) Uninitialized variable.
+ (issue #18640) (closes issue #18785) Reported by: pnlarsson
+ Patches: issue18785_enegaard.patch uploaded by enegaard (license
+ 1197) ........
+
+2011-05-06 15:18 +0000 [r317666] Matthew Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_lua.c: Add a datastore fixup to fix a pbx_lua crash.
+ (closes issue #19055) Reported by: jamhed Patches:
+ lua_datastore_fixup1.diff uploaded by mnicholson (license 96)
+ Tested by: mnicholson, jamhed
+
+2011-05-06 08:04 +0000 [r317575] Terry Wilson <twilson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 317574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011)
+ | 6 lines Re-fix queue round-robin This part of the change for
+ r315596 was incorrect. No bridge occurs when doing a roundrobin
+ dial and no one answers, so this code shouldn't have been
+ removed. ........
+
+2011-05-05 18:29 +0000 [r317255] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 317211 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011)
+ | 15 lines chan_sip: fix broken realtime peer count, fix memory
+ leak This patch addresses two bugs in chan_sip: 1) The count of
+ realtime peers and users was off. The increment checked the value
+ of the caching option, while the decrement did not. 2) Add a
+ missing regfree() for a regex. (closes issue #19108) Reported by:
+ vrban Patches: missing_regfree.patch uploaded by vrban (license
+ 756) sip_object_counter.patch uploaded by vrban (license 756)
+ ........
+
+2011-05-05 17:59 +0000 [r317195] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Set SO_KEEPALIVE on SIP TCP sockets so that
+ they eventually go away when a peer abruptly disappears. This
+ mostly occurs after a successful registration. (closes issue
+ #17544) Reported by: marcelloceschia Patches: (modified)
+ tcptls.patch uploaded by st (license 907)
+
+2011-05-05 14:56 +0000 [r317103] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/safe_asterisk, /: Merged revisions 317102 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011)
+ | 8 lines Disable console colourization inside safe_asterisk
+ checks. (closes issue #19213) Reported by: lefoyer Patches:
+ issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by
+ wdoekes (license 717) Tested by: wdoekes, lefoyer ........
+
+2011-05-04 16:10 +0000 [r316708] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 316707 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed,
+ 04 May 2011) | 8 lines If sox fails when processing a voicemail,
+ don't delete the original file. (closes issue #18111) Reported
+ by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright
+ (license 71) Tested by: seanbright ........
+
+2011-05-04 14:23 +0000 [r316616-316644] David Vossel <dvossel@digium.com>
+
+ * apps/app_chanspy.c: Fixes one-way-audio when chanspy activated
+ with the 'o' option (closes issue #18382) Reported by: jkister
+ Patches:
+ 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt
+ uploaded by malin (license ) Tested by: firstsip, Greenlightcrm,
+ malin, wdoekes, boroda, dvossel
+
+ * channels/chan_sip.c: Fixes session-timers=refuse not being
+ enforced for *caller* During handle_request_invite, the session
+ timer mode was retrieved from a cached variable. This patch
+ forces a peer lookup of the session timer mode in the case of an
+ incoming invite. (closes issue #18804) Reported by: wdoekes
+ Patches: issue18804_session_timer_refuse_caller.patch uploaded by
+ wdoekes (license 717) issue_18804_v2.diff uploaded by dvossel
+ (license 671)
+
+2011-05-04 02:23 +0000 [r316475] Sean Bright <sean@malleable.com>
+
+ * apps/app_meetme.c: Honor the C option to MeetMe when L is passed.
+ This fixes a case that r304773 and friends missed. (closes issue
+ #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff
+ uploaded by var (license 1227) Tested by: seanbright
+
+2011-05-03 21:29 +0000 [r316329] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 316328 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03
+ May 2011) | 10 lines Fixes chan_local crashs in local_fixup()
+ Thanks OEJ for tracking down the issue and submitting the patch.
+ (closes issue #19053) Reported by: oej Tested by: oej Review:
+ https://reviewboard.asterisk.org/r/1158/ ........
+
+2011-05-02 19:04 +0000 [r316093] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * funcs/func_curl.c: More possible crashes based upon invalid
+ inputs. (closes issue #18161) Reported by: wdoekes Patches:
+ 20110301__issue18161.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes
+
+2011-04-27 19:03 +0000 [r315893] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 315891 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr
+ 2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2.
+ This change optimizes the free_via() function and removes some
+ redundant null checking. It also fixes compliance with RFC 3261
+ section 18.2.2 by always using the port specified in the Via
+ header for routing responses (even when maddr is not set). Also
+ the htons() function is now used when setting the port.
+ Additional documentation comments have been added in various
+ places to make the logic in the code clearer. (closes issue
+ #18951) Reported by: jmls Patches:
+ issue18951_set_proper_port_from_via.patch uploaded by wdoekes
+ (license 717) (modified) ........
+
+2011-04-26 22:52 +0000 [r315643-315672] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 315671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011)
+ | 11 lines Make sure unregistering a peer unlinks it from the
+ peer container Instead of mostly copying the code from
+ expire_register, just use the function that "does the right
+ thing". (closes issue #16033) Reported by: kkm Patches:
+ 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
+ Tested by: kkm, tilghman, twilson ........
+
+ * apps/app_queue.c, apps/app_dial.c, /, main/features.c: Merged
+ revisions 315596 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011)
+ | 18 lines Allow transfer loops without allowing forwarding loops
+ We try to avoid the situation where two phones may be forwarded
+ to each other causing an infinite loop by storing each dialed
+ interface in a channel datastore and checking the list before
+ dialing out. This works, but currently breaks situations like A
+ calls B, A transfers B to C, B transfers C to A, and A transfers
+ C to B. Since human interaction is happening here and not an
+ automated forwarding loop, it should be allowed. This patch
+ removes the dialed_interfaces datastore when a call is bridged (a
+ suggestion from the brilliant mmichelson). If a call is being
+ bridged, it should be safe to assume that we aren't stuck in a
+ loop. Since we are now handling this is the bridge code, the
+ previous attempts at handling it in app_dial and app_queue are
+ removed. Review: https://reviewboard.asterisk.org/r/1195/
+ ........
+
+2011-04-26 19:22 +0000 [r315502] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * include/asterisk/select.h, /: Merged revisions 315501 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011)
+ | 14 lines Fix the bounds-checking code. The code that set the
+ bit within the select bitfield was correct, but the
+ bounds-checking code was not. The change to that line uses the
+ new _bitsize macro for clarity. Also, FD_ZERO macro did not
+ zero-out anything but the first word of the bitfield, so this
+ could have caused problems with modules using that macro with the
+ expanded bitfield. (closes issue #18773) Reported by: jamicque
+ Patches: 20110423__issue18773.diff.txt uploaded by tilghman
+ (license 14) Tested by: chris-mac ........
+
+2011-04-26 02:17 +0000 [r315393] Paul Belanger <pabelanger@digium.com>
+
+ * pbx/pbx_config.c: Add back CLI command 'dialplan save' (closes
+ issue #19140) Reported by: lmadsen Patches:
+ __20110419_dialplan_save.patch.txt uploaded by lmadsen (license
+ 10)
+
+2011-04-25 19:31 +0000 [r315212-315258] Russell Bryant <russell@digium.com>
+
+ * /, formats/format_wav.c: Merged revisions 315257 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25
+ Apr 2011) | 10 lines Be more flexible with unknown chunks in wav
+ files. This patch makes format_wav ignore unknown chunks instead
+ of erroring out on them. (closes issue #18306) Reported by:
+ jhirsch Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch
+ (license 1156) ........
+
+ * channels/chan_sip.c: Don't link non-cached realtime peers into
+ the peers_by_ip container. (closes issue #18924) Reported by:
+ wdoekes Patches:
+ issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded
+ by wdoekes (license 717)
+
+2011-04-25 07:11 +0000 [r315052] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_local.c, /: Merged revisions 315051 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25
+ Apr 2011) | 11 lines chan_local:check_bridge() misplaced
+ misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path
+ isn't followed, brigde remains locked. (closes issue #19176)
+ Reported by: alecdavis Patches: bug19176.diff.txt uploaded by
+ alecdavis (license 585) ........
+
+2011-04-22 20:49 +0000 [r314958] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 311203,314908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar
+ 2011) | 4 lines Don't hold the pvt lock while streaming a file.
+ ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48
+ -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and
+ the app threads from using the asterisk channel at the same time.
+ ABE-2756 ........
+
+2011-04-22 14:35 +0000 [r314776-314823] Russell Bryant <russell@digium.com>
+
+ * /: Recorded merge of revisions 314822 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r314822 | russell | 2011-04-22 09:34:23 -0500 (Fri, 22 Apr 2011)
+ | 11 lines Initialize buffers in getvar and getvarfull.
+ Initialize the buffers used to hold the result from GET VARIABLE
+ or GET VARIABLE FULL. The bug report shows func_read returning
+ garbage in the result. It assumed that the buffer passed in was
+ initialized, like many other functions do. In the more common
+ code path (through the dialplan), it is initialized, so just
+ initialize it here too. (closes issue #19050) Reported by: johnz
+ ........
+
+ * res/res_agi.c: Initialize buffers in getvar and getvarfull.
+ Initialize the buffers used to hold the result from GET VARIABLE
+ or GET VARIABLE FULL. The bug report shows func_read returning
+ garbage in the result. It assumed that the buffer passed in was
+ initialized, like many other functions do. In the more common
+ code path (through the dialplan), it is initialized, so just
+ initialize it here too. (closes issue #19050) Reported by: johnz
+
+ * main/features.c: Fix handling of some call parking config
+ options. This patch adjusts the handling of some call parking
+ config options to fix some issues that have already been
+ addressed in 1.8 and trunk. (closes issue #19167) Reported by:
+ bluecrow76 Patches:
+ asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff
+ uploaded by bluecrow76 (license 270)
+
+2011-04-21 18:22 +0000 [r314620] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/sip.conf.sample, configs/skinny.conf.sample,
+ configs/http.conf.sample, main/manager.c, /, channels/chan_sip.c,
+ channels/chan_skinny.c, main/http.c: Merged revisions 314607 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr
+ 2011) | 14 lines Added limits to the number of unauthenticated
+ sessions TCP based protocols are allowed to have open
+ simultaneously. Also added timeouts for unauthenticated sessions
+ where it made sense to do so. Unrelated, the manager interface
+ now properly checks if the user has the "system" privilege before
+ executing shell commands via the Originate action. AST-2011-005
+ AST-2011-006 (closes issue #18787) Reported by: kobaz (related to
+ issue #18996) Reported by: tzafrir ........
+
+2011-04-21 00:17 +0000 [r314549] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't allocate more space than necessary for
+ a sip_pkt This extra allocation is a hold-over from when
+ pkt->data was a character array. Now that it is an allocated
+ string, just allocate enough for the sip_pkt.
+
+2011-04-19 14:27 +0000 [r314202-314205] Leif Madsen <lmadsen@digium.com>
+
+ * funcs/func_channel.c: Remove duplicate documentation from
+ func_channel.c (closes issue #18970) Reported by: IgorG Patches:
+ func_channel.c.doc.diff uploaded by IgorG (license 20)
+
+ * apps/app_dial.c: Update seconds to milliseconds in ast_verb
+ output. (closes issue #19084) Reported by: smurfix Patches:
+ app_dial.patch uploaded by smurfix (license 547) Tested by:
+ lmadsen, smurfix
+
+2011-04-15 14:58 +0000 [r313859] Jonathan Rose <jrose@digium.com>
+
+ * main/cli.c: Fix a Tab Completion bug that occurs due to multiple
+ matches on a substring. Makes word_match function in cli.c repeat
+ a search for a command string until a proper match is found or
+ the string is searched to the last point. (closes issue #17494)
+ Reported by: ffossard Review:
+ https://reviewboard.asterisk.org/r/1180/
+
+2011-04-13 16:29 +0000 [r313579] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /, res/res_agi.c: Merged revisions 313545 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011)
+ | 41 lines Asterisk does not hangup a channel after endpoint
+ hangs up. If the call that the dialplan started an AGI script for
+ is hungup while the AGI script is in the middle of a command then
+ the AGI script is not notified of the hangup. There are many AGI
+ Exec commands that this can happen with. The reported
+ applications have been: Background, Wait, Read, and Dial. Also
+ the AGI Get Data command. * Don't wait on the Asterisk channel
+ after it has hung up. The channel is likely to never need
+ servicing again. * Restored the AGI script's ability to return
+ the AGI_RESULT_HANGUP value in run_agi(). It previously only
+ could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the
+ DeadAGI and AGI applications were merged. (closes issue #17954)
+ Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by
+ rmudgett (license 664) issue17954_v1.6.2.patch uploaded by
+ rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue
+ #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761
+ (closes issue #18935) Reported by: nvitaly Tested by: astmiv,
+ rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby
+ Tested by: rmudgett JIRA SWP-2727 Review:
+ https://reviewboard.asterisk.org/r/1165/ ........
+
+2011-04-12 18:44 +0000 [r313432-313435] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_dahdi.c: fixing stupid mistake with putting code
+ before variable declaration ........ Merged revisions 313433 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) |
+ 14 lines reload Chan_dahdi memory leak caused by variables
+ chan_dahdi reloading with variables set via setvar in
+ chan_dahdi.conf would stay in the dahdi_pvt structs for
+ individual channels (causing them to just continue adding the new
+ ones to the list) and also there was a memory leak causes by the
+ conf objects. This patch resolves both of these by using
+ ast_variables_destroy during the loading process. (closes issue
+ #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
+ jrose (license 1225) Tested by: tilghman, jrose Review:
+ https://reviewboard.asterisk.org/r/1170/ ........ ........
+
+ * channels/chan_dahdi.c: white space change ........ reload
+ Chan_dahdi memory leak caused by variables chan_dahdi reloading
+ with variables set via setvar in chan_dahdi.conf would stay in
+ the dahdi_pvt structs for individual channels (causing them to
+ just continue adding the new ones to the list) and also there was
+ a memory leak causes by the conf objects. This patch resolves
+ both of these by using ast_variables_destroy during the loading
+ process. (closes issue #17450) Reported by: nahuelgreco Patches:
+ patch.diff uploaded by jrose (license 1225) Tested by: tilghman,
+ jrose Review: https://reviewboard.asterisk.org/r/1170/ ........
+
+ * channels/chan_dahdi.c: fixes reload Chan_dahdi memory leak caused
+ by variables chan_dahdi reloading with variables set via setvar
+ in chan_dahdi.conf would stay in the dahdi_pvt structs for
+ individual channels (causing them to just continue adding the new
+ ones to the list) and also there was a memory leak causes by the
+ conf objects. This patch resolves both of these by using
+ ast_variables_destroy during the loading process. (closes issue
+ #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
+ jrose (license 1225) Tested by: tilghman, jrose Review:
+ https://reviewboard.asterisk.org/r/1170/
+
+2011-04-11 19:33 +0000 [r313278] Leif Madsen <lmadsen@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 313277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011)
+ | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093)
+ Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by
+ tzafrir (license 46) ........
+
+2011-04-11 15:32 +0000 [r313189] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 313188 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11
+ Apr 2011) | 25 lines Stuck channel using FEATD_MF if caller hangs
+ up at the right time. The cause was actually a caller hanging up
+ just at the end of the Feature Group D DTMF tones that setup the
+ call. The reason for this is a "guard timer" that's implemented
+ using ast_safe_sleep(100). If the caller happens to hang up AFTER
+ the final tone of the DTMF string but BEFORE the end of that
+ ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
+ This causes the code to bounce to the end of ss_thread(), but it
+ does NOT tear down the call properly. This should be a rare
+ occurrence because the caller has to hang up at EXACTLY the right
+ time. Nonetheless, it was happening quite regularly on the
+ reporter's system. It's not easily reproducible, unless you
+ purposely increase the guard-time to 2000 or more. Once you do
+ that, you can reproduce it every time by watching the DTMF debug
+ and hanging up just as it ends. Simply add an ast_hangup() before
+ goto quit. (closes issue #15671) Reported by: jcromes Patches:
+ issue15671.patch uploaded by pabelanger (license 224) Tested by:
+ jcromes ........
+
+2011-04-07 13:23 +0000 [r313047] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c: Makes parking lots clear and rebuild properly
+ when features reload is invoked from CLI Before, default
+ parkinglot in context parkedcalls with ext 700 would always be
+ present and when reload was invoked, the previous parkinglots
+ would not be cleared. (closes issue #18801) Reported by:
+ mickecarlsson Review: https://reviewboard.asterisk.org/r/1161/
+
+2011-04-07 10:26 +0000 [r313004] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_voicemail.c: app_voicemail: close_mailbox change
+ LOG_WARNING to LOG_NOTICE
+
+2011-04-05 14:13 +0000 [r312764] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /, configs/manager.conf.sample: Merged revisions
+ 312761 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr
+ 2011) | 8 lines Limit the number of unauthenticated manager
+ sessions and also limit the time they have to authenticate.
+ AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested
+ by: mnicholson ........
+
+2011-04-05 14:11 +0000 [r312762] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_meetme.c: Backporting trunk change to add verbosity to
+ 'L' option in meetme
+
+2011-04-04 16:00 +0000 [r312574] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 312573 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011)
+ | 38 lines Issues with ISDN calls changing B channels during call
+ negotiations. The handling of the PROCEEDING message was not
+ using the correct call structure if the B channel was changed.
+ (The same for PROGRESS.) The call was also not hungup if the new
+ B channel is not provisioned or is busy. * Made all call
+ connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS,
+ ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
+ using the correct structure and B channel. If there is any
+ problem with the operations then the call is now hungup with an
+ appropriate cause code. * Made miscellaneous messages
+ (INFORMATION, FACILITY, NOTIFY) find the correct structure by
+ looking for the call and not using the channel ID. NOTIFY is an
+ exception with versions of libpri before v1.4.11 because a call
+ pointer is not available for Asterisk to use. * Made all hangup
+ messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct
+ structure by looking for the call and not using the channel ID.
+ (closes issue #18313) Reported by: destiny6628 Tested by:
+ rmudgett JIRA SWP-2620 (closes issue #18231) Reported by:
+ destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue
+ #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The
+ issues fixed here are most likely causing this JIRA issue.) JIRA
+ DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
+ ........
+
+2011-04-01 10:51 +0000 [r312287] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/asterisk.c, include/asterisk/select.h, /: Merged revisions
+ 312285 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011)
+ | 7 lines Found some leaking file descriptors while looking at
+ ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej
+ Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman
+ (license 14) ........
+
+2011-04-01 09:16 +0000 [r312103-312213] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_voicemail.c: fix up bad merge46 extra 2 yuck: labels
+
+ * apps/app_voicemail.c, /: Merged revisions 312174 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01
+ Apr 2011) | 23 lines voicemail: get real last_message_index and
+ count_messages, ODBC resequence change last_message_index to read
+ the max msgnum stored in the database change count_messages to
+ actually count the number of messages. last_message_index change:
+ This fixed overwriting of the last message if msgnum=0 was
+ missing. Previously every incoming message would overwrite
+ msgnum=1. count_messages change: allows us to detect when
+ requencing is required in opneA_mailbox. resequence enabled for
+ ODBC storage: Assists with fixing up corrupt databases with gaps,
+ but only when a user actively opens there mailboxes. (closes
+ issue #18692,#18582,#19032) Reported by: elguero Patches: based
+ on odbc_resequence_mailbox2.1.diff uploaded by elguero (license
+ 37) Tested by: elguero, nivek, alecdavis Review:
+ https://reviewboard.asterisk.org/r/1153/ ........
+
+ * apps/app_voicemail.c, /: Merged revisions 312070 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01
+ Apr 2011) | 16 lines app_voicemail: close_mailbox needs to
+ respect additional messages while mailbox is open. close_mailbox
+ leave gaps in message sequence if messages are deleted and new
+ messages arrive during this time, this is because the shuffle
+ down to slot 0, only shuffles the number of pre-existing messages
+ when mailbox is opened, ignoring new arrivals. Fix: in
+ close_mailbox re-evaluate number of messages before the shuffle,
+ this then includes new arrivals. Happens on filebased or ODBC
+ storage. (issues #19032,#18582,#18692,#18998) Reported by:
+ alecdavis,tootai,afosorio Review:
+ https://reviewboard.asterisk.org/r/1153/ ........
+
+2011-03-29 13:17 +0000 [r311844] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c: When comebacktoorigin=no, Asterisk no longer
+ tries to dial extension <Originating Channe>@parkedcalltimeout
+ and instead dials s without going through fallback. (closes issue
+ #18650) Reported by: davidw Patches: patch.diff uploaded by jrose
+ (license 1225) https://reviewboard.asterisk.org/r/1150/
+
+2011-03-22 15:24 +0000 [r311496] David Vossel <dvossel@digium.com>
+
+ * apps/app_meetme.c: Fixes memory leak in MeetMe AMI action
+
+2011-03-17 14:58 +0000 [r311140] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c: Don't write items to the manager socket twice.
+ AST-2011-003 (closes issue 0018987) Reported by: ks-steven
+
+2011-03-17 10:45 +0000 [r311049] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, configs/indications.conf.sample: Merged revisions 311048 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar
+ 2011) | 12 lines Remove extra quote in indications.conf Picking
+ low hanging fruit. (closes issue #18971) Reported by: IgorG
+ Patches: based on indications.conf.sample.diff uploaded by IgorG
+ (license 20) Tested by: IgorG ........
+
+2011-03-16 19:46 +0000 [r310889-310998] Terry Wilson <twilson@digium.com>
+
+ * main/tcptls.c: Fix crash on fdopen failure See security advisory
+ AST-2011-004 (closes issue #18845) Reported by: cmaj Patches:
+ patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt
+ uploaded by cmaj (license 830)
+ patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt
+ uploaded by cmaj (license 830) Tested by: cmaj, twilson
+
+ * main/tcptls.c, main/manager.c: Revert patch with accidental
+ reversion of a previous patch
+
+ * main/tcptls.c, main/manager.c: Fix crash on fdopen failure See
+ security advisory AST-2011-004 (closes issue #18845) Reported by:
+ cmaj Patches:
+ patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt
+ uploaded by cmaj (license 830)
+ patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt
+ uploaded by cmaj (license 830) Tested by: cmaj, twilson
+
+ * main/manager.c: Don't keep trying to write to a closed connection
+ See security advisory AST-2011-003.
+
+ * /, main/features.c: Merged revisions 310888 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011)
+ | 29 lines Don't delay DTMF in core bridge while listening for
+ DTMF features This patch is mostly the work of Olle Johansson. I
+ did some cleanup and added the silence generating code if
+ transmit_silence is set. When a channel listens for DTMF in the
+ core bridge, the outbound DTMF is not sent until we have received
+ DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds
+ of DTMF to Asterisk, which sends no audio for those 4 seconds.
+ Some products see this delay and the time skew on RTP packets
+ that results and start ignoring the audio that is sent afterward.
+ With this change, the DTMF_BEGIN frame is inspected and checked.
+ If it matches a feature code, we wait for DTMF_END and activate
+ the feature as before. If transmit_silence=yes in asterisk.conf,
+ silence is sent if we paritally match a multi-digit feature. If
+ it doesn't match a feature, the frame is forwarded along with the
+ DTMF_END without delay. By doing it this way, DTMF is not
+ delayed. (closes issue #15642) Reported by: jasonshugart Patches:
+ issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license
+ 396) Tested by: globalnetinc, jde (closes issue #16625) Reported
+ by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/
+ Review: https://reviewboard.asterisk.org/r/1125/ ........
+
+2011-03-15 00:31 +0000 [r310780] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, main/utils.c: Merged revisions 310779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310779 | alecdavis | 2011-03-15 13:26:09 +1300 (Tue, 15 Mar
+ 2011) | 10 lines core show locks: display ThreadID in hexadecimal
+ Allow easier cross referencing of thread ID's with GDB backtraces
+ (closes issue #18968) Reported by: alecdavis Patches:
+ bug18968.diff.txt uploaded by alecdavis (license 585) ........
+
+2011-03-14 16:47 +0000 [r310635] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/callerid.c: Merged revisions 310633 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011)
+ | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and
+ TDM410 The last character in the caller id message is getting a
+ framing error. The checksum is the last character in the message.
+ A framing error in the checksum could be because: 1) The sender
+ did not send a full stop bit. 2) The sender cut off the FSK
+ carrier too soon. 3) The sender opted to send zero of the
+ specified zero to 10 trailing mark bits and round-off errors in
+ the code resulted in the code not being where it thought it was
+ in the demodulated bit stream. Bit 8 of 'b' is set when parity
+ error. Bit 9 of 'b' is set when framing error. Made ignore the
+ framing and parity error bits if the errored character is the
+ checksum. We can tolerate a framing/parity error there. The
+ checksum character validates the message. (closes issue #18474)
+ Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek
+ (license 636) (with modifications) Tested by: nivek ........
+
+2011-03-14 13:56 +0000 [r310585] Jonathan Rose <jrose@digium.com>
+
+ * funcs/func_volume.c: Adds 'p' as an option to func_volume. When
+ it is on, the old behavior with DTMF controlling volume
+ adjustment will be enforced. When it is off, DTMF will not be
+ processed by the function. Programmed by Jonathan Rose Reviewed
+ by David Vossel, Leif Madsen, and Russell Bryant
+ http://reviewboard.digium.internal/r/93/
+
+2011-03-12 20:24 +0000 [r310414-310448] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, pbx/pbx_ael.c: Recorded merge of revisions 310435 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12
+ Mar 2011) | 31 lines Add AELSub, which provides a stable entry
+ point into AEL subroutines. This commit needs some explanation,
+ given that we're adding a new application into an existing
+ release branch. This is generally a violation of our release
+ policy, except in very limited circumstances, and I believe this
+ is one of those circumstances. The problem that this solves is
+ one of the sanity of using multiple dialplan languages to define
+ a dialplan. In the case of the reporter, he or she is using AEL
+ is define subroutines, while using Realtime extensions to invoke
+ those subroutines. While you can do this, it's based upon the
+ reality of AEL using actual dialplan extensions; however, there
+ is no guarantee that the details of _how_ AEL is compiled into
+ extensions will remain stable. In fact, at the time of this
+ commit, it has already changed twice, once in a fundamental way.
+ Now normally, a new application would only be added to trunk.
+ However, this application is explicitly to create a stable
+ user-level API between versions, and adding it to trunk only will
+ not solve the user's problem of switching between 1.6.2 and 1.8,
+ nor will it help anybody switching from 1.8 to 1.10. Therefore,
+ it needs to go into existing release branches. For the sake of
+ consistency, and also because one of the changes was between 1.4
+ and 1.6.x, I am also electing to commit this to 1.4. (closes
+ issue #18910) Reported by: alexandrekeller Patches:
+ 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman
+ (license 14) 20110304__issue18919__1.4.diff.txt uploaded by
+ tilghman (license 14) Tested by: alexandrekeller ........
+
+ * funcs/func_odbc.c: Transactional handles should be used for the
+ insertbuf, if available. Also, fix a possible resource leak.
+ (closes issue #18943) Reported by: irroot
+
+2011-03-10 05:51 +0000 [r310141] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * apps/app_voicemail.c, res/res_config_odbc.c, /,
+ funcs/func_odbc.c: Merged revisions 310140 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011)
+ | 5 lines Initialize column size to 0 to deal with a potential
+ UnixODBC bug on 64-bit systems. (closes issue #18295) Reported
+ by: pruiz ........
+
+2011-03-07 22:04 +0000 [r309857] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 309856 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar
+ 2011) | 8 lines Bug fix for MixMonitor involving filenames with
+ '.' not in the extension Closes issue #18391) Reported by:
+ pabelanger Patches: bugfix.patch uploaded by jrose (license 1225)
+ Tested by: jrose ........
+
+2011-03-05 10:28 +0000 [r309677] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/asterisk.c: Missed part of the conversion when we started
+ passing ppid to astcanary. (closes issue #18850) Reported by:
+ viraptor Patches: canary_ppid.patch uploaded by viraptor (license
+ 543)
+
+2011-03-04 19:37 +0000 [r309494-309584] Matthew Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_lua.c: Restore mysterious lua_pushvalue() call removed in
+ r309494. The mystery has been solved.
+
+ * pbx/pbx_lua.c: Check for errors from fseek() when loading config
+ file, properly abort on errors from fread(), and supply a
+ traceback for errors generated when loading the config file.
+ Also, prepend a newline to traceback output so that the main
+ error message is on it's own line.
+
+ * pbx/pbx_lua.c: remove mysterious lua_pushvalue() that is never
+ used
+
+2011-03-04 00:42 +0000 [r309356] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c, /: Merged revisions 309355 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03
+ Mar 2011) | 9 lines fix small memory leak fix small memory leak
+ caused by a string allocation that wasn't freed (closes issue
+ #18907) Reported by: andy11 Patches:
+ asterisk_trunk-app_externalivr-leak.patch uploaded by andy11
+ (license 1224) ........
+
+2011-03-03 20:13 +0000 [r309348] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_directed_pickup.c: Update PickupChan documentation. The
+ PickupChan uses the ampersand as the argument separator. (closes
+ issue #18905) Reported by: vmikhnevych Tested by: vmikhnevych
+
+2011-03-02 19:53 +0000 [r309255] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Fix usage of "hasvoicemail=yes" and
+ "mailbox=" in users.conf for SIP. Since it's a duplicate, nothing
+ is going to be done, so delme doesn't need to be set at all.
+ Strangely, when this was added, this was being set to 1 in 1.6,
+ and 0 in trunk. (issue AST-439)
+
+2011-03-02 01:06 +0000 [r309251] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/ast_expr2.fl, configure, include/asterisk/autoconfig.h.in,
+ main/ast_expr2f.c, configure.ac: Revert previous 2 commits, and
+ instead conditionally redefine the same macro used in flex 2.5.35
+ that clashed with our workaround. Not surprisingly, the
+ workaround was exactly the same code as was provided by the Flex
+ maintainers, albeit in two different places, in different macros.
+ This should fix the FreeBSD builds, which have an older version
+ of Flex.
+
+2011-03-01 16:05 +0000 [r309083] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes thread blocking issue in the sip
+ TCP/TLS implementation. (closes issue #18497) Reported by: vois
+ Patches: issues_18497.diff uploaded by dvossel (license 671)
+ Tested by: vois, rossbeer, kowalma, Freddi_Fonet
+
+2011-02-28 11:07 +0000 [r308990-309034] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * configure, configure.ac: Clarify meaning, removing double
+ negative (stupid!)
+
+ * main/ast_expr2.fl, configure, include/asterisk/autoconfig.h.in,
+ main/ast_expr2f.c, configure.ac: A later version of flex already
+ includes the fwrite workaround code, which if used twice causes a
+ compilation error. Detect whether Flex will compile without the
+ workaround; if so, suppress our workaround code.
+
+ * funcs/func_odbc.c: Statements updating zero rows may return
+ SQL_NO_DATA. This is fine; it's handled. (closes issue #18815)
+ Reported by: irroot Patches: func_odbc.insert_nodata.patch
+ uploaded by irroot (license 52)
+
+2011-02-24 17:54 +0000 [r308814] Terry Wilson <twilson@digium.com>
+
+ * main/manager.c, /: Merged revisions 308813 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011)
+ | 12 lines Don't broadcast FullyBooted to every AMI connection
+ The FullyBooted event should not be sent to every AMI connection
+ every time someone connects via AMI. It should only be sent to
+ the user who just connected. (closes issue #18168) Reported by:
+ FeyFre Patches: bug0018168.patch uploaded by FeyFre (license
+ 1142) Tested by: FeyFre, twilson ........
+
+2011-02-24 14:59 +0000 [r308722] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 308721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb
+ 2011) | 2 lines silence gcc 4.2 compiler warning ........
+
+2011-02-24 03:38 +0000 [r308677-308678] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, channels/chan_sip.c: Use remotesecret to
+ authenticate with a remote party The remotesecret option was only
+ being used for outbound registration and not for placing calls.
+ This patch uses remotesecret on outbound calls if it is set,
+ otherwise secret is still used. Review:
+ https://reviewboard.asterisk.org/r/1107/
+
+ * include/asterisk/features.h, apps/app_dial.c, main/features.c:
+ Merge missing bugfix for issue #11583 This is the combination of
+ two commits that made it into 1.4, 1.6.0, 1.6.1, and trunk (and
+ therefor 1.8) but that was missed for 1.6.2. ........ r183386 |
+ dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
+ Cleaning up a few things in detect disconnect patch Initialized
+ ast_call_feature in detect_disconnect to avoid accessing
+ uninitialized memory. Cleaned up /param tags in features.h. No
+ longer send dynamic features in ast_feature_detect. issue #11583
+ ........ ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500
+ (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a
+ call is bridged feature.conf has a disconnect option. By default
+ this option is set to '*', but it could be anything. If a user
+ wishes to disconnect a call before the other side answers, only
+ '*' will work, regardless if the disconnect option is set to
+ something else. This is because features are unavailable until
+ bridging takes place. The default disconnect option, '*', was
+ hardcoded in app_dial, which doesn't make any sense from a user
+ perspective since they may expect it to be something different.
+ This patch allows features to be detected from outside of the
+ bridge, but not operated on. In this case, the disconnect feature
+ can be detected before briding and handled outside of features.c.
+ (closes issue #11583) Reported by: sobomax Patches:
+ patch-apps__app_dial.c uploaded by sobomax (license 359)
+ 11583.latest-patch uploaded by murf (license 17)
+ detect_disconnect.diff uploaded by dvossel (license 671) Tested
+ by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/
+ ........
+
+2011-04-25 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.18 Released.
+
+ * AST-2011-005, AST-2011-006
+
+2011-02-23 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.18-rc1 Released.
+
+2011-02-22 15:37 +0000 [r308528] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Add HTTP URI log, use ast_debug for console logging
+ Guessed the log levels based on info that level 3 is the soft
+ roof. Can we create a page / document to define the levels?
+
+2011-02-21 15:00 +0000 [r308414] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 308413 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb
+ 2011) | 5 lines Properly check the bounds of arrays when decoding
+ UDPTL packets. Also, remove broken support for receiving UDPTL
+ packets larger than 16k. That shouldn't ever happen anyway.
+ AST-2011-002 FAX-281 ........
+
+2011-02-19 14:03 +0000 [r308329] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Add CSS MIME Type Modern browsers are checking for
+ the MIME Type of pages and in some cases will not load a file if
+ the type is wrong.
+
+2011-02-15 23:33 +0000 [r308007] Jason Parker <jparker@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 308002 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) |
+ 10 lines Fix regression that changed behavior of queues when
+ ringing a queue member. This reverts r298596, which was to fix a
+ highly bizarre and contrived issue with a queue member that
+ called into his own queue being transferred back into his own
+ queue. I couldn't reproduce that issue in any way. I think one of
+ the other recent transfer fixes actually fixed this. (closes
+ issue #18747) Reported by: vrban ........
+
+2011-02-15 07:01 +0000 [r307792-307836] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * funcs/func_odbc.c: Need to retrieve the rows affected before
+ using the associated variable. (closes issue #18795) Reported by:
+ irroot Patches: 20110211__issue18795.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman
+
+ * res/res_odbc.c: Increment usage count at first reference, to
+ avoid a race condition with many threads creating connections all
+ at once. (issue #18156) Reported by: asgaroth Patches:
+ 20110214__issue18156.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2011-02-11 01:02 +0000 [r307624] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 307623 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r307623 | rmudgett | 2011-02-10 18:29:17 -0600 (Thu, 10
+ Feb 2011) | 13 lines Reentrancy problem if outgoing call gets
+ different B channel than requested. The chan_dahdi
+ pri_fixup_principle() routine needs to protect the Asterisk
+ channel with the channel lock when it changes the technology
+ private pointer to a new private structure. * Added lock
+ protection while pri_fixup_principle() moves a call from one
+ private structure to another. * Made some pri_fixup_principle()
+ messages more meaningful. Partial backport from v1.8 -r300714.
+ ........
+
+2011-02-10 22:35 +0000 [r307535] Jason Parker <jparker@digium.com>
+
+ * main/asterisk.c, contrib/init.d/rc.debian.asterisk, /: Merged
+ revisions 307534 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) |
+ 8 lines Remove color when executing commands via a remote
+ console. Essentially this makes '-x' imply '-n' on rasterisk.
+ This was done in a different and incomplete way previously, which
+ I'm reverting here. (issue #18776) Reported by: alecdavis
+ ........
+
+2011-02-09 21:48 +0000 [r307316] Andrew Latham <lathama@gmail.com>
+
+ * contrib/init.d/rc.debian.asterisk: Disable color during running
+ test (closes issue #18776) Reported by: alecdavis Patches:
+ ast_deb_init.diff uploaded by lathama (license 1028) Tested by:
+ andrel, lathama
+
+2011-02-09 19:52 +0000 [r307227] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Make sure to set parking dial context for
+ non-default parking lots. Since parking_con_dial isn't settable,
+ set all parking lots to "park-dial". (closes issue #17946)
+ Reported by: bluecrow76 Patches:
+ asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by
+ bluecrow76 (license 270) modified by me
+
+2011-02-08 20:14 +0000 [r306973] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 306972 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011)
+ | 2 lines Fix comparison for REFER Replaces tags with
+ pedantic=yes ........
+
+2011-02-08 19:41 +0000 [r306865-306966] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 306965 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08
+ Feb 2011) | 1 line fix this line again ........
+
+ * apps/app_voicemail.c, /: Merged revisions 306960 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08
+ Feb 2011) | 9 lines Backup file storing message duration is not
+ used with IMAP_STORAGE, remove code. The message duration is
+ stored in the body of the email when using IMAP_STORAGE, so
+ nothing needs to happen with the backup file. (closes issue
+ #18718) Reported by: kerframil ........
+
+ * apps/app_voicemail.c, /: Merged revisions 306864 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08
+ Feb 2011) | 1 line make this safer and fully correct, pointed out
+ by Steve Davis ........
+
+2011-02-07 22:40 +0000 [r306618-306673] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 306672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011)
+ | 10 lines Don't try to pickup a call in the middle of a
+ masquerade If A calls B which doesn't answer and C & D both try
+ to do a call pickup, it is possible for ast_pickup_call to answer
+ the call, then fail to masquerade one of the calls because the
+ other one is already in the process of masquerading. This patch
+ checks to see if the channel is in the process of masquerading
+ before call before selecting it for a pickup. Review:
+ https://reviewboard.asterisk.org/r/1094/ ........
+
+ * /, channels/chan_sip.c: Merged revisions 306617 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011)
+ | 10 lines Don't allow a REFER w/replaces to replace its own
+ dialog Asterisk currently accepts a REFER with a Refer-To with an
+ embedded Replaces header that matches the dialog of the REFER.
+ This would be a situation like A calls B, A calls C, A transfers
+ B to A, which is just silly. This patch makes the transfer fail
+ instead of making Asterisk freak out and forget to hang other
+ channels up. Review: https://reviewboard.asterisk.org/r/1093/
+ ........
+
+2011-02-04 19:21 +0000 [r306346] Jason Parker <jparker@digium.com>
+
+ * apps/app_queue.c: Don't fallthrough to 'unknown' in the 'ringing'
+ case. This could cause improper exits from the queue. (closes
+ issue #18499) Reported by: zaltar Patches: app_queue.patch
+ uploaded by zaltar (license 1148)
+
+2011-02-03 20:56 +0000 [r306126] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 306119 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03
+ Feb 2011) | 9 lines Set hangup cause in local_hangup When a call
+ involves a local channel (like SIP -> Local -> SIP), the hangup
+ cause was not being set. This resulted in SIP channels sometimes
+ getting a 503 error instead of a 486 when the far side sent a
+ busy. In Asterisk 1.8+ this also can cause issues with CCSS that
+ involve a local channel. This patch sets the hangupcause for one
+ side of the local channel to the other in local_hangup for
+ outbound calls. ........
+
+2011-02-03 20:49 +0000 [r306123] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Set exception on channel in parking thread when
+ POLLPRI event detected. This is done just to make the code be
+ equivalent to the old select code. As noted in 303106 the same
+ issue was already fixed in this branch, but the exception was not
+ set on the channel in the case of POLLPRI. The reason that this
+ did not cause a problem here is because in 122923 the check in
+ __ast_read to check the exception flag was removed. (related to
+ #18637)
+
+2011-02-03 15:41 +0000 [r305985] Andrew Latham <lathama@gmail.com>
+
+ * phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample:
+ res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
+ (issue #18713) Reported by: lathama Patches: snom_dir.diff
+ uploaded by lathama (license 1028) Tested by: lathama
+
+2011-02-03 00:15 +0000 [r305889] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, main/manager.c, /, channels/chan_sip.c,
+ apps/app_sendtext.c: Merged revisions 305888 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011)
+ | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null
+ terminator in the buffer length. When the frame is queued it is
+ copied. If the null terminator is not part of the frame buffer
+ length, the receiver could see garbage appended onto it. * Add
+ channel lock protection with ast_sendtext(). * Fixed AMI SendText
+ action ast_sendtext() return value check. ........
+
+2011-02-02 14:40 +0000 [r305648-305752] Andrew Latham <lathama@gmail.com>
+
+ * channels/chan_sip.c: Replace link to old doc with new wiki page.
+ Link to
+ https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
+
+ * configs/sip.conf.sample: SIP Configuration Documentation sip show
+ settings reports qualifyfreq in milliseconds. sip.conf configures
+ qualifyfreg in seconds.
+
+2011-02-01 17:02 +0000 [r305472] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 305471 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb
+ 2011) | 9 lines Close file descriptor for timing source when a
+ MOH class gets destroyed. (closes issue #18457) Reported by:
+ mcallist Patches: 18457-closetimer.diff uploaded by qwell
+ (license 4) 18457-closetimer_trunk.diff uploaded by qwell
+ (license 4) Tested by: qwell, loloski ........
+
+2011-01-31 23:50 +0000 [r305342] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 305341 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31
+ Jan 2011) | 7 lines Obtain the pri lock for PRI queue counters.
+ Need to obtain the pri lock when calling pri_dump_info_str() to
+ avoid a reentrancy problem when calculating the Q.921 Q count
+ statistic. JIRA AST-484 ........
+
+2011-01-31 22:59 +0000 [r305130-305253] Jason Parker <jparker@digium.com>
+
+ * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305252
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) |
+ 10 lines Prevent a crash when dialing a technology with no
+ destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers
+ already had code to prevent this. The attempt that app_dial was
+ making to prevent it was not correct, so I fixed that. (closes
+ issue #18371) Reported by: gbour Patches: 18371.patch uploaded by
+ gbour (license 1162) ........
+
+ * res/res_musiconhold.c, /: Merged revisions 305129 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan
+ 2011) | 2 lines Set file descriptors to -1 on creation, so that
+ we don't see weirdness later. ........
+
+2011-01-31 13:52 +0000 [r305082] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Asterisk HTTP response Content-type Address content
+ type for BSD and other platforms (closes issue #18456) Reported
+ by: alexo Patches: asterisk18_http.patch uploaded by alexo
+ (license 1175) Tested by: alexo
+
+2011-01-31 07:25 +0000 [r304978] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * apps/app_voicemail.c, /: Merged revisions 304952 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31
+ Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined.
+ ........
+
+2011-01-29 23:05 +0000 [r304659-304865] Sean Bright <sean@malleable.com>
+
+ * res/res_config_ldap.c: Plug some memory leaks in the LDAP
+ realtime driver. (closes issue #18435) Reported by: zaltar
+ Patches: res_config_ldap.patch uploaded by zaltar (license 1148)
+
+ * apps/app_meetme.c: If we fail to allocate our announcement
+ objects, make sure we don't leak objects. The majority of this
+ patch was committed already in r304726 and r304729. (issue
+ #18225) Reported by: kenji (issue #18444) Reported by: junky
+ (closes issue #18343) Reported by: kobaz Patches:
+ meetme-refs.diff uploaded by kobaz (license 834)
+
+ * apps/app_meetme.c: When we pass the S() or L() options to MeetMe,
+ make sure that we honor C as well. Without this patch, if the
+ user was kicked from the conference via the S() or L() mechanism,
+ we would just hang up on them even if we also passed C (continue
+ in dialplan when kicked). With this patch we honor the C flag in
+ those cases. (closes issue #17317) Reported by: var
+
+ * apps/app_meetme.c: Make sure that we unref the correct object
+ when ejecting the most recent caller. Currently, when we kick the
+ last user to enter, we decrement our own reference count which
+ results in a crash when we kick another user or when we exit the
+ conference ourselves. This will fix #18225 in 1.8 and trunk, but
+ that particular bug does not exist in 1.6.2. (closes issue
+ #18225) Reported by: kenji Patches: issue18225.patch uploaded by
+ seanbright (license 71) Tested by: seanbright
+
+ * apps/app_meetme.c: Fix user reference leak in MeetMe. We were
+ unlinking the user from the conferences user container, but not
+ decrementing the reference count of the user as well, resulting
+ in a leak. (closes issue #18444) Reported by: junky Tested by:
+ seanbright
+
+ * apps/app_meetme.c: Revert part of the previous commit that snuck
+ in.
+
+ * apps/app_meetme.c: Don't leak references if we can't create a
+ pseudo channel for mixing in MeetMe. If there was a problem
+ allocating a pseudo channel when building our meetme, we weren't
+ destroying our user container or destroying the mutexes that we
+ created.
+
+2011-01-27 17:01 +0000 [r304461-304465] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 304464 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan
+ 2011) | 9 lines Fix default prefix=/usr regression on non-Linux
+ systems. This partially reverts a change made in branches/1.4/
+ r267759, which will cause issue #17013 to be reopened. This issue
+ was pointed out by a user on #asterisk, who helpfully discovered
+ that paths were being set incorrectly. To truly understand what
+ was wrong, one should run: svn diff --force -c<this revision>
+ configure ........
+
+ * /, configure: Merged revisions 304460 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan 2011) |
+ 1 line Rerun bootstrap.sh with no changes, so that it is more
+ obvious what my next commit changes. ........
+
+2011-01-26 22:26 +0000 [r304338] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Change delimiter used internally for
+ GOTO_ON_BLINDXFR to commas to match 76703.
+
+2011-01-26 21:02 +0000 [r304250] Mark Michelson <mmichelson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 304242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed, 26 Jan
+ 2011) | 3 lines Get rid of unused 'verbose' field in ast_udptl
+ ........
+
+2011-01-26 21:01 +0000 [r304244-304249] Matthew Nicholson <mnicholson@digium.com>
+
+ * /: Merged revisions 304247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304247 | mnicholson | 2011-01-26 15:00:15 -0600 (Wed, 26 Jan
+ 2011) | 2 lines Convert from network to host byte ordering before
+ checking if an IP is a multicast address. ........
+
+ * /, channels/chan_sip.c: Merged revisions 304241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan
+ 2011) | 6 lines This patch modifies chan_sip to route responses
+ to the address the request came from. It also modifies chan_sip
+ to respect the maddr parameter in the Via header. ABE-2664
+ Review: https://reviewboard.asterisk.org/r/1059/ ........
+
+2011-01-26 20:22 +0000 [r304181] Sean Bright <sean@malleable.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 304159 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed, 26 Jan
+ 2011) | 1 line Make sure the sample queues.conf is properly
+ commented. ........
+
+2011-01-26 19:38 +0000 [r304149] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 304148 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed,
+ 26 Jan 2011) | 2 lines Update documentation for
+ DAHDISendCallreroutingFacility() application. ..........
+
+2011-01-26 01:24 +0000 [r304096] Sean Bright <sean@malleable.com>
+
+ * main/file.c: Per the man page, setvbuf() must be called before
+ any other operation on an open file. We use setvbuf() to
+ associate a buffer with a stream, but we have already written to
+ the open file. This works (by chance) on Linux, but fails on
+ other platforms, such as OpenSolaris. (closes issue #16610)
+ Reported by: bklang Patches: setvbuf.patch uploaded by crjw
+ (license 963) Tested by: bklang, asgaroth, efutch
+
+2011-01-25 23:25 +0000 [r304006] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 304005 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011)
+ | 8 lines DTMF attended transfers sometimes fail for no apparent
+ reason. The loop in feature_request_and_dial() can exit when
+ Party C has answered without processing an AST_CONTROL_ANSWER.
+ Also sometimes an AST_CONTROL_ANSWER never happens even though
+ Party C has answered. Don't hangup Party C if he is up or we
+ receive an AST_CONTROL_ANSWER. ........
+
+2011-01-25 22:02 +0000 [r303960] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 303906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011)
+ | 16 lines Guard against retransmitting BYEs indefinitely In the
+ case of an attended transfer (A calls B, A atxfers to C) where A
+ becomes unreachable before replying to Asterisk's BYE, Asterisk
+ can sometimes retransmit the BYE indefinitely. This is because
+ __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
+ SIP_ALREADYGONE and will then transmit a BYE. When this BYE times
+ out, it will not ever be marked as ALREADYGONE, so when
+ __sip_autodestruct is called again, we end up starting the cycle
+ over. This patch adds a call to sip_alreadygone(pkt->owner) in
+ retrans_pkt in the case of a BYE that has timed out. This should
+ prevent Asterisk from trying to transmit new BYE messages in the
+ future. Review: https://reviewboard.asterisk.org/r/1077/ ........
+
+2011-01-25 18:41 +0000 [r303858] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * channels/chan_sip.c: Fix "sip show user <tab>", so that it
+ actually shows results, instead of just completing the last
+ entry. (closes issue #16675) Reported by: pj
+
+2011-01-25 17:42 +0000 [r303769] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 303765 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25
+ Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages
+ breaks overlap dialing. Issue #16789 was a good idea.
+ Unfortunately, it breaks overlap dialing through Asterisk. There
+ is not enough information available at this point to know if
+ dialing is complete. The ast_exists_extension(),
+ ast_matchmore_extension(), and ast_canmatch_extension() calls are
+ not adequate to detect a dial through extension pattern of "_9!".
+ Workaround is to use the dialplan Proceeding() application early
+ in non-dial through extensions. * Effectively revert issue
+ #16789. * Allow outgoing overlap dialing to hear dialtone and
+ other early media. A PROGRESS "inband-information is now
+ available" message is now sent after the SETUP_ACKNOWLEDGE
+ message for non-digital calls. An AST_CONTROL_PROGRESS is now
+ generated for incoming SETUP_ACKNOWLEDGE messages for non-digital
+ calls. * Handling of the AST_CONTROL_CONGESTION in
+ chan_dahdi/sig_pri was inconsistent with the cause codes. * Added
+ better protection from sending out of sequence messages by
+ combining several flags into a single enum value representing
+ call progress level. * Added diagnostic messages for deferred
+ overlap digits handling corner cases. (closes issue #17085)
+ Reported by: shawkris (closes issue #18509) Reported by: wimpy
+ Patches: issue18509_early_media_v1.8_v3.patch uploaded by
+ rmudgett (license 664) Expanded upon
+ issue18509_early_media_v1.8_v3.patch to include analog and SS7
+ because of backporting requirements. Tested by: wimpy, rmudgett
+ ........
+
+2011-01-25 16:59 +0000 [r303677] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 303676 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25
+ Jan 2011) | 20 lines Fix voicemail sequencing for file based
+ storage. A previous change was made to account for when the
+ number of voicemail messages exceeds the max limit to be handled
+ properly, but it caused gaps in the messages to not be properly
+ handled. This has now been resolved. In later non 1.4 branches,
+ it appears that resequencing wasn't even occurring due from what
+ appears and accidental code removal. (closes issue #18498)
+ Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by
+ jpeeler (license 325) (closes issue #18486) Reported by: bluefox
+ Patches: bug18486.patch uploaded by jpeeler (license 325)
+ ........
+
+2011-01-24 20:49 +0000 [r303548] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, main/pbx.c, /, apps/app_meetme.c,
+ main/features.c, include/asterisk/channel.h: Merged revisions
+ 303546 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011)
+ | 31 lines Fix channel redirect out of MeetMe() and other issues
+ with channel softhangup. Mantis issue #18585 reports that a
+ channel redirect out of MeetMe() stopped working properly. This
+ issue includes a patch that resolves the issue by removing a call
+ to ast_check_hangup() from app_meetme.c. I left that in my patch,
+ as it doesn't need to be there. However, the rest of the patch
+ fixes this problem with or without the change to app_meetme. The
+ key difference between what happens before and after this patch
+ is the effect of the END_OF_Q control frame. After END_OF_Q is
+ hit in ast_read(), ast_read() will return NULL. With the
+ ast_check_hangup() removed, app_meetme sees this which causes it
+ to exit as intended. Checking ast_check_hangup() caused
+ app_meetme to exit earlier in the process, and the target of the
+ redirect saw the condition where ast_read() returned NULL.
+ Removing ast_check_hangup() works around the issue in app_meetme,
+ but doesn't solve the issue if another application did the same
+ thing. There are also other edge cases where if an application
+ finishes at the same time that a redirect happens, the target of
+ the redirect will think that the channel hung up. So, I made some
+ changes in pbx.c to resolve it at a deeper level. There are
+ already places that unset the SOFTHANGUP_ASYNCGOTO flag in an
+ attempt to abort the hangup process. My patch extends this to
+ remove the END_OF_Q frame from the channel's read queue, making
+ the "abort hangup" more complete. This same technique was used in
+ every place where a softhangup flag was cleared. (closes issue
+ #18585) Reported by: oej Tested by: oej, wedhorn, russell Review:
+ https://reviewboard.asterisk.org/r/1082/ ........
+
+2011-01-21 21:48 +0000 [r303285] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 303284 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan
+ 2011) | 8 lines Reset configuration before parsing users.conf.
+ Some values configured in chan_dahdi.conf were able to leak in to
+ users.conf configuration. This was surprising users, and
+ potentially setting non-sane "defaults". ASTNOW-125 ........
+
+2011-01-21 16:12 +0000 [r303273] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_dial.c: Fix changes to L() flag in Dial(). Tony
+ Mountifield pointed out an error I had in my patch. I was a bit
+ too aggressive on changing 'seconds' to 'milliseconds'. So I
+ decided to do some additioanl testing and have no changed just
+ the appropriate lines. One line says milliseconds, and the other
+ says seconds. Probably should change this to be either just
+ seconds or milliseconds, but I've spent too much time on this
+ already :) (issue #18264)
+
+2011-01-20 19:56 +0000 [r303106] Shaun Ruffell <sruffell@digium.com>
+
+ * main/features.c: main/features: Use POLLPRI when waiting for
+ events on parked channels. This change resolves a regression in
+ the 1.6.2 when converting from select to poll. The DAHDI timers
+ use POLLPRI to indicate that the timer fired, but features was
+ not waiting for that flag. The result was no audio for MOH when a
+ call was parked and res_timing_dahdi was in use. This patch is
+ slightly modified from the one on the mantis issue. It does not
+ set an exception on the channel if the POLLPRI flag is set.
+ (closes issue #18262) Reported by: francesco_r Patches:
+ patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
+ Tested by: francesco_r, rfrantik, one47
+
+2011-01-20 17:07 +0000 [r303008] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
+ 303007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011)
+ | 8 lines Add new queue strategy to preserve behavior for when
+ queue members moved to ao2. Add queue strategy called "rrordered"
+ to mimic old behavior from when queue members were stored in a
+ linked list. ABE-2707 ........
+
+2011-01-20 16:11 +0000 [r302920] Russell Bryant <russell@digium.com>
+
+ * apps/app_privacy.c: Resolve a compiler warning.
+
+2011-01-20 15:42 +0000 [r302917] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_dial.c, /: Option L() is milliseconds, not seconds. >
+ Change the verbose output of option L() to say milliseconds and
+ not seconds > as the value is in milliseconds. > > (closes issue
+ #18264) > Reported by: jacco > Patches: > app_dial_patch.txt
+ uploaded by lmadsen (license 10)
+
+2011-01-19 23:47 +0000 [r302833] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c: Support greetingsfolder as documented in
+ voicemail.conf.sample. (closes issue #17870) Reported by:
+ edhorton Patches:
+ __20100816-app_voicemail-greetingsfolder-support.txt uploaded by
+ lmadsen (license 10)
+
+2011-01-19 23:06 +0000 [r302788] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: Turn a noisy verbose message into a debug
+ message. This can drown your console if you're using the AMI over
+ HTTP.
+
+2011-01-19 21:25 +0000 [r302693] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 302671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011)
+ | 15 lines DTMF transfer plays the wrong sounds for wrong number
+ or other call failure. * Set the default for features.conf.sample
+ xferfailsound option to "beeperr" as documented instead of
+ "pbx-invalid" and corrected the use of it in DTMF blind transfer
+ (#1). * Improved DTMF blind transfer handling of wrong numbers.
+ Most of the concerns in this issue were taken care of by the
+ patch for issue 17999: Issues with DTMF triggered attended
+ transfers. (closes issue #18379) Reported by: gincantalupo Tested
+ by: rmudgett ........
+
+2011-01-19 21:22 +0000 [r302599-302675] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * include/asterisk/astdb.h, /: Merged revisions 302663 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19
+ Jan 2011) | 2 lines Add some API documentation ........
+
+ * main/app.c: Kill zombies. When we ast_safe_fork() with a non-zero
+ argument, we're expected to reap our own zombies. On a zero
+ argument, however, the zombies are only reaped when there aren't
+ any non-zero forked children alive. At other times, we accumulate
+ zombies. This code is forward ported from res_agi in 1.4, so that
+ forked children are always reaped, thus preventing an
+ accumulation of zombie processes. (closes issue #18515) Reported
+ by: ernied Patches: 20101221__issue18515.diff.txt uploaded by
+ tilghman (license 14) Tested by: ernied
+
+2011-01-19 19:02 +0000 [r302504-302554] Sean Bright <sean@malleable.com>
+
+ * main/utils.c: Don't call strlen() when we only need to look at
+ the next character or two. (closes issue #18042) Reported by:
+ wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded
+ by wdoekes (license 717)
+
+ * main/features.c: Remove an extraneous \r\n at the end of a
+ parking manager events. (closes issue #18363) Reported by:
+ clegall_proformatique Patches:
+ asterisk_1.8_295998_parking_manager_events_format.patch uploaded
+ by clegall proformatique (license 1139)
+
+ * res/res_agi.c: Properly handle partial reads from fgets() when
+ handling AGIs. When fgets() failed with EAGAIN, we were
+ continually decrementing the available space left in our buffer,
+ resulting in botched command handling. (closes issue #16032)
+ Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by
+ fnordian (license 110)
+
+ * main/utils.c: Make sure that h_length is set when we
+ short-circuit out of ast_gethostbyname. (closes issue #16135)
+ Reported by: thedavidfactor Patches: utils.patch uploaded by
+ thedavidfactor (license 903)
+
+2011-01-19 17:08 +0000 [r302461] Paul Belanger <pabelanger@digium.com>
+
+ * res/res_timing_timerfd.c: Handle 'Resource temporarily
+ unavailable' error more gracefully.
+
+2011-01-19 15:52 +0000 [r302416] Sean Bright <sean@malleable.com>
+
+ * configs/extensions.conf.sample: Remove references to
+ priorityjumping from the sample extensions.conf. Priority jumping
+ was removed from pbx_config in r68970. (closes issue #18622)
+ Reported by: kshumard Patches: extensions.conf.sample.patch
+ uploaded by kshumard (license 92)
+
+2011-01-18 21:40 +0000 [r302313] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 302311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan
+ 2011) | 4 lines URI encode the user part of the contact header.
+ ABE-2705 ........
+
+2011-01-18 20:13 +0000 [r302265] Jeff Peeler <jpeeler@digium.com>
+
+ * main/pbx.c: Convert device state callbacks to ao2 objects to fix
+ a deadlock in chan_sip. Lock scenario presented here: Thread 1
+ holds ast_rdlock_contexts &conlock holds handle_statechange hints
+ holds handle_statechange hint waiting for cb_extensionstate
+ Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds
+ handle_request_do &netlock holds find_call sip_pvt_ptr waiting
+ for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911
+ (ast_rdlock_contexts) Chan_sip has an established locking order
+ of locking the sip_pvt and then getting the context lock. So the
+ as stated by the summary, the operations in thread 2 have been
+ modified to no longer require the context lock. (closes issue
+ #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch
+ uploaded by one47 (license 23), modified by me Review:
+ https://reviewboard.asterisk.org/r/1072/
+
+2011-01-18 18:07 +0000 [r302173] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 302172 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
+ | 88 lines Issues with DTMF triggered attended transfers. Issue
+ #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
+ features.conf for attended transfer). 3) A hears MOH. B dial
+ number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
+ MOH. C ringing. 6) A hangup. C still ringing until
+ "atxfernoanswertimeout" expires. For v1.4 C will ring forever
+ until C answers the dead line. (Issue #17096) Problem: When A and
+ B hangup, C is still ringing. Issue #18395 SIP call limit of B is
+ 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
+ ringing 4. Timeout waiting for C to answer 5. Recall to B fails
+ because B has reached its call limit. Because B reached its call
+ limit, it cannot do anything until the transfer it started
+ completes. Issue #17273 Same scenario as issue 18395 but party B
+ is an FXS port. Party B cannot do anything until the transfer it
+ started completes. If B goes back off hook before C answers, B
+ hears ringback instead of the expected dialtone. ********** Note
+ for the issue #17273 and #18395 fix: DTMF attended transfer works
+ within the channel bridge. Unfortunately, when either party A or
+ B in the channel bridge hangs up, that channel is not completely
+ hung up until the transfer completes. This is a real problem
+ depending upon the channel technology involved. For chan_dahdi,
+ the channel is crippled until the hangup is complete. Either the
+ channel is not useable (analog) or the protocol disconnect
+ messages are held up (PRI/BRI/SS7) and the media is not released.
+ For chan_sip, a call limit of one is going to block that endpoint
+ from any further calls until the hangup is complete. For party A
+ this is a minor problem. The party A channel will only be in this
+ condition while party B is dialing and when party B and C are
+ conferring. The conversation between party B and C is expected to
+ be a short one. Party B is either asking a question of party C or
+ announcing party A. Also party A does not have much incentive to
+ hangup at this point. For party B this can be a major problem
+ during a blonde transfer. (A blonde transfer is our term for an
+ attended transfer that is converted into a blind transfer. :))
+ Party B could be the operator. When party B hangs up, he assumes
+ that he is out of the original call entirely. The party B channel
+ will be in this condition while party C is ringing, while
+ attempting to recall party B, and while waiting between call
+ attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
+ fix the problem. It will replace the party B channel technology
+ with a NULL channel driver to complete hanging up the party B
+ channel technology. The consequences of this code is that the 'h'
+ extension will not be able to access any channel technology
+ specific information like SIP statistics for the call.
+ ATXFER_NULL_TECH is not defined by default. ********** (closes
+ issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
+ SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
+ rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
+ shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
+ grecco Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1047/ ........
+
+2011-01-17 16:53 +0000 [r302049] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Merged revisions 293493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 [^] ........
+ r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010)
+ | 14 lines Only offer codecs both sides support for directmedia
+ When using directmedia, Asterisk needs to limit the codecs
+ offered to just the ones that both sides recognize, otherwise
+ they may end up sending audio that the other side doesn't
+ understand. (closes issue 0017403) Reported by: one47 Patches:
+ sip_codecs_simplified4 uploaded by one47 (license 23) Tested by:
+ one47, falves11 Review: https://reviewboard.asterisk.org/r/967/
+ [^] ........ Backporting a bugfix that should have been included.
+
+2011-02-22 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.17 Released.
+
+ * Merged in changes related to AST-2011-002
+
+2011-02-16 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.17-rc3 Released.
+
+ ------------------------------------------------------------------------
+ r308002 | qwell | 2011-02-15 17:32:21 -0600 (Tue, 15 Feb 2011) | 10
+ lines
+
+ Fix regression that changed behavior of queues when ringing a queue
+ member.
+
+ This reverts r298596, which was to fix a highly bizarre and contrived
+ issue with a queue member that called into his own queue being
+ transferred back into his own queue. I couldn't reproduce that issue
+ in any way. I think one of the other recent transfer fixes actually
+ fixed this.
+
+ (closes issue 0018747)
+ Reported by: vrban
+
+ ------------------------------------------------------------------------
+
+2011-01-20 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.17-rc2 Released.
+
+ ------------------------------------------------------------------------
+ r302172 | rmudgett | 2011-01-18 12:04:37 -0600 (Tue, 18 Jan 2011) | 88
+ lines
+
+ Issues with DTMF triggered attended transfers.
+
+ Issue 0017999
+ 1) A calls B. B answers.
+ 2) B using DTMF dial *2 (code in features.conf for attended transfer).
+ 3) A hears MOH. B dial number C
+ 4) C ringing. A hears MOH.
+ 5) B hangup. A still hears MOH. C ringing.
+ 6) A hangup. C still ringing until "atxfernoanswertimeout" expires.
+ For v1.4 C will ring forever until C answers the dead line. (Issue
+ 0017096)
+
+ Problem: When A and B hangup, C is still ringing.
+
+ Issue 0018395
+ SIP call limit of B is 1
+ 1. A call B, B answered
+ 2. B *2(atxfer) call C
+ 3. B hangup, C ringing
+ 4. Timeout waiting for C to answer
+ 5. Recall to B fails because B has reached its call limit.
+
+ Because B reached its call limit, it cannot do anything until the
+ transfer
+ it started completes.
+
+ Issue 0017273
+ Same scenario as issue 18395 but party B is an FXS port. Party B
+ cannot
+ do anything until the transfer it started completes. If B goes back
+ off
+ hook before C answers, B hears ringback instead of the expected
+ dialtone.
+
+ **********
+ Note for the issue 0017273 and 0018395 fix:
+
+ DTMF attended transfer works within the channel bridge. Unfortunately,
+ when either party A or B in the channel bridge hangs up, that channel
+ is
+ not completely hung up until the transfer completes. This is a real
+ problem depending upon the channel technology involved.
+
+ For chan_dahdi, the channel is crippled until the hangup is complete.
+ Either the channel is not useable (analog) or the protocol disconnect
+ messages are held up (PRI/BRI/SS7) and the media is not released.
+
+ For chan_sip, a call limit of one is going to block that endpoint from
+ any
+ further calls until the hangup is complete.
+
+ For party A this is a minor problem. The party A channel will only be
+ in
+ this condition while party B is dialing and when party B and C are
+ conferring. The conversation between party B and C is expected to be a
+ short one. Party B is either asking a question of party C or
+ announcing
+ party A. Also party A does not have much incentive to hangup at this
+ point.
+
+ For party B this can be a major problem during a blonde transfer. (A
+ blonde transfer is our term for an attended transfer that is converted
+ into a blind transfer. :)) Party B could be the operator. When party B
+ hangs up, he assumes that he is out of the original call entirely. The
+ party B channel will be in this condition while party C is ringing,
+ while
+ attempting to recall party B, and while waiting between call attempts.
+
+ WARNING:
+ The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will
+ replace the party B channel technology with a NULL channel driver to
+ complete hanging up the party B channel technology. The consequences
+ of
+ this code is that the 'h' extension will not be able to access any
+ channel
+ technology specific information like SIP statistics for the call.
+
+ ATXFER_NULL_TECH is not defined by default.
+ **********
+
+ (closes issue 0017999)
+ Reported by: iskatel
+ Tested by: rmudgett
+ JIRA SWP-2246
+
+ (closes issue 0017096)
+ Reported by: gelo
+ Tested by: rmudgett
+ JIRA SWP-1192
+
+ (closes issue 0018395)
+ Reported by: shihchuan
+ Tested by: rmudgett
+
+ (closes issue 0017273)
+ Reported by: grecco
+ Tested by: rmudgett
+
+ Review: https://reviewboard.asterisk.org/r/1047/ [^]
+
+ ------------------------------------------------------------------------
+
+ ------------------------------------------------------------------------
+ r303106 | sruffell | 2011-01-20 13:56:35 -0600 (Thu, 20 Jan 2011) | 15
+ lines
+
+ main/features: Use POLLPRI when waiting for events on parked channels.
+
+ This change resolves a regression in the 1.6.2 when converting from
+ select to poll. The DAHDI timers use POLLPRI to indicate that the
+ timer
+ fired, but features was not waiting for that flag. The result was no
+ audio for MOH when a call was parked and res_timing_dahdi was in use.
+
+ This patch is slightly modified from the one on the mantis issue. It
+ does
+ not set an exception on the channel if the POLLPRI flag is set.
+
+ (closes issue 0018262)
+ Reported by: francesco_r
+ Patches:
+ patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
+ Tested by: francesco_r, rfrantik, one47
+ ------------------------------------------------------------------------
+
+2011-01-14 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.17-rc1 Released.
+
+2011-01-14 20:03 +0000 [r301842-301848] lathama <lathama@localhost>:
+
+ * funcs/func_base64.c, funcs/func_aes.c: Add relationships to
+ function documentation. Fix amatuer type mistake
+
+ * funcs/func_base64.c, funcs/func_aes.c: Add relationships to
+ function documentation.
+
+2011-01-13 17:01 +0000 [r301730] Leif Madsen <lmadsen@digium.com>
+
+ * configs/phoneprov.conf.sample: Add static entry for split Polycom
+ 332 firmware. (closes issue #18607) Reported by: cjacobsen
+ Patches: polycom_331.diff uploaded by cjacobsen (license 1029)
+ Tested by: lathama
+
+2011-01-12 21:05 +0000 [r301682] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't reject all SUBSCRIBE auth requests
+ When merging another SUBSCRIBE fix from 1.4, some braces were put
+ in the wrong place. This patch fixes that. (closes issue #18597)
+ Reported by: thsgmbh
+
+2011-01-12 18:50 +0000 [r301594] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /: Removed a usleep(1) that shouldn't be
+ necessary in session_do, and removed the ms_t member from the
+ mansession_session structure. Merged revisions 301591 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan
+ 2011) | 5 lines Don't store the thread id for the manager session
+ in the structure we pass to the thread for the manager session.
+ ABE-2543 ........
+
+2011-01-12 18:11 +0000 [r301503] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 301502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011)
+ | 12 lines Fix CPU spike when pressing DTMF after agent login.
+ The problem here is that DTMF was being continuously deferred and
+ requeued since ast_safe_sleep is called in a loop. There are
+ serveral other places in the code that sleeps and then loops in a
+ similar fashion. Because of this fact I opted to not defer DTMF
+ any more, which will not affect the original fix:
+ https://reviewboard.asterisk.org/r/674 (closes issue #18130)
+ Reported by: rgj ........
+
+2011-01-11 19:14 +0000 [r301310] Paul Belanger <pabelanger@digium.com>
+
+ * configs/extensions.conf.sample: Fix a logic issue when passing
+ context ARG
+
+2011-01-11 18:42 +0000 [r301307] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/utils.c: Merged revisions 301305 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan
+ 2011) | 4 lines Prevent buffer overflows in ast_uri_encode()
+ ABE-2705 ........
+
+2011-01-09 21:38 +0000 [r301176-301220] Paul Belanger <pabelanger@digium.com>
+
+ * autoconf/ast_ext_lib.m4, configure, configure.ac: SOUND_CACHE_DIR
+ now defaults to empty Sounds files included in the Asterisk
+ tarball were being ignored and re-downloaded. Users wanting to
+ cache the files can still override the setting using the
+ --with-sounds-cache option. (closes issue #18589) Reported by:
+ pabelanger Patches: issue18589.patch uploaded by pabelanger
+ (license 224) Tested by: pabelanger Review:
+ https://reviewboard.asterisk.org/r/1074/
+
+ * apps/app_verbose.c: Indicate log level argument for Log() is not
+ optional (closes issue #18586) Reported by: kshumard Patches:
+ app_verbose.c.patch uploaded by kshumard (license 92)
+
+2011-01-07 20:52 +0000 [r301089] Jason Parker <jparker@digium.com>
+
+ * apps/app_meetme.c: Initialize useropts/adminopts in case there is
+ no column in the realtime DB. (closes issue #18182) Reported by:
+ dimas Patches: v1-18182.patch uploaded by dimas (license 88)
+ Tested by: dimas
+
+2011-01-07 19:57 +0000 [r300951-301046] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c: Fix regression causing forwarding
+ voicemails to not work with file storage. I had actually already
+ fixed this in 295200 in 1.4 and thought it wasn't missing in the
+ other branches for some reason. (closes issue #18358) Reported
+ by: cabal95
+
+ * apps/app_voicemail.c, /: Merged revisions 300918 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07
+ Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played
+ at the correct time. Specifically in the case of timing out but
+ not leaving voicemail nothing should be heard. And when leaving
+ voicemail it should be heard. ABE-2647 ........
+
+2011-01-05 18:54 +0000 [r300622] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/res_odbc.c, /: Merged revisions 300621 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011)
+ | 10 lines Use the sanity check in place of the
+ disconnect/connect cycle. The disconnect/connect cycle has the
+ potential to cause random crashes. (closes issue #18243) Reported
+ by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147)
+ Tested by: ks3 ........
+
+2011-01-05 16:28 +0000 [r300574] Paul Belanger <pabelanger@digium.com>
+
+ * cdr/cdr_sqlite.c: Change deprecated message to LOG_WARNING Also
+ removed latter part of message Discussed on #asterisk-dev
+
+2011-01-04 21:52 +0000 [r300431-300520] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_iax2.c, main/xmldoc.c, channels/chan_sip.c,
+ channels/chan_agent.c: Fix backwards and broken XML
+ documentation. (closes issue #18547) Reported by: jcovert
+ Patches: xmldoc.c.patch uploaded by jcovert (license 551)
+ chan_iax2.c.doc.patch uploaded by jcovert (license 551)
+ chan_sip.c.patch uploaded by jcovert (license 551)
+ chan_agent.c.patch uploaded by jcovert (license 551)
+
+ * configs/users.conf.sample: Add some documentation to
+ users.conf.sample. (closes issue #18531) Reported by: lathama
+ Patches: users.conf.sample2.diff uploaded by lathama (license
+ 1028) Tested by: lathama
+
+2011-01-04 20:59 +0000 [r300429] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
+ Merged revisions 300428 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011)
+ | 4 lines Update the autosupport script from Digium support.
+ (closes AST-395) ........
+
+2011-01-04 17:37 +0000 [r300298] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 300216 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011)
+ | 15 lines Don't authenticate SUBSCRIBE re-transmissions This
+ only skips authentication on retransmissions that are already
+ authenticated. A similar method is already used for INVITES. This
+ is the kind of thing we end up having to do when we don't have a
+ transaction layer... (closes issue #18075) Reported by: mdu113
+ Patches: diff.txt uploaded by twilson (license 396) Tested by:
+ twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/
+ ........
+
+2011-01-03 23:02 +0000 [r300165] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Use correct variable for atxfercallbackretries
+ config option. * Misc formatting changes.
+
+2010-12-28 18:51 +0000 [r299864] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_chanspy.c: Documentation typo
+
+2010-12-25 10:05 +0000 [r299625] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * channels/chan_local.c, /: Merged revisions 299624 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25
+ Dec 2010) | 5 lines Move check for extension existence below
+ variable inheritance, due to the possible use of an eswitch.
+ (closes issue #16228) Reported by: jlaguilar ........
+
+2010-12-23 03:02 +0000 [r299530-299533] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: do not use progress which is for PRI and
+ SS7, add mfcr2_progress member
+
+ * channels/chan_dahdi.c: Enqueue AST_CONTROL_PROGRESS after
+ AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue
+ #18438) Reported by: mariner7 Tested by: moy
+
+2010-12-22 20:03 +0000 [r299448] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c,
+ pbx/ael/ael-test/ref.ael-vtest25,
+ pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-test3:
+ Resolve warnings by disambiguating the "s" extension as used by
+ chan_dahdi from the "s" extension as used by the AEL macros.
+ (closes issue #18480) Reported by: nivek Patches:
+ 20101215__issue18480__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: nivek
+
+2010-12-20 21:25 +0000 [r299242] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 299194,299198,299220 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec
+ 2010) | 6 lines Respond as soon as possible with a 202 Accepted
+ to refer requests. This change also plugs a few memory leaks that
+ can occur when parking sip calls. ABE-2656 ........ r299198 |
+ mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2
+ lines Remove changes to via processing that were not supposed to
+ go into the last commit. ........ r299220 | mnicholson |
+ 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use
+ ast_free() instead of free() ABE-2656 ........
+
+2010-12-20 18:16 +0000 [r299130-299136] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * sample.call: Documentation fix
+
+ * cdr/cdr_pgsql.c: If a call was not answered, then the billsec was
+ calculated unusually large. Also, due to a copy and paste error,
+ a request for the answer field would have given the start value,
+ instead. (closes issue #18460) Reported by: joscas Patches:
+ 20101215__issue18460.diff.txt uploaded by tilghman (license 14)
+ Tested by: joscas
+
+2010-12-20 16:18 +0000 [r299087] Leif Madsen <lmadsen@digium.com>
+
+ * main/features.c: Note that Park() timeout is milliseconds.
+ (closes issue #15758) Reported by: mmurdock Tested by: mmurdock,
+ seanbright
+
+2010-12-20 09:13 +0000 [r299003] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_sip.c: Typos: recieved => received
+
+2010-12-18 00:08 +0000 [r298817-298962] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/say.c: Remove backtrace used for testing merge process
+
+ * main/astobj2.c, utils/conf2ael.c, include/asterisk/logger.h,
+ configure, build_tools/menuselect-deps.in, main/logger.c,
+ utils/ael_main.c, utils/hashtest2.c, makeopts.in,
+ utils/check_expr.c, utils/refcounter.c, include/asterisk/utils.h,
+ build_tools/cflags-devmode.xml, /, main/Makefile,
+ include/asterisk/autoconfig.h.in, main/say.c, configure.ac,
+ utils/hashtest.c, main/utils.c: Merged revisions 298905 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010)
+ | 6 lines Let Asterisk find better backtrace information with
+ libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will
+ use libbfd to search for better symbol information within both
+ the Asterisk binary, as well as loaded modules, to assist when
+ using inline backtraces to track down problems. ........
+
+ * configure, configure.ac: Also include PTHREAD_LIBS and
+ PTHREAD_CFLAGS for SQLite 3, as it's needed on some platforms.
+ (closes issue #18493) Reported by: pprindeville Patches:
+ asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347)
+ Tested by: pprindeville
+
+2010-12-16 23:30 +0000 [r298597-298684] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 298683 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16
+ Dec 2010) | 2 lines After recording only silence for a voicemail
+ prepending, restore backup files. ........
+
+ * apps/app_queue.c, /: Merged revisions 298596 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010)
+ | 7 lines Fix improper hangup when doing an attended transfer to
+ queue. Had to indicate ringing in wait_for_answer so the attended
+ transfer code would not try and hang up the local channel it
+ created, which would kill the call. ABE-2624 ........
+
+2010-12-16 09:04 +0000 [r298393-298481] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/res_config_odbc.c, /: Merged revisions 298480 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16
+ Dec 2010) | 14 lines Only increment the pointer once per loop,
+ otherwise we corrupt the value. (closes issue #18251) Reported
+ by: bcnit Patches: 20101110__issue18251.diff.txt uploaded by
+ tilghman (license 14) Tested by: trev, jthurman, elguero (closes
+ issue #18279) Reported by: zerohalo Patches:
+ 20101109__issue18279.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........
+
+ * funcs/func_dialgroup.c: Eliminate duplicates from container.
+ (closes issue #18091) Reported by: bunny Patches:
+ 20101006__issue18091.diff.txt uploaded by tilghman (license 14)
+ Tested by: bunny
+
+ * /, cdr/cdr_sqlite.c: Merged revisions 298392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010)
+ | 8 lines Unregister before shutting down the connection, to
+ avoid a race. (closes issue #18481) Reported by: pabelanger
+ Patches: 20101215__issue18481.diff.txt uploaded by tilghman
+ (license 14) Tested by: pabelanger ........
+
+2010-12-15 21:31 +0000 [r298346] Sean Bright <sean@malleable.com>
+
+ * main/astobj2.c, /: Merged revisions 298345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298345 | seanbright | 2010-12-15 16:28:29 -0500 (Wed, 15 Dec
+ 2010) | 6 lines Fix reference and container leaks when running
+ 'astobj2 test.' We need to make sure that ao2_iterator_destroy is
+ called once for each time that ao2_iterator_init is called. Also
+ make sure to unref a newly allocated object that we've linked
+ into a container. ........
+
+2010-12-13 17:04 +0000 [r298194] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 298193 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13
+ Dec 2010) | 19 lines Outgoing PRI/BRI calls cannot do DTMF
+ triggered transfers. Outgoing PRI/BRI calls cannot do DTMF
+ triggered transfers if a PROCEEDING message is not received. The
+ debug output shows that the DTMF begin event is seen, but the
+ DTMF end event is missing. When the DTMF begin happens, the call
+ is muted so we now have one way audio (until a DTMF end event is
+ somehow seen). * Made set the proceeding flag when the
+ PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin
+ and DTMF end events if we are overlap dialing and have not seen a
+ PROCEEDING message. * Added a debug message when absorbing a DTMF
+ event. JIRA SWP-2690 JIRA ABE-2697 ........
+
+2011-01-12 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.16 Released.
+
+2011-01-12 Leif Madsen <lmadsen@digium.com>
+
+ * Merge in changes for configure script to resolve issue for
+ Debian package builders.
+
+ ------------------------------------------------------------------------
+ r301220 | pabelanger | 2011-01-09 15:38:25 -0600 (Sun, 09 Jan 2011)
+ | 14 lines
+
+ SOUND_CACHE_DIR now defaults to empty
+
+ Sounds files included in the Asterisk tarball were being ignored and
+ re-downloaded. Users wanting to cache the files can still override
+ the setting
+ using the --with-sounds-cache option.
+
+ (closes issue 0018589)
+ Reported by: pabelanger
+ Patches:
+ issue18589.patch uploaded by pabelanger (license 224)
+ Tested by: pabelanger
+
+ Review: https://reviewboard.asterisk.org/r/1074/ [^]
+
+ ------------------------------------------------------------------------
+
+2010-12-13 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.16-rc1 Released.
+
+2010-12-10 16:24 +0000 [r298050] Tilghman Lesher <tlesher@digium.com>
+
+ * main/netsock.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Portability issue on OpenSolaris. Also detect the
+ required structure element, because OpenSolaris defines
+ SIOCGIFHWADDR, but without support for IP sockets. (closes issue
+ #18442) Reported by: ranjtech Patches:
+ 20101209__issue18442.diff.txt uploaded by tilghman (license 14)
+ Tested by: ranjtech
+
+2010-12-09 22:10 +0000 [r297960] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297959 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
+ | 14 lines Ignore spurious REGISTER requests If a REGISTER
+ request with a Call-ID matching an existing transaction is
+ received it was possible that the REGISTER request would
+ overwrite the initreq of the private structure. This info is used
+ to generate messages for other responses in the transaction. This
+ patch ignores REGISTER requests that match non-REGISTER
+ transactions. (closes issue #18051) Reported by: eeman Tested by:
+ twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
+
+2010-12-08 18:04 +0000 [r297908] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Use inheritance to get correct
+ results for SIPFROMDOMAIN. (from an internal Digium discussion)
+
+2010-12-07 22:58 +0000 [r297824] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 297823 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
+ | 12 lines Revert code that changed SSRC for DTMF. Some previous
+ behavior was attempted to be restored, but mistakingly I did not
+ realize that the previous behavior was incorrect. This fixes DTMF
+ not being detected since DTMF shouldn't cause the SSRC to change.
+ (related to issue #17404) (closes issue #18189) (closes issue
+ #18352) Reported by: marcbou Tested by: cmbaker82 ........
+
+2010-12-07 22:40 +0000 [r297713-297819] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/org.asterisk.muted.plist (added), Makefile,
+ utils/muted.c, /: Merged revisions 297818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
+ | 4 lines Use non-deprecated APIs for CoreAudio Review:
+ https://reviewboard.asterisk.org/r/1040/ ........
+
+ * apps/app_followme.c, /: Merged revisions 297689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
+ | 8 lines Don't create a Local channel if the target extension
+ does not exist. (closes issue #18126) Reported by: junky Patches:
+ followme.diff uploaded by junky (license 177) (partially
+ restructured by me to avoid a possible memory leak) ........
+
+2010-12-06 22:03 +0000 [r297605] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297603 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
+ | 12 lines Improve handling of REGISTER requests with multiple
+ contact headers. The changes here attempt to more strictly follow
+ RFC 3261 section 10.3. Basically the following will now cause a
+ 400 Bad Response to be returned, if: - multiple Contact headers
+ are present with one set to expire all bindings ("*") - wildcard
+ parameter is specified for Contact without Expires header or
+ Expires header is not set to zero. ABE-2442 ABE-2443 ........
+
+2010-12-03 17:40 +0000 [r297534] Sean Bright <sean@malleable.com>
+
+ * channels/chan_console.c: The CLI command should not contain
+ <placeholder>s, these are for descriptions.
+
+2010-12-02 20:06 +0000 [r297405] Paul Belanger <pabelanger@digium.com>
+
+ * Makefile, /: Merged revisions 297404 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
+ 2010) | 7 lines Resolve compile error under FreeBSD We now set
+ _ASTCFLAGS+=-march=i686 for i386 processors, still allowing
+ ASTCFLAGS to override the setting. Review:
+ https://reviewboard.asterisk.org/r/1043/ ........
+
+2010-12-02 18:07 +0000 [r297311] Terry Wilson <twilson@digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 297310 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
+ | 12 lines Initialize offset for adaptive jitter buffer When the
+ adaptive jitter buffer is enabled in sip.conf, the first frame
+ placed in the jitter buffer fails with something like:
+ jb_warning_output: Resyncing the jb. last_delay 0, this delay
+ -215886466, threshold 1000, new offset 215886466 This happens
+ because the offset is not initialized before calling jb_put().
+ This patch modifies jb_put_first_adaptive() to set the offset to
+ the frame's timestamp. Review:
+ https://reviewboard.asterisk.org/r/1041/ ........
+
+2010-12-02 13:16 +0000 [r297229] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 297228 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
+ | 6 lines Add "DAHDI" to a couple of app_meetme error messages.
+ This is in response to some questions on IRC. To the user, there
+ was nothing that made it obvious that this error had anything to
+ do with DAHDI not being loaded. ........
+
+2010-12-02 08:55 +0000 [r297186] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 297185 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5
+ lines If we get a NOTIFY from a non-existing subscription we
+ should answer with 481, not bad event. If we answer 481 the
+ subscription that we don't want will be cancelled. ........
+
+2010-12-01 17:52 +0000 [r297073] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
+ | 23 lines Fix not stopping MOH when transfered local channel
+ queue member is answered. The problem here is only present when
+ local channels are used with the MOH passthru option as well as
+ no optimization (/nm). I will describe the slightly bizarre
+ scenario that was used to test, where phones B and C are queue
+ members: Phone A dials into a queue with two members using local
+ channels and the above options. Phone B answers. Phone A blind
+ transfers phone B into the same queue. Phone A hangs up. Phone C
+ answers, but phone B didn't stop playing MOH. In this scenario,
+ the unhold frame that should have gotten to phone B never arrived
+ due to the masquerade from the blind transfer. This is usually
+ fine since app_queue manages the starting and stopping of MOH.
+ However, with the passthrough option enabled when app_queue
+ attempts to stop MOH it tries to do so on the local channel
+ rather than the real channel. The easiest solution was to just
+ make sure to send an unhold frame during the transfer since it
+ wouldn't make sense to have MOH playing after a transfer anyway.
+ This only modifies SIP transfers, but the other transfers did not
+ seem to be a problem. If DTMF based transfers were a problem it
+ might be okay to add ast_moh_stop to finishup, but I didn't want
+ to have to add that unless required. ABE-2624 ........
+
+2010-12-01 17:01 +0000 [r296950-296991] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/frame.h, /: Merged revisions 296990 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01
+ Dec 2010) | 5 lines Clarify documentation on how we store codec
+ preference lists. (closes issue #18397) Reported by: birgita
+ ........
+
+ * channels/chan_iax2.c: Missed initializations caused startup
+ errors on Mac OS X (and possibly others, too).
+
+2010-12-01 00:24 +0000 [r296869] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 296868 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30
+ Nov 2010) | 4 lines Properly restore backup information file when
+ hanging up during message prepending. ABE-2654 ........
+
+2010-11-29 22:54 +0000 [r296671] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 296670 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon,
+ 29 Nov 2010) | 5 lines Make sure nothing else is needed before
+ destroying the scheduler. (closes issue #18398) Reported by:
+ pabelanger ........
+
+2010-11-29 07:27 +0000 [r296533] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: I love standards. There are so many to choose from.
+ Except when there isn't one. Linux and *BSD disagree on the
+ elements within the ucred structure. Detect which one is in use
+ on the system. (closes issue #18384) Reported by: bjm Patches:
+ cred-diffs uploaded by bjm (license 473)
+ 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
+ (license 14) 20101127__issue18384__1.8.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman, bjm
+
+2010-11-27 10:39 +0000 [r296466] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: 18 characters is too short for most date/times
+ (20 is the usual, but we add more in case of greater precision).
+ (closes issue #18369) Reported by: tnakonz
+
+2010-11-26 12:23 +0000 [r296351] Olle Johansson <oej@edvina.net>
+
+ * /, main/say.c: Merged revisions 296309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
+ lines Fix bugs in saying numbers using the Swedish language
+ syntax (closes issue #18355) Reported by: oej Patch by: oej Much
+ help from Peter Lindahl. Testing by the ClearIT team during a
+ coffee break. Review: https://reviewboard.asterisk.org/r/1033/
+ ........
+
+2010-11-24 23:28 +0000 [r296221] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 296213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
+ | 6 lines Make Asterisk less crashy. Since we might not put a new
+ translation path on the channel, go ahead and set it to NULL
+ right after destroying the old one to ensure we don't try to free
+ an invalid translation path later on. ........
+
+2010-11-24 22:42 +0000 [r296166] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 296165 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24
+ Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port
+ after FXS port gets a CallWaiting pip. The FXS connected phone
+ has to have CW/CID support to fail, as it will send back a DTMF
+ 'A' or 'D' when it's ready to receive CallerID. A normal phone
+ with no CID never fails. Also the SIP phone does not hear MOH
+ when the CW call is answered. The DTMF end frame is suppressed
+ when the phone acknowledges the CW signal for CID. The problem is
+ the DTMF begin frame needs to be suppressed as well. The DTMF
+ begin frame is causing SIP to start sending the DTMF RTP frames.
+ Since the DTMF end frame is suppressed, SIP will not stop sending
+ those DTMF RTP packets. * Suppress the DTMF begin and end frames
+ when the channel driver is looking for DTMF digits. * Fixed a
+ couple issues caused by not cleaning up the CID spill if you
+ answer the CW call while it is sending the CID spill. * Fixed not
+ sending CW/CID spill to the phone when the call is natively
+ bridged. (Fixed by not using native bridge if CW/CID is
+ possible.) * Suppress received audio when sending CW/CID spills.
+ The other parties involved do not need to hear the CW/CID spills
+ and may be confused if the CW call is for them. (closes issue
+ #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ NOTE: * v1.4 does not have the main problem fixed by suppressing
+ the DTMF start frames. The other three items fixed are relevant.
+ * If you really must restore native bridging between analog
+ ports, you need to disable CW/CID either by configuring
+ chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
+ dialing the number to temporarily disable CW. ........
+
+2010-11-24 20:23 +0000 [r296001-296083] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 296082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
+ | 12 lines Fix false reporting of an error by set_format(). In
+ the case that the native format was able to be changed to match
+ the new requested format, the code proceeded to attempt to build
+ a translation path, anyway. The result would be NULL, since no
+ translation path is necessary and resulted in this function
+ thinking an error has occurred. This case is now specifically
+ caught and no attempt to build a translation path is attempted.
+ Thanks to our automated tests and bamboo.asterisk.org for
+ catching this problem and making a whole lot of noise when things
+ started failing. :-) ........
+
+ * apps/app_dial.c, main/channel.c, /: Merged revisions 296000 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
+ | 38 lines Handle failures building translation paths more
+ effectively. The problem scenario occurred on a heavily loaded
+ system that was using the codec_dahdi module and exceeded the
+ hardware transcoding capacity. The failure mode at that point was
+ not good. The report came in to us as an Asterisk lock-up. The
+ "core show locks" shows a ton of threads locked up (but no
+ obvious deadlock). Upon deeper investigation, when the system is
+ in this state, the CPU was maxed out. The CPU was being consumed
+ by the Asterisk logger spewing messages on every audio frame for
+ calls set up after transcoder capacity was reached. The purpose
+ of this patch is to make Asterisk handle failures to create a
+ translation path in a more graceful manner. If we can't
+ translate, then the call just needs to be dropped, as it's not
+ going to work. These are the changes: 1) In set_format() of
+ channel.c (which is called by set_read_format() and
+ set_write_format()), it was ignoring if
+ ast_translator_build_path() failed and returned NULL. It now pays
+ attention to that case and returns a result reflecting failure.
+ With this change in place, the bridging code will immediately
+ detect a failure and end the bridge instead of proceeding to try
+ to bridge frames that can't be translated and making channel
+ drivers freak out by sending them frames in a format they weren't
+ expecting. 2) In ast_indicate_data() of channel.c, failure of
+ ast_playtones_start() was ignored. It is now reflected in the
+ return value of the function. This didn't turn out to have any
+ affect on the bug, but seemed like a good change to leave in. 3)
+ In app_dial(), when only sending a call to a single endpoint, it
+ will attempt to do some bridging of its own of early audio. It
+ uses make_compatible() when it's going to do this. However, it
+ ignored failure from make compatible. So, even with the fix from
+ #1, if there was early audio going through app_dial, there would
+ still be a period of invalid frames passing through. After
+ detecting failure here, Dial() exits. ABE-2658 ........
+
+2010-11-23 09:36 +0000 [r295907] Olle Johansson <oej@edvina.net>
+
+ * /, main/say.c: Merged revisions 295906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
+ lines Fix support of saynumber(1,n) in the Swedish language
+ (closes issue #18353) Reported by: oej Review:
+ https://reviewboard.asterisk.org/r/1031/ ........
+
+2010-11-22 20:02 +0000 [r295868] Sean Bright <sean@malleable.com>
+
+ * configs/chan_dahdi.conf.sample: Change some documentation to
+ suggest dahdi_monitor instead of ztmonitor.
+
+2010-11-22 19:28 +0000 [r295843] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/frame.h, main/channel.c, main/pbx.c, /,
+ apps/app_macro.c, include/asterisk/channel.h: Merged revisions
+ 295790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
+ | 46 lines The channel redirect function (CLI or AMI) hangs up
+ the call instead of redirecting the call. To recreate the
+ problem: 1) Party A calls Party B 2) Invoke CLI "channel
+ redirect" command to redirect channel call leg associated with A.
+ 3) All associated channels are hung up. Note that if the CLI
+ command were done on the channel call leg associated with B it
+ works. This regression was a result of the fix for issue #16946
+ (https://reviewboard.asterisk.org/r/740/). The regression affects
+ all features that use an async goto to execute the dialplan
+ because of an external event: Channel redirect, AMI redirect, SIP
+ REFER, and FAX detection. The struct ast_channel._softhangup code
+ is a mess. The variable is used for several purposes that do not
+ necessarily result in the call being hung up. I have added
+ doxygen comments to describe how the various _softhangup bits are
+ used. I have corrected all the places where the variable was
+ tested in a non-bit oriented manner. The primary fix is the new
+ AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
+ the soft hangup requests that do not normally result in a hangup
+ do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
+ Reported by: SantaFox (closes issue #18185) Reported by:
+ kwemheuer (closes issue #18211) Reported by: zahir_koradia
+ (closes issue #18230) Reported by: vmarrone (closes issue #18299)
+ Reported by: mbrevda (closes issue #18322) Reported by: nerbos
+ Review: https://reviewboard.asterisk.org/r/1013/ ........
+
+2010-11-20 00:45 +0000 [r295710] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h, main/event.c: Fix cache of device state
+ changes for multiple servers. This patch addresses a regression
+ where device states across multiple servers were not being
+ processing completely correctly. The code works to determine the
+ overall state by looking at the last known state of a device on
+ each server. However, there was a regression due to some invasive
+ rewrites of how the cache works that led to the cache only
+ storing the last device state change for a device, regardless of
+ which server it was on. The code is set up to cache device state
+ change events by ensuring that each event in the cache has a
+ unique device name + entity ID (server ID). The code that was
+ responsible for comparing raw information elements (which EID is)
+ always returned a match due to a memcmp() with a length of 0.
+ There isn't much code to fix the actual bug. This patch also
+ introduces a new CLI command that was very useful for debugging
+ this problem. The command allows you to dump the contents of the
+ event cache. (closes issue #18284) Reported by: klaus3000
+ Patches: issue18284.rev1.txt uploaded by russell (license 2)
+ Tested by: russell, klaus3000 (closes issue #18280) Reported by:
+ klaus3000 Review: https://reviewboard.asterisk.org/r/1012/
+
+2010-11-19 21:55 +0000 [r295672] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 295628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010)
+ | 8 lines Discard responses with more than one Via This is not a
+ perfect solution as headers that are joined via commas are not
+ detected. This is a parsing issue that to fix "correctly" would
+ necessitate a new SIP parser. Review:
+ https://reviewboard.asterisk.org/r/1019/ ........
+
+2010-11-18 17:51 +0000 [r295440] Paul Belanger <pabelanger@digium.com>
+
+ * res/res_jabber.c, include/asterisk/jabber.h: Fix compiler
+ warnings when using openssl-dev 1.0.0+ Review:
+ https://reviewboard.asterisk.org/r/1016/
+
+2010-11-16 22:57 +0000 [r295281] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Merged revisions 295280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010)
+ | 1 line Dead code elimination in channel.c:ast_channel_bridge()
+ variable who. ........
+
+2010-12-02 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.15 Released.
+
+2010-11-15 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.15-rc1
+
+2010-11-15 18:24 +0000 [r294988-295062] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_expr.c (added), /: Merged revisions 295026 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010)
+ | 2 lines Create test verifying results of expression parser
+ ........
+
+ * funcs/func_curl.c: It is possible to crash Asterisk by feeding
+ the curl engine invalid data. (closes issue #18161) Reported by:
+ wdoekes Patches: 20101029__issue18161.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman
+
+2010-11-12 21:14 +0000 [r294904-294910] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c: Return correct error code if lock path
+ fails. The recent changes to open_mailbox actually caused it to
+ be fixed, but let's be consistent. Reported by alecdavis in
+ asterisk-dev.
+
+ * apps/app_voicemail.c, /: Merged revisions 294903 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12
+ Nov 2010) | 16 lines Fix regression causing abort in voicemail
+ after opening a mailbox with no mesgs. In order to be more safe,
+ some error handling code was changed to respect more error
+ conditions including the potential memory allocation failure for
+ deleted and heard message tracking introduced in 293004. However,
+ last_message_index returns -1 for zero messages (perhaps as
+ expected) and was triggering the stricter error checking. Because
+ last_message_index is only called directly in one place, just
+ return 0 from open_mailbox (for file based storage) when no
+ messages are detected unless a real error has occurred. (closes
+ issue #18240) Reported by: leobrown Patches:
+ bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
+ Tested by: pabelanger ........
+
+2010-11-12 02:44 +0000 [r294822] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 294821 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11
+ Nov 2010) | 11 lines Asterisk is getting a "No D-channels
+ available!" warning message every 4 seconds. Asterisk is just
+ whining too much with this message: "No D-channels available!
+ Using Primary channel XXX as D-channel anyway!". Filtered the
+ message so it only comes out once if there is no D channel
+ available without an intervening D channel available period.
+ (closes issue #17270) Reported by: jmls ........
+
+2010-11-11 21:57 +0000 [r294639-294733] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
+ | 18 lines Fix problem with qualify option packets for realtime
+ peers never stopping. The option packets not only never stopped,
+ but if a realtime peer was not in the peer list multiple options
+ dialogs could accumulate over time. This scenario has the
+ potential to progress to the point of saturating a link just from
+ options packets. The fix was to ensure that the poke scheduler
+ checks to see if a peer is in the peer list before continuing to
+ poke. The reason a peer must be in the peer list to be able to
+ properly manage an options dialog is because otherwise the call
+ pointer is lost when the peer is regenerated from the database,
+ which is how existing qualify dialogs are detected. (closes issue
+ #16382) (closes issue #17779) Reported by: lftsy Patches:
+ bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
+ zerohalo ........
+
+ * main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
+ revisions 294384 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
+ | 47 lines Fix a deadlock in device state change processing.
+ Copied from some notes from the original author (Russell):
+ Deadlock scenario: Thread 1: device state change thread Holds -
+ rdlock on contexts Holds - hints lock Waiting on channels
+ container lock Thread 2: SIP monitor thread Holds the "iflock"
+ Holds a sip_pvt lock Holds channel container lock Waiting for a
+ channel lock Thread 3: A channel thread (chan_local in this case)
+ Holds 2 channel locks acquired within app_dial Holds a 3rd
+ channel lock it got inside of chan_local Holds a local_pvt lock
+ Waiting on a rdlock of the contexts lock A bunch of other threads
+ waiting on a wrlock of the contexts lock To address this
+ deadlock, some locking order rules must be put in place and
+ enforced. Existing relevant rules: 1) channel lock before a pvt
+ lock 2) contexts lock before hints lock 3) channels container
+ before a channel What's missing is some enforcement of the order
+ when you involve more than any two. To fix this problem, I put in
+ some code that ensures that (at least in the code paths involved
+ in this bug) the locks in (3) come before the locks in (2). To
+ change the operation of thread 1 to comply, I converted the
+ storage of hints to an astobj2 container. This allows processing
+ of hints without holding the hints container lock. So, in the
+ code path that led to thread 1's state, it no longer holds either
+ the contexts or hints lock while it attempts to lock the channels
+ container. (closes issue #18165) Reported by: antonio ABE-2583
+ ........
+
+2010-11-10 23:16 +0000 [r294571] Tilghman Lesher <tlesher@digium.com>
+
+ * main/features.c: Actually pay attention to documented settings in
+ features.conf. (closes issue #16757) Reported by: voxter Patches:
+ 20101012__issue16757.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/994/
+
+2010-11-10 12:41 +0000 [r294500] Russell Bryant <russell@digium.com>
+
+ * main/devicestate.c: Improve a debug message to be more readable
+ and consistent. (closes issue #18282) Reported by: klaus3000
+ Patches: ast_devstate2str-patch.txt uploaded by klaus3000
+ (license 65)
+
+2010-11-09 20:27 +0000 [r294429] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Detect GMime properly on systems where
+ gmime flags and libs are configured with pkg-config. (closes
+ issue #16155) Reported by: jcollie Patches:
+ 20100917__issue16155.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-11-08 22:30 +0000 [r294277-294312] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_timing_timerfd.c: add missing unlock not present in
+ 294277
+
+ * main/timing.c, main/channel.c, res/res_timing_timerfd.c,
+ include/asterisk/timing.h: Fix playback failure when using IAX
+ with the timerfd module. To fix this issue the alert pipe will
+ now be used when the timerfd module is in use. There appeared to
+ be a race that was not solved by adding locking in the timerfd
+ module, but needed to be there anyway. The race was between the
+ timer being put in non-continuous mode in ast_read on the channel
+ thread and the IAX frame scheduler queuing a frame which would
+ enable continuous mode before the non-continuous mode event was
+ read. This race for now is simply avoided. (closes issue #18110)
+ Reported by: tpanton Tested by: tpanton I put tested by tpanton
+ because it was tested on his hardware. Thanks for the remote
+ access to debug this issue!
+
+2010-11-08 20:50 +0000 [r294242] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Go off hold when we get an empty reinvite
+ telling us to. (closes issue 0014448) Reported by: frawd (closes
+ issue #17878) Reported by: frawd
+
+2010-11-05 00:06 +0000 [r293969] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c, /: Merged revisions 293968 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04
+ Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio
+ when receiving unexpected frame sizes. dahdi-linux 2.4.0
+ (specifically commit 9034) added the capability for the wctc4xxp
+ to return more than a single packet of data in response to a
+ read. However, when decoding packets, codec_dahdi was still
+ assuming that the default number of samples was in each read. In
+ other words, each packet your provider sent you, regardless of
+ size, would result in 20 ms of decoded data (30 ms if decoding
+ G723). If your provider was sending 60 ms packets then
+ codec_dahdi would end up stripping 40 ms of data from each
+ transcoded frame resulting in "choppy" audio. This would only
+ affect systems where G729 packets are arriving in sizes greater
+ than 20ms or G723 packets arriving in sizes greater than 30ms.
+ DAHDI-744. ........
+
+2010-11-03 18:31 +0000 [r293806] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 293805 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03
+ Nov 2010) | 20 lines Party A in an analog 3-way call would
+ continue to hear ringback after party C answers. All parties are
+ analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A
+ flash hooks to bring C into 3-way call before C answers. (A and B
+ hear ringback) 4) C answers 5) A continues to hear ringback
+ during the 3-way call. (All parties can hear each other.) * Fixed
+ use of wrong variable in dahdi_bridge() that stopped ringback on
+ the wrong subchannel. * Made several debug messages have more
+ information. A similar issue happens if B and C are SIP channels.
+ B continues to hear ringback. For some reason this only affects
+ v1.8 and trunk. * Don't start ringback on the real and 3-way
+ subchannels when creating the 3-way conference. Removing this
+ code is benign on v1.6.2 and earlier. ........
+
+2010-11-02 23:07 +0000 [r293723] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
+ | 8 lines Add enabled/disabled information for rtautoclear sip
+ show settings output. When setting to zero/"no", the numeric
+ default was shown making it not obvious the disabled setting was
+ respected. (closes issue #18123) Reported by: zerohalo ........
+
+2010-11-02 21:26 +0000 [r293647] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 293639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02
+ Nov 2010) | 6 lines Make warning message have more useful
+ information in it. Change "Unable to get index, and nullok is not
+ asserted" to "Unable to get index for '<channel-name>' on channel
+ <number> (<function>(), line <number>)". ........
+
+2010-10-30 01:49 +0000 [r293340-293417] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 293416 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some more code that serves no purpose.
+ ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 293339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some code that serves no purpose.
+ ........
+
+2010-10-28 19:54 +0000 [r293195-293196] Tilghman Lesher <tlesher@digium.com>
+
+ * main/ast_expr2.c, main/ast_expr2.h: Merged revisions 293194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........
+
+ * /, res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c,
+ res/ael/ael.tab.h: Merged revisions 293194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........
+
+2010-10-28 16:09 +0000 [r293158] Jeff Peeler <jpeeler@digium.com>
+
+ * funcs/func_strings.c: Fix infinite loop in FILTER(). Specifically
+ when you're using characters above \x7f or invalid character
+ escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
+ Patches: issue18060_func_strings_filter_infinite_loop.patch
+ uploaded by wdoekes (license 717) Tested by: wdoekes
+
+2010-10-26 18:33 +0000 [r293118] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 293004 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25
+ Oct 2010) | 29 lines Fix inprocess_container in voicemail to
+ correctly restrict max messages. The comparison function logic
+ was off, so the number of sessions for a given mailbox were not
+ being incremented properly. This problem caused the maximum
+ number of messages per folder to not be respected when
+ simultaneously leaving multiple voicemails just below the
+ threshold. These problems should be fixed by the above, but just
+ in case: Fixed resequence_mailbox to rely on the actual number of
+ detected number of files in a directory rather than just assuming
+ only 10 messages more than the maximum had been left. Also if
+ more messages than the maximum are deleted they are actually
+ removed now. The second purpose of this commit should have been
+ separated out probably, but is related to the above. Again, if
+ the number of messages in a given voicemail folder exceeds the
+ maximum set limit make sure to allocate enough space for the
+ deleted and heard index tracking array. A few random fixes: There
+ was a forgotten decrement of the inprocess count in
+ imap_store_file. When using IMAP storage, do not look in the
+ directory where file based storage messages may still reside and
+ influence the message count. Ensure to use only the first format
+ in sendmail. ABE-2516 ........
+
+2010-10-25 19:06 +0000 [r292867] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 292866 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25
+ Oct 2010) | 27 lines This patch turns chan_local pvts into
+ astobj2 objects. chan_local does some dangerous things involving
+ deadlock avoidance. tech_pvt functions like hangup and
+ queue_frame are provided with a locked channel upon entry. Those
+ functions are completely safe as long as you don't attempt to
+ give up that channel lock, but that is impossible to guarantee
+ due to the required deadlock avoidance necessary to lock both the
+ tech_pvt and both channels involved. In the past, we have tried
+ to account for this by doing things like setting a "glare" flag
+ that indicates what function should destroy the pvt. This was
+ used in local_hangup and local_queue_frame to decided who should
+ destroy the pvt if they collided in separate threads. I have
+ removed the need to do this by converting all chan_local
+ tech_pvts to astobj2. This means we can ref a pvt before deadlock
+ avoidance and not have to worry about that pvt possibly getting
+ destroyed under us. It also cleans up where we destroy the
+ tech_pvt. The only unlink from the tech_pvt container occurs in
+ local_hangup now, which is where it should occur. Since there
+ still may be thread collisions on some functions like
+ local_hangup after deadlock avoidance, I have added some checks
+ to detect those collisions and exit appropriately. I think this
+ patch is going to solve quite a bit of weirdness we have had with
+ local channels in the past. ........
+
+2010-10-22 21:16 +0000 [r292786] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/asterisk.ldif, channels/chan_sip.c,
+ configs/res_ldap.conf.sample: Update the LDIF file for LDAP. The
+ LDIF file asterisk.ldif was quite a bit out of date from the
+ asterisk.ldap-schema file, so I've now updated that to be in
+ sync. The asterisk.ldif file being out of sync was a problem on
+ my systems where I was doing an ldapadd to import the schema into
+ the LDAP database, and the existing file would cause problems and
+ ERROR messages when registering. Additional documention has been
+ added based on feedback in the issue I'm closing. (closes issue
+ #13861) Reported by: scramatte Patches: ldap-update.txt uploaded
+ by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
+ rgenthner
+
+2010-10-21 13:11 +0000 [r292556] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample: Change res_ldap.sample.conf to
+ match the schema. (closes issue #17376) Reported by: jcovert
+ Patches: res_ldap.conf.sample.patch uploaded by jcovert (license
+ 551)
+
+2010-10-21 00:05 +0000 [r292412] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_dial.c, /: Merged revisions 292411 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct
+ 2010) | 10 lines Record priv-recordintro as sln, not gsm This
+ removes the gsm->sln step when transcoding priv-recordintro.
+ (closes issue #18176) Reported by: pabelanger Patches:
+ chan_sip.diff uploaded by pabelanger (license 224) ........
+
+2010-10-18 22:01 +0000 [r292229] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/Makefile: Fix typo in the sounds/Makefile. (Issue #17426)
+
+2010-10-18 21:54 +0000 [r292226] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 292223 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18
+ Oct 2010) | 11 lines Fix improper operator key acceptance and
+ clean up temp recording files. This is a fix for when pressing
+ the operator key after recording an unavailable, busy, name, or
+ temporary message in mailbox options. The operator key should not
+ be accepted here, but should be allowed during the message
+ recording. If the operator key is pressed during ensure the file
+ is saved or deleted as apporopriate. Also, ensure removal of
+ temporary recorded files after an early hang up or when message
+ acceptance confirmation times out. ABE-2518 ........
+
+2010-10-18 21:50 +0000 [r292224] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/Makefile, /, sounds/sounds.xml: Merged revisions 292222
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
+ | 9 lines Add support for the new English (Australian Accent)
+ sound files. (closes issue #17426) Reported by: camsown Patches:
+ core-sounds-en_AU.txt uploaded by camsown (license 1050)
+ add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
+ by: camsown, lmadsen, jtodd, qwell ........
+
+2010-10-16 10:03 +0000 [r292049] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * res/res_musiconhold.c, configs/musiconhold.conf.sample: Base
+ directory for MOH should be ASTDATADIR If the directive
+ 'directory' is relative, make it relative to the datadir, rather
+ than to the varlibdir. In the sample configuration it is relative
+ ('moh'). This has no effect unless you have actively set the
+ datadir explicitly (at build time or at run time). (closes issue
+ #16906) Patches: moh_datadir uploaded by tzafrir (license 46)
+ Review: https://reviewboard.asterisk.org/r/974/
+
+2010-10-15 19:35 +0000 [r291939] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * configs/gtalk.conf.sample, /: Merged revisions 291938 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct
+ 2010) | 2 lines Clean up formatting. ........
+
+2010-10-15 16:16 +0000 [r291904] Terry Wilson <twilson@digium.com>
+
+ * res/res_jabber.c: Don't crash or deadlock on module unload We
+ can't hold the lock while pthread_join is called since
+ aji_log_hook will attempt to lock from the other therad. We
+ reorder the pthread_join and ast_aji_disconnect so that we don't
+ do an SSL_read() while SSL_shutdown is running, causing a crash.
+
+2010-10-13 23:36 +0000 [r291655] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 291643 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13
+ Oct 2010) | 20 lines Deadlock between dahdi_exception() and
+ dahdi_indicate(). There is a deadlock between dahdi_exception()
+ and dahdi_indicate() for analog ports. The call-waiting and
+ three-way-calling feature can experience deadlock if these
+ features are trying to do something and an event from the bridged
+ channel happens at the same time. Deadlock avoidance code added
+ to obtain necessary channel locks before attemting an operation
+ with call-waiting and three-way-calling. (closes issue #16847)
+ Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ Review: https://reviewboard.asterisk.org/r/971/ ........
+
+2010-10-13 22:58 +0000 [r291580] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 291577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
+ | 21 lines Don't ignore frames that have been queued when
+ softhangup'd When an outgoing call is answered and hung up by the
+ far end *very* quickly, we may not read any frames and therefor
+ end up with a call that displays the wrong
+ disposition/DIALSTATUS. The reason is because ast_queue_hangup()
+ immediately sets the _softhangup flag on the channel and then
+ queues the HANGUP control frame, but __ast_read refuses to read
+ any frames if ast_check_hangup() indicates that a hangup request
+ has been made (which it will if _softhangup is set). So, we end
+ up losing control frames. This change makes __ast_read continue
+ to read frames even if a soft hangup has been requested. It
+ queues a hangup frame to make sure that __ast_read() will still
+ eventually return NULL. Much thanks to David Vossel for all of
+ the reviews, discussion, and help! (closes issue #16946) Reported
+ by: davidw Review: https://reviewboard.asterisk.org/r/740/
+ ........
+
+2010-10-13 15:29 +0000 [r291393] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 291392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
+ | 6 lines Lock pvt so pvt->owner can't disappear when queueing up
+ a frame. This fixes a crash due to a hangup race condition.
+ ABE-2601 ........
+
+2010-10-12 17:20 +0000 [r291280] Leif Madsen <lmadsen@digium.com>
+
+ * configs/phoneprov.conf.sample: Add undocumented variables to
+ phoneprov.conf.sample (closes issue #18107) Reported by: lathama
+ Patches: phoneprov.conf.sample.diff uploaded by lathama (license
+ 1028)
+
+2010-10-12 17:05 +0000 [r291264] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/acl.c: Merged revisions 291263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010)
+ | 2 lines Oops, incorrect range (although unallocated at ARIN)
+ ........
+
+2010-10-12 16:07 +0000 [r291229] Leif Madsen <lmadsen@digium.com>
+
+ * configs/manager.conf.sample: Add documention that mentions
+ options are defined but not used. (Issue #18101)
+
+2010-10-11 18:39 +0000 [r291073-291111] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Make exit from handle_request_do()
+ consistent.
+
+ * /, channels/chan_sip.c: Merged revisions 291109 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010)
+ | 1 line Add missing unlock to an exception condition in
+ reload_config(). ........
+
+ * main/cli.c: Fixed infinite loop in verbose/debug message output.
+ Setting the module/filename specific message level and then
+ changing it resulted in the linked list being looped on itself.
+ Traversing this linked list is an infinite loop if what you are
+ looking for is not in the list. Also plugged some CLI parsing
+ holes in the associated CLI command: * Removing a nonexistent
+ module from the list actually added it with a level of zero. *
+ Setting the non-module specific level to zero is now equivalent
+ to setting it to "off" as documented.
+
+2010-10-08 02:45 +0000 [r290863] Jeff Peeler <jpeeler@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 290862 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
+ | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
+ at control console. A recent change was made to avoid a race
+ condition on shutdown which only called the end functions from
+ the console thread. However, when pressing Ctrl-C the quit
+ handler is called from the signal handler thread. (closes issue
+ #17698) Reported by: jmls ........
+
+2010-10-07 20:57 +0000 [r290751] Jason Parker <jparker@digium.com>
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 290750 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
+ 9 lines Allow PRI to build properly when using --with-pri. Use
+ the directories found for the parent when using lib dependencies.
+ (closes issue #17314) Reported by: tzafrir Patches:
+ 17314-withdeps.diff uploaded by qwell (license 4) ........
+
+2010-10-07 10:53 +0000 [r290712] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Don't crash when Set() is called without a value.
+ Review: https://reviewboard.asterisk.org/r/949/
+
+2010-10-06 13:48 +0000 [r290396-290575] Tilghman Lesher <tlesher@digium.com>
+
+ * main/file.c: Allow streaming audio from a pipe. (closes issue
+ #18001) Reported by: jamicque Patches:
+ 20100926__issue18001.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque
+
+ * res/res_jabber.c, /: Merged revisions 290392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
+ | 8 lines Fix a crash by ensuring that we don't alter memory
+ after it's freed. (closes issue #17387) Reported by: jmls
+ Patches: 20100726__issue17387.diff.txt uploaded by tilghman
+ (license 14) Tested by: jmls ........
+
+2010-10-05 19:54 +0000 [r290375] David Vossel <dvossel@digium.com>
+
+ * apps/app_directed_pickup.c: Fixes PickupChan() not working with
+ full channel name. (closes issue #18011) Reported by: schern
+ Patches: app_directed_pickup.c.2.patch uploaded by schern
+ (license 995) app_directed_pickup.c.trunk.patch uploaded by
+ schern (license 995) Tested by: schern, dvossel
+
+2010-10-05 17:42 +0000 [r290324] Richard Mudgett <rmudgett@digium.com>
+
+ * contrib/valgrind.supp, /: Merged revisions 290323 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r290323 | rmudgett | 2010-10-05 12:41:18 -0500
+ (Tue, 05 Oct 2010) | 11 lines Merged revision 258974 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r258974
+ | diruggles | 2010-04-26 14:05:47 -0500 (Mon, 26 Apr 2010) | 4
+ lines Line 24 missed in compatibility fix in revision 233577
+ added a "fun:" prefix line 24 .......... ................
+
+2010-10-04 23:14 +0000 [r290101-290254] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
+ pbx/ael/ael-test/ref.ael-vtest17,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
+ pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5:
+ Change new pattern matcher to regard dashes the same as the old
+ pattern matcher -- as visual candy to be ignored. Also change the
+ AEL parser to not generate dashes within extensions, as those
+ dashes would be ignored. Update the AEL tests to match this
+ behavior. (closes issue #17366) Reported by: murf Patches:
+ 20100727__issue17366.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+ * /, configure, configure.ac: Merged revisions 290177 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
+ Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
+
+ * /, configure, configure.ac: Merged revisions 290100 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
+ Oct 2010) | 2 lines Automatically re-run configure test for
+ menuselect, when the relevant makeopts settings change. ........
+
+2010-10-02 08:52 +0000 [r289950] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 289949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
+ lines Add documentation for undocumented option to AMI action
+ originate ........
+
+2010-10-02 04:45 +0000 [r289874] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 289873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01
+ Oct 2010) | 8 lines When forwarding a message, a prepend means
+ that the filesystem will always have a better copy. (closes issue
+ #17803) Reported by: dpetersen Patches:
+ 20100923__issue17803.diff.txt uploaded by tilghman (license 14)
+ Tested by: dpetersen ........
+
+2010-10-01 23:01 +0000 [r289798] Jeff Peeler <jpeeler@digium.com>
+
+ * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
+ Merged revisions 289797 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
+ | 15 lines Change RFC2833 DTMF event duration on end to report
+ actual elapsed time. The scenario here is with a non P2P early
+ media session. The reported time length of DTMF presses are
+ coming up short when sending to the remote side. Currently the
+ event duration is a running total that is incremented when
+ sending continuation packets. These continuation packets are only
+ triggered upon incoming media from the remote side, which means
+ that the running total probably is not going to end up matching
+ the actual length of time Asterisk received DTMF. This patch
+ changes the end event duration to be lengthened if it is detected
+ that the end event is going to come up short. Review:
+ https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
+
+2010-10-01 17:09 +0000 [r289704] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
+ 289703 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
+ 2010) | 6 lines Disable debugging by default and reformat .config
+ file. Review: https://reviewboard.asterisk.org/r/929/ ........
+
+2010-10-01 16:21 +0000 [r289700] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
+ | 14 lines Ensure user portion of SIP URI matches dialplan when
+ using encoded characters. This commit takes a simliar approach to
+ 288112 and checks the dialplan to determine the proper action for
+ an incoming contact header as to whether or not it should be
+ decoded or not. sip_new was blindly always decoding the
+ extension, which also caused the outgoing contact header to be
+ incorrect as well as failing to match the encoded extension in
+ the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
+ bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
+ wdoekes ........
+
+2010-10-01 09:42 +0000 [r289622] schmitds <schmitds@localhost>:
+
+ * channels/chan_sip.c: don't iterate through all dialogs to find
+ and delete old subscribes On every incoming subscribe there is a
+ iteration through all dialogs to find old subscribes and delete
+ them. This is slow and not RFC conform. This was only needed in
+ 1.2 cause a subscribe was not deleted when a dialog was
+ destroyed, after 1.4 a subscribe get removed when its dialog is
+ destroyed. (closes issue #17950) Reported by: schmidts Tested by:
+ schmidts Review: https://reviewboard.asterisk.org/r/901/
+
+2010-09-30 19:51 +0000 [r289553] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Properly handle channel allocation failures
+ duing invites with replaces. ABE-2588
+
+2010-09-30 17:09 +0000 [r289501] Brett Bryant <bbryant@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 289500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289500 | bbryant | 2010-09-30 13:08:20 -0400 (Thu, 30 Sep 2010)
+ | 11 lines res_agi.c:handle_getvariablefull() could recursively
+ lock a channel and not release it if an argument is the current
+ channel's name. (closes issue #17970) Reported by: mdu113
+ Patches: res_agi.c.diff3 uploaded by mdu113 (license 582) Tested
+ by: mdu113 Review: https://reviewboard.asterisk.org/r/947/
+ ........
+
+2010-09-30 15:37 +0000 [r289425] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_sms.c: Merged revisions 289424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
+ | 8 lines Fix a crash in app_sms. Since the data being passed to
+ the generator callback is on the stack of the SMS() application,
+ we must ensure that the generator is stopped before the
+ application exits. ABE-2587 ........
+
+2010-09-29 21:03 +0000 [r289339] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 289338 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
+ 8 lines Allow a manager originate to succeed on forwarded
+ devices. The timeout to wait for an answer was being set to 0
+ when a device forwarded to another extension. We don't always
+ need the timeout set like this, so make it an optional parameter,
+ and don't use it in this case. ABE-2544 ........
+
+2010-09-29 20:24 +0000 [r289334] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample: Update sample documentation to note
+ md5secret requirements.
+
+2010-09-29 20:15 +0000 [r289332] Russell Bryant <russell@digium.com>
+
+ * res/res_config_ldap.c: Don't completely ignore md5secret from
+ LDAP if the value does not begin with {md5}. This fixes a problem
+ that lmadsen ran in to where md5secret was not working for him.
+
+2010-09-29 15:04 +0000 [r289178] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 289177 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
+ 2010) | 8 lines Set the caller id on CDRs when it is set on the
+ parent channel. (closes issue #17569) Reported by: tbelder
+ Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
+ tbelder ........
+
+2010-09-28 18:14 +0000 [r289095] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 289094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010)
+ | 14 lines Fixes an issue with the Newchannel AMI event during
+ the Masquerading process. Fixes an issue with the Newchannel AMI
+ event during the Masquerading process, where no Newchannel AMI
+ event was generated for the psuedo channel used during the
+ masquerading process. (closes issue #17987) Reported by:
+ RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish
+ (license 1122) Tested by: RadicAlish Review:
+ https://reviewboard.asterisk.org/r/937/ ........
+
+2010-09-24 15:37 +0000 [r288747] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 288746 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24
+ Sep 2010) | 5 lines Don't fail a masquerade if it is already
+ being hung up This avoids noise on some Local channel situations
+ where we don't use /n. Thanks to Alec Davis for the suggestion.
+ ........
+
+2010-09-24 13:53 +0000 [r288637-288712] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Solaris won't printf a NULL. (closes issue
+ #18041) Reported by: asgaroth
+
+ * cdr/cdr_pgsql.c, /, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac,
+ include/asterisk/channel.h: Merged revisions 288636 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23
+ Sep 2010) | 2 lines Solaris compatibility fixes ........
+
+2010-09-22 23:10 +0000 [r288500] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 288499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22
+ Sep 2010) | 8 lines Don't let a Local channel get bridged to
+ itself If a local channel gets bridged to itself, it becomes
+ orphaned with no devices left to actually tell it to hang up.
+ This patch modifies local_fixup() to detect this case and deny
+ it. Review: https://reviewboard.asterisk.org/r/934 ........
+
+2010-09-22 17:49 +0000 [r288344-288417] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
+ | 5 lines RFC3261 section 12.2 explicitly says out of order
+ requests are responded with a 500 Server Internal Error response.
+ ABE-2458 ........
+
+ * /, channels/chan_sip.c: Merged revisions 288343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010)
+ | 2 lines During check_pendings, if the dialog is terminated with
+ a CANCEL, change the invitestate to INV_CANCEL like in
+ sip_hangup. ........
+
+2010-09-22 16:44 +0000 [r288340] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 288339 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
+ | 11 lines Fix a 100% CPU consumption problem when setting
+ console=yes in asterisk.conf. The handling of -c and console=yes
+ should be the same, but they were not. When you specify -c, it
+ sets both a flag for console module and for asterisk not to
+ fork() off into the background. The handling of console=yes only
+ set console mode, so you would end up with a background process()
+ trying to run the Asterisk console and freaking out since it
+ didn't have anything to read input from. Thanks to beagles for
+ reporting and helping debug the problem! ........
+
+2010-09-22 15:11 +0000 [r288267] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /, UPGRADE.txt:
+ Merged revisions 288265-288266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
+ | 9 lines Allow the encoding to be set, in case local charset
+ does not agree with database. (closes issue #16940) Reported by:
+ jamicque Patches: 20100827__issue16940.diff.txt uploaded by
+ tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+ r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
+ | 5 lines Document addition of encoding parameter. (issue #16940)
+ Reported by: jamicque ........
+
+2010-09-22 00:03 +0000 [r288193] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288192 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21
+ Sep 2010) | 26 lines In chan_iax2.c:schedule_delivery() calls
+ ast_bridged_channel() on an unlocked channel. Near the beginning
+ of schedule_delivery(), ast_bridged_channel() is called on
+ iaxs[fr->callno]->owner. However, the channel is not locked,
+ which can result in ast_bridged_channel() crashing should
+ owner->tech change to a technology that doesn't implement
+ bridged_channel. I also fixed the other calls to
+ ast_bridged_channel() in chan_iax2.c since the owner lock was not
+ held there either. Converted the existing channel deadlock
+ avoidance to use iax2_lock_owner(). Using the new function
+ simplified some awkward code. In the process of fixing the
+ locking on ast_bridged_channel(), I also found a memory leak in
+ socket_process() for v1.6.2 and v1.8. The local struct variable
+ ies.vars is not freed on early/abnormal function exits. (closes
+ issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
+ uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/926/ ........
+
+2010-09-21 22:22 +0000 [r288147] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_iax2.c: Setup timer before set_config(). (closes
+ issue #18019) Reported by: Netview Patches: issue_0018019.patch
+ uploaded by pabelanger (license 224) Tested by: Netview
+
+2010-09-21 21:59 +0000 [r288113] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
+ | 15 lines Try both the encoded and unencoded subscription URI
+ for a match in hints. When a phone sends an encoded URI for a
+ subscription, the URI is not matched with the actual hint that is
+ in decoded format. For example, if we have an extension with a
+ hint that is named: "#5601" or "*5601", the subscription will
+ work fine if the phone subscribes with an already decoded URI,
+ but when it's decoded like "%255601" or "%2A5601", Asterisk is
+ unable to match it with the correct hint. (closes issue #17785)
+ Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
+ uploaded by tilghman (license 14) Tested by: ramonpeek ........
+
+2010-09-21 19:46 +0000 [r288006] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 288005 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
+ | 8 lines Add a check to fix a rare segmentation fault you'd get
+ if ast_frdup couldn't allocate memory on the first frame being
+ queued in ast_queue_frame. (closes issue #17882) Reported by:
+ seanbright Tested by: seanbright ........
+
+2010-09-21 19:07 +0000 [r287934] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 287933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010)
+ | 2 lines Less than zero is an error, not any non-zero value.
+ ........
+
+2010-09-20 23:58 +0000 [r287759] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 287758 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
+ | 16 lines Fix misvalidation of meetme pins in conjunction with
+ the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
+ user and admin pin setup for your conference, using the user pin
+ would gain you admin priviledges. Also, when no user pin was set,
+ an admin pin was, the 'a' MeetMe flag wasn't used, and the user
+ tried to enter a conference then they were still prompted for a
+ pin and forced to hit #. (closes issue #17908) Reported by: kuj
+ Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
+ kuj Review: [full review board URL with trailing slash] ........
+
+2010-09-20 23:16 +0000 [r287685] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/channel.c: ast_channel_masquerade: Avoid recursive
+ masquerades. Check all 4 combinations of (original/clonechan) *
+ (masq/masqr). Initially original->masq and clonechan->masqr were
+ only checked. It's possible with multiple masq's planned - and
+ not yet executed, that the 'original' chan could already have
+ another masq'd into it - thus original->masqr would be set, that
+ masqr would lost. Likewise for the clonechan->masq. (closes issue
+ #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
+ based on bug16057.diff4.txt uploaded by alecdavis (license 585)
+ Tested by: ramonpeek, davidw, alecdavis
+
+2010-09-20 21:28 +0000 [r287642] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Don't crash when parking a non-bridged
+ call. (closes issue #17680) Reported by: jmhunter Patches:
+ chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by:
+ jmhunter, DEA
+
+2010-11-02 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.14 Released.
+
+2010-09-20 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.14-rc1 Released.
+
+2010-09-20 15:56 +0000 [r287556-287558] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Use ast_str when processing hint state changes
+ Merged revisions 287555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+ 2010) | 5 lines Use ast_dynamic_str when processing hint state
+ changes (related to issue #17928) Reported by: mdu113 ........
+
+ * /: Revert r287556.
+
+ * /: Use ast_str when processing hint state changes Merged
+ revisions 287555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+ 2010) | 5 lines Use ast_dynamic_str when processing hint state
+ changes (related to issue #17928) Reported by: mdu113 ........
+
+2010-09-19 16:06 +0000 [r287470] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 287469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
+ lines Make sure we always free variables properly in manager
+ originate. (closes issue #17891) reported, solved and tested by
+ oej Review: https://reviewboard.asterisk.org/r/869/ ........
+
+2010-09-17 21:08 +0000 [r287387] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 287386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
+ | 7 lines Blank columns should get set on reload, not ignored.
+ (closes issue #16893) Reported by: haakon Patches:
+ 20100818__issue16893.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2010-09-17 13:36 +0000 [r287308] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287307 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
+ 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
+ processing in ast_hint_state_changed(). (related to issue #17928)
+ Reported by: mdu113 ........
+
+2010-09-16 22:12 +0000 [r287198] Jason Parker <jparker@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 287197 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287197 | qwell | 2010-09-16 17:12:30 -0500 (Thu, 16 Sep 2010) |
+ 7 lines Add LSB headers for Debian init script, since Debian will
+ complain if it isn't there. Headers were taken from trunk.
+ (closes issue #17958) Reported by: javyer ........
+
+2010-09-16 20:06 +0000 [r287115-287119] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287118 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't limit hint processing in
+ ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
+ (closes issue #17928) Reported by: mdu113 Patches:
+ 20100831__issue17928.diff.txt uploaded by tilghman (license 14)
+ Tested by: mdu113 ........
+
+ * main/cdr.c, /: Merged revisions 287114 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't stop printing cdr variables if we encounter
+ one with a blank name or value. (closes issue #17900) Reported
+ by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson ........
+
+2010-09-15 20:28 +0000 [r286998] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 286941 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15
+ Sep 2010) | 7 lines Ensure mailbox is not filled to capacity
+ before doing message forwarding. Specifically, before prompting
+ to record a prepended message the capacity is checked first. If
+ the mailbox is full the extension will be reprompted. ABE-2517
+ ........
+
+2010-09-14 19:27 +0000 [r286681-286757] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286756 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
+ 2010) | 13 lines Don't clear the username from a realtime
+ database when a registration expires. Non-realtime chan_sip does
+ not clear the username from memory when a registration expiries
+ so realtime probably shouldn't either. (closes issue #17551)
+ Reported by: ricardolandim Patches:
+ reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
+ 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
+ (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
+ mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: ricardolandim,
+ mnicholson ........
+
+ * main/channel.c, /: Merged revisions 286679 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
+ 2010) | 7 lines Only drop duplicate answer frames if the channel
+ is bridged. Back in r3710 ast_read() was modified to drop answer
+ frames on channels that were in the UP state. This modification
+ prevented bridges that were up before the answer from being
+ broken and reestablished by an ANSWER control frame. That change
+ also prevents pickup of channels called from the ast_dial
+ framework from working properly. The ast_dial framework expects
+ to see an ANSWER frame after dialing and the pickup code queues
+ one but ast_read() drops it. This new change only drops ANSWER
+ frames when the channel is bridged, allowing the answer queued by
+ the pickup code to properly pass through ast_read() on to the
+ ast_dial framework. ABE-2473 (related to issue #2342) ........
+
+2010-09-14 05:06 +0000 [r286527-286587] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/realtime/mysql/voicemail_messages.sql (added),
+ contrib/realtime/mysql/voicemail_data.sql (added): Add
+ documentation on missing backend tables for Voicemail
+
+ * main/features.c: C precedence got me
+
+ * main/features.c: Refactor conversion to ast_poll() to fix
+ callparking regression.
+
+2010-09-13 19:38 +0000 [r286456] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Remove "Internal IP" from sip show settings,
+ as it's not at all useful to display. (closes issue #17840)
+ Reported by: oej
+
+2010-09-11 17:05 +0000 [r286268] Olle Johansson <oej@edvina.net>
+
+ * /, main/file.c: Merged revisions 286267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
+ lines Handle error response when we can't make file compatible
+ Review: https://reviewboard.asterisk.org/r/911/ ........
+
+2010-09-10 22:56 +0000 [r286223] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 286222 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10
+ Sep 2010) | 1 line Return -1 if chan_local doesn't support an
+ option ........
+
+2010-09-10 20:55 +0000 [r286117] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 286114 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri,
+ 10 Sep 2010) | 4 lines Load iax.conf before registering any
+ functions/applications/actions. Review:
+ https://reviewboard.asterisk.org/r/914/ ........
+
+2010-09-10 20:42 +0000 [r286116] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 286113 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10
+ Sep 2010) | 11 lines An outgoing call may not get hung up if a
+ pre-connect incoming ISDN call is disconnected. If the ISDN link
+ a pre-connect incoming call is using fails or is reset, the
+ outgoing leg may not hang up or be delayed in hanging up.
+ (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
+ PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
+ PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
+ incoming call leg hangs up before connecting for any reason. It
+ makes no sense to send a BUSY or CONGESTION control frame to the
+ outgoing call leg under these circumstances. ........
+
+2010-09-10 20:35 +0000 [r286115] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/pbx.h, include/asterisk/frame.h,
+ channels/chan_local.c, /, funcs/func_channel.c,
+ include/asterisk/channel.h: Merged revisions 286059 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10
+ Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a
+ Local channel Having Local (/n) channels as queue members and
+ setting the language in the extension with
+ Set(CHANNEL(language)=fr) sets the language on the Local/...,2
+ channel. Hold time report playbacks happen on the Local/...,1
+ channel and therefor do not play in the specified language. This
+ patch modifies func_channel_write to call the setoption callback
+ and pass the CHANNEL() write info to the callback. chan_local
+ uses this information to look up the other side of the channel
+ and apply the same changes to it. (closes issue #17673) Reported
+ by: Guggemand Review: https://reviewboard.asterisk.org/r/903/
+ ........
+
+2010-09-10 18:30 +0000 [r285930-286024] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_heap.c, /, main/test.c: Merged revisions 286023 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010)
+ | 2 lines Missing newline ........
+
+ * include/asterisk/select.h: Another fix for Mac OS X. While trying
+ to fix this the "right" way, I wandered into dependency hell. Two
+ hours later, I backed out, and just removed the offending code.
+ ast_inline_api only goes one level deep and then it breaks. Ouch.
+
+ * tests/test_poll.c, include/asterisk/select.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 285889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
+ | 7 lines Fix Mac OS X build. This also fixes a rather grievous
+ calculation error for the offset of ast_fdset, which was masked
+ on Linux and FreeBSD, because these platforms check the first 256
+ FDs regardless of the bitmask setting (due to backwards
+ compatibility). ........
+
+2010-09-09 22:49 +0000 [r285818] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, codecs/gsm/Makefile: Merged revisions 285817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
+ 2010) | 8 lines GCC 4.2.x optimizations result in improper
+ behavior of GSM codec (closes issue #17688) Reported by:
+ pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
+ pprindeville (license 347) Tested by: mkeuter, pprindeville
+ ........
+
+2010-09-09 20:09 +0000 [r285744] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /: Merged revisions 285742 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
+ 9 lines Transmit silence when reading DTMF in ast_readstring.
+ Otherwise, you could get issues with DTMF timeouts causing
+ hangups. (closes issue #17370) Reported by: makoto Patches:
+ channel-readstring-silence-generator.patch uploaded by makoto
+ (license 38) ........
+
+2010-09-09 18:50 +0000 [r285639-285710] Brett Bryant <bbryant@digium.com>
+
+ * main/pbx.c: Fixes an issue with dialplan pattern matching where
+ the specificity for pattern ranges and pattern special characters
+ was inconsistent. (closes issue #16903) Reported by: Nick_Lewis
+ Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
+ 657) Tested by: Nick_Lewis
+
+ * res/res_musiconhold.c, /: Merged revisions 285638 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09
+ Sep 2010) | 7 lines Fixes an issue with MOH where it doesn't
+ recover cleanly when it can't play a file and would just stop,
+ instead of continuing to find the next playable file in the MOH
+ class. (closes issue #17807) Reported by: kshumard Review:
+ https://reviewboard.asterisk.org/r/910/ ........
+
+2010-09-08 22:11 +0000 [r285563-285567] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 285566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010)
+ | 2 lines In retrans_pkt, do not unlock pvt until the end of the
+ function on a transmit failure. ........
+
+ * channels/chan_sip.c: Fixes interoperability problems with session
+ timer behavior in Asterisk. CHANGES: 1. Never put "timer" in
+ "Require" header. This is not to our benefit and RFC 4028 section
+ 7.1 even warns against it. It is possible for one endpoint to
+ perform session-timer refreshes while the other endpoint does not
+ support them. If in this case the end point performing the
+ refreshing puts "timer" in the Require field during a refresh,
+ the dialog will likely get terminated by the other end. 2. Change
+ the behavior of 'session-timer=accept' in sip.conf (which is the
+ default behavior of Asterisk with no session timer configuration
+ specified) to only run session-timers as result of an incoming
+ INVITE request if the INVITE contains an "Session-Expires"
+ header... Asterisk is currently treating having the "timer"
+ option in the "Supported" header as a request for session timers
+ by the UAC. I do not agree with this. Session timers should only
+ be negotiated in "accept" mode when the incoming INVITE supplies
+ a "Session-Expires" header, otherwise RFC 4028 says we should
+ treat a request containing no "Session-Expires" header as a
+ session with no expiration. Below I have outlined some situations
+ and what Asterisk's behavior is. The table reflects the behavior
+ changes implemented by this patch. SITUATIONS: -Asterisk as UAS
+ 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
+ "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
+ "Session-Expires". 200 Ok Response HAS "Session-Expires" header
+ 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
+ "Session-Expires" header 5. Outgoing INVITE: HAS
+ "Session-Expires". Active - Asterisk will have an active refresh
+ timer regardless if the other endpoint does. Inactive - Asterisk
+ does not have an active refresh timer regardless if the other
+ endpoint does. XXXXXXX - Not possible for mode.
+ ______________________________________ |SITUATIONS |
+ 'session-timer' MODES | |___________|________________________| |
+ | originate | accept | |-----------|------------|-----------| |1.
+ | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
+ Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
+ -------------------------------------- (closes issue #17005)
+ Reported by: alexrecarey
+
+2010-09-08 20:56 +0000 [r285532] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_meetme.c: Fixes a bug with MeetMe where after announcing
+ the amount of time left in a conference, if music on hold was
+ playing, it doesn't restart. (closes issue #17408) Reported by:
+ sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
+ sysreq (license 1009) Tested by: sysreq
+
+2010-09-08 20:42 +0000 [r285526-285529] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c, include/asterisk/astobj2.h: Follow coding
+ guidelines in moh rescan fix. Also fix the documentation that got
+ me in trouble.
+
+ * res/res_musiconhold.c: Fixes issue where moh files were no longer
+ rescanned during a reload. (closes issue #16744) Reported by: pj
+ Patches: 16744-reload.diff uploaded by qwell (license 4) Tested
+ by: qwell
+
+2010-09-07 20:31 +0000 [r285267-285366] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 285365 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
+ | 9 lines Catch invalid extensions at the parser, instead of
+ making the core deal with them. (closes issue #17794) Reported
+ by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
+ by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: PavelL ........
+
+ * main/poll.c, /: Merged revisions 285266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
+ | 4 lines Use poll, if indicated to do so, in the ast_poll2
+ implementation. This fixes the unit tests on FreeBSD 8.0.
+ ........
+
+2010-09-07 17:49 +0000 [r285196] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 285194 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07
+ Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes
+ with passwords that don't precede a comma would throw unnecessary
+ error messages. (closes issue #15726) Reported by: 298 Patches:
+ M15726.diff uploaded by junky (license 177) Tested by: junky
+ Review: [full review board URL with trailing slash] ........
+
+2010-09-06 06:55 +0000 [r285089] Tilghman Lesher <tlesher@digium.com>
+
+ * makeopts.in, /, BSDmakefile (added): Merged revisions 285088 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010)
+ | 2 lines Silly convenience script for BSD platforms. ........
+
+2010-09-03 18:15 +0000 [r284958] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_iax2.c: This is a patch provided for issue #17935
+ to add the ActionID to the IAXregistry AMI response. (closes
+ issue #17935) Reported by: alexkuklin Patches: iaxshowreg
+ uploaded by alexkuklin (license 1115) Tested by: alexkuklin
+
+2010-09-03 16:20 +0000 [r284897] Terry Wilson <twilson@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 284881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
+ | 5 lines Properly detect when a sound file doesn't exist
+ ast_fileexists returns -1 for error and 0 for a non-existant
+ file. The existing code treated missing files as though they
+ existed. ........
+
+2010-09-02 20:54 +0000 [r284778] Brett Bryant <bbryant@digium.com>
+
+ * main/manager.c, /: Merged revisions 284777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010)
+ | 7 lines Fixes a bug in manager.c where the default
+ configuration values weren't reset when the manager configuration
+ was reloaded. (closes issue #17917) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/883/ ........
+
+2010-09-02 16:48 +0000 [r284704] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284703 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
+ | 7 lines Removed relatedpeer code from sip_autodestruct Handling
+ of the relatedpeer structure associated with a sip_pvt should be
+ done during the final sip_destruction function, not in
+ sip_autodestruct. ........
+
+2010-09-02 16:07 +0000 [r284399-284665] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_usbradio.c: Fixing build.
+
+ * apps/app_queue.c: Don't reset queue stats on a module reload.
+ (closes issue #17535) Reported by: raarts Patches:
+ 20100819__issue17535.diff.txt uploaded by tilghman (license 14)
+
+ * configure, include/asterisk/autoconfig.h.in: Failed to rerun
+ bootstrap.sh after last commit
+
+ * res/res_jabber.c, main/rtp.c, main/poll.c,
+ include/asterisk/select.h (added), channels/chan_usbradio.c,
+ channels/chan_phone.c, channels/chan_misdn.c, main/features.c,
+ include/asterisk/poll-compat.h, tests/test_poll.c (added),
+ main/asterisk.c, utils/clicompat.c, res/res_ais.c, /,
+ configure.ac, channels/console_video.c,
+ include/asterisk/channel.h: Merged revisions 284478 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01
+ Sep 2010) | 11 lines Ensure that all areas that previously used
+ select(2) now use poll(2), with implementations that need poll(2)
+ implemented with select(2) safe against 1024-bit overflows. This
+ is a followup to the fix for the pthread timer in 1.6.2 and
+ beyond, fixing a potential crash bug in all supported releases.
+ (closes issue #17678) Reported by: russell Branch:
+ https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
+ Review: https://reviewboard.asterisk.org/r/824/ ........
+
+ * res/res_config_pgsql.c: Don't warn on floats and timestamps
+ (closes issue #17082) Reported by: coolmig
+
+ * /, channels/chan_sip.c: Merged revisions 284393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
+ | 7 lines Don't send a devstate change on poke_noanswer if the
+ state did not change. (closes issue #17741) Reported by: schmidts
+ Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
+ ........
+
+2010-08-31 18:59 +0000 [r284317] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 284316 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31
+ Aug 2010) | 7 lines Update say.conf.sample to match the rules in
+ say.c (closes issue #17835) Reported by: RoadKill Patches:
+ say.conf.sample.patch.rules uploaded by RoadKill (license 933)
+ Tested by: RoadKill ........
+
+2010-08-30 22:27 +0000 [r284280] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_festival.c: Fix 3 coding errors: 1) After we close FD,
+ we should not be trying to write to it. 2) Call _exit(0), not
+ exit(0), to avoid running shutdown routines in a child. 3) Use
+ endian, not processor, detection to ensure bytes are written in
+ the correct order. (closes issue #15706) Reported by: modelnine
+ Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by
+ modelnine (license 865) Tested by: gmartinez
+
+2010-08-27 22:27 +0000 [r284002] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
+ | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
+ (closes issue #17758) Reported by: ibc Patches:
+ multiple_accept_headers_1.4.diff uploaded by dvossel (license
+ 671) ........
+
+2010-08-27 20:30 +0000 [r283881] Jason Parker <jparker@digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+ revisions 283880 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
+ 8 lines Fix issue with decoding ^-escaped characters in realtime.
+ (closes issue #17790) Reported by: denzs Patches:
+ 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
+ denzs ........
+
+2010-08-26 15:24 +0000 [r283381-283691] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
+ | 19 lines Fixed how Asterisk destroys a dialog on channel hangup
+ before invite receives a response. If an ast_channel with a SIP
+ tech pvt hangs up before the sip dialog gets a response to its
+ outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
+ not rfc compliant and results in confusion at the other endpoint.
+ sip_pretend_ack will ack and remove all the packets in the
+ retransmit queue. This means that the INVITE will stop
+ retransmitting, and that any response to that INVITE that comes
+ after the pretend_ack occurs will be ignored. Instead of faking
+ any sort of acknowledgement for an outgoing INVITE during an
+ internal hangup, we should let the protocol stack process the
+ INVITE transaction and terminate the dialog properly. This is
+ achieved by setting the PENDING_BYE flag. When this flag is used,
+ once the dialog proceeds to an escapable state the transaction
+ will either be canceled with a SIP_CANCEL or completed followed
+ immediately by a BYE. Attempting to do this any other way is
+ incorrect. If the endpoint is not responding to the INVITE
+ request, the INVITE must continue to be retransmitted until it
+ times out which will result in the dialog being destroyed.
+ ........
+
+ * channels/chan_sip.c: Add to and from tags to NOTIFY dialog-info
+ xml body so pickup can occur. When pedantic mode is used, the
+ dialog-info xml generated during a ringing event must contain the
+ to and from tag values. Otherwise if a pickup occurs using INVITE
+ with replaces, Astrisk will not be able to locate the
+ subscription.
+
+ * channels/chan_sip.c: Asterisk will not advertise session timers
+ are supported when 'session-timers=refuse' is used. Asterisk now
+ dynamically builds the "Supported" header depending on what is
+ enabled/disabled in sip.conf. Session timers used to always be
+ advertised as being supported even when they were disabled in the
+ configuration. This caused problems with some end points. (issue
+ #17005)
+
+ * /, channels/chan_sip.c: Merged revisions 283380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
+ | 11 lines This fix makes sure the ast_channel hangs up correctly
+ when the dialog's PENDING_BYE flag is set. When the pending bye
+ flag is used, it is possible that the dialog will terminate and
+ leave the sip_pvt->owner channel up. This is because we never
+ hangup the ast_channel after sending the SIP_BYE request. When we
+ receive the response for the SIP_BYE we set need_destroy which we
+ would expect to destroy the dialog on the next do_monitor loop,
+ but this is not the case. The dialog will only be destroyed once
+ the owner is hungup even with the need_destroy flag set. This
+ patch sets the softhangup flag on the ast_channel when a SIP_BYE
+ request is sent as a result of the pending bye flag. ........
+
+2010-08-23 21:32 +0000 [r283318] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_odbc.c, cdr/cdr_adaptive_odbc.c: CDR drivers depend upon
+ res_odbc, not directly on the ODBC libraries
+
+2010-09-15 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.13 released.
+
+ * Incorrect .version and ChangeLog files updated. Re-release
+ of Asterisk 1.6.2.12 with corrections and version
+ number bump.
+
+2010-09-15 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.12 released.
+
+2010-08-23 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.12-rc1 Released.
+
+2010-08-20 16:48 +0000 [r283049-283124] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 283123 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
+ (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
+ | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
+ line Reference correct struct member for unlikely event
+ PRI_EVENT_CONFIG_ERR. .......... ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 283048 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20
+ Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending
+ ALERTING is a protocol error The PRI layer in chan_dadhi will
+ check if a PROGRESS message has already been sent, and not allow
+ sending another (although that is technically allowed by the Q931
+ spec), however it does not protect against sending an ALERTING
+ and then sending a PROGRESS message, which is a violation of the
+ specification. Most switches don't seem to care too deeply about
+ this, but some do, and will disconnect the call when receiving
+ this invalid sequence. Protocol specification reference:
+ T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
+ protocol control (network side) point-point (sheet 3 of 8)"
+ (closes issue #17874) Reported by: nic_bellamy Patches:
+ asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
+ nic bellamy (license 299)
+ asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299)
+ asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299) ........
+
+2010-08-19 21:05 +0000 [r282890-282894] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
+ | 11 lines tos_sip option was not being set correctly When
+ tos_sip is used, the tos of the sip socket is only set correctly
+ if the socket binding changes on a reload. If the binding stays
+ the same but the TOS changes, the new tos value would not take
+ into effect. This patch fixes that. (closes issue #17712)
+ Reported by: nickb ........
+
+ * channels/chan_sip.c: fixes sip peer memory leaks in the
+ peer_by_ip table (issue #17798)
+
+2010-08-19 19:44 +0000 [r282859] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Merged revisions 277944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
+ 2010) | 16 lines Regression with T.38 negotiation Prior to
+ 1.4.26.3 T.38 negotiation worked properly, in the case of the
+ reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+ Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+ by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+ samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
+
+2010-08-19 02:14 +0000 [r282730] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 282729 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
+ Aug 2010) | 2 lines Add some documentation about codec
+ negotiation to sip.conf ........
+
+2010-08-18 14:28 +0000 [r282668] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes crash with notifycid (closes issue
+ #17868) Reported by: francesco_r Patches: issue_17868.diff
+ uploaded by dvossel (license 671) Tested by: francesco_r
+
+2010-08-18 07:43 +0000 [r282607] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Don't warn on callerid when completely
+ text, instead of numeric with localdialplan prefixes. (closes
+ issue #16770) Reported by: jamicque Patches:
+ 20100413__issue16770.diff.txt uploaded by tilghman (license 14)
+ 20100811__issue16770.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque
+
+2010-08-17 21:35 +0000 [r282576] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes no default transport for temp peer
+ creation in chan_sip (closes issue #17829) Reported by: falves11
+ Patches: issue_17829.rev1.txt uploaded by russell (license 2)
+ issue_17829.diff uploaded by dvossel (license 671) Tested by:
+ falves11
+
+2010-08-16 18:00 +0000 [r282469] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/asterisk.tex, doc/tex/sounds.tex (added): Add information
+ about creating sounds files using the sounds tools publically
+ available so that others can create their own sounds prompts
+ using the same tools we use to generate sounds releases. This
+ allows people creating their own prompts to sound consistent with
+ the prompts available from the open source project. SWP-595
+
+2010-08-16 17:32 +0000 [r282467] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 282430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
+ | 16 lines Send a SRCCHANGE indication when we masquerade
+ Masquerading a channel means that the src of the audio is
+ potentially changing, so send a SRCCHANGE so that RTP-based media
+ streams can get a new SSRC generated to reflect the change.
+ Original patch by addix (along with lots of testing--thanks!).
+ (closes issue #17007) Reported by: addix Patches:
+ 1001-reset-SSRC-original-channel.diff uploaded by addix (license
+ 1006) srcchange.diff uploaded by twilson (license 396) Tested by:
+ addix, twilson Review: https://reviewboard.asterisk.org/r/862/
+ ........
+
+2010-08-13 18:54 +0000 [r282235] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: only do magic pickup when notifycid is
+ enabled A new way of doing BLF pickup was introduced into 1.6.2.
+ This feature adds a call-id value into the XML of a SIP_NOTIFY
+ message sent to alert a subscriber that a device is ringing. This
+ option should only be enabled when the new 'notifycid' option is
+ set... but this was not the case. Instead the call-id value was
+ included for every RINGING Notify message, which caused a
+ regression for people who used other methods for call pickup.
+ (closes issue #17633) Reported by: urosh Patches: chan_sip.txt
+ uploaded by urosh (license ) blf_cid_issue.diff uploaded by
+ dvossel (license 671) Tested by: dvossel, urosh, okrief,
+ alecdavis
+
+2010-08-12 22:50 +0000 [r282130] Jason Parker <jparker@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 282129 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) |
+ 1 line Register CLI commands before parsing config, in case there
+ is a config error. ........
+
+2010-08-12 03:01 +0000 [r281912] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 281911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
+ | 20 lines Ensure SSRC is changed when media source is changed to
+ resolve audio delay. This change causes the SSRC to change right
+ before the channels are bridged, which is what used to happen. It
+ seems that fixes were made to attempt limiting SSRC changes,
+ targeted mainly at sending DTMF. DTMF is not affecting the SSRC
+ with this change. There are two other control frames sent in
+ ast_channel_bridge that probably should also be changed to
+ AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
+ up to the discretion of resolving issue #17007. For reference -
+ old review implementing new control frame SRCCHANGE:
+ https://reviewboard.asterisk.org/r/540 (closes issue #17404)
+ Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
+ (license 325) Tested by: sdolloff ........
+
+2010-08-11 21:09 +0000 [r281763-281873] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 281819 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11
+ Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes
+ issue #17836) Reported by: RoadKill Patches:
+ say.conf.sample.patch.dk uploaded by RoadKill (license 933)
+ ........
+
+ * configs/say.conf.sample, /: Merged revisions 281762 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11
+ Aug 2010) | 6 lines Allow say.conf to handle large numbers ending
+ with multiple zeros. (closes issue #17833) Reported by: RoadKill
+ Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
+ (license 933) ........
+
+2010-08-11 15:17 +0000 [r281722] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_readexten.c: Only set status TIMEOUT, if we have no
+ digits. (closes issue #15188) Reported by: jcovert Patches:
+ app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
+ 551)
+
+2010-08-10 18:04 +0000 [r281567-281574] Russell Bryant <russell@digium.com>
+
+ * main/sched.c: Don't move the time threshold for running scheduled
+ events on every iteration. Instead, only calculate the time
+ threshold each time ast_sched_runq() is called. (closes issue
+ #17742) Reported by: schmidts Patches: sched.c.patch uploaded by
+ schmidts (license 1077)
+
+ * apps/app_dial.c, /: Merged revisions 281566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Reset visible indication after answer. (closes issue
+ #17641) Reported by: klaus3000 Patches:
+ ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+ klaus3000 (license 65) Tested by: schmidts ........
+
+2010-08-09 20:46 +0000 [r281430] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes SIP peers memory leak We zeroed out
+ the peer's addr before it was removed from the peers_by_ip
+ container. This made it impossible to be removed from the
+ container as the addr is the key used by the container to find
+ the peer. (closes issue #17774) Reported by: kkm Patches:
+ 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
+ 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
+
+2010-08-09 20:07 +0000 [r281391] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 281390 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09
+ Aug 2010) | 13 lines Prevent loss of Caller ID information set on
+ local channel after masquerade. Caller ID set on the channel
+ before a masquerade occurs when using a local channel would cause
+ the information to be lost. The problem was that the information
+ was set on a channel destined to be hung up. The somewhat
+ confusing fix is to detect if any Caller ID has been set on the
+ channel and if so preswap the Caller ID data so that basically
+ the masquerade puts the data back. (closes issue #17138) Reported
+ by: kobaz Review: https://reviewboard.asterisk.org/r/847/
+ ........
+
+2010-08-05 13:11 +0000 [r281051] Russell Bryant <russell@digium.com>
+
+ * main/cdr.c: Cleanup default option value handling for cdr.conf
+ [general]. The default values would differ depending on whether
+ or not cdr.conf exists. That is no longer the case. Apply a
+ default value to the unanswered option. Define all default values
+ as named constants.
+
+2010-08-05 07:40 +0000 [r280983] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280982
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
+ | 8 lines Change context lock back to a mutex, because
+ functionality depends upon the lock being recursive. (closes
+ issue #17643) Reported by: zerohalo Patches:
+ 20100726__issue17643.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........
+
+2010-08-03 20:52 +0000 [r280671-280812] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_callerid.c, channels/chan_dahdi.c, /: Merged revisions
+ 280811 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010)
+ | 9 lines Prevent DAHDI channels from overriding the callerid,
+ once it's been set by the user. (closes issue #16661) Reported
+ by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by
+ tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: jstapleton ........
+
+ * doc/asterisk.sgml, doc/asterisk.8, doc/Makefile (added): Document
+ -B and -W flags and regenerate manpage from sgml
+
+ * apps/app_voicemail.c: Allow the pipe, but also allow the comma
+
+2010-08-02 21:14 +0000 [r280669] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Change SIP NOTIFY requests to expect a
+ response so authentication will work. This changes the request to
+ be sent with the transmit type XMIT_RELIABLE so that sip_ack
+ doesn't return false and cause the 401 to be ignored in cases
+ where authentication is required. (closes issue #14255) Reported
+ by: zktech
+
+2010-07-29 21:07 +0000 [r280556] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_curl.c: Off-by-one error (closes issue #17590)
+ Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded
+ by tilghman (license 14)
+
+2010-07-29 20:42 +0000 [r280449-280551] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes wrong SRV query for TLS connection
+ (closes issue #17612) Reported by: marcelloceschia Patches:
+ chan-sip_srvQuery.patch uploaded by marcelloceschia (license
+ 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
+ chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
+ (license 1079) Tested by: marcelloceschia, st, pabelanger
+
+ * main/channel.c, /: Merged revisions 280448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
+ | 12 lines fixes issue with translator frame not getting freed A
+ translator frame even if it local storage so the translation path
+ can be freed. This issue prevented g729 licenses from being freed
+ up. (closes issue #17630) Reported by: manvirr Patches:
+ encoder_fix.diff uploaded by dvossel (license 671) Tested by:
+ manvirr, dvossel ........
+
+2010-07-29 16:01 +0000 [r280345] Jean Galarneau <jgalarneau@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 280341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
+ 2 lines Fix a dsp structure leak occuring when a local channel is
+ put into a meetme conference, then masquaraded away. ABE-2422
+ ........
+
+2010-07-29 13:45 +0000 [r280306] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_local.c: Implement support for
+ ast_channel_queryoption on local channels. Currently only
+ AST_OPTION_T38_STATE is supported. ABE-2229 Review:
+ https://reviewboard.asterisk.org/r/813/
+
+2010-07-28 20:02 +0000 [r280231] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile: Work around some silly behavior on BSD. A
+ non-zero exit from a subshell should make the build fail. (closes
+ issue #17621)
+
+2010-07-28 19:57 +0000 [r280229] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Add missing enum value "unknown" to the
+ SS7 called_nai and calling_nai config options.
+
+2010-07-28 19:54 +0000 [r280193-280227] Jason Parker <jparker@digium.com>
+
+ * build_tools/sha1sum-sh (added): Add sha1sum-sh in case there is
+ no util on the system.
+
+ * sounds/Makefile: Remove unnecessary subshells. Attempt to make
+ checksumming work. Also improves readability. (issue #17621)
+ Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
+
+2010-07-28 16:51 +0000 [r280160] Sean Bright <sean@malleable.com>
+
+ * apps/app_queue.c: Plug a reference leak in app_queue when adding
+ members dynamically. (closes issue #17738) Reported by:
+ bobwienholt Patches: issue17738.patch uploaded by bobwienholt
+ (license 950) Tested by: bobwienholt, seanbright
+
+2010-07-28 13:51 +0000 [r280089] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/live_ast, /: Merged revisions 280088 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
+ Jul 2010) | 1 line Update help text to be less confusing.
+ ........
+
+2010-07-27 20:54 +0000 [r279946] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c, main/channel.c, /,
+ include/asterisk/audiohook.h: Merged revisions 279945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
+ | 19 lines remove empty audiohook write list on channel If a
+ channel has an audiohook write list created on it, that list
+ stays on the channel until the channel is destroyed. There is no
+ reason to keep that list on the channel if it becomes empty. If
+ it is empty that just means we are doing needless translating for
+ every ast_read and ast_write. This patch removes the audiohook
+ list from the channel once it is detected to be empty on either a
+ read or write. If a audiohook is added back to the channel after
+ this list is destroyed, the list just gets recreated as if it
+ never existed to begin with. (closes issue #17630) Reported by:
+ manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
+
+2010-07-27 17:54 +0000 [r279849-279883] Jason Parker <jparker@digium.com>
+
+ * makeopts.in, configure, configure.ac: Add SHA1SUM to configure,
+ since we require it for sounds/
+
+ * sounds/Makefile: Remove aptly-named EMPTY and BS vars, since they
+ aren't used anymore.
+
+ * sounds/Makefile: Simply sounds/Makefile some more.
+
+2010-07-27 15:13 +0000 [r279784] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix bad behavior of dynamic_exclude_static
+ option in sip.conf. We were attempting to create a contactdeny
+ rule based on the peer's IP address before the peer's IP address
+ had been set. By moving the processing further down in the
+ function, we can ensure stuff works as we expect for it to.
+ (closes issue #17717) Reported by: mmichelson Patches:
+ 17717.patch uploaded by mmichelson (license 60) Tested by:
+ DennisD
+
+2010-07-26 22:59 +0000 [r279657] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile (added), sounds/Makefile.380 (removed),
+ configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+ (removed), configure.ac: Really fix sounds Makefile (and make it
+ readableish). There was a rather large syntax error that should
+ have caused ALL versions of GNU make to fail. I don't know how it
+ worked.
+
+2010-07-26 21:18 +0000 [r279609] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Dunno why this worked on my machine, but
+ it works better this way.
+
+2010-07-26 20:25 +0000 [r279597] Gavin Henry <ghenry@suretecsystems.com>
+
+ * res/res_config_ldap.c: Apply all patches in:
+ https://issues.asterisk.org/view.php?id=13573 (closes issue
+ #13573) Reported by: navkumar Patches:
+ res_config_ldap-category.diff uploaded by navkumar (license 580)
+ res_config_ldap.patch uploaded by bencer (license 961)
+ res_config_ldap uploaded by bencer (license 961) Tested by:
+ suretec
+
+2010-07-26 19:15 +0000 [r279561] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile (removed), configure, sounds/Makefile.380
+ (added), sounds/Makefile.381 (added), configure.ac: Use a special
+ Makefile for noobs who still have GNU Make 3.80. (Closes issue
+ #17716) Reported by: farisraouf
+
+2010-07-26 15:41 +0000 [r279501] Sean Bright <sean@malleable.com>
+
+ * autoconf/ast_ext_lib.m4: Expand the correct value within
+ AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
+
+2010-07-24 23:58 +0000 [r279347] Bradley Latus <brad.latus@gmail.com>
+
+ * doc/asterisk.8: Minor update to man page
+
+2010-07-23 22:11 +0000 [r279207] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279206 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
+ | 7 lines SIP promiscuous redirect could fail to dial the
+ redirect. The ast_channel was created with one variable to
+ ast_request() but the call to ast_call() that initiates the
+ outgoing call was using a different variable. The two variables
+ are not equivalent if the call_forward string included a channel
+ technology specifier. e.g., SIP/200 ........
+
+2010-07-23 18:29 +0000 [r279112] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Backport sip_uri_params_cmp() fix from trunk
+ to 1.6.2.
+
+2010-07-23 18:22 +0000 [r279072-279088] Russell Bryant <russell@digium.com>
+
+ * /: remove old properties
+
+ * /: Add branch-1.4-merged and branch-1.4-blocked properties to
+ 1.6.2 branch.
+
+2010-07-23 17:06 +0000 [r278983-278986] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
+ revisions 278985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010)
+ | 12 lines Merged revisions 278984 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
+ | 5 lines Establish a maximum version for openh323 (i.e. not
+ opal), because chan_h323 will fail to load, even if it links.
+ (issue #17679) Reported by: am ........ ................
+
+ * main/asterisk.c, /: Merged revisions 278982 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010)
+ | 15 lines Merged revisions 278981 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
+ | 8 lines Avoid race with consolethread on shutdown (on parallel
+ processors). (closes issue #17080) Reported by: sybasesql
+ Patches: 20100721__issue17080.diff.txt uploaded by tilghman
+ (license 14) Tested by: sybasesql ........ ................
+
+2010-07-23 15:23 +0000 [r278934] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: Two more typos to cancell.
+
+2010-07-22 19:52 +0000 [r278709] Jeff Peeler <jpeeler@digium.com>
+
+ * main/xmldoc.c, /: Merged revisions 278708 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r278708 |
+ jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines
+ Add method for finding XML doc files for systems that don't
+ support GLOB_BRACE. In particular, Solaris and perhaps others do
+ not support the above mentioned GNU extension. In this case the
+ paths are simply expanded without the braces and the calls to
+ glob are made separately. Note: I could not explain memory
+ allocation failures that were being reported from within libxml
+ itself when making calls to glob without using GLOB_NOCHECK. This
+ is the only reason why that flag is being used. (closes issue
+ #15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded
+ by snuffy (license 35), modified by me ........
+
+2010-07-22 19:32 +0000 [r278703] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: DNID does not get cleard on a new call
+ when using immediate=yes with ISDN signaling. When you are using
+ chan_dahdi ISDN signaling with immediate=yes and a call comes in
+ without a DNID then you get the DNID of a previous call.
+ Chan_dahdi does not touch the DNID field on a new call if it does
+ not have a DNID. Made always copy the DNID from the new call. The
+ patches backport the relevant changes from trunk -r210387.
+ (closes issue #17568) Reported by: wuwu Patches:
+ issue17568_v1.4.patch uploaded by rmudgett (license 664)
+ issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
+
+2010-08-10 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.11 Released.
+
+2010-07-26 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.11-rc2 Released.
+
+2010-07-26 Leif Madsen <lmadsen@digium.com>
+
+ * qwell, asterisk, branch-1.6.2, r279657 ***
+ Really fix sounds Makefile (and make it readableish).
+ There was a rather large syntax error that should have
+ caused ALL versions of GNU make to fail.
+ I don't know how it worked.
+
+ (Closes issue #17716)
+
+2010-07-22 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.11-rc1 Released.
+
+2010-07-22 15:00 +0000 [r278621] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 278620 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul
+ 2010) | 19 lines Merged revisions 278618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
+ 2010) | 13 lines Allow PLC to function properly when channels use
+ SLIN for audio. If a channel involved in a bridge was using SLIN
+ audio, then translation paths were not guaranteed to be set up
+ properly since in all likelihood the number of translation steps
+ was only 1. This patch enforces the transcode_via_slin behavior
+ if transcode_via_slin or generic_plc is enabled and one of the
+ formats to make compatible is SLIN. AST-352 ........
+ ................
+
+2010-07-21 18:22 +0000 [r278524] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: Fix invalid test for rxisoffhook in FXO
+ channels This fixes some cases of no outgoing calls on FXO before
+ an incoming call. Remove an unnecessary testing of an "off-hook"
+ bit from DAHDI for FXO (KS/GS) channels.In some cases the bit
+ would not be initialized properly before the first inbound call
+ and thus prevent an outgoing call. If those tests are actually
+ required by anybody, they should define DAHDI_CHECK_HOOKSTATE in
+ channels/sig_analog.c . (closes issue #14577) Reported by: jkroon
+ Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd
+ (license 610) Tested by: frawd Review:
+ https://reviewboard.asterisk.org/r/699/
+
+2010-07-21 16:20 +0000 [r278479] Russell Bryant <russell@digium.com>
+
+ * /, res/res_timing_pthread.c: Merged revisions 278465 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010)
+ | 41 lines Use poll() instead of select() in res_timing_pthread
+ to avoid stack corruption. This code did not properly check
+ FD_SETSIZE to ensure that it did not try to select() on fds that
+ were too large. Switching to poll() removes the limitation on the
+ maximum fd value. (closes issue #15915) Reported by: keiron
+ (closes issue #17187) Reported by: Eddie Edwards (closes issue
+ #16494) Reported by: Hubguru (closes issue #15731) Reported by:
+ flop (closes issue #12917) Reported by: falves11 (closes issue
+ #14920) Reported by: vrban (closes issue #17199) Reported by:
+ aleksey2000 (closes issue #15406) Reported by: kowalma (closes
+ issue #17438) Reported by: dcabot (closes issue #17325) Reported
+ by: glwgoes (closes issue #17118) Reported by: erikje possibly
+ other issues, too ... ........
+
+2010-07-21 15:58 +0000 [r278025-278464] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 278463 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 |
+ tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11
+ lines Ensure realtime conferences are treated the same as static
+ conferences when trying to find an empty one. Also, parse the
+ useropts properly, when retrieving from realtime, and add them to
+ the existing flags. (closes issue #17502) Reported by: kenji
+ Patches: 20100720__issue17502.diff.txt uploaded by tilghman
+ (license 14) Tested by: kenji ........
+
+ * apps/app_voicemail.c, /: Merged revisions 278275 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500
+ (Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010)
+ | 7 lines Delete IMAP messages in reverse order, to ensure
+ reordering after each expunge does not cause deletion of the
+ wrong message. (closes issue #16350) Reported by: noahisaac
+ Patches: 20100623__issue16350.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+ * main/autoservice.c, /, main/features.c,
+ include/asterisk/channel.h: Merged revisions 278272 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r278272 | tilghman | 2010-07-20 17:26:23 -0500
+ (Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010)
+ | 4 lines Do not queue up DTMF frames while a call is on hold.
+ (Fixes ABE-2110) ........ ................
+
+ * main/manager.c, /: Merged revisions 278024 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010)
+ | 14 lines Merged revisions 278023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
+ | 7 lines Off-by-one error (closes issue #16506) Reported by:
+ nik600 Patches: 20100629__issue16506.diff.txt uploaded by
+ tilghman (license 14) ........ ................
+
+2010-07-19 21:21 +0000 [r277966] Jean Galarneau <jgalarneau@digium.com>
+
+ * /, main/features.c: Merged revisions 277945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) |
+ 15 lines Merged revisions 277906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
+ 7 lines Avoid trying to pickup a parked extension before the park
+ operation is completed. A crash could occur if the extension is
+ picked up while the parking extension is being announced. Testing
+ pu->notquiteyet while searching for a parked extension resolves
+ this crash. (ABE-2418) ........ ................
+
+2010-07-17 17:52 +0000 [r277774-277777] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c: Merge issues...
+
+ * /, autoconf/ast_func_fork.m4, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 277775 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r277775 | tilghman | 2010-07-17 12:42:32 -0500
+ (Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
+ | 5 lines Remove uclibc cross-compile triplet, as uclibc has a
+ working fork()... it's only uclinux that does not. (closes issue
+ #17616) Reported by: pprindeville ........ ................
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+ revisions 277773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010)
+ | 15 lines Merged revisions 277568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010)
+ | 8 lines Since we split values at the semicolon, we should store
+ values with a semicolon as an encoded value. (closes issue
+ #17369) Reported by: gkservice Patches:
+ 20100625__issue17369.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........ ................
+
+2010-07-16 23:37 +0000 [r277666] Tim Ringenbach <tim.ringenbach@gmail.com>
+
+ * /, main/features.c: Merged revisions 277657 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul
+ 2010) | 16 lines Merged revisions 277625 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
+ 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
+ attended transfer. ast_bridge_call() clears
+ AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
+ ast_bridge_call() is called for a second bridge on the same
+ channel, and it clears that flag, which still needs to get set
+ for when the original ast_bridge_call() gets control back and
+ checks it. Review: https://reviewboard.asterisk.org/r/741
+ ........ ................
+
+2010-07-16 21:31 +0000 [r277563] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 277530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul
+ 2010) | 11 lines Merged revisions 277497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
+ 2010) | 4 lines Default to no udptl error correction so that
+ error correction will be disabled in the event that the remote
+ end indicates that they do not support the error correction mode
+ we requested. FAX-128 ........ ................
+
+2010-07-16 21:16 +0000 [r277489] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 277488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 |
+ jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines
+ Fix reporting estimated queue hold time. Just say the number of
+ seconds (after minutes) rather than doing some incorrect
+ calculation with respect to minutes. (closes issue #17498)
+ Reported by: corruptor Patches: holdesecs_bug.diff uploaded by
+ corruptor (license 253) ........
+
+2010-07-16 20:35 +0000 [r277485] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 277467 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500
+ (Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010)
+ | 15 lines priexclusive in chan_dahdi.conf ignored when reloading
+ dahdi module During a reload, the priexclusive and outsignalling
+ parameters are not read in from the config file as intended.
+ Unfortunately, they get set to defaults as a result. This patch
+ makes sure that they do not get set to defaults during a reload.
+ (closes issue #17441) Reported by: mtryfoss Patches:
+ issue17441_v1.4.patch uploaded by rmudgett (license 664)
+ issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
+ issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
+ by: rmudgett ........ ................
+
+2010-07-16 20:30 +0000 [r277478] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
+ (added), /: Merged revisions 277452 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277452 |
+ tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines
+ Add documentation for MOH realtime fields ........
+
+2010-07-16 19:24 +0000 [r277377] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 277366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 |
+ jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines
+ Add missing handling for ringing state for use with queue empty
+ options. (closes issue #17471) Reported by: jazzy Patches:
+ app_queue.c.diff uploaded by jazzy (license 1056) ........
+
+2010-07-16 18:33 +0000 [r277338] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 277331 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul
+ 2010) | 15 lines Merged revisions 277327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul
+ 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as
+ extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
+ Reported by: francesco_r Patches: pbx.c.patch uploaded by
+ viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
+ ........ ................
+
+2010-07-16 18:14 +0000 [r277264] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 277263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010)
+ | 12 lines Merged revisions 277261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010)
+ | 5 lines If variable gotten is not set, will segfault on
+ Solaris. (closes issue #17636) Reported by: bklang ........
+ ................
+
+2010-07-16 17:31 +0000 [r277256] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 277250 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277250 | mnicholson | 2010-07-16 12:30:39 -0500 (Fri, 16 Jul
+ 2010) | 11 lines Merged revisions 277247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul
+ 2010) | 4 lines For pass through DTMF tones, measure the actual
+ duration between the begin and end packets on the wire. If it is
+ detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
+ emulation. AST-362 ........ ................
+
+2010-07-16 17:18 +0000 [r277188] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_amd.c: Merged revisions 277183 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul
+ 2010) | 15 lines Merged revisions 277182 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul
+ 2010) | 8 lines Total analysis time error with SIP and silence
+ suppression When using app_amd with SIP providers that have
+ silence suppression on, the iTotalTime count increases
+ exponentially. (closes issue #17656) Reported by: juls ........
+ ................
+
+2010-07-16 15:21 +0000 [r277144] Sean Bright <sean@malleable.com>
+
+ * /, main/translate.c: Merged revisions 277143 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277143 |
+ seanbright | 2010-07-16 11:20:40 -0400 (Fri, 16 Jul 2010) | 8
+ lines Avoid crashing when installing a duplicate translation path
+ with a lower cost. (closes issue #17092) Reported by: moy
+ Patches: translate.rev254273.patch uploaded by moy (license 222)
+ Tested by: moy ........
+
+2010-07-15 20:42 +0000 [r276572-276809] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 276788 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r276788 |
+ jpeeler | 2010-07-15 15:21:03 -0500 (Thu, 15 Jul 2010) | 6 lines
+ Correct not setting the bindport before attempting to open the
+ socket. Related to changes from 276571, I was accidentally
+ testing with a port set in my configuration causing me to miss
+ this. Also moved the TCP handling as well to occur before
+ build_peer is called. ........
+
+ * main/channel.c, /: Merged revisions 276653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r276653 | jpeeler | 2010-07-15 08:51:11 -0500 (Thu, 15 Jul 2010)
+ | 9 lines Merged revisions 276652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010)
+ | 2 lines In a perfect world, the frame source would never be
+ NULL. In the meantime, don't crash when it is. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 276571 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r276571 |
+ jpeeler | 2010-07-14 17:58:24 -0500 (Wed, 14 Jul 2010) | 21 lines
+ Fix MWI notification transmission problems over SIP. MWI updates
+ were not being sent if no messages were found in the event cache.
+ This was corrected since a phone may need to clear its MWI status
+ configured previously from another mailbox. Upon module or sip
+ reload, MWI updates could not be sent due to the sipsock socket
+ not being set early enough in reload_config. The code handling
+ the descriptor assignment and such has simply been moved before
+ the call to build_peer. Issuing a sip reload cleared the IP
+ address of the peer, but skipped checking the database for
+ registration information. The database is now checked both for
+ sip reload and actually reloading the module. If a transmission
+ occurs before the do_monitor thread has started, do not attempt
+ to send a signal to it. (closes issue #17398) Reported by: ip-rob
+ ........
+
+2010-07-14 20:16 +0000 [r276442] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/loader.c, /: Merged revisions 276441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r276441 |
+ kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4
+ lines Don't try to call an embedded module's backup_globals()
+ function until after confirming it exists. ........
+
+2010-07-14 11:52 +0000 [r276269] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 276268 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r276268 | lmadsen | 2010-07-14 06:51:48 -0500
+ (Wed, 14 Jul 2010) | 9 lines Merged revisions 276267 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14
+ Jul 2010) | 1 line Update documentation for voicemail.conf
+ externpass option. ........ ................
+
+2010-07-13 19:11 +0000 [r276125] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 276124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r276124 | russell | 2010-07-13 14:09:42 -0500 (Tue, 13 Jul 2010)
+ | 9 lines Merged revisions 276123 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010)
+ | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr
+ instead of peer_cdr in the last commit). ........
+ ................
+
+2010-07-13 19:01 +0000 [r276121] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 276074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010)
+ | 19 lines Merged revisions 275773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010)
+ | 12 lines Make user removals and traversals thread safe in
+ meetme. Race conditions present in meetme involving the user list
+ where a lack of locking has the potential for a user to be
+ removed during a traversal or as in the case of the reporter
+ after checking if the list is empty could cause a crash. Fixing
+ this was done by convering the userlist to an ao2 container.
+ (closes issue #17390) Reported by: Vince Review:
+ https://reviewboard.asterisk.org/r/746/ ........ ................
+
+2010-07-13 16:55 +0000 [r275996] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 275995 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275995 | russell | 2010-07-13 11:53:44 -0500 (Tue, 13 Jul 2010)
+ | 21 lines Merged revisions 275994 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010)
+ | 14 lines Access peer->cdr directly instead of through a saved
+ off reference. At this point in the code, it is possible that
+ peer_cdr may be invalid. Specifically, in the blind transfer
+ code, CDRs are swapped between channels. So, peer_cdr is no
+ longer == peer->cdr. The scenario that exposed a crash in this
+ code was a blind transfer that hit the system call limit, causing
+ the transferee channel to get destroyed after the transfer
+ attempt failed. Even if it succeeds and this code doesn't crash,
+ this code was still trying to reset a CDR on a channel that was
+ now owned by a different thread, which is a BadThing(tm).
+ (ABE-2417) ........ ................
+
+2010-07-13 14:49 +0000 [r275911] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/realtime/mysql, contrib/realtime/oracle,
+ contrib/scripts/sip-friends.sql (removed),
+ contrib/realtime/mysql/sipfriends.sql,
+ contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql
+ (removed), contrib/realtime/mysql/meetme.sql,
+ contrib/realtime/sqlserver, contrib/scripts/realtime_pgsql.sql
+ (removed), contrib/scripts/iax-friends.sql (removed), /,
+ contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql
+ (removed), contrib/realtime (added), contrib/realtime/postgresql,
+ contrib/realtime/postgresql/realtime.sql: Merged revisions 275910
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r275910 | tilghman | 2010-07-13 09:48:40 -0500
+ (Tue, 13 Jul 2010) | 9 lines Merged revisions 275909 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13
+ Jul 2010) | 2 lines Move SQL scripts into their own
+ database-specific directories. ........ ................
+
+2010-07-12 17:26 +0000 [r275706] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 275682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275682 | jpeeler | 2010-07-12 12:21:01 -0500 (Mon, 12 Jul 2010)
+ | 18 lines Merged revisions 275665 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010)
+ | 11 lines Change ast_write to not stop generator when called
+ from ast_prod. For SIP channels configured with the
+ progressinband option on, the ringback was being immediately
+ stopped. This problem was due to ast_prod being moved for a
+ deadlock fix in 259858. Prodding the channel after setting up the
+ generator triggered the check in ast_write to stop the generator.
+ The fix here should write the frame the same as was done before
+ the call to ast_prod was moved. (closes issue #17372) Reported
+ by: tech_admin ........ ................
+
+2010-07-12 15:38 +0000 [r275627] Leif Madsen <lmadsen@digium.com>
+
+ * cdr/cdr_pgsql.c, /: Merged revisions 275626 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r275626 |
+ lmadsen | 2010-07-12 10:37:01 -0500 (Mon, 12 Jul 2010) | 11 lines
+ cdr_pgsql does not detect when a table is found. This change adds
+ an ERROR message to let you know when a failure exists to get the
+ columns from the pgsql database, which typically means that the
+ table does not exist. (closes issue #17478) Reported by: kobaz
+ Patches: cdr_pgsql.patch uploaded by kobaz (license 834) Tested
+ by: kobaz, russell, lmadsen ........
+
+2010-07-10 15:11 +0000 [r275311-275469] Russell Bryant <russell@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 245192 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245192 |
+ mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21
+ lines Remove useless sip options related to hash table size.
+ First off, these options weren't actually doing anything. By the
+ time the options were parsed, the peer and dialog containers had
+ already been allocated with their default values. Second, hash
+ table size is something that doesn't really make sense to change
+ in a config file. If a user is that interested in changing the
+ hashtable size, he can modify the source itself. I have removed
+ the parsing of the hash_peer, hash_user, and hash_dialog options.
+ I have removed the hash_user_size variable altogether since it is
+ not used at all. I also changed hash_peer_size and
+ hash_dialog_size to be constant, and have changed the symbols to
+ be in all caps as constants typically are. I have also removed
+ the entire section in sip.conf.sample regarding configurable
+ hashtable sizes. ........ (merge to 1.6.2 inspired by issue
+ #17553)
+
+ * /: unblock a rev
+
+ * configs/features.conf.sample, /, main/features.c: Merged
+ revisions 275424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r275424 |
+ russell | 2010-07-09 16:57:21 -0500 (Fri, 09 Jul 2010) | 27 lines
+ Fix some issues related to dynamic feature groups in
+ features.conf. The bridge handling code did not properly consider
+ feature groups when setting parameters that would affect whether
+ or not a native bridge would be attempted. If DYNAMIC_FEATURES
+ only include a feature group, a native bridge would occur that
+ may prevent features from working. Fix a bug in verbose output
+ that would show the key mapping as empty if it was using the
+ default mapping and not a custom mapping in the feature group.
+ Add feature groups to the output of "features show". Adjust the
+ feature execution logic to match that of the logic when executing
+ a feature that was not configured through a feature group. Update
+ features.conf.sample to show that an '=' is still required if
+ using the default key mapping from [applicationmap]. Finally,
+ clean up a little bit of formatting to better coform to coding
+ guidelines while in the area. (closes issue #17589) Reported by:
+ lmadsen Patches: issue_17589.rev4.txt uploaded by russell
+ (license 2) Tested by: russell, lmadsen ........
+
+ * /, main/features.c: Merged revisions 275310 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r275310 |
+ russell | 2010-07-09 14:58:06 -0500 (Fri, 09 Jul 2010) | 2 lines
+ Add missing ao2_iterator_destroy(). ........
+
+2010-07-09 19:23 +0000 [r275260] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: Merged revisions 275249 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275249 | pabelanger | 2010-07-09 15:21:27 -0400 (Fri, 09 Jul
+ 2010) | 15 lines Merged revisions 275241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul
+ 2010) | 8 lines Fix logging message for stale nonce. (closes
+ issue #17582) Reported by: kenner Patches: chan_sip.c.diff
+ uploaded by kenner (license 1040) Tested by: lmadsen ........
+ ................
+
+2010-07-09 18:24 +0000 [r275191] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/loader.c, /: Merged revisions 275186 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275186 | mnicholson | 2010-07-09 13:24:03 -0500 (Fri, 09 Jul
+ 2010) | 9 lines Merged revisions 275182 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul
+ 2010) | 2 lines give a better error message when attempting to
+ unload a module that is not loaded ........ ................
+
+2010-07-09 18:11 +0000 [r275148] Russell Bryant <russell@digium.com>
+
+ * configs/features.conf.sample, /: Merged revisions 275147 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r275147 | russell | 2010-07-09 13:11:13 -0500 (Fri, 09
+ Jul 2010) | 2 lines Move parking lot sample config out from the
+ middle of dynamic features sample config. ........
+
+2010-07-09 17:51 +0000 [r275029-275145] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/loader.c, /: Merged revisions 275144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul
+ 2010) | 9 lines Merged revisions 275143 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul
+ 2010) | 2 lines don't unload modules that returned
+ AST_MODULE_LOAD_DECLINE when they were loaded ........
+ ................
+
+ * apps/app_dial.c, /: Merged revisions 275028 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul
+ 2010) | 15 lines Merged revisions 275027 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul
+ 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels
+ going into the pbx via the G option in app_dial (closes issue
+ #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: jamicque,
+ mnicholson ........ ................
+
+2010-07-09 15:39 +0000 [r275023] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/test.h, /, main/test.c: Merged revisions 275022
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r275022 | russell | 2010-07-09 10:35:53 -0500
+ (Fri, 09 Jul 2010) | 11 lines Merged revisions 275021 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010)
+ | 4 lines Document that a leading and trailing slash is expected
+ for test categories. Also, emit a warning if a test is registered
+ without one of these. ........ ................
+
+2010-07-07 18:34 +0000 [r274627-274640] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 274639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r274639 | rmudgett | 2010-07-07 13:32:35 -0500 (Wed, 07 Jul 2010)
+ | 1 line Add missing conditional around chan_dahdi
+ mfcr2_skip_category config parameter. ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 274595 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r274595 | rmudgett | 2010-07-07 13:20:00 -0500
+ (Wed, 07 Jul 2010) | 9 lines Merged revisions 274579 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07
+ Jul 2010) | 1 line Close the DAHDI FD on error when processing
+ chan_dahdi toneduration config parameter. ........
+ ................
+
+2010-07-07 06:16 +0000 [r274419] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 274418 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r274418 | tilghman | 2010-07-07 01:15:43 -0500
+ (Wed, 07 Jul 2010) | 15 lines Merged revisions 274417 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010)
+ | 8 lines Correct how 100, 200, 300, etc. is said. Also add the
+ crazy British numbers. (closes issue #16102) Reported by: Delvar
+ Patches: say.conf.fix.patch uploaded by Delvar (license 908)
+ (plus a few additional fixes and simplifications by me) ........
+ ................
+
+2010-07-06 23:06 +0000 [r274360] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, channels/chan_sip.c: Merged revisions
+ 274284 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010)
+ | 18 lines Merged revisions 274280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010)
+ | 9 lines Add option to not do a call forward on 482 Loop
+ Detected Asterisk has always set up a forwarded call when
+ receiving a 482 Loop Detected. This prevents handling the call
+ failure by just continuing on in the dialplan. Since this would
+ be a change in behavior, the new option to disable this behavior
+ is forwardloopdetected which defaults to 'yes'. Review:
+ https://reviewboard.asterisk.org/r/764/ ........ ................
+
+2010-07-06 22:30 +0000 [r274347] Jeff Peeler <jpeeler@digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 274316 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r274316 | jpeeler | 2010-07-06 17:23:35 -0500
+ (Tue, 06 Jul 2010) | 14 lines Merged revisions 274283 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010)
+ | 7 lines Correct sip.conf.sample comments for prematuremedia
+ option. (closes issue #17513) Reported by: festr Patches: patch
+ uploaded by festr (license 443) ........ ................
+
+2010-07-06 22:10 +0000 [r274282] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 274281 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r274281 | tilghman | 2010-07-06 17:09:23 -0500 (Tue, 06 Jul 2010)
+ | 2 lines Status shows all non-CRC4 lines as "yellow", even if
+ "yellow" was not in the bitfield. ........
+
+2010-07-06 14:33 +0000 [r274168] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Merged revisions 274164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r274164 | mmichelson | 2010-07-06 09:31:13 -0500 (Tue, 06 Jul
+ 2010) | 22 lines Merged revisions 274157 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul
+ 2010) | 16 lines Fix problem with RFC 2833 DTMF not being
+ accepted. A recent check was added to ensure that we did not
+ erroneously detect duplicate DTMF when we received packets out of
+ order. The problem was that the check did not account for the
+ fact that the seqno of an RTP stream will roll over back to 0
+ after hitting 65535. Now, we have a secondary check that will
+ ensure that the seqno rolling over will not cause us to stop
+ accepting DTMF. (closes issue #17571) Reported by: mdeneen
+ Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license
+ 60) Tested by: richardf, maxochoa, JJCinAZ ........
+ ................
+
+2010-07-05 13:55 +0000 [r273888] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/config.c, /: Merged revisions 273886 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273886 | pabelanger | 2010-07-05 09:53:44 -0400 (Mon, 05 Jul
+ 2010) | 15 lines Merged revisions 273884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul
+ 2010) | 8 lines Remove extra line breaks from 'core show config
+ mappings' (closes issue #17583) Reported by: pabelanger Patches:
+ issue17583.patch uploaded by pabelanger (license 224) Tested by:
+ lmadsen ........ ................
+
+2010-07-03 02:43 +0000 [r273716-273831] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /, channels/chan_agent.c,
+ channels/chan_h323.c, include/asterisk/lock.h: Merged revisions
+ 273830 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010)
+ | 16 lines Merged revisions 273793 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010)
+ | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock
+ fails, to help catch potentially large software bugs. (closes
+ issue #17407) Reported by: pdf Patches:
+ 20100527__issue17407.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/751/ ........
+ ................
+
+ * main/autoservice.c, /: Merged revisions 273718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273718 | tilghman | 2010-07-02 12:10:59 -0500 (Fri, 02 Jul 2010)
+ | 15 lines Merged revisions 273717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010)
+ | 8 lines Autoservice loop optimization causes a busy loop, when
+ channels are serviced while in hangup. (closes issue #17564)
+ Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt
+ uploaded by tilghman (license 14) Tested by: ramonpeek ........
+ ................
+
+ * apps/app_queue.c, /: Merged revisions 273714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273714 |
+ tilghman | 2010-07-02 11:57:28 -0500 (Fri, 02 Jul 2010) | 2 lines
+ The switch fallthrough could create some errorneous situations,
+ so best to force directly to the default case. ........
+
+2010-07-02 15:59 +0000 [r273642] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_iax2.c, apps/app_voicemail.c,
+ channels/chan_dahdi.c, channels/chan_sip.c, res/res_agi.c: Fix
+ typos reported by Lintian
+
+2010-07-01 22:17 +0000 [r273571] Russell Bryant <russell@digium.com>
+
+ * main/datastore.c, /: Merged revisions 273566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273566 | russell | 2010-07-01 17:16:23 -0500 (Thu, 01 Jul 2010)
+ | 14 lines Merged revisions 273565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010)
+ | 7 lines Don't return a partially initialized datastore. If
+ memory allocation fails in ast_strdup(), don't return a partially
+ initialized datastore. Bad things may happen. (related to
+ ABE-2415) ........ ................
+
+2010-07-01 20:29 +0000 [r273356-273529] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 273522 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273522 | jpeeler | 2010-07-01 15:28:15 -0500 (Thu, 01 Jul 2010)
+ | 21 lines Merged revisions 273474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010)
+ | 14 lines Allow admin user to join conference without using
+ admin mode and no user pin. Configuring the conference in
+ meetme.conf like the following: conf => 2345,,6666 did not prompt
+ for pin when used without admin mode. This meant that the
+ conference could not be joined as an admin even if the user knew
+ the correct pin. The original bug report was submitted claiming
+ that the blank user pin should deny entry into the conference. I
+ think a better way to handle this would be with a feature
+ enhancement that used the following syntax: conf => 2345,X,6666 -
+ where X denotes no acceptable pin allowed (closes issue #15704)
+ Reported by: modelnine ........ ................
+
+ * /, apps/app_meetme.c: Merged revisions 273355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273355 | jpeeler | 2010-07-01 10:12:31 -0500 (Thu, 01 Jul 2010)
+ | 19 lines Merged revisions 273354 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010)
+ | 12 lines Ensure channel placed in meetme in ringing state is
+ properly hung up. An outgoing channel placed in meetme while
+ still ringing which was then hung up would not exit meetme and
+ the channel was not properly destroyed. Specifically checking for
+ this scenario by looking at the appropriate control frames
+ resolves the issue. (closes issue #15871) Reported by: Ivan
+ Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
+ (license 229) ........ ................
+
+2010-07-01 14:39 +0000 [r273271-273353] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /: Merged revisions 273352 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273352 |
+ mnicholson | 2010-07-01 09:37:37 -0500 (Thu, 01 Jul 2010) | 2
+ lines Fixed whitespace problems ........
+
+ * main/manager.c, /: Merged revisions 273350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273350 |
+ mnicholson | 2010-07-01 09:34:31 -0500 (Thu, 01 Jul 2010) | 2
+ lines Altered my comment about TCP_NODELAY ........
+
+ * main/manager.c, /: Merged revisions 273270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273270 |
+ mnicholson | 2010-06-30 13:48:21 -0500 (Wed, 30 Jun 2010) | 2
+ lines Set TCP_NODELAY on manager TCP sockets to prevent delays on
+ outgoing packets. This regression was introduced in r48338.
+ AST-359 ........
+
+2010-06-30 17:32 +0000 [r273193-273234] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/rtp.c, /: Merged revisions 273233 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273233 |
+ pabelanger | 2010-06-30 13:28:04 -0400 (Wed, 30 Jun 2010) | 11
+ lines Fix rt(c)p set debug ip taking wrong argument Also clean up
+ some coding errors. (closes issue #17469) Reported by: wdoekes
+ Patches: astsvn-rtp-set-debug-ip.patch uploaded by wdoekes
+ (license 717) Tested by: wdoekes, pabelanger ........
+
+ * /: Revert previous commit; res_rtp_asterisk.c does not exist.
+
+ * /: Unblock revisions 218107 ........ r218107 | mvanbaak |
+ 2009-09-12 15:08:16 +0200 (Sat, 12 Sep 2009) | 8 lines use the
+ actual given ip address for 'rtp set debug ip <foo>' instead of
+ the word 'ip' (closes issue 0015711) Reported by: davidw Patches:
+ 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
+ Tested by: davidw ........
+
+2010-06-30 01:07 +0000 [r273056-273145] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 273144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273144 |
+ tilghman | 2010-06-29 20:07:02 -0500 (Tue, 29 Jun 2010) | 8 lines
+ Permission checking for the system application is backwards.
+ (closes issue #17550) Reported by: kenner Patches: manager.c.diff
+ uploaded by kenner (license 1040) Tested by: kenner ........
+
+ * main/config.c, /: Merged revisions 273142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273142 |
+ tilghman | 2010-06-29 20:01:14 -0500 (Tue, 29 Jun 2010) | 5 lines
+ Don't attempt to proceed if our internal parser indicates an
+ invalid file. (closes issue #17560) Reported by: Nick_Lewis
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 273078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273078 | tilghman | 2010-06-29 18:20:40 -0500 (Tue, 29 Jun 2010)
+ | 17 lines Merged revisions 273060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010)
+ | 10 lines Allow the "useragent" value to be restored into memory
+ from the realtime backend. This value is purely informational. It
+ does not alter configuration at all. (closes issue #16029)
+ Reported by: Guggemand Patches: realtime-useragent.patch uploaded
+ by Guggemand (license 897) Tested by: Guggemand ........
+ ................
+
+ * main/channel.c, /: Merged revisions 273058 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r273058 | tilghman | 2010-06-29 17:59:51 -0500 (Tue, 29 Jun 2010)
+ | 11 lines Recorded merge of revisions 273057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010)
+ | 4 lines _Really_ skip the channel... don't just retry for
+ another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........
+ ................
+
+ * main/pbx.c, /: Merged revisions 273054 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r273054 |
+ tilghman | 2010-06-29 17:39:22 -0500 (Tue, 29 Jun 2010) | 11
+ lines Send DialPlanComplete as a response, not as a separate
+ event. Otherwise, it goes to all manager sessions and may exclude
+ the current session, if the Events mask excludes it. (closes
+ issue #17504) Reported by: rrb3942 Patches:
+ showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested
+ by: rrb3942 ........
+
+2010-06-29 16:43 +0000 [r272972] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 253357 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253357 |
+ russell | 2010-03-18 13:18:43 -0500 (Thu, 18 Mar 2010) | 8 lines
+ Increase CLI command output timeout for asterisk -rx to 60
+ seconds. (closes issue #17049) Reported by: russell Tested by:
+ russell Review: https://reviewboard.asterisk.org/r/573/ ........
+
+2010-07-22 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.10
+
+ * Included a fix for res_timing_pthread per the description below:
+
+ r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines
+
+ Use poll() instead of select() in res_timing_pthread to avoid stack corruption.
+ This code did not properly check FD_SETSIZE to ensure that it did not try to
+ select() on fds that were too large. Switching to poll() removes the limitation
+ on the maximum fd value.
+
+2010-07-07 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.10-rc2
+
+ * Fix problem with RFC 2833 DTMF not being accepted.
+
+ A recent check was added to ensure that we did not erroneously
+ detect duplicate DTMF when we received packets out of order.
+ The problem was that the check did not account for the fact that
+ the seqno of an RTP stream will roll over back to 0 after hitting
+ 65535. Now, we have a secondary check that will ensure that the
+ seqno rolling over will not cause us to stop accepting DTMF.
+
+ (closes issue 0017571)
+ Reported by: mdeneen
+ Patches:
+ rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
+ Tested by: richardf, maxochoa, JJCinAZ
+
+ * Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx
+ via the G option in app_dial
+
+ (closes issue 0017592)
+ Reported by: jamicque
+ Patches:
+ G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: jamicque, mnicholson
+
+2010-06-29 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.10-rc1
+
+2010-06-28 21:51 +0000 [r272924-272927] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 272926 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272926 | tilghman | 2010-06-28 16:50:57 -0500 (Mon, 28 Jun 2010)
+ | 15 lines Merged revisions 272925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010)
+ | 8 lines Don't change ownership/group/permissions on run
+ directory, if it already exists. (closes issue #17076) Reported
+ by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth ........
+ ................
+
+ * main/config.c, /: Merged revisions 272923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272923 | tilghman | 2010-06-28 16:42:52 -0500 (Mon, 28 Jun 2010)
+ | 19 lines Merged revisions 272921-272922 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010)
+ | 8 lines Change the way that we read include files, to
+ accommodate for changes in GCC 4.4. (closes issue #17472)
+ Reported by: seandarcy Patches: config2.patch uploaded by nivan
+ (license 1066) Tested by: nivan ........ r272922 | tilghman |
+ 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim
+ trailing blanks on #includes ........ ................
+
+2010-06-28 18:50 +0000 [r272882] Russell Bryant <russell@digium.com>
+
+ * tests/test_astobj2.c (added): Backport applicable parts of
+ test_astobj2 from trunk.
+
+2010-06-28 17:37 +0000 [r272806] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 272805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272805 | mmichelson | 2010-06-28 12:33:12 -0500 (Mon, 28 Jun
+ 2010) | 11 lines Merged revisions 272804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun
+ 2010) | 5 lines Decode URI in contact header of 302 response.
+ ABE-2352 ........ ................
+
+2010-06-28 15:36 +0000 [r272685-272686] Russell Bryant <russell@digium.com>
+
+ * doc/tex/chan-mobile.tex (removed): remove accidentally added
+ file.
+
+ * doc/tex/cdrdriver.tex, doc/tex/asterisk.tex, /,
+ doc/tex/chan-mobile.tex (added): Merged revisions 272684 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r272684 | russell | 2010-06-28 10:33:32 -0500 (Mon, 28
+ Jun 2010) | 2 lines Use the underscore package so that
+ underscores do not need to be escaped. ........
+
+2010-06-25 20:20 +0000 [r272556-272577] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272568 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r272568 | tilghman | 2010-06-25 15:18:47 -0500
+ (Fri, 25 Jun 2010) | 12 lines Merged revisions 272562 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010)
+ | 5 lines Make the structure of the table specified before match
+ the queries and results. (closes issue #17557) Reported by: cmaj
+ ........ ................
+
+ * sounds/Makefile, /: Merged revisions 272533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272533 |
+ tilghman | 2010-06-25 14:17:16 -0500 (Fri, 25 Jun 2010) | 2 lines
+ Symlink sounds files, to save disk space, when multiple
+ tarballs/checkouts are on the same system. ........
+
+2010-06-25 18:58 +0000 [r272531] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/_private.h, tests/test_sched.c, main/asterisk.c,
+ include/asterisk/test.h (added), build_tools/cflags-devmode.xml,
+ tests/test_heap.c, tests/test_skel.c, main/Makefile, main/test.c
+ (added): Backport unit test API from trunk. Also, update existing
+ test modules that were already in this branch but had been
+ converted to the unit test API in trunk. Review:
+ https://reviewboard.asterisk.org/r/748/
+
+2010-06-24 22:19 +0000 [r272459] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 272447 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r272447 | rmudgett | 2010-06-24 17:11:26 -0500
+ (Thu, 24 Jun 2010) | 17 lines Merged revisions 272446 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010)
+ | 10 lines ss_thread calls pri_grab without lock during overlap
+ dial Recent changes to chan_dahdi with relation to overlap
+ dialing call pri_grab without first obtaining a lock. (closes
+ issue #17414) Reported by: pdf Patches: bug17414.patch uploaded
+ by jpeeler (license 325) ........ ................
+
+2010-06-23 23:40 +0000 [r272440] Terry Wilson <twilson@digium.com>
+
+ * autoconf/ast_ext_tool_check.m4, /, configure: Merged revisions
+ 272254,272256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272254 |
+ twilson | 2010-06-23 15:53:48 -0500 (Wed, 23 Jun 2010) | 10 lines
+ Honor the --with-${library}=path for AST_EXT_TOOL_CHECK (closes
+ issue #16991) Reported by: pprindeville Patches:
+ with_netsnmp.patch.txt uploaded by twilson (license 396) Tested
+ by: twilson Review: https://reviewboard.asterisk.org/r/739/
+ ........ r272256 | twilson | 2010-06-23 15:59:17 -0500 (Wed, 23
+ Jun 2010) | 2 lines Update configure when changing autconf m4
+ files... ........
+
+2010-06-23 23:14 +0000 [r272371] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 272370 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r272370 | russell | 2010-06-23 18:09:28 -0500 (Wed, 23 Jun 2010)
+ | 23 lines Resolve some errors produced during module unload of
+ chan_iax2. The external test suite stops Asterisk using the "core
+ stop gracefully" command. The logs from the tests show that there
+ are a number of problems with Asterisk trying to cleanly shut
+ down. This patch addresses the following type of error that comes
+ from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129
+ __ast_pthread_mutex_destroy: chan_iax2.c line 11371
+ (iax2_process_thread_cleanup): Error destroying mutex
+ &thread->lock: Device or resource busy For an example in the
+ context of a build, see:
+ http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary
+ purpose of this patch is to change the thread pool shutdown
+ procedure to be more explicit to ensure that the thread exits
+ from a point where it is not holding a lock. While testing that,
+ I encountered various crashes due to the order of operations in
+ unload_module() being problematic. I reordered some things there,
+ as well. Review: https://reviewboard.asterisk.org/r/736/ ........
+
+2010-06-23 22:37 +0000 [r272369] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 272368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272368 | mnicholson | 2010-06-23 17:36:49 -0500 (Wed, 23 Jun
+ 2010) | 16 lines Merged revisions 272367 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 This version
+ of the patch only adds AgentComplete for attended transfers. It
+ was already present for blind transfers. ........ r272367 |
+ mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8
+ lines Send AgentComplete manager events in the event of blind and
+ attended transfers. (closes issue #16819) Reported by: elbriga
+ Patches: app_queue.diff uploaded by elbriga (license 482)
+ ........ ................
+
+2010-06-23 21:54 +0000 [r272333] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 272332 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r272332 | tilghman | 2010-06-23 16:53:49 -0500 (Wed, 23 Jun 2010)
+ | 8 lines If there is realtime configuration, it does not get
+ re-read on reload unless the config file also changes. (closes
+ issue #16982) Reported by: dmitri Patches: res_musiconhold.patch
+ uploaded by dmitri (license 1001) Tested by: atis ........
+
+2010-06-23 21:15 +0000 [r272263] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_meetme.c: Revert previous commit, ast_test_flag64 does
+ not exist in 1.6.2
+
+2010-06-23 21:09 +0000 [r272262] Tilghman Lesher <tlesher@digium.com>
+
+ * res/ael/ael.flex, /, res/ael/ael.tab.c, res/ael/ael.y,
+ res/ael/ael_lex.c: Merged revisions 272260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272260 |
+ tilghman | 2010-06-23 16:06:40 -0500 (Wed, 23 Jun 2010) | 8 lines
+ Ensure a NULL file while debugging cannot crash AEL. (closes
+ issue #17215) Reported by: vazir Patches:
+ 20100518__issue17215.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+2010-06-23 21:07 +0000 [r272253-272261] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_meetme.c: Merged revisions 272259 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272259 |
+ pabelanger | 2010-06-23 17:06:15 -0400 (Wed, 23 Jun 2010) | 2
+ lines Fix previous merge. ast_test_flag != ast_test_flag64
+ ........
+
+ * /, apps/app_meetme.c: Merged revisions 272257 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272257 | pabelanger | 2010-06-23 17:00:00 -0400 (Wed, 23 Jun
+ 2010) | 19 lines Merged revisions 272255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun
+ 2010) | 12 lines First caller into a dynamic conference now enter
+ pin once. If MeetMe is configured to use dynamic conference
+ numbers, then the first caller (which creates the conference) had
+ to enter the PIN number twice. (closes issue #15878) Reported by:
+ shawkris Patches: issue15878.patch uploaded by pabelanger
+ (license 224) Tested by: pabelanger ........ ................
+
+ * main/manager.c, /: Merged revisions 272252 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272252 |
+ pabelanger | 2010-06-23 16:35:45 -0400 (Wed, 23 Jun 2010) | 8
+ lines Correct manager variable 'EventList' case. (closes issue
+ #17520) Reported by: kobaz Patches: manager.patch uploaded by
+ kobaz (license 834) Tested by: lmadsen ........
+
+2010-06-23 18:41 +0000 [r272124-272149] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 272146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272146 |
+ twilson | 2010-06-23 13:39:20 -0500 (Wed, 23 Jun 2010) | 2 lines
+ Don't start the sla thread unless we realy need it ........
+
+ * /, apps/app_meetme.c: Merged revisions 272109 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272109 |
+ twilson | 2010-06-23 12:21:40 -0500 (Wed, 23 Jun 2010) | 12 lines
+ Make sure reload updates SLA config Even if there are no stations
+ or trunks defined, we need to start the sla thread to make sure
+ we get the reload event. Also, when doing a reload we need to
+ remove the existing trunks and stations or they end up hanging
+ around. (closes issue #16818) Reported by: mbonin Patches:
+ sla_reload.patch uploaded by twilson (license 396) Tested by:
+ twilson ........
+
+2010-06-22 22:14 +0000 [r272015] David Vossel <dvossel@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 272014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272014 |
+ dvossel | 2010-06-22 17:11:50 -0500 (Tue, 22 Jun 2010) | 5 lines
+ fixes issue with 'dialplan remove extension blah' segfaulting
+ with tab completion (closes issue #17440) Reported by: kobaz
+ ........
+
+2010-06-22 17:37 +0000 [r271904] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 271903 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun
+ 2010) | 15 lines Merged revisions 271902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun
+ 2010) | 8 lines Decrease the module ref count in sip_hangup when
+ SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the
+ ref count correct. (closes issue #16815) Reported by: rain
+ Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
+ (modified) Tested by: rain ........ ................
+
+2010-06-22 16:30 +0000 [r271869] Russell Bryant <russell@digium.com>
+
+ * /, res/ais/clm.c, res/ais/evt.c: Merged revisions 271867 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r271867 | russell | 2010-06-22 11:28:03 -0500 (Tue, 22
+ Jun 2010) | 7 lines Resolve some errors that occur on a graceful
+ shutdown. Don't Finalize() if Initialize() did not succeed. This
+ resulted in an error about trying to Finalize() an invalid
+ handle. Also trim some trailing whitespace while in the area.
+ ........
+
+2010-06-22 15:49 +0000 [r271832] David Vossel <dvossel@digium.com>
+
+ * /, main/features.c: Merged revisions 271831 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r271831 |
+ dvossel | 2010-06-22 10:46:22 -0500 (Tue, 22 Jun 2010) | 10 lines
+ fixes attended transfer behavior when both transferee and
+ transferer hung up If both the transferer and transferee of a
+ attended transfer hangup before the new channel picks up, the new
+ channel should be hung up as well as it has no endpoint to talk
+ to. This mirrors the expected behavior used in 1.4. (closes issue
+ #17444) Reported by: corruptor ........
+
+2010-06-22 15:00 +0000 [r271691-271763] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions
+ 271762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271762 | mnicholson | 2010-06-22 09:54:58 -0500 (Tue, 22 Jun
+ 2010) | 15 lines Merged revisions 271761 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun
+ 2010) | 9 lines Allow users to specify a port for dundi peers.
+ (closes issue #17056) Reported by: klaus3000 Patches:
+ dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
+ Tested by: klaus3000 ........ ................
+
+ * include/asterisk/strings.h, configs/sip_notify.conf.sample, /,
+ channels/chan_sip.c: Merged revisions 271690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun
+ 2010) | 18 lines Merged revisions 271689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun
+ 2010) | 8 lines Modify chan_sip's packet generation api to
+ automatically calculate the Content-Length. This is done by
+ storing packet content in a buffer until it is actually time to
+ send the packet, at which time the size of the packet is
+ calculated. This change was made to ensure that the
+ Content-Length is always correct. (closes issue #17326) Reported
+ by: kenner Tested by: mnicholson, kenner Review:
+ https://reviewboard.asterisk.org/r/693/ ........ This change also
+ adds an ast_str_copy_string() function (similar to
+ ast_copy_string), that copies one ast_str into another, properly
+ handling embedded nulls. ................
+
+2010-06-21 20:48 +0000 [r271555] Jeff Peeler <jpeeler@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 271554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271554 | jpeeler | 2010-06-21 15:46:53 -0500 (Mon, 21 Jun 2010)
+ | 14 lines Merged revisions 271552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010)
+ | 7 lines Do not use sizeof to calculate size of a heap allocated
+ character array. Change left out from 271399. (closes issue
+ #16053) Reported by: diLLec ........ ................
+
+2010-06-18 21:33 +0000 [r271338-271484] Jeff Peeler <jpeeler@digium.com>
+
+ * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged
+ revisions 271483 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010)
+ | 18 lines Merged revisions 271399 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010)
+ | 11 lines Fix crash when parsing some heavily nested statements
+ in AEL on reload. Due to the recursion used when compiling AEL in
+ gen_prios, all the stack space was being consumed when parsing
+ some AEL that contained nesting 13 levels deep. Changing a few
+ large buffers to be heap allocated fixed the crash, although I
+ did not test how many more levels can now be safely used. (closes
+ issue #16053) Reported by: diLLec Tested by: jpeeler ........
+ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 269307 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010)
+ | 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling
+ dahdi_indicate() within dahdi_fixup() while the owner pointers
+ are in a potentially inconsistent state is a potentially bad
+ thing in principle. However, calling dahdi_indicate() when the
+ channel private lock is already held can cause a deadlock if the
+ PRI lock is needed because dahdi_indicate() will also get the
+ channel private lock. The pri_grab() function assumes that the
+ channel private lock is held once to avoid deadlock. ........
+
+2010-06-17 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.9 Released.
+
+2010-06-10 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.9-rc3 Released.
+
+2010-06-10 Tilghman Lesher <tlesher@digium.com>
+
+ * Ensure signals are not blocked inside other signal handlers.
+
+ This eliminates the annoying <beep> on the console.
+
+ (closes issue 0017477)
+ Reported by: jvandal
+ Patches:
+ 20100610__issue17477.diff.txt uploaded by tilghman (license 14
+
+2010-06-09 Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Fix Debian init script to not use -c.
+
+ When using the init script as-is currently, it could cause issues on Debian
+ such as high CPU usage. This fix has worked for several people so I'm
+ implementing the change. We now handle color displays properly.
+
+ (closes issue 0016784)
+ Reported by: pabelanger
+ Patches:
+ 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: pabelanger, tilghman
+
+2010-06-07 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.9-rc2 Released.
+
+2010-06-07 Tilghman Lesher <tlesher@digium.com>
+
+ * Fix crash in DTMF detection.
+
+ What I did not originally see in my previous commit was that even
+ though the next digit could be detected before the previous was
+ considered ended, the detection of the next digit effectively ends
+ the detection of the previous. Therefore, the length moves in
+ lockstep with the digit, and no separate counter is needed for the
+ length alone.
+
+ (closes issue 0017371)
+ Reported by: alecdavis
+
+ (closes issue 0017474)
+ Reported by: kenner
+
+2010-06-01 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.9-rc1 Released.
+
+2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 266592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010)
+ | 18 lines Merged revisions 266585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
+ | 11 lines Prevent CLI prompt from distorting output of lines
+ shorter than the prompt. Uses the VT100 method of clearing the
+ line from the cursor position to the end of the line: Esc-0K
+ (closes issue #17160) Reported by: coolmig Patches:
+ 20100531__issue17160.diff.txt uploaded by tilghman (license 14)
+ Tested by: coolmig ........ ................
+
+2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_agi.c: Fix typo in documentation (closes issue #17395)
+ Reported by: pabelanger Patches: res_agi.c.patch uploaded by
+ pabelanger (license 224)
+
+2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r266438 | tilghman | 2010-05-29 23:44:28 -0500
+ (Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29
+ May 2010) | 2 lines Reverting patch and reopening issue #16784,
+ as patch breaks color display. ........ ................
+
+2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 266337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 |
+ tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line
+ Only report swap on platforms which can examine those statistics
+ ........
+
+2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 |
+ dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines
+ fixes crash when creation of UDPTL fails (closes issue #17264)
+ Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff
+ uploaded by dvossel (license 671)
+ issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
+ (license 671) Tested by: falves11 ........
+
+2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged
+ revisions 266146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010)
+ | 21 lines Merged revisions 266142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
+ | 14 lines Use sigaction for signals which should persist past
+ the initial trigger, not signal. If you call signal() in a
+ Solaris signal handler, instead of just resetting the signal
+ handler, it causes the signal to refire, because the signal is
+ not marked as handled prior to the signal handler being called.
+ This effectively causes Solaris to immediately exceed the
+ threadstack in recursive signal handlers and crash. (closes issue
+ #17000) Reported by: rmcgilvr Patches:
+ 20100526__issue17000.diff.txt uploaded by tilghman (license 14)
+ Tested by: rmcgilvr ........ ................
+
+2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 |
+ dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines
+ fixes failed SIP Directed pickup resulting in dead channel
+ (closes issue #17339) Reported by: one47 Patches:
+ sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
+ one47, dvossel ........
+
+2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r265923 | tilghman | 2010-05-26 11:23:28 -0500
+ (Wed, 26 May 2010) | 14 lines Merged revisions 265910 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010)
+ | 7 lines Not finding rows in the DB does not rise to the level
+ of a warning. (closes issue #17062) Reported by: drookie Patches:
+ 20100525__issue17062.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+ * configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged
+ revisions 265894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 |
+ tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines
+ Construct socket name, according to the Postgres docs, and
+ document as such. (closes issue #17392) Reported by: dps Patches:
+ 20100525__issue17392.diff.txt uploaded by tilghman (license 14)
+ Tested by: dps ........
+
+2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Recorded merge of revisions 265842 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed,
+ 26 May 2010) | 9 lines Re-enable "always" option for videosupport
+ option in sip.conf. (closes issue #17016) Reported by: twilson
+ Patches: 17016.patch uploaded by mmichelson (license 60) Tested
+ by: devmod ........
+
+2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 |
+ tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines
+ Use configure to determine the prefixes and include directories
+ properly. This ensures cross-platform compatibility, even among
+ Linux distributions, which don't always put headers in the same
+ place. (closes issue #17391) Reported by: loloski ........
+
+2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 |
+ mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12
+ lines Properly use peer's outboundproxy for outbound REGISTERs.
+ The logic used in transmit_register to get the outboundproxy for
+ a peer was flawed since this value would be overridden shortly
+ afterwards when create_addr was called. In addition, this also
+ fixes some logic used when parsing users.conf so that the peer
+ name is placed in the internally-generated register string so
+ that an outboundproxy set in the Asterisk GUI will be used for
+ outbound REGISTERs. ........
+
+2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel@digium.com>
+
+ * channels/chan_dahdi.c: fixes build issue with zaptel (closes
+ issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff
+ uploaded by dvossel (license 671) Tested by: aragon
+
+2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 265611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May
+ 2010) | 15 lines Merged revisions 265610 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
+ 2010) | 8 lines Don't mark the cdr records of unanswered queue
+ calls with "NOANSWER". This restores the behavior prior to
+ r258670. (closes issue #17334) Reported by: jvandal Patches:
+ queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
+ by: aragon, jvandal ........ ................
+
+2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, Makefile,
+ doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged
+ revisions 265320,265467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 |
+ twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
+ Add the FullyBooted AMI event It is possible to connect to the
+ manager interface before all Asterisk modules are loaded. To
+ ensure that an application does not send AMI actions that might
+ require a module that has not yet loaded, the application can
+ listen for the FullyBooted manager event. It will be sent upon
+ connection if all modules have been loaded, or as soon as loading
+ is complete. The event: Event: FullyBooted Privilege: system,all
+ Status: Fully Booted Review:
+ https://reviewboard.asterisk.org/r/639/ ........ r265467 |
+ twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
+ Merge the rest of the FullyBooted patch ........
+
+2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/h323/ast_h323.cxx: Merged revisions 265451 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon,
+ 24 May 2010) | 8 lines Print openh323 log to the Asterisk
+ console. (closes issue #17109) Reported by: under Patches:
+ logstream.diff uploaded by under (license 914) ........
+
+ * /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 |
+ mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11
+ lines Allow type=user SIP endpoints to be loaded properly from
+ realtime. (closes issue #16021) Reported by: Guggemand Patches:
+ realtime-type-fix.patch uploaded by Guggemand (license 897)
+ (altered by me slightly to avoid ref leaks) Tested by: Guggemand
+ ........
+
+2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 265273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 |
+ dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines
+ fixes segfault when using generic plc ........
+
+2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 265316 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 |
+ tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines
+ On systems with a LOT of RAM, a signed integer sometimes printed
+ negative. (closes issue #16837) Reported by: jlpedrosa Patches:
+ 20100504__issue16837.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix memory hogging behavior of app_queue. From
+ reviewboard: This review request is for the patch on issue 17081.
+ A user reported that he saw increasing numbers of allocations
+ stemming from app_queue.c when he would run the "queue show" CLI
+ command. The user reported that he was using approximately 40
+ realtime queues and as he ran the CLI command more and more, the
+ memory usage would shoot up. As it turns out, there was a memory
+ leak and a separate usage of memory that, while not really a
+ leak, was very irresponsible. Both memory problems can be
+ attributed to the function init_queue(). When the "queue show"
+ command is run, all realtime queues have the init_queue()
+ function called on the in-memory queue. The idea is to place the
+ queue in its default state and then overwrite options specified
+ in the realtime backend as we read them. The first problem, the
+ memory leak, had to do with the fact that the string field for
+ the name of the first periodic announcement file was being
+ re-created every time init_queue was called. This patch corrects
+ the behavior by only calling ast_str_create if the memory has not
+ already been allocated. The other problem is a bit more
+ complicated. The majority of the strings in the call_queue
+ structure were changed to use the ast_string_fields API for 1.6.0
+ and beyond. init_queue resets all string fields on the queue to
+ their default values. Then, later in the realtime queue loading
+ process, these string fields are set to their configured values.
+ For those unfamiliar with string fields, frequent resizing of a
+ string like this is not what the string fields API is designed
+ for. The result of this constant resizing is that as the queue
+ gets loaded, eventually space for the string runs out and so a
+ new memory pool, at twice the size of the previously allocated
+ one, is created for the string fields. The reporter of issue
+ 17081 wrote a script that ran the "queue show" CLI command 2100
+ times. By the end, each of his 40 queues was taking about a
+ megabyte of memory apiece just for their string fields. My fix
+ for this problem is to revert the call_queue structure from using
+ string fields. In my patch here, I have moved the queue back to
+ using fixed-sized buffers. I ran the script provided by the
+ reporter of 17081 and determined that I no longer saw the
+ steadily-increasing memory usage that I had seen before applying
+ the patch. (closes issue #17081) Reported by: wliegel Patches:
+ 17081v2.patch uploaded by mmichelson (license 60) Tested by:
+ wliegel, mmichelson Review:
+ https://reviewboard.asterisk.org/r/651/
+
+ * apps/app_queue.c, include/asterisk/file.h, /: Merged revisions
+ 265090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May
+ 2010) | 15 lines Merged revisions 265089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
+ 2010) | 8 lines Don't hang up on a queue caller if the file we
+ attempt to play does not exist. This also fixes a documentation
+ mistake in file.h that made my original attempt to correct this
+ problem not work correctly. (closes issue #17061) Reported by:
+ RoadKill ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 |
+ mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7
+ lines Be sure to set the sin_family on the proxy when allocating.
+ (closes issue #17157) Reported by: stuarth ........
+
+ * /, include/asterisk/channel.h: Merged revisions 265000 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500
+ (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri,
+ 21 May 2010) | 3 lines Fix grammatical error in comment. ........
+ ................
+
+ * main/channel.c, main/autoservice.c, /,
+ include/asterisk/channel.h: Merged revisions 264997 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500
+ (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May
+ 2010) | 32 lines Allow ast_safe_sleep to defer specific frames
+ until after the sleep has concluded. From reviewboard Background:
+ A Digium customer discovered a somewhat odd bug. The setup is
+ that parties A and B are bridged, and party A places party B on
+ hold. While party B is listening to hold music, he mashes a bunch
+ of DTMF. Party A takes party B off hold while this is happening,
+ but party B continues to hear hold music. I could reproduce this
+ about 1 in 5 times. The issue: When DTMF features are enabled and
+ a user presses keys, the channel that the DTMF is streamed to is
+ placed in an ast_safe_sleep for 100 ms, the duration of the
+ emulated tone. If an AST_CONTROL_UNHOLD frame is read from the
+ channel during the sleep, the frame is dropped. Thus the unhold
+ indication is never made to the channel that was originally
+ placed on hold. The fix: Originally, I discussed with Kevin
+ possible ways of fixing the specific problem reported. However,
+ we determined that the same type of problem could happen in other
+ situations where ast_safe_sleep() is used. Using autoservice as a
+ model, I modified ast_safe_sleep_conditional() to defer specific
+ frame types so they can be re-queued once the sleep has finished.
+ I made a common function for determining if a frame should be
+ deferred so that there are not two identical switch blocks to
+ maintain. Review: https://reviewboard.asterisk.org/r/674/
+ ........ ................
+
+2010-05-20 23:34 +0000 [r264829] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/callerid.c: Merged revisions 264828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010)
+ | 13 lines Merged revisions 264820 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010)
+ | 6 lines ast_callerid_parse() had a path that left name
+ uninitialized. Several callers of ast_callerid_parse() do not
+ initialize the name parameter before calling thus there is the
+ potential to use an uninitialized pointer. ........
+ ................
+
+2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 264779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 |
+ tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines
+ Let ExtensionState resolve dynamic hints. (closes issue #16623)
+ Reported by: tilghman Patches: 20100116__issue16623.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen ........
+
+ * apps/app_stack.c, /: Merged revisions 264752 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 |
+ tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines
+ Error message fix. (closes issue #17356) Reported by: kenner
+ Patches: app_stack.c.diff uploaded by kenner (license 1040)
+ ........
+
+2010-05-19 22:10 +0000 [r264453] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/_private.h, include/asterisk/options.h,
+ main/asterisk.c, main/loader.c, main/channel.c, /,
+ channels/chan_sip.c: Merged revisions 264452 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 |
+ mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86
+ lines Fix transcode_via_sln option with SIP calls and improve PLC
+ usage. From reviewboard: The problem here is a bit complex, so
+ try to bear with me... It was noticed by a Digium customer that
+ generic PLC (as configured in codecs.conf) did not appear to
+ actually be having any sort of benefit when packet loss was
+ introduced on an RTP stream. I reproduced this issue myself by
+ streaming a file across an RTP stream and dropping approx. 5% of
+ the RTP packets. I saw no real difference between when PLC was
+ enabled or disabled when using wireshark to analyze the RTP
+ streams. After analyzing what was going on, it became clear that
+ one of the problems faced was that when running my tests, the
+ translation paths were being set up in such a way that PLC could
+ not possibly work as expected. To illustrate, if packets are lost
+ on channel A's read stream, then we expect that PLC will be
+ applied to channel B's write stream. The problem is that generic
+ PLC can only be done when there is a translation path that moves
+ from some codec to SLINEAR. When I would run my tests, I found
+ that every single time, read and write translation paths would be
+ set up on channel A instead of channel B. There appeared to be no
+ real way to predict which channel the translation paths would be
+ set up on. This is where Kevin swooped in to let me know about
+ the transcode_via_sln option in asterisk.conf. It is supposed to
+ work by placing a read translation path on both channels from the
+ channel's rawreadformat to SLINEAR. It also will place a write
+ translation path on both channels from SLINEAR to the channel's
+ rawwriteformat. Using this option allows one to predictably set
+ up translation paths on all channels. There are two problems with
+ this, though. First and foremost, the transcode_via_sln option
+ did not appear to be working properly when I was placing a SIP
+ call between two endpoints which did not share any common
+ formats. Second, even if this option were to work, for PLC to be
+ applied, there had to be a write translation path that would go
+ from some format to SLINEAR. It would not work properly if the
+ starting format of translation was SLINEAR. The one-line change
+ presented in this review request in chan_sip.c fixed the first
+ issue for me. The problem was that in sip_request_call, the
+ jointcapability of the outbound channel was being set to the
+ format passed to sip_request_call. This is nativeformats of the
+ inbound channel. Because of this, when
+ ast_channel_make_compatible was called by app_dial, both channels
+ already had compatibly read and write formats. Thus, no
+ translation path was set up at the time. My change is to set the
+ jointcapability of the sip_pvt created during sip_request_call to
+ the intersection of the inbound channel's nativeformats and the
+ configured peer capability that we determined during the earlier
+ call to create_addr. Doing this got the translation paths set up
+ as expected when using transcode_via_sln. The changes presented
+ in channel.c fixed the second issue for me. First and foremost,
+ when Asterisk is started, we'll read codecs.conf to see the value
+ of the genericplc option. If this option is set, and ast_write is
+ called for a frame with no data, then we will attempt to fill in
+ the missing samples for the frame. The implementation uses a
+ channel datastore for maintaining the PLC state and for creating
+ a buffer to store PLC samples in. Even when we receive a frame
+ with data, we'll call plc_rx so that the PLC state will have
+ knowledge of the previous voice frame, which it can use as a
+ basis for when it comes time to actually do a PLC fill-in. So,
+ reviewers, now I ask for your help. First off, there's the one
+ line change in chan_sip that I have put in. Is it right? By my
+ logic it seems correct, but I'm sure someone can tell me why it
+ is not going to work. This is probably the change I'm least
+ concerned about, though. What concerns me much more is the set of
+ changes in channel.c. First off, am I even doing it right? When I
+ run tests, I can clearly see that when PLC is activated, I see a
+ significant increase in RTP traffic where I would expect it to
+ be. However, in my humble opinion, the audio sounds kind of
+ crappy whenever the PLC fill-in is done. It sounds worse to me
+ than when no PLC is used at all. I need someone to review the
+ logic I have used to be sure that I'm not misusing anything. As
+ far as I can see my pointer arithmetic is correct, and my use of
+ AST_FRIENDLY_OFFSET should be correct as well, but I'm sure
+ someone can point out somewhere where I've done something
+ incorrectly. As I was writing this review request up, I decided
+ to give the code a test run under valgrind, and I find that for
+ some reason, calls to plc_rx are causing some invalid reads.
+ Apparently I'm reading past the end of a buffer somehow. I'll
+ have to dig around a bit to see why that is the case. If it's
+ obvious to someone reviewing, speak up! Finally, I have one other
+ proposal that is not reflected in my code review. Since without
+ transcode_via_sln set, one cannot predict or control where a
+ translation path will be up, it seems to me that the current
+ practice of using PLC only when transcoding to SLINEAR is not
+ useful. I recommend that once it has been determined that the
+ method used in this code review is correct and works as expected,
+ then the code in translate.c that invokes PLC should be removed.
+ Review: https://reviewboard.asterisk.org/r/622/ ........
+
+2010-05-19 20:31 +0000 [r264405] David Vossel <dvossel@digium.com>
+
+ * main/udptl.c, /: Merged revisions 264400 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 |
+ dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines
+ fixes infinite loop during udptl.c's decode_open_type When
+ decode_length returns the length there is a check to see if that
+ length is negative, if so the decode loop breaks as this means
+ the limit has been reached. The problem here is that length is an
+ unsigned int, so length can never be negative. This resulted in
+ an infinite loop. (issue #17352) ........
+
+2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 264379 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 |
+ mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4
+ lines Cast an unsigned int to a signed int when comparing it with
+ 0. (AST-377) ........
+
+ * apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500
+ (Wed, 19 May 2010) | 12 lines Merged revisions 264334 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May
+ 2010) | 5 lines Set quieted flag when receiving a dtmf tone
+ during playback in speechbackground. (closes issue #16966)
+ Reported by: asackheim ........ ................
+
+2010-05-19 19:25 +0000 [r264332] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 |
+ dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines
+ fixes crash in check_rtp_timeout During deadlock avoidance the
+ sip dialog pvt is locked and unlocked. When this occurs we have
+ no guarantee the pvt's owner is still valid. We were trying to
+ access the pvt's owner after this without checking to see if it
+ still existed first. (closes issue #17271) Reported by: under
+ Patches: check_rtp_timeout.diff uploaded by under (license 914)
+ Tested by: dvossel ........
+
+2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/options.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 264249 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010)
+ | 24 lines Merged revisions 264248 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010)
+ | 17 lines Internal timing is now on by default, if you're using
+ DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is
+ that this version ensures that a timer is always available,
+ whereas in previous versions, it was possible for DAHDI to be
+ loaded, but have no drivers to actually generate timing. If
+ internal_timing was turned on in this circumstance, a complete
+ lack of audio would result. This is the reason why
+ internal_timing was not on by default. However, now that DAHDI
+ ensures the availability of a timer, there is no reason for this
+ setting to be off (and in fact, it solves a great many initial
+ user problems). (closes issue #15932) Reported by: dimas Patches:
+ 20100519__issue15932.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........ ................
+
+ * main/dsp.c, /: Merged revisions 264204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 |
+ tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines
+ Keep track of digit duration, when we're decoding inband to pass
+ DTMF frames. (closes issue #17235) Reported by: frawd Patches:
+ new_dtmf_dsp_len.patch uploaded by frawd (license 610)
+ 20100518__issue17235.diff.txt uploaded by tilghman (license 14)
+ Tested by: frawd ........
+
+2010-05-19 14:47 +0000 [r264115] David Vossel <dvossel@digium.com>
+
+ * main/rtp.c, /: Merged revisions 264114 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 |
+ dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines
+ fixes crash during dtmf During the processing of Cisco dtmf the
+ dtmf samples were not being calculated correctly. In an attempt
+ to determine what sample rate was being used, a NULL frame was
+ processed which caused a crash. This patch resolves this. (closes
+ issue #17248) Reported by: falves11 Patches: issue_17248.diff
+ uploaded by dvossel (license 671) ........
+
+2010-05-19 08:15 +0000 [r264032] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, configs/indications.conf.sample: Merged revisions 264031 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19
+ May 2010) | 8 lines fix incorrectly typed indications for [nz]
+ stutter and dialrecall (closes issue #17359) Reported by:
+ alecdavis Patches: bug17359.diff.txt uploaded by alecdavis
+ (license 585) ........
+
+2010-05-19 06:41 +0000 [r263951] Tilghman Lesher <tlesher@digium.com>
+
+ * main/dsp.c, /: Merged revisions 263950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010)
+ | 15 lines Merged revisions 263949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010)
+ | 8 lines Because progress is called multiple times, across
+ several frames, we must persist states when detecting multitone
+ sequences. (closes issue #16749) Reported by: dant Patches:
+ dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
+ dant ........ ................
+
+2010-05-18 22:49 +0000 [r263906] David Vossel <dvossel@digium.com>
+
+ * main/strings.c, /: Merged revisions 263904 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 |
+ dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines
+ fixes segfault on logging (closes issue #17331) Reported by:
+ under Patches: utils.diff uploaded by under (license 914)
+ segfault_on_logging.diff uploaded by dvossel (license 671) Tested
+ by: under, dvossel ........
+
+2010-05-18 19:41 +0000 [r263809] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_directory.c, /: Merged revisions 263807 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500
+ (Tue, 18 May 2010) | 17 lines Merged revisions 263769 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010)
+ | 10 lines Modify directory name reading to be interrupted with
+ operator or pound escape. In the case of accidentally entering
+ the wrong first three letters for the reading, users could be
+ very frustrated if the name listing is very long. This allows
+ interrupting the reading by pressing 0 or #. 0 will attempt to
+ execute a configured operator (o) extension and # will exit and
+ proceed in the dialplan. ABE-2200 ........ ................
+
+2010-05-17 22:10 +0000 [r263642] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/devicestate.c: Merged revisions 263640 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May
+ 2010) | 16 lines Merged revisions 263639 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May
+ 2010) | 10 lines Fix logic error when checking for a devstate
+ provider. When using strsep, if one of the list of specified
+ separators is not found, it is the first parameter to strsep
+ which is now NULL, not the pointer returned by strsep. This issue
+ isn't especially severe in that the worst it is likely to do is
+ waste some cycles when a device with no '/' and no ':' is passed
+ to ast_device_state. ........ ................
+
+2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010)
+ | 9 lines With IMAP backend, messages in INBOX were counted twice
+ for MWI. (closes issue #17135) Reported by: edhorton Patches:
+ 20100513__issue17135.diff.txt uploaded by tilghman (license 14)
+ 17135_2.diff uploaded by ebroad (license 878) Tested by:
+ edhorton, ebroad ........
+
+ * main/app.c: Don't close 'n', just close 'above_n'. (closes issue
+ #17345) Reported by: wdoekes
+
+2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen <lmadsen@digium.com>
+
+ * main/manager.c, /: Merged revisions 263457 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010)
+ | 19 lines Recorded merge of revisions 263456 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010)
+ | 11 lines Manager cookies are not compatible with RFC2109. The
+ Version field in the cookies we're setting contain quotes around
+ the version number which is not compatible with RFC2109 and
+ breaks some implementations. (closes issue #17231) Reported by:
+ ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
+ ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
+ ecarruda (license 559) Tested by: ecarruda, russell ........
+ ................
+
+ * sounds/Makefile, /: Merged revisions 263375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010)
+ | 16 lines Merged revisions 263374 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010)
+ | 8 lines Update link to new version of core sounds. The latest
+ version of the core sounds files 1.4.19 now includes the missing
+ queue-minute sound file which is called by app_queue but which
+ has been missing. (closes issue #17123) Reported by: n8ideas
+ ........ ................
+
+2010-05-17 13:03 +0000 [r263293] David Vossel <dvossel@digium.com>
+
+ * CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer
+ policy dialstring option
+
+2010-05-15 23:41 +0000 [r263202] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 |
+ tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines
+ Make the Makefile logic more explicit and move the Snow Leopard
+ logic down to where it's not executed on non-Darwin systems.
+ (closes issue #17028) Reported by: pabelanger Patches:
+ issue17028_20100315.patch uploaded by seanbright (license 71)
+ 20100315__issue17028.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman, pabelanger ........
+
+2010-05-13 22:13 +0000 [r263070] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010)
+ | 1 line Fix inverted logic in cli command: ss7 set debug on/off
+ ........
+
+2010-05-13 15:36 +0000 [r262898] Russell Bryant <russell@digium.com>
+
+ * channels/chan_console.c, /: Merged revisions 262897 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010)
+ | 4 lines Fix an off by one error that causes a crash. Thanks to
+ Raymond Burke for pointing it out. ........
+
+2010-05-12 20:01 +0000 [r262801] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/loader.c, main/cli.c, /: Merged revisions 262800 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed,
+ 12 May 2010) | 8 lines Notify CLI when modules is loaded /
+ unloaded (closes issue #17308) Reported by: pabelanger Patches:
+ cli.modules.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger, russell ........
+
+2010-05-12 19:53 +0000 [r262797-262799] Leif Madsen <lmadsen@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 262798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 |
+ lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines
+ Revert previous WARNING message removal. Marquis42 suggested a
+ better method of doing what I wanted because I ended up removing
+ the WARNING message for all instances when really I just wanted
+ to remove it for the 'return' keyword, not everything. (issue
+ #17145) ........
+
+ * res/ael/pval.c, /: Merged revisions 262796 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 |
+ lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines
+ Remove unnecessary WARNING message in ael/pval.c (closes issue
+ #17145) Reported by: okrief ........
+
+2010-05-12 18:03 +0000 [r262746] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010)
+ | 17 lines Merged revisions 262662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
+ | 11 lines fixes app_meetme dsp error We attempted to detect
+ silence after translating a frame from signed linear. This caused
+ a flooding of errors. To resolve this the code to detect silence
+ was moved before the translation. (closes issue #17133) Reported
+ by: jsdyer ........ ................
+
+2010-05-12 16:29 +0000 [r262516-262659] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 |
+ tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines
+ Ensure the arguments are initialized. Also miscellaneous CG
+ cleanup. (closes issue #16576) Reported by: uxbod Patches:
+ 20100505__issue16576.diff.txt uploaded by tilghman (license 14)
+ Tested by: uxbod ........
+
+ * /, include/asterisk/causes.h: Merged revisions 262513 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11
+ May 2010) | 7 lines Move cause 200 to cause 26, as specified in
+ Q.850. Also cleanup the formatting and add a few more that seem
+ like good candidates. (closes issue #16157) Reported by: wimpy
+ ........
+
+2010-05-11 19:58 +0000 [r262425] Jason Parker <jparker@digium.com>
+
+ * /, res/Makefile: Merged revisions 262422 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) |
+ 18 lines Merged revisions 262421 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
+ 11 lines Use a less silly method for modifying a flex-generated
+ file. The sed syntax that was used wasn't actually valid, causing
+ some versions to choke. This is the method that is used in 1.6.x+
+ for similar changes. (closes issue #16696) Reported by: bklang
+ Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
+ by: qwell ........ ................
+
+2010-05-11 19:41 +0000 [r262415-262420] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 |
+ pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8
+ lines Improve logging by displaying line number (closes issue
+ #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded
+ by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 |
+ pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8
+ lines Improve logging information for misconfigured contexts
+ (closes issue #17238) Reported by: pprindeville Patches:
+ chan_sip-bug17238.patch uploaded by pprindeville (license 347)
+ Tested by: pprindeville ........
+
+2010-05-11 17:25 +0000 [r262340] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r262330 | tilghman | 2010-05-11 12:23:51 -0500
+ (Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11
+ May 2010) | 2 lines Fix issue #17302 a slightly different way
+ (mad props to Qwell) ........ ................
+
+2010-05-10 19:06 +0000 [r262237-262241] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 262240 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10
+ May 2010) | 9 lines fixes PickupChan application (closes issue
+ #16863) Reported by: schern Patches: app_directed_pickup.c.patch
+ uploaded by schern (license 995) for_trunk.diff uploaded by
+ cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama,
+ rickead2000, dvossel ........
+
+ * channels/chan_console.c, /: Merged revisions 262236 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010)
+ | 11 lines fixes crash in chan_console There is a race condition
+ between console_hangup() and start_stream(). It is possible for
+ console_hangup() to be called and then the stream thread to begin
+ after the hangup. To avoid this a check in start_stream() to make
+ sure the pvt-owner still exists while the pvt lock is held is
+ made. If the owner is gone that means the channel hung up and
+ start_stream should be aborted. ........
+
+2010-05-10 16:39 +0000 [r262155] Tilghman Lesher <tlesher@digium.com>
+
+ * /, Makefile.rules: Merged revisions 262152 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010)
+ | 17 lines Merged revisions 262151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
+ | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
+ issue #17297) Reported by: jcovert Patches:
+ 20100506__issue17297.diff.txt uploaded by tilghman (license 14)
+ (closes issue #17302) Reported by: jcovert ........
+ ................
+
+2010-05-09 02:17 +0000 [r261916-262105] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4,
+ autoconf/ast_c_define_check.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 262102 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08
+ May 2010) | 5 lines Cleanup a bit more by getting rid of useless
+ version defines. Also make library detection use passed CFLAGS.
+ (closes issue #17309) Reported by: stuarth ........
+
+ * /, configure, configure.ac: Merged revisions 262048 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010)
+ | 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only
+ ........
+
+ * /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 |
+ tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines
+ Double free crash (closes issue #17245) Reported by:
+ thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by
+ tilghman (license 14) Tested by: murraytm ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 261913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 |
+ tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14
+ lines Use the detected pthread building flags in every place,
+ instead of hardcoding -lpthread. We nicely detect the right flags
+ on each system for building Asterisk with pthreads, then ignore
+ it for every other build option that requires us to build with
+ pthreads. This caused some items to return a false negative. Also
+ cleanup some minor naming issues that caused "library library"
+ redundancy in the output. (closes issue #17303) Reported by:
+ stuarth Patches: 20100507__issue17303.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth ........
+
+2010-05-07 16:08 +0000 [r261868] Leif Madsen <lmadsen@digium.com>
+
+ * UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 |
+ lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines
+ Update UPGRADE-1.6.txt stating insecure=very has been removed.
+ (closes issue #17282) Reported by: stuarth Tested by: stuarth
+ ........
+
+2010-05-06 20:13 +0000 [r261739] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500
+ (Thu, 06 May 2010) | 15 lines Merged revisions 261735 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010)
+ | 8 lines Only allow the operator key to be accepted after
+ leaving a voicemail. Or rather disallow the operator key from
+ being accepted when not offered, such as after finishing a
+ recording from within the mailbox options menu. ABE-2121 SWP-1267
+ ........ ................
+
+2010-05-06 17:08 +0000 [r261612] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 261609 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) |
+ 11 lines Merged revisions 261608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
+ 4 lines Use the versioned MOH tarballs, now that we have them.
+ This makes for more reproducibility. Prompted by a discussion in
+ #asterisk-dev ........ ................
+
+2010-05-06 15:43 +0000 [r261563] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 |
+ tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines
+ Permit more lines within a SIP body to be parsed. The example
+ given within the related issue showed 120 lines, which was mostly
+ a result of the body being XML. (closes issue #17179) Reported
+ by: khw ........
+
+2010-06-01 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.8 Released.
+
+2010-05-26 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.8-rc2 Released.
+
+2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson@digium.com>
+
+ * Merged r265610 from 1.4:
+
+ Don't mark the cdr records of unanswered queue calls with "NOANSWER".
+ This restores the behavior prior to r258670.
+
+ (closes issue #17334)
+ Reported by: jvandal
+ Patches:
+ queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, jvandal
+
+2010-05-06 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.8-rc1 Released
+
+2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c: Add test case that ensures the heap handles
+ arbitrary removals properly. (issue #17277) Reported by:
+ cappucinoking Patches: test_heap.diff uploaded by cappucinoking
+ (license 1036) Tested by: cappucinoking, russell
+
+ * /, main/heap.c: Merged revisions 261496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 |
+ russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines
+ Fix handling of removing nodes from the middle of a heap. This
+ bug surfaced in 1.6.2 and does not affect code in any other
+ released version of Asterisk. It manifested itself as SIP qualify
+ not happening when it should, causing peers to go unreachable.
+ This was debugged down to scheduler entries sometimes not getting
+ executed when they were supposed to, which was in turn caused by
+ an error in the heap code. The problem only sometimes occurs, and
+ it is due to the logic for removing an entry in the heap from an
+ arbitrary location (not just popping off the top). The scheduler
+ performs this operation frequently when entries are removed
+ before they run (when ast_sched_del() is used). In a normal pop
+ off of the top of the heap, a node is taken off the bottom,
+ placed at the top, and then bubbled down until the max heap
+ property is restored (see max_heapify()). This same logic was
+ used for removing an arbitrary node from the middle of the heap.
+ Unfortunately, that logic is full of fail. This patch fixes that
+ by fully restoring the max heap property when a node is thrown
+ into the middle of the heap. Instead of just pushing it down as
+ appropriate, it first pushes it up as high as it will go, and
+ _then_ pushes it down. Lastly, fix a minor problem in
+ ast_heap_verify(), which is only used for debugging. If a parent
+ and child node have the same value, that is not an error. The
+ only error is if a parent's value is less than its children. A
+ huge thanks goes out to cappucinoking for debugging this down to
+ the scheduler, and then producing an ast_heap test case that
+ demonstrated the breakage. That made it very easy for me to focus
+ on the heap logic and produce a fix. Open source projects are
+ awesome. (closes issue #16936) Reported by: ib2 Tested by:
+ cappucinoking, crjw (closes issue #17277) Reported by:
+ cappucinoking Patches: heap-fix.rev2.diff uploaded by russell
+ (license 2) Tested by: cappucinoking, russell ........
+
+2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) |
+ 4 lines When failing to configure, don't destroy 'cfg' twice
+ Fixes a crash when some config section had an incorrect channel
+ config. ........
+
+2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May
+ 2010) | 19 lines Merged revisions 261274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
+ 2010) | 12 lines Registration fix for SIP realtime. Make sure
+ realtime fields are not empty. (closes issue #17266) Reported by:
+ Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
+ Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
+ https://reviewboard.asterisk.org/r/643/ ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 261232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 |
+ pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10
+ lines 'queue reset stats' erroneously clears wrapuptime
+ configuration. Resets each member's lastcall to 0 now. (closes
+ issue #17262, #16519) Reported by: rain Patches:
+ wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
+ by: rain ........
+
+2010-05-04 23:55 +0000 [r261098] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 261095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010)
+ | 18 lines Merged revisions 261093-261094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010)
+ | 7 lines Protect against overflow, when calculating how long to
+ wait for a frame. (closes issue #17128) Reported by: under
+ Patches: d.diff uploaded by under (license 914) ........ r261094
+ | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2
+ lines Add a tiny corner case to the previous commit ........
+ ................
+
+2010-05-04 19:01 +0000 [r260927] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500
+ (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010)
+ | 12 lines Voicemail transfer to operator should occur
+ immediately, not after main menu. There were two scenarios in the
+ advanced options that while using the operator=yes and review=yes
+ options, the transfer occurred only after exiting the main menu
+ (after sending a reply or leaving a message for an extension).
+ Now after the audio is processed for the reply or message the
+ transfer occurs immediately as expected. ABE-2107 ABE-2108
+ ........ ................
+
+2010-05-04 16:58 +0000 [r260884] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/sip.conf.sample, include/asterisk/frame.h,
+ main/channel.c, /, channels/chan_sip.c: Merged revisions 254450
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25
+ Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that
+ arrive before a T.38-capable application is executing on a
+ channel. This patch addresses an issue found during working with
+ end-users using res_fax. If an incoming call is answered in the
+ dialplan, or jumps to the 'fax' extension due to reception of a
+ CNG tone (with faxdetect enabled), and then the remote endpoint
+ sends a T.38 re-INVITE, it is possible for the channel's T.38
+ state to be 'T38_STATE_NEGOTIATING' when the application starts
+ up. Unfortunately, even if the application wants to use T.38, it
+ can't respond to the peer's negotiation request, because the
+ AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
+ originally has been lost, and the application needs the content
+ of that frame to be able to formulate a reply. This patch adds a
+ new 'request' type to AST_CONTROL_T38_PARAMETERS,
+ AST_T38_REQUEST_PARMS. If the application sends this request,
+ chan_sip will re-send the original control frame (with
+ AST_T38_REQUEST_NEGOTIATE as the request type), and the
+ application can respond as normal. If this occurs within the five
+ second timeout in chan_sip, the automatic cancellation of the
+ peer reinvite will be stopped, and the application will 'own' the
+ negotiation process from that point onwards. This also improves
+ the code path in chan_sip to allow sip_indicate(), when called
+ for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
+ response, which should have been in place before since the
+ control frame *can* fail to be processed properly. It also
+ modifies ast_indicate() to return whatever result the channel
+ driver returned for this control frame, rather than converting
+ all non-zero results into '-1'. Finally, the new request type
+ intentionally returns a positive value, so that an application
+ that sends AST_T38_REQUEST_PARMS can know for certain whether the
+ channel driver accepted it and will be replying with a control
+ frame of its own, or whether it was ignored (if the
+ sip_indicate()/ast_indicate() path had properly supported failure
+ responses before, this would not be necessary). This patch also
+ modifies res_fax to take advantage of the new request. In
+ addition, this patch makes sip_t38_abort() actually lock the
+ private structure before doing its work... bad programmer, no
+ donut. This patch also enhances chan_sip's 'faxdetect' support to
+ allow triggering on T.38 re-INVITEs received as well as CNG tone
+ detection. Review: https://reviewboard.asterisk.org/r/556/
+ ........
+
+2010-05-04 15:51 +0000 [r260746-260805] Jason Parker <jparker@digium.com>
+
+ * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260802 | qwell | 2010-05-04 10:49:57 -0500
+ (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
+ 2010) | 1 line Fix fallout from removing from configure script.
+ Pointed out by philipp64 on #asterisk-dev ........
+ ................
+
+ * /: Fix merge props
+
+2010-05-03 17:42 +0000 [r260743] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile, /: Merged revisions 260661-260662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
+ 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
+ libdir when executing mkpkgconfig allowing non-root installs to
+ work. (closes issue #17268) Reported by: pabelanger Patches:
+ issue17268.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
+ -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
+ part. Thanks Qwell. ........
+
+2010-05-03 14:59 +0000 [r260571] Leif Madsen <lmadsen@digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500
+ (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03
+ May 2010) | 1 line Minor typo pointed out by pabelanger on IRC.
+ ........ ................
+
+2010-04-30 22:48 +0000 [r260441] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500
+ (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
+ | 11 lines Ensure channel state is not incorrectly set in the
+ case of a very early answer. The needringing bit was being read
+ in dahdi_read after answering thereby setting the state to
+ ringing from up. This clears needringing upon answering so that
+ is no longer possible. (closes issue #17067) Reported by: tzafrir
+ Patches: needringing.diff uploaded by tzafrir (license 46)
+ ........ ................
+
+2010-04-30 20:22 +0000 [r260373] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500
+ (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr
+ 2010) | 18 lines Fix potential crash from race condition due to
+ accessing channel data without the channel locked. In
+ res_musiconhold.c, there are several places where a channel's
+ stream's existence is checked prior to calling ast_closestream on
+ it. The issue here is that in several cases, the channel was not
+ locked while checking the stream. The result was that if two
+ threads checked the state of the channel's stream at
+ approximately the same time, then there could be a situation
+ where both threads attempt to call ast_closestream on the
+ channel's stream. The result here is that the refcount for the
+ stream would go below 0, resulting in a crash. I have added
+ proper channel locking to res_musiconhold.c to ensure that we do
+ not try to check chan->stream without the channel locked. A
+ Digium customer has been using this patch for several weeks and
+ has not had any crashes since applying the patch. ABE-2147
+ ........ ................
+
+2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/app.c: Merged revisions 260292 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 |
+ tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13
+ lines Don't allow file descriptors to go above 64k, when we're
+ closing them in a fork(2). This saves time, when, even though the
+ system allows the process limit to be that high, the practical
+ limit is much lower. (closes issue #17223) Reported by:
+ dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
+ tilghman (license 14) Tested by: dbackeberg ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 260280 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30
+ Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan
+ context. (closes issue #17263) Reported by: pprindeville Patches:
+ freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
+ ........
+
+2010-04-29 23:13 +0000 [r260234] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500
+ (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
+ | 26 lines DTMF CallerID detection problems. The code handling
+ DTMF CallerID drops digits on long CallerID numbers and may
+ timeout waiting for the first ring with shorter numbers. The DTMF
+ emulation mode was not turned off when processing DTMF CallerID.
+ When the emulation code gets behind in processing the DTMF digits
+ it can skip a digit. For shorter numbers, the timeout may have
+ been too short. I increased it from 2 seconds to 4 seconds. Four
+ seconds is a typical time between rings for many countries.
+ (closes issue #16460) Reported by: sum Patches: issue16460.patch
+ uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
+ uploaded by rmudgett (license 664) Tested by: sum, rmudgett
+ Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
+ AST-334 JIRA SWP-901 ........ ................
+
+2010-04-29 18:18 +0000 [r260156] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 260148 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29
+ Apr 2010) | 2 lines Pattern match fail. ........
+
+2010-04-29 15:35 +0000 [r260051] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c, /, include/asterisk/audiohook.h: Merged
+ revisions 260050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010)
+ | 21 lines Merged revisions 260049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
+ | 14 lines Fixes crash in audiohook_write_list The middle_frame
+ in the audiohook_write_list function was being freed if a
+ audiohook manipulator returned a failure. This is incorrect
+ logic. This patch resolves this and adds detailed descriptions of
+ how this function should work and why manipulator failures must
+ be ignored. (closes issue #17052) Reported by: dvossel Tested by:
+ dvossel (closes issue #16196) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/623/ ........ ................
+
+2010-04-28 22:36 +0000 [r259959] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 |
+ mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11
+ lines Don't override peer context with domain context. (closes
+ issue #17040) Reported by: pprindeville Patches:
+ asterisk-1.6-bugid17040.patch uploaded by pprindeville (license
+ 347) Tested by: pprindeville Review:
+ https://reviewboard.asterisk.org/r/565/ ........
+
+2010-04-28 21:26 +0000 [r259899] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 259870
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500
+ (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
+ | 33 lines resolves deadlocks in chan_local Issue_1. In the
+ local_hangup() 3 locks must be held at the same time... pvt,
+ pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
+ the channel to hangup is the outbound chan_local channel, but
+ when it is not the outbound channel we have an issue... We
+ attempt to do deadlock avoidance only on the tech pvt, when both
+ the tech pvt and the pvt->owner are locked coming into that loop.
+ By never giving up the pvt->owner channel deadlock avoidance is
+ not entirely possible. This patch resolves that by doing deadlock
+ avoidance on both the pvt->owner and the pvt when trying to get
+ the pvt->chan lock. Issue_2. ast_prod() is used in
+ ast_activate_generator() to queue a frame on the channel and make
+ the channel's read function get called. This function is used in
+ ast_activate_generator() while the channel is locked, which
+ mean's the channel will have a lock both from the generator code
+ and the frame_queue code by the time it gets to chan_local.c's
+ local_queue_frame code... local_queue_frame contains some of the
+ same crazy deadlock avoidance that local_hangup requires, and
+ this recursive lock prevents that deadlock avoidance from
+ happening correctly. This patch removes ast_prod() from the
+ channel lock so only one lock is held during the
+ local_queue_frame function. (closes issue #17185) Reported by:
+ schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
+ (license 671) issue_17185_v2.diff uploaded by dvossel (license
+ 671) Tested by: schmoozecom, GameGamer43 Review:
+ https://reviewboard.asterisk.org/r/631/ ........ ................
+
+2010-04-28 21:09 +0000 [r259854] Leif Madsen <lmadsen@digium.com>
+
+ * config.guess: Merged revisions 259853 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010)
+ | 14 lines Merged revisions 259852 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
+ | 6 lines Update config.guess. Updating config.guess because
+ after installing Ubuntu Server 9.10 and running all the update
+ scripts, running ./configure would not continue because it was
+ unable to determine what kind of system I had. After updating
+ config.guess things started working again. ........
+ ................
+
+2010-04-28 20:34 +0000 [r259781-259851] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 259848 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259848 | qwell | 2010-04-28 15:32:14 -0500
+ (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
+ 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
+ systems without install can use install-sh from our source dir.
+ ........ ................
+
+ * makeopts.in, /: Merged revisions 259837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) |
+ 9 lines Merged revisions 259833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
+ 1 line Missed this when removing $ID ........ ................
+
+ * Makefile, /, configure, configure.ac: Merged revisions 259760 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259760 | qwell | 2010-04-28 14:19:54 -0500
+ (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
+ 7 lines Remove usage of `id` since it isn't useful and was
+ causing breakge. Solaris `id` doesn't support the -u argument.
+ Instead of figuring out how to fix this to work on Solaris, I
+ decided to check why it was necessary and where else it was used.
+ It was only used in one place, and it hasn't been needed for a
+ very long time (I question whether it was ever needed). ........
+ ................
+
+2010-04-28 17:19 +0000 [r259681] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500
+ (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010)
+ | 4 lines Do not play goodbye prompt after timeout of message
+ review. ABE-2124 ........ ................
+
+2010-04-27 22:46 +0000 [r259616] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500
+ (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010)
+ | 11 lines DAHDI "WARNING" message is confusing and vague
+ "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
+ failed: Success" Changed the warning to "Failed to decode
+ CallerID on channel 'name'". The message before it is likely more
+ specific about why the CallerID decode failed. SWP-501 AST-283
+ ........ ................
+
+2010-04-27 21:50 +0000 [r259528] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/Makefile: Merged revisions 259527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010)
+ | 23 lines Merged revisions 259526 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
+ | 15 lines Update sounds files. * Add additional sounds prompts
+ for say_enumeration * Update the English conference sounds
+ prompts so they are better quality and all sound more consistent
+ * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
+ to include all present sound files Both core (en, fr, es) and
+ extra (en, fr) sounds files have been updated. (closes issue
+ #16200) Reported by: murf (closes issue #17137) Reported by:
+ lmadsen ........ ................
+
+2010-04-27 21:25 +0000 [r259356-259486] Jason Parker <jparker@digium.com>
+
+ * main/editline/configure.in, /, main/editline/configure,
+ main/editline/Makefile.in: Merged revisions 259439 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) |
+ 5 lines Add gar to the check for AR for those silly OSes
+ (Solaris) that don't have ar. autoconf2.13 couldn't handle
+ AC_PROG_GREP, so I removed it. This is fine, since we don't need
+ to use anything that the configure script doesn't. ........
+
+ * /: Unblock revision 259439.
+
+ * /, configure, configure.ac: Merged revisions 259353 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259353 | qwell | 2010-04-27 14:31:55 -0500
+ (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) |
+ 5 lines Support the silly OSes that don't have ar and strip.
+ Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't
+ specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to
+ AC_CHECK_TOOLS. ........ ................
+
+2010-04-27 19:03 +0000 [r259310] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 259307 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010)
+ | 21 lines Merged revisions 259270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
+ | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
+ #7321 implements a new chan_dahdi configuration option. However,
+ a change mentioned in the issue was never implemented. This is
+ the change that will allow the feature to work. I added a note to
+ chan_dahdi.conf.sample about the feature. (closes issue #17143)
+ Reported by: djensen99 Patches: diff.txt uploaded by djensen99
+ (license NA) (One line change) Tested by: djensen99 ........
+ ................
+
+2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 259105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr
+ 2010) | 9 lines Merged revisions 259104 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
+ 2010) | 3 lines Let compilation succeed warning-free when
+ DONT_OPTIMIZE is turned off. ........ ................
+
+ * main/channel.c, /: Merged revisions 259023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr
+ 2010) | 19 lines Merged revisions 259018 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
+ 2010) | 13 lines Prevent Newchannel manager events for dummy
+ channels. No Newchannel manager event will be fired for channels
+ that are allocated to not match a registered technology type.
+ Thus bogus channels allocated solely for variable substitution or
+ CDR operations do not result in a Newchannel event. (closes issue
+ #16957) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/601 ........ ................
+
+2010-04-26 16:00 +0000 [r258935] Leif Madsen <lmadsen@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 |
+ lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines
+ Small error in the T.140 RTP port verbose log. (closes issue
+ #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff
+ uploaded by frawd (license 610) Tested by: russell ........
+
+2010-04-25 18:14 +0000 [r258779] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010)
+ | 13 lines Merged revisions 258775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
+ | 6 lines When StopMonitor is called, ensure that it will not be
+ restarted by a channel event. (closes issue #16590) Reported by:
+ kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
+ (license 888) ........ ................
+
+2010-04-22 22:15 +0000 [r258676] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
+ 258671,258675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr
+ 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
+ 2009) | 8 lines 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 ........ r258670 |
+ mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
+ lines Fix broken CDR behavior. This change allows a CDR record
+ previously marked with disposition ANSWERED to be set as BUSY or
+ NO ANSWER. Additionally this change partially reverts r235635 and
+ does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
+ from ast_call(). To preserve proper CDR behavior, the
+ AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
+ ast_bridge_call(). (closes issue #16797) Reported by:
+ VarnishedOtter Tested by: mnicholson ........ (closes issue
+ #16222) Reported by: telles Tested by: mnicholson
+ ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500
+ (Thu, 22 Apr 2010) | 2 lines Fix previous commit.
+ ................
+
+2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 258632 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only
+ merge the bug fixes, not the unit test. ........ r258632 |
+ russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines
+ Add ast_event subscription unit test and fix some ast_event API
+ bugs. This patch introduces another test in test_event.c that
+ exercises most of the subscription related ast_event API calls. I
+ made some minor additions to the existing event allocation test
+ to increase API coverage by the test code. Finally, I made a list
+ in a comment of API calls not yet touched by the test module as a
+ to-do list for future test development. During the development of
+ this test code, I discovered a number of bugs in the event API.
+ 1) subscriptions to AST_EVENT_ALL were not handled appropriately
+ in a couple of different places. The API allows a subscription to
+ all event types, but with IE parameters, just as if it was a
+ subscription to a specific event type. However, the parameters
+ were being ignored. This affected ast_event_check_subscriber()
+ and event distribution to subscribers. 2) Some of the logic in
+ ast_event_check_subscriber() for checking subscriptions against
+ query parameters was wrong. Review:
+ https://reviewboard.asterisk.org/r/617/ ........
+
+ * /, doc/tex/channelvariables.tex: Merged revisions 258515 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22
+ Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........
+
+2010-04-21 22:11 +0000 [r258436] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500
+ (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010)
+ | 8 lines Fix looping forever when no input received in certain
+ voicemail menu scenarios. Specifically, prompting for an
+ extension (when leaving or forwarding a message) or when
+ prompting for a digit (when saving a message or changing
+ folders). ABE-2122 SWP-1268 ........ ................
+
+2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/asterisk.tex: Remove missed line in previous merge.
+ (issue #17220)
+
+ * configure: Forgot to merge the updated configure script. (issue
+ #17220)
+
+ * doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in,
+ doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile,
+ configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
+ build_tools/prep_tarball: Merged revisions 258351 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010)
+ | 20 lines Add ability to generate ASCII documentation from the
+ TeX files. These changes add the ability to run 'make
+ asterisk.txt' just like the existing 'make asterisk.pdf' commands
+ to generate a text document from the TeX files we have in the
+ doc/tex/ directory. I've also updated a few of the .tex files
+ because they weren't properly escaping certain characters so they
+ would show up as Unicode characters (like [U+021C]). Made changes
+ to the configure scripts so it would detect the catdvi program
+ which is required to convert the .dvi file generated by latex.
+ I've also added a few lines to the build_tools/prep_tarball
+ script so that the text documentation gets generated and added to
+ future tarballs of Asterisk releases. (closes issue #17220)
+ Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
+ lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
+ (license 224) Tested by: lmadsen, pabelanger ........
+
+2010-04-21 18:19 +0000 [r258314] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 |
+ dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines
+ fixes issue with double "sip:" in header field This is a clear
+ mistake in logic. Future discussions about how to avoid having to
+ handle uri's like this should take place in the future, but this
+ fix needs to go in for now. (closes issue #15847) Reported by:
+ ebroad Patches: doublesip.patch uploaded by ebroad (license 878)
+ ........
+
+2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/cli_aliases.conf.sample: Merged revisions 258149 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20
+ Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on
+ asterisk-users. ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 258147 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20
+ Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers
+ (http://www.freenum.org). Minor tweaks and documentation added by
+ me. (closes issue #17058) Reported by: pprindeville Patches:
+ freenum.patch#5 uploaded by pprindeville (license 347) Tested by:
+ lmadsen ........
+
+2010-04-20 18:04 +0000 [r258108] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500
+ (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010)
+ | 11 lines Play correct prompt when voicemail store failure
+ occurs after attempted forward. If a user's mailbox was full and
+ a message was attempted to be forwarded to said box, warnings on
+ the console would indicate failure. However, the played prompt
+ was that of success (vm-msgsaved). Now storage failure is taken
+ into account and the correct prompt (vm-mailboxfull) is played
+ when appropriate. ABE-2123 SWP-1262 ........ ................
+
+2010-04-20 18:02 +0000 [r258107] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20
+ Apr 2010) | 7 lines Add missing 'useragent' field to
+ sip-friends.sql file. (closes issue #17171) Reported by: thehar
+ Patches: sip-friends.patch uploaded by thehar (license 831)
+ Tested by: pabelanger, thehar ........
+
+2010-04-19 21:58 +0000 [r257948-257950] Jason Parker <jparker@digium.com>
+
+ * main/indications.c, /: Merged revisions 257949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 |
+ qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line
+ Change log message to match severity. ........
+
+ * main/indications.c, /: Merged revisions 257947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 |
+ qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines
+ Don't consider a missing indications.conf to be a critical error.
+ There were many changes in revision 176627 which would avoid the
+ error that a missing config would have caused. Other than this,
+ there are no other config files (including asterisk.conf,
+ surprisingly) that are required. ........
+
+2010-04-19 18:30 +0000 [r257850] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 257810 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 |
+ twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines
+ Fix incomplete CDR merge from r195881 Because res/res_features.c
+ was removed and main/cdr.c added, these changes didn't make it to
+ trunk and the 1.6.x branches ........
+
+2010-04-18 17:28 +0000 [r257771] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18
+ Apr 2010) | 2 lines Removing unused configuration parameters
+ ........
+
+2010-04-16 21:47 +0000 [r257740] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500
+ (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010)
+ | 21 lines Make the mixmonitor thread process audio frames faster
+ Mantis issue 17078 reports MixMonitor recordings have shorter
+ durations than the call duration. This was because the mixmonitor
+ thread was not processing frames from the audiohook fast enough.
+ The mixmonitor thread would slowly fall behind the most recent
+ audio frame and when the channel hangs up, the mixmonitor thread
+ would exit without processing the same number of frames as the
+ channel; leaving the mixmonitor recording shorter than actual
+ call duration. This revision fixes this issue by moving the
+ ast_audiohook_trigger_wait() and the subsequent audiohook.status
+ check into the block where the ast_audiohook_read_frame()
+ function returns NULL. (closes issue #17078) Reported by:
+ geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license
+ 733) Tested by: dhubbard, geoff2010 Review:
+ https://reviewboard.asterisk.org/r/611/ ........ ................
+
+2010-04-15 21:34 +0000 [r257510-257597] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, /, main/app.c: Merged revisions 257560
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500
+ (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
+ | 6 lines Allow application options with arguments to contain
+ parentheses, through a variety of escaping techniques. Fixes
+ SWP-1194 (ABE-2143). Review:
+ https://reviewboard.asterisk.org/r/604/ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010)
+ | 20 lines Merged revisions 257467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
+ | 13 lines Don't recreate peer, when responding to a repeated
+ deregistration attempt. When a reply to a deregistration is lost
+ in transmit, the client retries the deregistration. Previously,
+ this would cause a realtime/autocreate peer to be loaded back
+ into memory, after it had already been correctly purged. Instead,
+ we just want to resend the reply without loading the peer.
+ (closes issue #16908) Reported by: kkm Patches:
+ 20100412__issue16908.diff.txt uploaded by tilghman (license 14)
+ Tested by: kkm ........ ................
+
+2010-04-15 19:42 +0000 [r257344-257428] Leif Madsen <lmadsen@digium.com>
+
+ * doc/backtrace.txt: Merged revisions 257427 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010)
+ | 21 lines Merged revisions 257426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
+ | 13 lines Update backtrace.txt documentation. Update the
+ backtrace.txt documentation so it conforms to the same layout as
+ other documents we've been working on recently. Additionally, add
+ a bunch of new information about gathering backtraces for crashes
+ and deadlocks, along with ways of verifying your file before
+ uploading it. Create a couple of one line commands for people to
+ generate the files we need. (closes issue #17190) Reported by:
+ lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
+ (license 10) Tested by: lmadsen, pabelanger ........
+ ................
+
+ * doc/backtrace.txt: Merged revisions 257343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010)
+ | 9 lines Merged revisions 257342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
+ | 1 line Update address of the bug tracker. ........
+ ................
+
+2010-04-14 23:00 +0000 [r257265] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/features.conf.sample, /, main/features.c: Merged
+ revisions 257262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 |
+ tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15
+ lines Yet another issue where the conversion of the application
+ delimiter to comma caused an issue. Application arguments within
+ the feature map could possibly contain a comma, which conflicts
+ with the syntax of the features.conf configuration file. This
+ patch allows the argument to be wrapped in parentheses or quoted,
+ to allow the application arguments to be interpreted as a single
+ configuration parameter. (closes issue #16646) Reported by:
+ pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/547/ ........
+
+2010-04-13 19:20 +0000 [r257210] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 |
+ tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10
+ lines Also unref the pvt when we delete the provisional keepalive
+ job. (closes issue #16774) Reported by: kowalma Patches:
+ 20100315__issue16774.diff.txt uploaded by tilghman (license 14)
+ Tested by: falves11, jamicque Review:
+ https://reviewboard.asterisk.org/r/591/ ........
+
+2010-04-13 18:43 +0000 [r257184] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /, configs/manager.conf.sample: Merged revisions
+ 257146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr
+ 2010) | 16 lines Merged revisions 257070 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
+ 2010) | 9 lines Add an option to restore past broken behavor of
+ the Events manager action Before r238915, certain values for the
+ EventMask parameter of the Events action would result in no
+ response being returned. This patch adds an option to restore
+ that broken behavior. Also while fixing this bug I discovered
+ that passing an empty EventMasks parameter would also result in
+ no response being returned, this has been fixed as well while
+ being preserved when the broken behavior is requested. (closes
+ issue #17023) Reported by: nblasgen Review:
+ https://reviewboard.asterisk.org/r/602/ ........ ................
+
+2010-04-13 16:38 +0000 [r257068] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010)
+ | 8 lines Ensure that we can have commas within cdr values.
+ (closes issue #17001) Reported by: snuffy Patches:
+ 20100412__issue17001.diff.txt uploaded by tilghman (license 14)
+ Tested by: snuffy ........
+
+2010-04-12 17:30 +0000 [r256822-256902] Leif Madsen <lmadsen@digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt (added): Merged revisions
+ 256901 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010)
+ | 23 lines Merged revisions 256900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
+ | 15 lines Add How-To document on collecting debugging info for
+ issues.asterisk.org Paul Belanger has been helping a lot with bug
+ tracking recently and created this document that we can now point
+ to when additional debugging information is required. This
+ document will help those filing issues to know how to get the
+ information required when filing their issues. This will make
+ things easier on the developers. Initial text and changes by
+ pabelanger. Tweaks and editing by myself. (closes issue #17159)
+ Reported by: pabelanger Patches:
+ HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
+ (license 10) Tested by: tzafrir, pabelanger, lmadsen ........
+ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010)
+ | 3 lines Remove silly debug message that is not useful. (issue
+ #17159) ........
+
+ * /, main/logger.c: Merged revisions 256821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 |
+ lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines
+ CLI command logger set level auto complete. A simple patch to
+ enable auto tab complete. (closes issue #17152) Reported by:
+ pabelanger Patches: 0017152.patch uploaded by pabelanger (license
+ 224) ........
+
+2010-04-08 22:03 +0000 [r256483] Tilghman Lesher <tlesher@digium.com>
+
+ * main/app.c: Backport /proc/%d/fd method of closing file
+ descriptors to 1.6.2.
+
+2010-04-06 19:40 +0000 [r256373] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Merged revisions 256370 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010)
+ | 2 lines Mac OS X does not support comparing a mutex to its
+ initializer. Create a test for this. ........
+
+2010-04-06 18:53 +0000 [r256268-256368] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty
+ after the first call. The bug is exposed if MFC/R2 support is
+ built into asterisk (i.e., openr2.h is present in the include
+ path). Code that unconditionally clears the CallerID name and
+ number is included. Also fixed a malformed if test in mkintf()
+ added by issue 15883. Converted the if statement to a switch
+ statement for clarity. Regression of the issue 15883 fix. (closes
+ issue #16968) Reported by: grecco Patches: issue16968.patch
+ uploaded by rmudgett (license 664) (closes issue #16747) Reported
+ by: viniciusfontes
+
+ * channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500
+ (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010)
+ | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by
+ PRI lock. SWP-1231 ABE-2163 ........ ................
+
+2010-05-03 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.7 Released
+
+2010-04-29 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.7-rc3 Released
+
+2010-04-29 10:31 +0000 [r260053] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
+ audiohook_write_list. (closes issue 0017052) Reported by: dvossel
+ Tested by: dvossel. (closes issue 0016196) Reported by: atis.
+ Review: https://reviewboard.asterisk.org/r/623/
+
+2010-04-28 10:31 +0000 [r259899] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, main/channel.c: Resolves deadlocks in
+ chan_local. (closes issue 0017185) Reported by: schmoozecom
+ Patches: issue_17185_v1.diff uploaded by dvossel (license 671)
+ issue_17185_v2.diff uploaded by dvossel (license 671) Tested
+ by: schmoozecom, GameGamer43
+ Review: https://reviewboard.asterisk.org/r/631/
+
+2010-04-13 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.7-rc2 Released
+
+2010-04-13 [r257210] Tilghman Lesher <tlesher@digium.com>
+
+ Also unref the pvt when we delete the provisional keepalive job.
+
+ (closes issue #16774)
+ Reported by: kowalma
+ Patches:
+ 20100315__issue16774.diff.txt uploaded by tilghman (license 14)
+ Tested by: falves11, jamicque
+
+ Review: https://reviewboard.asterisk.org/r/591/
+
+2010-04-05 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.7-rc1 Released
+
+2010-04-05 15:15 +0000 [r256162] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/localchannel.tex, /: Merged revisions 256161 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010)
+ | 1 line Fix for localchannel.tex to allow PDFs to be generated
+ again. ........
+
+2010-04-02 23:56 +0000 [r256013-256020] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 256019 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r256019 |
+ russell | 2010-04-02 18:55:57 -0500 (Fri, 02 Apr 2010) | 10 lines
+ Export MEETMEBOOKID and fix pin-less conferences with realtime
+ conferences (closes issue #16866) Reported by: DEA Patches:
+ rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA
+ Review: https://reviewboard.asterisk.org/r/582/ ........
+
+ * channels/chan_local.c, /: Merged revisions 256015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r256015 | russell | 2010-04-02 18:46:45 -0500
+ (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010)
+ | 9 lines Resolve a deadlock that occurs due to a pointless call
+ to ast_bridged_channel() (closes issue #16840) Reported by:
+ bzing2 Patches: patch.txt uploaded by bzing2 (license 902)
+ issue_16840.rev1.diff uploaded by russell (license 2) Tested by:
+ bzing2, russell ........ ................
+
+ * main/channel.c, /: Merged revisions 256010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010)
+ | 9 lines Merged revisions 256009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010)
+ | 2 lines Remove extremely verbose debug message. ........
+ ................
+
+2010-04-02 20:20 +0000 [r255955] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 255952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 |
+ tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines
+ Pass the PID of the Asterisk process, not the PID of the canary.
+ (closes issue #17065) Reported by: globalnetinc Patches:
+ astcanary.patch uploaded by makoto (license 38) Tested by: frawd,
+ globalnetinc ........
+
+2010-04-01 18:21 +0000 [r255676-255816] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010)
+ | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue
+ #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt
+ uploaded by tilghman (license 14) ........
+
+ * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500
+ (Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010)
+ | 15 lines Ensure line terminators in email are consistent. Fixes
+ an issue with certain Mail Transport Agents, where attachments
+ are not interpreted correctly. (closes issue #16557) Reported by:
+ jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by
+ tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt
+ uploaded by tilghman (license 14)
+ 20100308__issue16557__trunk.diff.txt uploaded by tilghman
+ (license 14) Tested by: ebroad, zktech Reviewboard:
+ https://reviewboard.asterisk.org/r/544/ ........ ................
+
+2010-03-31 17:49 +0000 [r255505] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample, apps/app_dial.c: Merged revisions 255504
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31
+ Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T'
+ can be used. (closes issue #17021) Reported by: kovzol Tested by:
+ lmadsen, kovzol, davidw, ebroad ........
+
+2010-03-30 20:58 +0000 [r255326-255413] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r255410 | russell | 2010-03-30 15:56:26 -0500
+ (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30
+ Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does
+ not start. ........ ................
+
+ * /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010)
+ | 9 lines Merged revisions 255322 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010)
+ | 2 lines Don't make Asterisk not start if pbx_dundi fails to
+ initialize. ........ ................
+
+2010-03-26 19:28 +0000 [r255023-255067] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 255066 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010)
+ | 6 lines Replace some documentation from 1.6.x back into trunk.
+ This documentation associated wth tlsbindaddr is still useful so
+ lets synchronize it between trunk and 1.6.x branches. (issue
+ #17054) ........
+
+ * configs/sip.conf.sample, /: Merged revisions 255021 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010)
+ | 8 lines Update confusing documentation for tlsbindaddr. Update
+ some confusing documentation for the tlsbindaddr option in
+ sip.conf.sample. Point at a link instead which has better
+ documentation. (closes issue #17054) Reported by: klaus3000
+ ........
+
+2010-03-25 20:43 +0000 [r254770-254805] Jason Parker <jparker@digium.com>
+
+ * utils/Makefile, /: Merged revisions 254802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) |
+ 9 lines Merged revisions 254800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) |
+ 1 line Don't remove local copies of utils in uninstall. ........
+ ................
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
+ issue with out-of-tree modules. Take 2, without ABI breakage this
+ time. Review: https://reviewboard.asterisk.org/r/588/
+
+2010-03-25 20:09 +0000 [r254721] Russell Bryant <russell@digium.com>
+
+ * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010)
+ | 2 lines chan_usbradio depends on alsa. ........
+
+2010-03-25 17:47 +0000 [r254556] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500
+ (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar
+ 2010) | 5 lines Add doxygen for acl.h Review:
+ https://reviewboard.asterisk.org/r/528 ........ ................
+
+2010-03-25 17:21 +0000 [r254548] Sean Bright <sean@malleable.com>
+
+ * channels/chan_sip.c: Initialize stream to avoid a compilation
+ error.
+
+2010-03-25 17:12 +0000 [r254542] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix potential crashes from trying to
+ reference nonexistent RTP streams.
+
+2010-03-25 16:26 +0000 [r254499] Terry Wilson <twilson@digium.com>
+
+ * /, main/file.c: Merged revisions 254453 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010)
+ | 9 lines Merged revisions 254451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010)
+ | 2 lines Handle new SRCCHANGE control message here too ........
+ ................
+
+2010-03-25 16:22 +0000 [r254482] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500
+ (Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar
+ 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection.
+ Here is a copy and paste of the details from my request on
+ reviewboard that dealt with these changes: Fix 1. The first
+ change in place is to fix Mantis issue 15811, which deals with a
+ situation where Asterisk will incorrectly interpret out of order
+ RFC2833 frames as duplicate DTMF digits. For instance, we would
+ receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3:
+ DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1
+ seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch
+ when we received the frame with seqno 5, we would interpret this
+ as a new DTMF 1. With this patch, we will check the seqno of the
+ incoming digit and not process the frame if the seqno is lower
+ than the last recorded seqno. Note that we do not record the
+ seqno of the dropped DTMF frame for future processing. While the
+ above situation is what was designed to be fixed, the patch is
+ written in such a way that the following would also be fixed too:
+ seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end)
+ seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno
+ 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In
+ this second situation, the beginning of the DTMF 2 arrives before
+ the final end frame of the DTMF 1. With the patch, seqno 12 is no
+ processed and thus we properly interpret the DTMF. Fix 2. The
+ second change in place is to fix an issue like the following:
+ seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
+ lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
+ *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
+ code in place that was supposed to properly end the previously
+ unended DTMF 1. The problem was that the code was essentially a
+ no-op. The code would set up an end frame for the DTMF 1 but
+ would immediately overwrite the frame with the begin for DTMF 2.
+ I changed process_dtmf_rfc2833() so that instead of returning a
+ single frame, it is given as an output parameter a list of
+ frames. Each frame that needs to be returned is appended to this
+ list. Fix 3. The final change is a minor one where an
+ AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
+ DTMF or an RFC 3389 frame and no frame was returned, then we
+ would return &ast_null_frame. The problem is that earlier in the
+ function, we may have generated an AST_CONTROL_SRCCHANGE frame
+ and put it in the list of frames we wish to return. This frame
+ would be lost in such a case. The patch fixes this problem
+ ........ ................
+
+2010-03-25 15:21 +0000 [r254447] Leif Madsen <lmadsen@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 254446 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 |
+ lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines
+ handle_speechset has 4 arguments. Update code to reflect that
+ handle_speechset has 4 arguments. (closes issue #17093) Reported
+ by: gpatri Patches: res_agi.patch uploaded by gpatri (license
+ 1014) Tested by: pabelanger, mmichelson ........
+
+2010-03-24 17:19 +0000 [r254288] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010)
+ | 78 lines Merged revisions 254235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010)
+ | 72 lines Ensure that monitor recordings are written to the
+ correct location (again) This is an extension to 248860. As such
+ the dialplan test has been extended: ; non absolute path, not
+ combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
+ exten => 5040, n, dial(sip/5001) ; absolute path, not combined
+ exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
+ 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
+ monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
+ exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
+ dial(sip/5001) ; combined: changemonitor from no path to non
+ absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
+ exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
+ wasn't possible before exten => 5044, n, dial(sip/5001) ; non
+ absolute path, combined exten => 5045, 1,
+ monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
+ dial(sip/5001) ; absolute path, combined exten => 5046, 1,
+ monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
+ dial(sip/5001) ; no path, combined exten => 5047, 1,
+ monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to absolute (leaves
+ tmp/jeff) exten => 5048, 1,
+ monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
+ changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5049, 1,
+ monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
+ changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
+ dial(sip/5001) ; combined: changemonitor from no path to absolute
+ exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
+ changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to no path
+ (leaves /tmp/jeff) exten => 5051, 1,
+ monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
+ changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
+ not combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
+ exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to non
+ absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
+ 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
+ dial(sip/5001) ; not combined: changemonitor from non absolute to
+ absolute (leaves tmp/jeff) exten => 5054, 1,
+ monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
+ changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
+ dial(sip/5001) ; not combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5055, 1,
+ monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
+ changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to
+ absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
+ 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
+ n, dial(sip/5001) ; not combined: changemonitor from absolute to
+ no path (leaves /tmp/jeff) exten => 5057, 1,
+ monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
+ changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
+ ........ ................
+
+2010-03-23 22:05 +0000 [r254131] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * tests/Makefile, /: Merged revisions 254001 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk ........ r254001 |
+ tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines
+ Change the name of the category 'TEST' to match the name of the
+ subdir ........
+
+2010-03-23 21:20 +0000 [r254068] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 254050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 |
+ jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines
+ Exit native bridging early for greater timing accuracy with
+ warnings This changes native bridging to break one millisecond
+ early so that the more accurate timeval calculations done in the
+ generic bridge can be performed using the bridge config.
+ Currently the time between exiting native bridging slightly late
+ can sometimes cause a large enough discrepancy for warnings to be
+ missed. For the record, 1.4 does not attempt to native bridge at
+ all when warnings are enabled. (closes issue #15815) Reported by:
+ adomjan Review: https://reviewboard.asterisk.org/r/577/ ........
+
+2010-03-22 19:55 +0000 [r253801] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 253800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar
+ 2010) | 11 lines Merged revisions 253799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar
+ 2010) | 4 lines Unconditionally copy the caller's account code to
+ the called party. (related to issue #16331) ........
+ ................
+
+2010-03-22 19:06 +0000 [r253714-253760] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22
+ Mar 2010) | 2 lines Update query should be an UPDATE, not a
+ SELECT. ........
+
+ * /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22
+ Mar 2010) | 4 lines Return the list for later manipulation. This
+ fixes an issue with the update procedure. Debugging with
+ mmichelson. ........
+
+ * configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged
+ revisions 253712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 |
+ tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines
+ Accomodate equal signs in DSNs and add documentation, based upon
+ mmichelson's feedback. ........
+
+2010-03-20 17:33 +0000 [r253595-253620] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_pgsql.c, main/stdtime/localtime.c, main/tcptls.c, /,
+ main/features.c: Merged revisions 253540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253540 |
+ russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines
+ Resolve more compiler warnings on FreeBSD. ........
+
+ * apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20
+ Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD.
+ ........
+
+ * /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 |
+ russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines
+ Resolve a compiler warning on FreeBSD. ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 253536 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010)
+ | 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix
+ build issues I had with this module on FreeBSD. ........
+
+2010-03-19 08:05 +0000 [r253492] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/astobj2.c, /: Merged revisions 253490 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 |
+ alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19
+ lines prevent segfault if bad magic number is encountered.
+ internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic
+ number', but internal_ao2_ref continues on, causing segfault.
+ Although AO2_MAGIC number is checked by INTERNAL_OBJ before
+ internal_ao2_ref is called, A02_MAGIC is being destroyed (or a
+ wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ.
+ internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad
+ magic number. (issue #17037) Reported by: alecdavis Patches:
+ bug17037.diff.txt uploaded by alecdavis (license 585) Tested by:
+ alecdavis ........
+
+2010-03-18 17:54 +0000 [r253257-253346] Leif Madsen <lmadsen@digium.com>
+
+ * /, apps/app_userevent.c: Merged revisions 253345 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253345 | lmadsen | 2010-03-18 12:52:35 -0500 (Thu, 18 Mar 2010)
+ | 7 lines Change usage of pipe to comma in UserEvent docs. Change
+ the example usage of pipe as a separator to comma in the
+ UserEvent documentation. (closes issue #16961) Reported by:
+ jlpedrosa ........
+
+ * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010)
+ | 9 lines Update to new Local channel documentation. Add same
+ changes as commit to 1.4, but convert to TeX. (issue #16963)
+ Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz
+ (license 834) ........
+
+2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_mgcp.c, channels/chan_sip.c,
+ include/asterisk/rtp.h: Revert API change in release branches
+ This re-renames ast_rtp_update_source to ast_rtp_new_source
+
+2010-03-17 00:41 +0000 [r253029-253033] Leif Madsen <lmadsen@digium.com>
+
+ * main/xmldoc.c, /: Merged revisions 253032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253032 |
+ lmadsen | 2010-03-16 19:40:51 -0500 (Tue, 16 Mar 2010) | 1 line
+ Fix a typo. ........
+
+ * configs/say.conf.sample, /: Merged revisions 253028 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500
+ (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010)
+ | 6 lines Add french snipset to say.conf. Add the french snipset
+ to say.conf. (Closes issue #15799) ........ ................
+
+2010-03-16 23:54 +0000 [r252978] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 |
+ tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines
+ Mask out previous arguments on each nested invocation of Gosub.
+ (closes issue #16758) Reported by: wdoekes Patches:
+ 20100316__issue16758.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/561/ ........
+
+2010-03-16 19:38 +0000 [r252850] Sean Bright <sean@malleable.com>
+
+ * res/res_clialiases.c, /: Merged revisions 252848 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r252848 | seanbright | 2010-03-16 15:36:24 -0400 (Tue, 16 Mar
+ 2010) | 10 lines Include an extra newline after "Aliased CLI
+ command" to get back the prompt. The other issue mentioned in
+ this bug will be more difficult to resolve since we have no idea
+ (right now) of knowing if the command that is aliased has been
+ installed yet. (issue #16978) Reported by: jw-asterisk Tested by:
+ seanbright ........
+
+2010-03-16 19:02 +0000 [r252770] Russell Bryant <russell@digium.com>
+
+ * utils/Makefile, /: Merged revisions 252767 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010)
+ | 13 lines Merged revisions 252766 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010)
+ | 6 lines Don't treat warnings as errors for muted. muted
+ supports OS X, but uses functions marked as deprecated in 10.6.
+ However, the functions are still supported, so just ignore the
+ warnings for now and allow the build to proceed. ........
+ ................
+
+2010-03-16 18:49 +0000 [r252763] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.ael.sample, /: Merged revisions 252762 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500
+ (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010)
+ | 7 lines Additional extensions.ael global variable fixes. Fixing
+ up a couple more overlapping global variable namespaces shared
+ with extensions.conf.sample. Also noticed a few of the lines that
+ were commented out didn't have the closing semi-colon so I added
+ that as well. (issue #17035) ........ ................
+
+2010-03-15 21:59 +0000 [r252626] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 |
+ seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4
+ lines Resolve a crash in SLATrunk when the specified trunk
+ doesn't exist. Reported by philipp64 in #asterisk-dev. ........
+
+2010-03-15 21:54 +0000 [r252622] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
+ 252619 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010)
+ | 9 lines Merged revisions 252617 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010)
+ | 2 lines Uh, yeah. Umask. I'm stupid. ........ ................
+
+2010-03-15 20:53 +0000 [r252535] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.ael.sample: Merged revisions 252534 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500
+ (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010)
+ | 7 lines Update extensions.ael file to not overlap
+ extensions.conf. Updated the extensions.ael file so the global
+ variables don't overlap those that we have in extensions.conf
+ (sample files). This way unexpected things won't happed hopefully
+ if both pbx_ael and res_config are loaded. (closes issue #17035)
+ Reported by: pprindeville ........ ................
+
+2010-03-15 05:04 +0000 [r252365-252444] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 |
+ tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines
+ THIS IS NOT PYTHON. Indentation doesn't matter, only braces do.
+ (closes issue #17025) Reported by: smurfix Patches: sip.patch
+ uploaded by smurfix (license 547) ........
+
+ * main/asterisk.c, Makefile,
+ contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged
+ revisions 252362 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010)
+ | 11 lines Merged revisions 252361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010)
+ | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard:
+ https://reviewboard.asterisk.org/r/551/ ........ ................
+
+2010-03-14 17:48 +0000 [r252317] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar
+ 2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They
+ added a sqlite3_log() function which was conflicting with our
+ function names. (closes issue #17017) Reported by: alephlg
+ ........
+
+2010-03-13 00:32 +0000 [r252137-252178] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c: Remove unusued field
+
+ * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c,
+ channels/chan_mgcp.c, main/channel.c, /, channels/chan_sip.c,
+ channels/chan_skinny.c, include/asterisk/rtp.h,
+ channels/chan_h323.c: Merged revisions 252089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
+ twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
+ Only change the RTP ssrc when we see that it has changed This
+ change basically reverts the change reviewed in
+ https://reviewboard.asterisk.org/r/374/ and instead limits the
+ updating of the RTP synchronization source to only those times
+ when we detect that the other side of the conversation has
+ changed the ssrc. The problem is that SRCUPDATE control frames
+ are sent many times where we don't want a new ssrc, including
+ whenever Asterisk has to send DTMF in a normal bridge. This is
+ also not the first time that this mistake has been made. The
+ initial implementation of the ast_rtp_new_source function also
+ changed the ssrc--and then it was removed because of this same
+ issue. Then, we put it back in again to fix a different issue.
+ This patch attempts to only change the ssrc when we see that the
+ other side of the conversation has changed the ssrc. It also
+ renames some functions to make their purpose more clear. Review:
+ https://reviewboard.asterisk.org/r/540/ ........
+
+2010-03-12 22:05 +0000 [r252090] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 252088 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r252088 | moy | 2010-03-12 16:57:40 -0500 (Fri, 12 Mar 2010) | 1
+ line add missing mfcr2_skip_category setting ........
+
+2010-03-12 19:50 +0000 [r251994] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010)
+ | 8 lines Don't override a user option with the global option.
+ (closes issue #16849) Reported by: ip-rob Patches:
+ 20100311__issue16849.diff.txt uploaded by tilghman (license 14)
+ Tested by: ip-rob ........
+
+2010-03-12 19:49 +0000 [r251991] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 251946 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r251946 | rmudgett | 2010-03-12 13:05:40 -0600 (Fri, 12 Mar 2010)
+ | 1 line Doxegen this chan_dahdi lock. ........
+
+2010-03-11 21:08 +0000 [r251879-251887] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_exec.c, /: Merged revisions 251884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 |
+ tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines
+ Because ExecIf needs to reprocess arguments, it's best if we
+ don't remove quotes during parsing. (closes issue #16905)
+ Reported by: ip-rob Patches: 20100303__issue16905.diff.txt
+ uploaded by tilghman (license 14) Tested by: ip-rob ........
+
+ * apps/app_system.c, /: Merged revisions 251877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 |
+ tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines
+ If the argument to the system application is quoted, ensure we
+ remove the quotes before trying to execute. (closes issue #16842)
+ Reported by: ip-rob Patches: 20100310__issue16842.diff.txt
+ uploaded by tilghman (license 14) Tested by: ip-rob ........
+
+2010-03-11 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.6 released
+
+2010-03-05 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.6-rc2 released
+
+2010-03-05 Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman
+ | 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in
+ ODBC query. (closes issue #16953) Reported by: elguero Patches:
+ app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37)
+ ........
+
+2010-03-04 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.6-rc1 released
+
+2010-03-03 21:24 +0000 [r250610] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/localchannel.tex, /: Merged revisions 250609 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010)
+ | 11 lines Update existing Local channel documentation. A
+ complete re-write of the Local channel documentation has been
+ performed, with the existing information from localchannel.txt
+ and localchannel.tex merged in. (closes issue #16637) Reported
+ by: kobaz Patches: localchannel.tex uploaded by lmadsen (license
+ 10) localchannel.txt uploaded by lmadsen (license 10) Tested by:
+ lmadsen, jsmith, mmichelson ........
+
+2010-03-03 19:13 +0000 [r250484] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600
+ (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010)
+ | 15 lines Make sure to clear red alarm after polarity reversal.
+ From the issue: The automatic overnight line tests (or manual
+ ones) used on UK (BT) lines causes a red alarm on a dahdi /
+ TDM400P connected channel. This is because the line uses voltage
+ tests (battery loss) and polarity reversal. The polarity reversal
+ causes chan_dahdi to initiate v23 CallerID processing but during
+ this the event DAHDI_EVENT_NOALARM is ignored so that the alarm
+ is never cleared. (closes issue #14163) Reported by: jedi98
+ Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license
+ 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........
+ ................
+
+2010-03-03 18:05 +0000 [r250253-250396] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 250395 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600
+ (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010)
+ | 16 lines fixes problem with duplicate TXREQ packets When
+ Asterisk receives an IAX2 TXREQ packet, try_transfer() will call
+ store_by_transfercallno() to link the chan_iax2_pvt struct into
+ iax_transfercallno_pvts. If a duplicate TXREQ packet is received
+ for the same call, the pvt struct will be linked into
+ iax_transfercallno_pvts multiple times. This patch fixes this.
+ Thanks rain for debugging this and providing a patch! (closes
+ issue #16904) Reported by: rain Patches:
+ iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
+ by: rain, dvossel ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 |
+ dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines
+ fixes signed to unsigned int comparision issue for FaxMaxDatagram
+ value. ........
+
+2010-03-02 21:10 +0000 [r249953-250052] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010)
+ | 8 lines Update IMAP documentation. Update the IMAP
+ documentation to make it clear that storing voicemails in the
+ same folder as a large number of emails could potentially cause
+ significant slow downs when writing or retrieving voicemails.
+ (issue #16704) Reported by: TimeHider Tested by: lmadsen,
+ TimeHider ........
+
+ * configs/cdr.conf.sample: Merged revisions 250045 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500
+ (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010)
+ | 7 lines Update documentation to clarify purpose of unanswered
+ option. (closes issue #16267) Reported by: elsto Patches:
+ cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested
+ by: davidw, elsto ........ ................
+
+ * doc/tex/configuration.tex, /: Merged revisions 250037 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02
+ Mar 2010) | 4 lines Update documentation to not imply we support
+ overriding options. (closes issue #16855) Reported by: davidw
+ ........
+
+ * apps/app_directory.c, /: Merged revisions 249950 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010)
+ | 4 lines Fix literal values wrapped in documentation. (closes
+ issue #16145) Reported by: tilghman ........
+
+2010-03-02 19:50 +0000 [r249952] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * UPGRADE-1.6.txt, main/editline/makelist.in, apps/app_echo.c,
+ UPGRADE.txt: revert ability to exit echo app caused a regression,
+ as only supported VOICE, not VIDEO etc. (issue #16880)
+
+2010-03-02 19:26 +0000 [r249916-249933] Leif Madsen <lmadsen@digium.com>
+
+ * /, main/features.c: Merged revisions 249925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 |
+ lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines
+ Add missing description of the PARKINGLOT variable in XML
+ documentation. (closes issue #16743) Reported by: snuffy Patches:
+ parkingdoc.diff uploaded by snuffy (license 35) ........
+
+ * /, pbx/pbx_dundi.c: Merged revisions 249912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 |
+ lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines
+ Convert some DUNDI functions to XML documentation. (closes issue
+ #16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by
+ snuffy (license 35) ........
+
+2010-03-02 19:12 +0000 [r249895] David Vossel <dvossel@digium.com>
+
+ * channels/chan_console.c, channels/chan_gtalk.c,
+ channels/chan_oss.c, channels/misdn_config.c,
+ include/asterisk/abstract_jb.h, configs/alsa.conf.sample,
+ channels/chan_jingle.c, channels/chan_usbradio.c,
+ channels/chan_dahdi.c, channels/chan_skinny.c,
+ configs/mgcp.conf.sample, main/abstract_jb.c,
+ channels/chan_h323.c, channels/chan_alsa.c,
+ configs/sip.conf.sample, channels/chan_mgcp.c,
+ channels/chan_unistim.c, configs/console.conf.sample,
+ configs/chan_dahdi.conf.sample, channels/chan_local.c,
+ configs/oss.conf.sample, channels/chan_sip.c, /,
+ configs/usbradio.conf.sample, configs/misdn.conf.sample: Merged
+ revisions 249893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 |
+ dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
+ fixes adaptive jitterbuffer configuration When configuring the
+ adaptive jitterbuffer, the target_extra value not only could not
+ be set from the configuration, but was not even being set to its
+ proper default. This value is required in order for the adaptive
+ jitterbuffer to work correctly. To resolve this a config option
+ has been added to expose this value to the conf files, and a
+ default value is provided when no config specific value is
+ present. ........
+
+2010-03-02 19:09 +0000 [r249894] Leif Madsen <lmadsen@digium.com>
+
+ * /, apps/app_confbridge.c: Merged revisions 249892 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010)
+ | 1 line Fix several XML documentation validate errors. ........
+
+2010-03-02 09:05 +0000 [r249844] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_echo.c: fixes ability to exit echo app when called from
+ a ISDN channel, null frames prevent '#' exit. Now only echo back
+ VOICE and DTMF frames (issue #16880) Reported by: alecdavis
+ Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis
+
+2010-03-01 19:40 +0000 [r249675] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500
+ (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar
+ 2010) | 11 lines Fix crash in app_voicemail related to message
+ counting. We were passing a 'struct inprocess **' and treating it
+ like a 'struct inprocess *' causing a segfault. (closes issue
+ #16921) Reported by: whardier Patches: 20100301_issue16921.patch
+ uploaded by seanbright (license 71) Tested by: whardier ........
+ ................
+
+2010-03-01 18:47 +0000 [r249625] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010)
+ | 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP
+ compile once again. ........
+
+2010-03-01 17:25 +0000 [r249580] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 249538 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600
+ (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010)
+ | 11 lines Modify queued frames from local channels to not set
+ the other side to up In this case, attended transfers were broken
+ due to ast_feature_request_and_dial detecting the channel being
+ set to up before the answer frame could be read and therefore
+ failing to mark the channel as ready. This fix is a regression
+ fix for 244785, which should continue to work properly as well.
+ (closes issue #16816) Reported by: jamhed Tested by: jamhed,
+ corruptor ........ ................
+
+2010-02-28 20:51 +0000 [r249407-249493] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010)
+ | 5 lines Fix unit test that Alec Davis broke. (closes issue
+ #16927) Reported by: alecdavis ........
+
+ * apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions
+ 249405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 |
+ tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines
+ Properly document voicemail API documents. Also fix a crash
+ reported via the -dev list. ........
+
+2010-02-27 23:04 +0000 [r249321] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_dahdi.c: overlap receiving: automatically send CALL
+ PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
+ user has determined that sufficient call information has been
+ received the user shall stop T302 and send CALL PROCEEDING to the
+ network. Previously timeouts were possible if the dialplan took a
+ long time to issue any response back to the network. Verified
+ that our local TELCO also does the same. (issue #16789) Reported
+ by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis
+
+2010-02-27 14:10 +0000 [r249238] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 249235 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500
+ (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27
+ Feb 2010) | 1 line add a reference to the now-published IAX2 RFC
+ ........ ................
+
+2010-02-26 18:49 +0000 [r249190] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010)
+ | 18 lines Cleanups to fix bugs in the VM count API functions. -
+ Urgent voicemails were not attached, because the attachment code
+ looked in the wrong folder. - Urgent voicemails were sometimes
+ counted twice when displaying the count of new messages. -
+ Backends were inconsistent as to which voicemails each API
+ counted. (closes issue #15654) Reported by: tomo1657 Patches:
+ 20100225__issue15654.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman (closes issue #16448) Reported by: hevad
+ Review: https://reviewboard.asterisk.org/r/525/ ........
+
+2010-02-26 17:06 +0000 [r249104] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb
+ 2010) | 14 lines Merged revisions 249100 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb
+ 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488.
+ (closes issue #16792) Reported by: vrban Patches: t38_606.patch
+ uploaded by vrban (license 756) ........ ................
+
+2010-02-25 23:12 +0000 [r248955] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 248952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010)
+ | 24 lines Merged revisions 248860 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010)
+ | 18 lines Ensure that monitor recordings are written to the
+ correct location (again) This is an extension to 248757. As such
+ the dialplan test has been extended: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) exten => 5043, 1,
+ monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
+ changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
+ exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
+ changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
+ design and emits a warning exten => 5044, n, dial(sip/5001)
+ ........ ................
+
+2010-02-25 22:42 +0000 [r248949] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/acl.c: Merged revisions 248946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 |
+ mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5
+ lines Fix incorrect ACL behavior when CIDR notation of "/0" is
+ used. AST-2010-003 ........
+
+2010-02-25 21:25 +0000 [r248864] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 248861 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010)
+ | 22 lines Merged revisions 248859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010)
+ | 15 lines Some platforms clear /var/run at boot, which makes
+ connecting a remote console... difficult. Previously, we only
+ created the default /var/run/asterisk directory at install time.
+ While we could create it in the init script, that would not work
+ for those who start asterisk manually from the command line. So
+ the safest thing to do is to create it as part of the Asterisk
+ boot process. This also changes the ownership of the directory,
+ because the pid and ctl files are created after we setuid/setgid.
+ (closes issue #16802) Reported by: Brian Patches:
+ 20100224__issue16802.diff.txt uploaded by tilghman (license 14)
+ Tested by: tzafrir ........ ................
+
+2010-02-25 18:52 +0000 [r248797] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 248793 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010)
+ | 22 lines Merged revisions 248757 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010)
+ | 15 lines Ensure that monitor recordings are written to the
+ correct location. Recordings should be placed in the monitor
+ directory when a non-absolute path is used. Exact dialplan used
+ for testing: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) ABE-2101 ........
+ ................
+
+2010-02-24 21:29 +0000 [r248643] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/logger.c: Merged revisions 248584 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010)
+ | 14 lines Merged revisions 248582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010)
+ | 7 lines Remove color code sequences from verbose messages that
+ go to logfiles. (closes issue #16786) Reported by: dodo Patches:
+ logger2.patch uploaded by dodo (license 989) Tested by: tilghman
+ ........ ................
+
+2010-02-23 16:37 +0000 [r248398] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010)
+ | 15 lines Merged revisions 248396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010)
+ | 9 lines fixes invite with replaces deadlock (closes issue
+ #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
+ uploaded by dvossel (license 671) Tested by: pwalker, dvossel
+ ........ ................
+
+2010-02-19 19:07 +0000 [r248011] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_console.c, main/loader.c, /: Merged revisions
+ 228798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228798 |
+ tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14
+ lines Fix various problems detected with Valgrind. * chan_console
+ accessed pvts after deallocation. * The module loader did not
+ check usecount on shutdown, which led to chan_iax2 reading a
+ timer that was already unloaded. (closes issue #16062) Reported
+ by: alexanderheinz Patches: 20091109__issue16062.diff.txt
+ uploaded by tilghman (license 14) Tested by: tilghman ........
+
+2010-02-19 19:00 +0000 [r248005] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 248003 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1
+ line mfcr2 issue 0016844 - Fix portability bit fields and make
+ mfcr2_immediate_accept work again, reported and patched by
+ korihor ........
+
+2010-02-19 18:45 +0000 [r248004] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600
+ (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600
+ (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
+ 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
+ consistent with other channel technologies. The processing of
+ DTMF tones on the receiving side of an ISDN channel is
+ inconsistent with the way it is handled in other channels,
+ especially DAHDI analog. This causes DTMF tones sent from an ISDN
+ phone to be doubled at the connected party. We are using the
+ following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
+ Option one is necessary because the asterisk DSP DTMF detection
+ is better than mISDN's internal DSP. Not as many false positives.
+ Option two is necessary to transmit DTMF tones end to end when
+ mISDN channels are connected to SIP channels with out of band
+ DTMF for example. The symptom is that DTMF tones sent by an ISDN
+ phone are doubled on the way through asterisk when two mISDN
+ channels are connected with a Local channel in between or if it
+ is bridged to an analog channel. The doubling of DTMF tones is
+ because DTMF is passed inband to asterisk by the mISDN channel
+ and passed out of band once again after the release of the DTMF
+ tone. Passing it inband is wrong. Neither an analog channel nor
+ SIP channel passes DTMF inband if configured to inband DTMF.
+ Analog and SIP channels filter out the DTMF tones because they
+ use the voice frames returned by ast_dsp_process. But chan_misdn
+ passes the unfiltered input voice frames instead. To overcome one
+ aspect of the problem, the doubling of DTMF tones when two mISDN
+ channels are directly bridged, someone made an 'optimization',
+ where in that case the DTMF tone passed out-of-band to the peer
+ channel is not translated to an inband tone at the transmit side.
+ This optimization is bad because it does not work in general. For
+ example, analog channels or mISDN channels when bridged through
+ an intermediary local channel will generate DTMF tones from
+ out-of-band information. Also, of course, it must not be done
+ when there is no inband DTMF available. This patch fixes the
+ issue. Now chan_misdn will filter the received inband DTMF signal
+ the same as other channel types. Another change included: No need
+ to build an extra translation path because ast_process_dsp does
+ it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
+ ................ ................
+
+2010-02-19 17:41 +0000 [r247916] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 247915 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 |
+ dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines
+ handle_request_invite revise comment, fix coding guideline issues
+ I'm working with this code right now trying to analyze a
+ deadlock. This change is just to clean up a few things before I
+ make a more complex patch. ........
+
+2010-02-18 23:15 +0000 [r247792-247845] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_speech.c, /: Merged revisions 247841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 |
+ tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines
+ Revert an errant part of a previous cleanup, to fix a memory
+ corruption issue. (closes issue #16368) Reported by: thirionjwf
+ Patches: res_speech.c.patch uploaded by thirionjwf (license 955)
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 |
+ tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17
+ lines If the peer record is from realtime, it could be set to 0,
+ due to MySQL not representing NULL well in integer columns. NULL
+ means the value is not specified for the column, which normally
+ means the driver uses whatever is the default value. However, on
+ MySQL, placing a NULL in either a float or integer column results
+ in a retrieval of the 0 value. Hence, users get an errant error
+ on load. This patch suppresses that error and makes the value as
+ if it was not there. Note that this cannot be done in the
+ realtime driver, because the lack of difference between NULL and
+ 0 can only be intepreted correctly by the driver itself. If we
+ did it in the realtime driver, then it would be effectively
+ impossible to set any realtime field to 0, because it would act
+ as if the field were unspecified and possibly take on a different
+ value. (closes issue #16683) Reported by: wdoekes ........
+
+2010-02-18 21:25 +0000 [r247737-247776] David Vossel <dvossel@digium.com>
+
+ * /, bridges/bridge_softmix.c: Merged revisions 247770 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010)
+ | 9 lines fixes confbridge crash when no timing module is loaded.
+ (closes issue #16471) Reported by: kjotte Patches: M16471.diff
+ uploaded by junky (license 177) Tested by: kjotte, junky ........
+
+ * apps/app_queue.c, /: Merged revisions 247736 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 |
+ dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines
+ fixes Queue with C option crash (closes issue #16475) Reported
+ by: okrief Patches: queue_crash.diff uploaded by dvossel (license
+ 671) ........
+
+2010-02-18 19:41 +0000 [r247653] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 247652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb
+ 2010) | 13 lines Merged revisions 247651 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb
+ 2010) | 6 lines Copy the calling party's account code to the
+ called party if they don't already have one. (closes issue
+ #16331) Reported by: bluefox Tested by: mnicholson ........
+ ................
+
+2010-02-18 16:58 +0000 [r247506-247512] Leif Madsen <lmadsen@digium.com>
+
+ * README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500
+ (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18
+ Feb 2010) | 1 line Add additional link to best practices document
+ per jsmith. ........ ................
+
+ * README-SERIOUSLY.bestpractices.txt (added): Merged revisions
+ 247503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010)
+ | 18 lines Merged revisions 247502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010)
+ | 10 lines Add best practices documentation. (issue #16808)
+ Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis
+ Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/507/ ........ ................
+
+2010-02-18 04:21 +0000 [r247426] Russell Bryant <russell@digium.com>
+
+ * sounds/Makefile, Makefile, /: Merged revisions 247423 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r247423 | russell | 2010-02-17 22:20:11 -0600
+ (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010)
+ | 10 lines Tweak argument handling for wget in the sounds
+ Makefile. 1) Fix the check to see if we are using wget to not be
+ full of fail. The configure script populates this variable with
+ the absolute path to wget if it is found, so it didn't work. 2)
+ Allow some extra arguments to be passed in for wget. This is just
+ a simple change to allow our Bamboo build script to tell wget to
+ be quiet and not fill up our logs with download status output.
+ ........ ................
+
+2010-02-17 21:32 +0000 [r246989-247337] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/strings.h, main/strings.c, /: Merged revisions
+ 247335 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 |
+ mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20
+ lines Fix two problems in ast_str functions found while writing a
+ unit test. 1. The documentation for ast_str_set and
+ ast_str_append state that the max_len parameter may be -1 in
+ order to limit the size of the ast_str to its current allocated
+ size. The problem was that the max_len parameter in all cases was
+ a size_t, which is unsigned. Thus a -1 was interpreted as
+ UINT_MAX instead of -1. Changing the max_len parameter to be
+ ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an
+ off-by-one error in the case where we attempted to write a string
+ larger than the current allotted size to a string when -1 was
+ passed as the max_len parameter. When trying to write more than
+ the allotted size, the ast_str's __AST_STR_USED was set to 1
+ higher than it should have been. Thanks to Tilghman for quickly
+ spotting the offending line of code. Oh, and the unit test that I
+ referenced in the top line of this commit will be added to
+ reviewboard shortly. Sit tight... ........
+
+ * apps/app_queue.c, /: Merged revisions 247169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb
+ 2010) | 9 lines Merged revisions 247168 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb
+ 2010) | 3 lines Make sure that when autofill is disabled that
+ callers not in the front of the queue cannot place calls.
+ ........ ................
+
+ * main/strings.c, /: Merged revisions 247076 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247076 |
+ mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12
+ lines Add va_end calls to __ast_str_helper. According to the man
+ page for stdarg(3), "Each invocation of va_copy() must be matched
+ by a corresponding invocation of va_end() in the same function."
+ There were several cases in __ast_str_helper where va_copy was
+ not matched with a corresponding call to va_end. ........
+
+ * include/asterisk/strings.h, /: Merged revisions 246985 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue,
+ 16 Feb 2010) | 3 lines Add some clarifying documentation to the
+ ast_str_set and ast_str_append functions. ........
+
+2010-02-16 21:03 +0000 [r246900-246982] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 246980 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 |
+ dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines
+ warning message if openssl support is missing while attempting
+ tls connection (closes issue #16673) Reported by: michaesc
+ Patches: tls_error_msg.diff uploaded by dvossel (license 671)
+ ........
+
+ * main/channel.c: fixes merge error with Monitor calculation fix
+
+ * main/channel.c, /: Merged revisions 246899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 |
+ dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines
+ fixes sample rate conversion issue with Monitor application When
+ using ast_seekstream with the read/write streams of a monitor,
+ the number of samples we are seeking must be of the same rate as
+ the stream or the jump calculation will be incorrect. This patch
+ adds logic to correctly convert the number of samples to jump to
+ the sample rate the read/write stream is using. For example, if
+ the call is G722 (16khz) and the read/write stream is recording a
+ 8khz wav, seeking 320 samples of 16khz audio is not the same as
+ seeking 320 samples of 8khz audio when performing the
+ ast_seekstream on the stream. ABE-2044 ........
+
+2010-02-15 23:45 +0000 [r246713] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, /: Merged revisions 246710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010)
+ | 12 lines Merged revisions 246709 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010)
+ | 5 lines Make the menuselect instructions correct by allowing
+ 'make menuselect' to actually solve dependency problems.
+ (Previously, it would fail out again with the same message about
+ running 'make menuselect', which was NOT at all helpful.)
+ ........ ................
+
+2010-02-12 23:33 +0000 [r246547] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 246546 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010)
+ | 21 lines Merged revisions 246545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010)
+ | 16 lines lock channel during datastore removal On channel
+ destruction the channel's datastores are removed and destroyed.
+ Since there are public API calls to find and remove datastores on
+ a channel, a lock should be held whenever datastores are removed
+ and destroyed. This resolves a crash caused by a race condition
+ in app_chanspy.c. (closes issue #16678) Reported by:
+ tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
+ tim ringenbach (license 540) Tested by: dvossel ........
+ ................
+
+2010-02-12 19:08 +0000 [r246464] Jason Parker <jparker@digium.com>
+
+ * main/channel.c: Fix some silly formatting that made my head hurt.
+
+2010-02-10 21:28 +0000 [r246199-246207] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 246204 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Fussy compiler on another machine... ........
+
+ * /, funcs/func_strings.c: Merged revisions 246200 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Fix weird issue with unit tests on optimized build -
+ turned out to be a signing issue. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_agi.c: Merged revisions 246030 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r246030 |
+ tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12
+ lines Solaris doesn't like outputting a NULL to a %s in format
+ strings. Detect all platforms that don't like that, either, and
+ ensure that when documentation is missing, we pass a non-NULL
+ pointer when outputting the corresponding documentation. (closes
+ issue #16689) Reported by: bklang Patches:
+ 20100209__issue16689__with_tests.diff.txt uploaded by tilghman
+ (license 14) Review: https://reviewboard.asterisk.org/r/497/
+ ........
+
+2010-02-10 17:51 +0000 [r246117] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 246116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010)
+ | 14 lines Merged revisions 246115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010)
+ | 8 lines fixes random deadlock in app_queue with use_weight
+ during reload (closes issue #16677) Reported by: tim_ringenbach
+ Patches: app_queue_use_weight_deadlock.diff uploaded by tim
+ ringenbach (license 540) ........ ................
+
+2010-02-10 16:58 +0000 [r246073] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 246070 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010)
+ | 22 lines Change channel state on local channels for
+ busy,answer,ring. Previously local channels channel state never
+ changed. This became problematic when the state of the other side
+ of the local channel was lost, for example during a masquerade.
+ Changing the state of the local channel allows for the scenario
+ to be detected when the channel state is set to ringing, but the
+ peer isn't ringing. The specific problem scenario is described in
+ 164201. Although this was noted on one of the issues, here is the
+ tested dialplan verified to work: exten =>
+ 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten =>
+ *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
+ exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
+ *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did
+ not exten =>
+ 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
+ issue #14992) Reported by: davidw ........
+
+2010-02-10 15:38 +0000 [r245948-246025] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 246022 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Enable warnings on atypical conditions for the FILTER
+ function (suggested by mmichelson on the -dev list). ........
+
+ * configs/extensions.conf.sample, /, funcs/func_strings.c: Merged
+ revisions 245945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010)
+ | 9 lines Merged revisions 245944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Include examples of FILTER usage in extension patterns
+ where a "." may be a risk. ........ ................
+
+2010-02-09 23:11 +0000 [r245794] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 245793 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600
+ (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010)
+ | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 =
+ 32768 which is the maximum allowed iax2 callnumber. Creating the
+ iaxs and iaxsl array of size 32768 means the maximum callnumber
+ is actually out of bounds. This causes a nasty crash. (closes
+ issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded
+ by dvossel (license 671) ........ ................
+
+2010-02-09 18:09 +0000 [r245732] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 245729 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 |
+ tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines
+ Ensure frames are only freed once. (closes issue #16361) Reported
+ by: vlad Patches: 20100208__issue16361.diff.txt uploaded by
+ tilghman (license 14) Tested by: kenny, bloodoff, misaksen
+ ........
+
+2010-02-09 17:43 +0000 [r245728] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 245727 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245727 |
+ mnicholson | 2010-02-09 11:40:04 -0600 (Tue, 09 Feb 2010) | 2
+ lines This commit removes an extra newline in T.38 generated SDP
+ packets. This bug was caused by the fix introduced in r243860.
+ (closes issue #16766) Reported by: raivisr Patches:
+ t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: raivisr ........
+
+2010-02-09 16:26 +0000 [r245683] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 245680 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 |
+ kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8
+ lines Don't offer MMR or JBIG transcoding during T.38
+ negotiation. After further discussion with Steve Underwood, we
+ should not (yet) be offering to receive MMR or JBIG transcoded
+ streams from T.38 endpoints. A future spandsp release will
+ support those features, and then they can be enabled during
+ negotiation ........
+
+2010-02-08 23:47 +0000 [r245626] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 245624 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245624 |
+ russell | 2010-02-08 17:43:00 -0600 (Mon, 08 Feb 2010) | 5 lines
+ Fix return value of get_ie_str() and get_ie_str_hash() for
+ non-existent IE. I found this bug while developing a unit test
+ for event allocation. Testing is awesome. ........
+
+2010-02-08 22:46 +0000 [r245581] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/Makefile, /, main/Makefile: Merged revisions 245578 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08
+ Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and
+ channels/ Makefiles. They were previously passed correctly, but
+ they simply weren't used. This caused issues with various
+ platforms whose builds needed to pass special linker flags via
+ the configure script. (closes issue #16596) Reported by:
+ pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by
+ pprindeville (license 347) Tested by: tilghman ........
+
+2010-02-08 20:43 +0000 [r245500] Jason Parker <jparker@digium.com>
+
+ * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r245497 | qwell | 2010-02-08 14:41:05 -0600
+ (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) |
+ 4 lines Remove reference of documentation in source directory.
+ People don't always build Asterisk from source (distro packages,
+ anybody?). ........ ................
+
+2010-02-05 19:27 +0000 [r245097] Jeff Peeler <jpeeler@digium.com>
+
+ * contrib/firmware (removed), /, LICENSE: Merged revisions 245090
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600
+ (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb
+ 2010) | 5 lines Remove contrib/firmware directory as it is empty
+ Remove explicit license for IAXy firmware as it is no longer
+ included in the tree ........ ................
+
+2010-02-05 17:10 +0000 [r244930] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c, /: Merged revisions 244927 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb
+ 2010) | 9 lines Merged revisions 244926 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb
+ 2010) | 1 line Update main copyright date. ........
+ ................
+
+2010-02-03 19:28 +0000 [r244555] Mark Michelson <mmichelson@digium.com>
+
+ * main/sched.c, /: Merged revisions 244547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244547 |
+ mmichelson | 2010-02-03 13:26:53 -0600 (Wed, 03 Feb 2010) | 3
+ lines Initialize counters in ast_sched_report so that resulting
+ data is not bogus. ........
+
+2010-02-03 18:47 +0000 [r244508] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /, main/ast_expr2f.c: Merged revisions
+ 244505 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244505 |
+ tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines
+ The chanvar= setting should inherit the entire list of variables,
+ not just the first one. (closes issue #16359) Reported by: raarts
+ Patches: dahdi-setvars.diff uploaded by raarts (license 937)
+ Tested by: raarts ........
+
+2010-02-02 22:29 +0000 [r244445] David Vossel <dvossel@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 244443 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 |
+ dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines
+ fixes crash during T.38 negotiation caused by invalid or missing
+ FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported
+ by: krn (closes issue #16724) Reported by: barthpbx (closes issue
+ #16517) Reported by: bklang (closes issue #16485) Reported by:
+ elsto ........
+
+2010-02-02 20:35 +0000 [r244395] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 244393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244393 |
+ tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18
+ lines Properly respect GOSUB_RESULT as to what to do with the
+ master channel. Previously, we would parse GOSUB_RESULT, but not
+ actually do anything with it. (closes issue #16686) Reported by:
+ bklang Patches: app_dial-respect-gosub_result.patch uploaded by
+ bklang (license 919) (with modifications) ........
+
+2010-02-02 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.2
+
+ * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can
+ remotely crash Asterisk by modifying the FaxMaxDatagram field of
+ the SDP to contain either a negative or exceptionally large value.
+ The same crash occurs when the FaxMaxDatagram field is omitted from
+ the SDP as well.
+
+2010-01-14 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.1
+
+2010-01-08 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.1-rc1
+
+2010-01-07 21:17 +0000 [r238499] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_console.c, channels/chan_oss.c, main/poll.c,
+ channels/chan_usbradio.c, include/asterisk/utils.h, /,
+ channels/chan_sip.c, channels/chan_alsa.c: Merged revisions
+ 209400 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 |
+ kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3
+ lines Define side-effect-safe MIN and MAX macros and remove
+ duplicate definitions from various files. (closes issue #16251)
+ Reported by: asgaroth ........
+
+2010-01-07 20:17 +0000 [r238362-238416] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 238412 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600
+ (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010)
+ | 10 lines fixes crash in "scheduled_destroy" in chan_iax A
+ signed short was used to represent a callnumber. This is makes it
+ possible to attempt to access the iaxs array with a negative
+ index. (closes issue #16565) Reported by: jensvb ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 |
+ dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines
+ Change in sip show channels display format allowing more digits
+ for CID (closes issue #16459) Reported by: Rzadzins Patches:
+ chan_sip_longer_cid.patch uploaded by Rzadzins (license 953)
+ ........
+
+ * apps/app_queue.c, /: Merged revisions 238361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 |
+ dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines
+ cli 'queue show' formatting fix. queue name was truncated over 12
+ characters (closes issue #16078) Reported by: RoadKill Patches:
+ quequename_limit.patch uploaded by ppyy (license 906) Tested by:
+ dvossel ........
+
+2010-01-07 09:49 +0000 [r238349] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 238313 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) |
+ 2 lines Document the usefulness of explicit udp:// in the
+ register string ........
+
+2010-01-06 21:48 +0000 [r238234] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010)
+ | 11 lines Merged revisions 238230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010)
+ | 4 lines Revise documentation on disposition values to the
+ actual values used. (closes issue #16289) Reported by: wdoekes
+ ........ ................
+
+2010-01-06 20:40 +0000 [r238137-238185] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 238181 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 |
+ jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines
+ Fix misreverting from 177158. (closes issue #15725) Reported by:
+ shanermn Patches: v1-15725.patch uploaded by dimas (license 88)
+ Tested by: shanermn ........
+
+ * /, main/features.c: Merged revisions 238134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 |
+ jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines
+ Fix channel name comparison for bridge application. The channel
+ name comparison was not comparing the whole string and therefore
+ if one channel name was a substring of the other, the bridge
+ would fail. (closes issue #16528) Reported by: telecos82 Patches:
+ res_features_r236843.diff uploaded by telecos82 (license 687)
+ ........
+
+2010-01-06 15:22 +0000 [r238013] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010)
+ | 14 lines Merged revisions 238009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010)
+ | 7 lines Resolve a crash due to an ast_frame not being fully
+ initialized. (closes issue #16531) Reported by: john8675309
+ (closes SWP-615) ........ ................
+
+2010-01-06 06:54 +0000 [r237969] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 237968 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237968 |
+ tilghman | 2010-01-06 00:53:23 -0600 (Wed, 06 Jan 2010) | 2 lines
+ Whoa, duplicate setting (dead code). ........
+
+2010-01-05 23:10 +0000 [r237924] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_test.c: Add a wait to ensure TestServer thinks it has
+ finished sending the final digit. This was previously committed
+ to 1.4, 1.6.0, 1.6.1, and trunk just after 1.6.2 was created (and
+ missed). 1.6.2 also needs this patch to resolve the bug. (closes
+ issue #16550) Reported by: opticron Patches: apptest.diff
+ uploaded by opticron (license 267)
+
+2010-01-05 23:09 +0000 [r237840-237921] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 237920 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 |
+ dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines
+ fixes holdtime playback issue in app_queue When reporting hold
+ time, the number of seconds should be mod 60. Otherwise audio
+ playback could be something like "2 minutes 123 seconds" rather
+ than "2 minutes 3 seconds". Also, the "minute" sound file is
+ missing, so for the moment until that file can be created the
+ "minutes" file is used instead. (closes issue #16168) Reported
+ by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by
+ nickilo (license ) Tested by: nickilo, wonderg ........
+
+ * main/pbx.c, /: Merged revisions 237839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 |
+ dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines
+ fixes subscriptions being lost after 'module reload' During a
+ module reload if multiple extension configs are present, such as
+ both extensions.conf and extensions.ael, watchers for one
+ config's hints will be lost during the merging of the other
+ config. This happens because hint watchers are only preserved for
+ the current config being merged. The old context list is
+ destroyed after the merging takes place, meaning any watchers
+ that were not perserved will be removed. Now all hints are
+ preserved during merging regardless of what config file is being
+ merged. These hints are only restored if they are present within
+ the new context list. (closes issue #16093) Reported by: jlaroff
+ ........
+
+2010-01-05 17:25 +0000 [r237743] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c: Merged revisions 237699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010)
+ | 14 lines Merged revisions 237697 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010)
+ | 7 lines Change a NOTICE log message to DEBUG where it belongs.
+ (closes issue #16479) Reported by: alexrecarey (closes SWP-577)
+ ........ ................
+
+2010-01-05 16:09 +0000 [r237657] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 237656 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010)
+ | 6 lines Make CLI command 'mixmonitor start|stop <channel> work
+ again. (closes issue #16534) Reported by: jlaguilar Fix as
+ suggested by jlaguilar in the bugreport ........
+
+2010-01-04 21:52 +0000 [r237409-237577] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: Merged revisions 237574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010)
+ | 13 lines Merged revisions 237573 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010)
+ | 6 lines Bounds checking for input string (closes issue #16407)
+ Reported by: qwell Patches: 20100104__issue16407.diff.txt
+ uploaded by tilghman (license 14) ........ ................
+
+ * main/pbx.c, /: Merged revisions 237494 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010)
+ | 15 lines Merged revisions 237493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010)
+ | 8 lines Regression in issue #15421 - Pattern matching (closes
+ issue #16482) Reported by: wdoekes Patches:
+ astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
+ 20091223__issue16482.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes, tilghman ........ ................
+
+ * main/config.c, /: Merged revisions 237414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 |
+ tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines
+ Oops, didn't compile (thanks, kpfleming) ........
+
+ * main/config.c, /: Merged revisions 237410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 |
+ tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines
+ Further reduce the encoded blank values back to blank in the
+ realtime API. (closes issue #16533) Reported by: sergee Patches:
+ 200100104__issue16533.diff.txt uploaded by tilghman (license 14)
+ Tested by: sergee ........
+
+ * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged
+ revisions 237406 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010)
+ | 23 lines Merged revisions 237405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010)
+ | 16 lines Add a flag to disable the Background behavior, for AGI
+ users. This is in a section of code that relates to two other
+ issues, namely issue #14011 and issue #14940), one of which was
+ the behavior of Background when called with a context argument
+ that matched the current context. This fix broke FreePBX,
+ however, in a post-Dial situation. Needless to say, this is an
+ extremely difficult collision of several different issues. While
+ the use of an exception flag is ugly, fixing all of the issues
+ linked is rather difficult (although if someone would like to
+ propose a better solution, we're happy to entertain that
+ suggestion). (closes issue #16434) Reported by: rickead2000
+ Patches: 20091217__issue16434.diff.txt uploaded by tilghman
+ (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by
+ tilghman (license 14) Tested by: rickead2000 ........
+ ................
+
+2010-01-04 16:50 +0000 [r237328] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 237327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237327 |
+ dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines
+ app_queue segfaults if realtime field uniqueid is NULL (closes
+ issue #16385) Reported by: haakon Patches: app_queue.c.patch
+ uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by
+ dvossel (license 671) Tested by: haakon ........
+
+2010-01-04 16:27 +0000 [r237326] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 237323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 |
+ jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines
+ Fix timeout for AGI command speech recognize. (closes issue
+ #16297) Reported by: semond ........
+
+2010-01-04 16:21 +0000 [r237322] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 237319 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600
+ (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010)
+ | 3 lines It's also possible for the Local channel to directly
+ execute an Application. Reviewboard:
+ https://reviewboard.asterisk.org/r/452/ ........ ................
+
+2010-01-02 10:03 +0000 [r237139] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10
+ lines Merged revisions 237135 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2
+ lines Release memory of the contact acl before unloading module
+ ........ ................
+
+2009-12-30 22:00 +0000 [r236985] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 236982 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600
+ (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009)
+ | 9 lines Don't queue frames to channels that have no means to
+ process them. (closes issue #15609) Reported by: aragon Patches:
+ 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by
+ tilghman (license 14) Tested by: aragon Review:
+ https://reviewboard.asterisk.org/r/452/ ........ ................
+
+2009-12-30 21:13 +0000 [r236899-236905] Jeff Peeler <jpeeler@digium.com>
+
+ * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 |
+ jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines
+ One more LOW_MEMORY compile fix. ........
+
+ * main/cli.c, /: Merged revisions 236893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236893 |
+ jpeeler | 2009-12-30 14:34:41 -0600 (Wed, 30 Dec 2009) | 11 lines
+ Fix compiling with LOW_MEMORY. Modified handle_verbose to be
+ LOW_MEMORY aware. (closes issue #16381) Reported by:
+ michael_iedema Patches: ast_complete_source_filename.patch
+ uploaded by michael iedema (license 942) modified by me ........
+
+2009-12-30 17:56 +0000 [r236804-236850] Tilghman Lesher <tlesher@digium.com>
+
+ * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009)
+ | 4 lines When the field is blank, don't warn about the field
+ being unable to be coerced, just skip the column. (closes
+ http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html)
+ Reported by Nic Colledge on the -dev list, fixed by me. ........
+
+ * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 |
+ tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines
+ Shut down the SIP session timers more gracefully, in order to
+ prevent a possible crash. (closes issue #16452) Reported by:
+ corruptor Patches: 20091221__issue16452.diff.txt uploaded by
+ tilghman (license 14) Tested by: corruptor ........
+
+2009-12-28 22:13 +0000 [r236716] Jason Parker <jparker@digium.com>
+
+ * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec
+ 2009) | 8 lines Allow "REMAINDER" to function properly in
+ expressions. (closes issue #16427) Reported by: wdoekes Patches:
+ ast16-reminder-remainder.patch uploaded by wdoekes (license 717)
+ Tested by: wdoekes ........
+
+2009-12-28 17:40 +0000 [r236670] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009)
+ | 4 lines Use recommended option, not deprecated option. (closes
+ issue #16515) Reported by: ManChicken ........
+
+2009-12-28 15:31 +0000 [r236513-236635] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/threadstorage.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 236613 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec
+ 2009) | 14 lines Merged revisions 236585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec
+ 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT
+ requires extra braces. There was conditional code (based on build
+ platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that
+ was removed since it is fixed in newer versions of
+ Solaris/OpenSolaris, but I am still running into it on Solaris 10
+ x86 so add a configure-time check for it. ........
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec
+ 2009) | 19 lines Merged revisions 236509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec
+ 2009) | 12 lines Avoid a crash with large numbers of MeetMe
+ conferences. Similar to changes made to Queue(), when we have
+ large numbers of conferences in meetme.conf (1000s) and we use
+ alloca()/strdupa(), we can blow out the stack and crash, so
+ instead just use a single fixed buffer. (closes issue #16509)
+ Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
+ by seanbright (license 71) Tested by: seanbright ........
+ ................
+
+2009-12-27 18:22 +0000 [r236437] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600
+ (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27
+ Dec 2009) | 2 lines Turn on colors in the daemon, since there's
+ many requests for it on Ubuntu. ........ ................
+
+2009-12-26 15:32 +0000 [r236361] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 236358 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec
+ 2009) | 9 lines Merged revisions 236357 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec
+ 2009) | 1 line update to latest releases with zero uid/gid
+ ........ ................
+
+2009-12-23 18:27 +0000 [r236189-236303] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 |
+ tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines
+ AGI may be invoked from outside the dialplan (closes issue
+ #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt
+ uploaded by tilghman (license 14) Tested by: atis ........
+
+ * /, res/res_agi.c: Merged revisions 236186 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009)
+ | 11 lines Merged revisions 236184 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009)
+ | 4 lines If EXEC only gets a single argument, don't crash when
+ the second is used. (closes issue #16504) Reported by: bklang
+ ........ ................
+
+2009-12-22 17:04 +0000 [r236064] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009)
+ | 18 lines Merged revisions 236062 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009)
+ | 11 lines fixes issue with p->method incorrectly set to ACK It
+ is possible for a second ACK to come in for a retransmitted
+ message. If an ack does not match an unacked message in our
+ queue, restore the previous p->method as this ACK is completely
+ ignored. (closes issue #16295) Reported by: omolenkamp Patches:
+ issue16295_v2.diff uploaded by dvossel (license 671) ........
+ ................
+
+2009-12-21 19:58 +0000 [r235944] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009)
+ | 20 lines Merged revisions 235940 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009)
+ | 13 lines Change Monitor to not assume file to write to does not
+ contain pathing. 227944 changed the fname_base argument to always
+ append the configured monitor path. This change was necessary to
+ properly compare files for uniqueness. If a full path is given
+ though, nothing needs to be appended and that is handled
+ correctly now. (closes issue #16377) (closes issue #16376)
+ Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch
+ uploaded by dant (license 670) ........ ................
+
+2009-12-21 17:11 +0000 [r235826] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/features.c: Merged revisions 235822 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009)
+ | 15 lines Merged revisions 235821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009)
+ | 8 lines Send parking lot announcement to the channel which
+ parked the call, not the park-ee. (closes issue #16234) Reported
+ by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded
+ by tilghman (license 14) 20091221__issue16234__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: yeshuawatso ........
+ ................
+
+2009-12-20 08:58 +0000 [r235775] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c: restarts busydetector (if enabled) when DTMF is
+ received after call is bridged. (closes issue #16389) Reported
+ by: alecdavis Tested by: alecdavis Patch
+ dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
+
+2009-12-18 23:04 +0000 [r235665] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h: Merged revisions
+ 235660 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009)
+ | 55 lines Merged revisions 235635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009)
+ | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is
+ simple in that it reorders the disposition defines so that the
+ fix for issue 12946 works properly (the default CDR disposition
+ was changed to AST_CDR_NOANSWER). Also, the
+ AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all
+ CDR records are written. The side effects of CDR changes are
+ scary, so I'm documenting the test cases performed to attempt to
+ catch any regressions. The following tests were all performed
+ using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls
+ B (busy) Hangup C Hangup A (Both SIP and features) A calls B A
+ blind transfers to C Hangup C (Both SIP and features) A calls B A
+ attended transfers to C Hangup C A calls B A attended transfers
+ to C (SIP) C blind transfers to A (features) Hangup A All of the
+ test scenario CDRs matched. The following tests were performed
+ just with the patch to ensure proper operation (with
+ unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten
+ =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w)
+ (closes issue #16180) Reported by: aatef Patches: bug16180.patch
+ uploaded by jpeeler (license 325) ........ ................
+
+2009-12-18 22:42 +0000 [r235576-235659] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 235656 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600
+ (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18
+ Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion
+ ........ ................
+
+ * /, configure, configure.ac: Merged revisions 235573 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r235573 | tilghman | 2009-12-18 15:19:43 -0600
+ (Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18
+ Dec 2009) | 2 lines Point to the typical missing package, not the
+ cryptic "termcap support". ........ ................
+
+2009-12-17 23:22 +0000 [r235522] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 235521 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r235521 |
+ file | 2009-12-17 19:21:07 -0400 (Thu, 17 Dec 2009) | 3 lines
+ Remove some old code for going to the 'fax' extension when a T.38
+ switchover occurs. This would have already happened when we
+ detected the CNG tone so this was basically a noop. ........
+
+2009-12-17 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0
+
+2009-12-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc8
+
+2009-12-08 18:33 +0000 [r233731] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 233718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009)
+ | 8 lines Find another ref leak and change how we manage module
+ references. (closes issue #16388) Reported by: parisioa Patches:
+ 20091208__issue16388.diff.txt uploaded by tilghman (license 14)
+ Tested by: parisioa, tilghman Review:
+ https://reviewboard.asterisk.org/r/442/ ........
+
+2009-12-08 18:04 +0000 [r233694] Russell Bryant <russell@digium.com>
+
+ * formats/format_sln16.c, formats/format_wav_gsm.c,
+ formats/format_siren7.c, formats/format_ilbc.c,
+ formats/format_vox.c, formats/format_pcm.c,
+ formats/format_h263.c, formats/format_g723.c,
+ formats/format_h264.c, formats/format_siren14.c,
+ formats/format_jpeg.c, formats/format_g726.c,
+ formats/format_gsm.c, formats/format_g729.c, /,
+ formats/format_sln.c, formats/format_wav.c,
+ formats/format_ogg_vorbis.c: Merged revisions 233692 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009)
+ | 16 lines Set a module load priority for format modules. A
+ recent change to app_voicemail made it such that the module now
+ assumes that all format modules are available while processing
+ voicemail configuration. However, when autoloading modules, it
+ was possible that app_voicemail was loaded before the format
+ modules. Since format modules don't depend on anything, set a
+ module load priority on them to ensure that they get loaded first
+ when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2.
+ The fix for 1.4 and 1.6.0 will require a different approach since
+ the module load priority functionality is not present in the
+ module API. (issue #16412) Reported by: jiddings ........
+
+2009-12-08 07:41 +0000 [r233689] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6.
+
+2009-12-07 23:54 +0000 [r233615] Atis Lezdins <atis@iq-labs.net>
+
+ * contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8
+ lines Fix compatibility with valgrind 3.3 and older. (noticed in
+ issue #16388) Reported by: parisioa Patches: valgrind.supp
+ uloaded by atis (license 242) Tested by: atis, parisioa ........
+
+2009-12-07 23:29 +0000 [r233473-233612] David Vossel <dvossel@digium.com>
+
+ * /, main/utils.c: Merged revisions 233611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 |
+ dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines
+ fixes incorrect logic in ast_uri_encode issue #16299 ........
+
+ * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009)
+ | 15 lines Merged revisions 233471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009)
+ | 9 lines fixes missing Contact header angle brackets (closes
+ issue #16298) Reported by: mgernoth Patches:
+ reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
+ by: dvossel ........ ................
+
+2009-12-07 16:16 +0000 [r233396] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 |
+ mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8
+ lines Do not reject SDP packets describing only non audio
+ streams. (closes issue #16387) Reported by: zalex1953 Patches:
+ media-level-c-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: mnicholson, zalex1953 ........
+
+2009-12-04 21:55 +0000 [r233281] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600
+ (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009)
+ | 7 lines clarify requirecalltoken option in iax.sample.conf
+ (closes issue #16223) Reported by: bklang Patches:
+ clarify-iax-requirecalltoken.patch uploaded by bklang (license
+ 919) ........ ................
+
+2009-12-04 21:07 +0000 [r233240] Matthias Nick <mnick@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 233093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 |
+ mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines
+ Parse global variables or expressions in hint extensions Parse
+ global variables or expressions in hint extensions. Like: exten
+ => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166)
+ Reported by: rmudgett Tested by: mnick, rmudgett ........
+
+2009-12-04 17:36 +0000 [r233165] David Vossel <dvossel@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600
+ (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009)
+ | 6 lines document and rename strip_control() in app_voicemail
+ (closes issue #16291) Reported by: wdoekes ........
+ ................
+
+2009-12-04 17:23 +0000 [r233130] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 233100 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009)
+ | 14 lines Merged revisions 233092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009)
+ | 7 lines Only do frame payload check for HOLD frames. This code
+ was added for helping to debug the source of invalid HOLD frames.
+ However, a side effect of this is that it will incorrectly report
+ errors for frames that have an integer payload. Make the check
+ for this block specific to the HOLD frame case. ........
+ ................
+
+2009-12-04 15:57 +0000 [r233049] Matthias Nick <mnick@digium.com>
+
+ * main/dsp.c, /: Merged revisions 233046 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) |
+ 17 lines Merged revisions 233014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) |
+ 11 lines Warning message gets displayed only once Added
+ additional field 'int display_inband_dtmf_warning', which when
+ set to '1' displays the warning ('Inband DTMF is not supported on
+ codec %s. Use RFC2833'), and when set to '0' doesn't display the
+ warning. Otherwise you would get hundreds of warnings every
+ second. (closes issue #15769) Reported by: falves11 Patches:
+ patch_15769_14.txt uploaded by mnick (license 874) Tested by:
+ mnick, falves11 ........ ................
+
+2009-12-03 21:03 +0000 [r232866] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600
+ (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009)
+ | 8 lines Deprecate "cz" in favor of "cs". Also, change the use
+ of language codes so that language registers as a prefix, rather
+ than an exact match. (closes issue #16272) Reported by: patrol-cz
+ Patches: 20091203__issue16272.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+2009-12-03 15:14 +0000 [r232813] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 |
+ diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12
+ lines Prevent double closing of FDs by EIVR This caused a problem
+ when asterisk was under heavy load and running both AGI and EIVR
+ applications. EIVR would close an FD at which point it would be
+ considered freed and be used by a new AGI instance the second
+ close would then close the FD now in use by AGI. (closes issue
+ #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec
+ Review: https://reviewboard.asterisk.org/r/436/ ........
+
+2009-12-03 00:20 +0000 [r232675-232678] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Oops, really remove it this time
+
+ * res/res_musiconhold.c, /: Recorded merge of revisions
+ 232660-232661 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232660 |
+ tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 Dec 2009) | 19
+ lines Fix multiple issues with musiconhold, which led to classes
+ not getting destroyed properly. * Classes are now tracked past
+ removal from the core container, and module removal is actively
+ prevented until all references are freed. * A hanging reference
+ stored in the channel has been removed. This could have caused a
+ mismatch and the music state not properly cleared, if two or more
+ reloads occurred between MOH being stopped and MOH being
+ restarted. * In certain circumstances, duplicate classes were
+ possible. * A race existed at reload time between a process being
+ killed and the thread responsible for reading from the related
+ pipe respawning that process. * Several reference counts have
+ also been corrected. At least one could have caused deleted
+ classes to stick around forever, consuming resources. This
+ originally manifested as MOH external processes that were not
+ killed at reload time. (closes issue #16279, closes issue #16207)
+ Reported by: parisioa, dcabot Patches:
+ 20091202__issue16279__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: parisioa, tilghman ........ r232661 | tilghman |
+ 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove
+ debugging line ........
+
+2009-12-02 23:28 +0000 [r232658] David Vossel <dvossel@digium.com>
+
+ * CHANGES, /, UPGRADE.txt: Merged revisions 232657 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r232657 | dvossel | 2009-12-02 17:27:45 -0600 (Wed, 02 Dec 2009)
+ | 6 lines update CHANGES and UPGRADE.txt for early media behavior
+ change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by:
+ miki ........
+
+2009-12-02 22:05 +0000 [r232579-232585] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, /: Merged revisions 232582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009)
+ | 14 lines Merged revisions 232581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009)
+ | 7 lines Send ack (response/message) after receiving manager
+ action userevent (closes issue #16264) Reported by: dimas
+ Patches: event-ack.patch uploaded by dimas (license 88) ........
+ ................
+
+ * main/manager.c, /: Merged revisions 232576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 |
+ jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines
+ Make manager response to "Action: events" finish with empty line
+ (closes issue #16275) Reported by: vnovy Patches: manager.c.diff
+ uploaded by vnovy (license 922) ........
+
+2009-12-02 17:11 +0000 [r232359] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) |
+ 12 lines Merged revisions 232355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5
+ lines Fix a bug where if you hung up very quickly after calling
+ AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
+ (closes issue #16239) Reported by: CGMChris ........
+ ................
+
+2009-12-02 17:01 +0000 [r232352] David Vossel <dvossel@digium.com>
+
+ * /, main/acl.c: Merged revisions 232351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009)
+ | 12 lines Merged revisions 232350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009)
+ | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in
+ strace. (closes issue #16290) Reported by: wdoekes ........
+ ................
+
+2009-12-02 16:43 +0000 [r232348] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 |
+ file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add
+ support for handling the 415 Unsupported media type response like
+ we do for a 488 Not acceptable here response. (closes issue
+ #16186) Reported by: atis Patches: sip_t38_response_415.patch
+ uploaded by atis (license 242) ........
+
+2009-12-02 15:43 +0000 [r232270] David Vossel <dvossel@digium.com>
+
+ * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600
+ (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009)
+ | 9 lines fixes segfault in func_groupcount closes issue #16337)
+ Reported by: Parantido Patches: issue_16337.diff uploaded by
+ dvossel (license 671) Tested by: Parantido, dvossel ........
+ ................
+
+2009-12-02 14:55 +0000 [r232232] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 232230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 |
+ file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix
+ a bug where a scheduled item ID would get retained on
+ registrations in a certain scenario causing code to execute
+ during reload that should not. (issue AST-263) ........
+
+2009-12-02 00:52 +0000 [r232094] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600
+ (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009)
+ | 10 lines Do not modify the gain settings on data calls. (The
+ digital flag actually represents a data call.) (closes issue
+ #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis ........
+ ................
+
+2009-12-01 23:40 +0000 [r232011-232015] Russell Bryant <russell@digium.com>
+
+ * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 |
+ russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines
+ Fix a build error on FreeBSD. ........
+
+ * /, main/file.c: Merged revisions 232008 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009)
+ | 9 lines Merged revisions 232007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009)
+ | 2 lines Fix a warning pointed out by buildbot. ........
+ ................
+
+2009-12-01 22:03 +0000 [r231930] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 231927 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009)
+ | 19 lines Merged revisions 231911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009)
+ | 12 lines Fix crash with invalid frame data The crash was
+ happening as a result of a frame containing an invalid data
+ pointer, but was set with data length of zero. The few times the
+ issue was reproduced it _seemed_ that the frame was queued
+ properly, that is the data pointer was set to NULL. I never could
+ reproduce the crash so as a last resort the crash has been fixed,
+ but a check in __ast_read has been added to give as much
+ information about the source of problematic frames in the future.
+ (closes issue #16058) Reported by: atis ........ ................
+
+2009-12-01 21:21 +0000 [r231870] David Vossel <dvossel@digium.com>
+
+ * main/pbx.c, /: Merged revisions 231867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009)
+ | 9 lines Merged revisions 231853 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009)
+ | 3 lines WaitExten m option with no parameters generates frame
+ with zero datalen but non-null data ptr ........ ................
+
+2009-12-01 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc7
+
+2009-12-01 15:48 +0000 [r231743] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/file.c: Merged revisions 231741 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec
+ 2009) | 9 lines Merged revisions 231740 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec
+ 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce()
+ and return an error if no know formats are found. ........
+ ................
+
+2009-11-30 21:59 +0000 [r231695-231696] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 231692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 |
+ kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22
+ lines Another round of UDPTL stack fixes/improvements: 1) Allow
+ users of UDPTL stack to associate a character-string tag with a
+ UDPTL session, so that log/error/debug messages generated by the
+ UDPTL stack can be 'connected' to the endpoint that caused them
+ to be generated. 2) Improve comments (and process) of calculating
+ the far end's maximum IFP size when redundancy mode is in use for
+ error correction. 3) When an IFP larger than the calculated 'far
+ max IFP' size is presented for writing, truncate it rather than
+ putting in the buffer and allowing the buffer to overflow; this
+ will cause the ends to retrain to a lower bit rate that produces
+ IFPs of an appropriate size if possible, and if not possible, the
+ FAX transfer will fail completely. In these cases, it is due to
+ the one endpoint supplying a T38FaxMaxDatagram value that is
+ improperly calculated and is too low to be of use; we have
+ configuration options available to override this behavior. 4)
+ Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no
+ longer needed. ........
+
+ * pbx/pbx_config.c: Backport a tiny fix from trunk that makes GCC
+ 4.4.x happier.
+
+2009-11-30 21:36 +0000 [r231689] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c,
+ main/app.c: Merged revisions 231688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov
+ 2009) | 15 lines Merged revisions 231614 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
+ 2009) | 8 lines Remove duplicate entries from voicemail format
+ lists. This prevents app_voicemail from entering an infinite loop
+ when the same format is specified twice in the format list.
+ (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/429/ ........
+ ................
+
+2009-11-30 20:47 +0000 [r231605] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 |
+ file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines
+ When receiving SDP that matches the version of the last one do
+ not treat it as a fatal error. (closes issue #16238) Reported by:
+ seandarcy ........
+
+2009-11-30 18:57 +0000 [r231505-231558] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 231556 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 |
+ dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines
+ app_queue crashes randomly, often during call-transfers This
+ patch adds a ref to the queue_ent object's parent call_queue in
+ queue_exec() so the call_queue won't be destroyed while the the
+ queue_ent still holds a pointer to it. (closes issue 0015686)
+ Tested by: dvossel, aragon ........
+
+ * main/rtp.c, /: Merged revisions 231491 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009)
+ | 17 lines Merged revisions 231441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009)
+ | 11 lines fixes crash caused by RTP comfort noise payload
+ greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
+ by: amorsen Patches: issue16242.diff uploaded by oej (license
+ 306) Tested by: amorsen, oej, dvossel ........ ................
+
+2009-11-25 22:34 +0000 [r231302] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 231299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009)
+ | 9 lines Merged revisions 231298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009)
+ | 2 lines After a frame duplication failure, unlock the channel
+ before returning. ........ ................
+
+2009-11-25 15:48 +0000 [r231191] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 |
+ mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4
+ lines Load pbx_lua with global symbols to allow linking with
+ other lua libraries. Found by Maxim Litnitskiy. ........
+
+2009-11-24 20:36 +0000 [r231136] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 231134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 |
+ tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines
+ Found a few places where queue refcounts were counted
+ incorrectly. Also add debug statements. (closes issue #15982,
+ closes issue #15984) Reported by: atis Patches:
+ 20091111__issue15982.diff.txt uploaded by tilghman (license 14)
+ Tested by: atis ........
+
+2009-11-24 18:54 +0000 [r231098] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 231095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 |
+ jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines
+ Fix erroneous hangup extension execution ast_spawn_extension
+ behaves differently from 1.4 in that hangups and extensions that
+ do not exist do not return an error, whereas in 1.6 it does. This
+ is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN
+ flag gets set properly. (closes issue #16106) Reported by:
+ ajohnson Tested by: ajohnson ........
+
+2009-11-23 15:48 +0000 [r230884] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 230881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 |
+ file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines
+ Change fax detection in chan_sip so it behaves as one would
+ expect. Internally the way T.38 is negotiated has changed and the
+ option no longer reflects a behavior that is valid. It will now
+ look for a CNG tone on received calls and if present send the
+ call to the 'fax' extension. It is then up to the application or
+ channel to request the switch over to T.38. ........
+
+2009-11-23 15:38 +0000 [r230796-230880] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov
+ 2009) | 9 lines Merged revisions 230839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov
+ 2009) | 1 line Correct fix for issue #16268... the reporter's
+ original patch was very close to correct. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov
+ 2009) | 12 lines Merged revisions 230772 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov
+ 2009) | 5 lines Ensure that SDP parsing does not ignore the last
+ line of the SDP. (closes issue #16268) Reported by: sgimeno
+ ........ ................
+
+2009-11-20 22:36 +0000 [r230727] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 230726 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009)
+ | 7 lines fixes iax2 show cache locking error, thanks alecdavis!
+ (closes issue #16094) Reported by: alecdavis Patches:
+ bug16094.diff.txt uploaded by alecdavis (license 585) Tested by:
+ alecdavis, dvossel ........
+
+2009-11-20 21:07 +0000 [r230629] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 230628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov
+ 2009) | 15 lines Merged revisions 230627 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov
+ 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR
+ if it exists. This is necessary for the recordagentcalls option
+ in chan_agent to store the recorded file name in the bridge CDR.
+ (closes issue #14590) Reported by: msetim Patches:
+ queue_agent_userfield.patch uploaded by Laureano (license 265)
+ Tested by: Laureano, mnicholson ........ ................
+
+2009-11-20 17:31 +0000 [r230510-230585] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c, /, include/asterisk/audiohook.h: Merged
+ revisions 230583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 |
+ dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines
+ audiohook signal trigger on every status change (issue #14618)
+ Review: https://reviewboard.asterisk.org/r/434/ ........
+
+ * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600
+ (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009)
+ | 10 lines fixes MixMonitor thread not exiting when
+ StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
+ Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, AlexMS Review:
+ https://reviewboard.asterisk.org/r/424/ ........ ................
+
+2009-11-16 16:41 +0000 [r230250-230384] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 230381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230381 |
+ kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line
+ Fix another buglet in T.38 session teardown at the end of FAX
+ sessions. ........
+
+ * /, apps/app_fax.c: Merged revisions 230343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230343 |
+ kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2
+ lines Ensure that only one end of a T.38 session initiates
+ teardown at completion. ........
+
+ * channels/chan_iax2.c, /: Merged revisions 230247 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r230247 | kpfleming | 2009-11-15 11:23:02 -0600
+ (Sun, 15 Nov 2009) | 12 lines Merged revisions 230246 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov
+ 2009) | 6 lines Correct mistaken option name in error message.
+ The configuration option for allowing hosts to make
+ non-token-based calls is 'calltokenoptional', not
+ 'calltokenignore'. (reported on asterisk-users) ........
+ ................
+
+2009-11-13 22:01 +0000 [r229969-230148] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 230145 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) |
+ 15 lines Merged revisions 230144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8
+ lines Respect the maddr parameter in the Via header. (closes
+ issue #14446) Reported by: frawd Patches: via_maddr.patch
+ uploaded by frawd (license 610) Tested by: frawd ........
+ ................
+
+ * channels/chan_local.c, /: Merged revisions 230039 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r230039 | file | 2009-11-13 13:44:53 -0600 (Fri,
+ 13 Nov 2009) | 16 lines Merged revisions 230038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9
+ lines Fix a crash caused by two threads thinking they should both
+ free the chan_local private structure when only one should.
+ (closes issue #15314) Reported by: sroberts Patches:
+ Issue15314_Move_Nulling_owner.patch uploaded by davidw (license
+ 780) Tested by: davidw, lottc ........ ................
+
+ * configs/extensions.conf.sample, /, apps/app_chanisavail.c: Merged
+ revisions 229966 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) |
+ 13 lines Merged revisions 229965 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6
+ lines Document a limitation in the AVAILSTATUS variable from
+ ChanIsAvail and provide a workaround for it that does not change
+ existing behavior. (closes issue #14426) Reported by: macli
+ ........ ................
+
+2009-11-13 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc6
+
+2009-11-13 15:57 +0000 [r229915] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 |
+ file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix
+ T.38 negotiation regression introduced with the SDP parser
+ changes. ........
+
+2009-11-12 23:31 +0000 [r229752] Jason Parker <jparker@digium.com>
+
+ * channels/chan_oss.c, /: Merged revisions 229750 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 |
+ qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix
+ mute toggling on OSS channels. ........
+
+2009-11-12 16:47 +0000 [r229671] David Vossel <dvossel@digium.com>
+
+ * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600
+ (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009)
+ | 6 lines fixes merging error, datastore was being freed in the
+ wrong function. (closes issue #16219) Reported by: aragon
+ ........ ................
+
+2009-11-11 20:49 +0000 [r229570] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229568 |
+ diruggles | 2009-11-11 15:47:06 -0500 (Wed, 11 Nov 2009) | 9
+ lines Remove non-functional feature from ExternalIVR
+ documentation Remove non-functional socket implementation of
+ ExternalIVR from documentation (closes issue #16225) Reported by:
+ thedavidfactor Patches: externalivr.txt.20091111.1542.patch
+ uploaded by thedavidfactor (license 903) ........
+
+2009-11-11 19:56 +0000 [r229492-229502] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Merged revisions 229499 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009)
+ | 15 lines Merged revisions 229498 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009)
+ | 8 lines Solaris doesn't like NULL going to ast_log Solaris will
+ crash if NULL is passed to ast_log. This simple patch simply uses
+ S_OR to get around this. (closes issue #15392) Reported by:
+ yrashk ........ ................
+
+ * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009)
+ | 7 lines Flags not initialized in app_softhangup.c, causing
+ undefined behavior Trivial patch [kobaz] to initialize an
+ ast_flags = {0} (closes issue #16129) Reported by: kobaz ........
+
+2009-11-10 22:17 +0000 [r229366] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 229361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009)
+ | 19 lines Merged revisions 229360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009)
+ | 12 lines If two pattern classes start with the same digit and
+ have the same number of characters, they will compare equal. The
+ example given in the issue report is that of [234] and [246],
+ which have these characteristics, yet they are clearly not
+ equivalent. The code still uses these two characteristics, yet
+ when the two scores compare equal, an additional check will be
+ done to compare all characters within the class to verify
+ equality. (closes issue #15421) Reported by: jsmith Patches:
+ 20091109__issue15421__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: jsmith, thedavidfactor ........ ................
+
+2009-11-10 22:04 +0000 [r229359] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov
+ 2009) | 16 lines Merged revisions 229355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov
+ 2009) | 9 lines Fix ExternalIVR Documentation Remove
+ documentation for event that doesn't function (closes issue
+ #16220) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
+ (license 903) ........ ................
+
+2009-11-10 21:33 +0000 [r229354] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 |
+ tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines
+ When GOSUB is invoked within an AGI, it may not exit correctly.
+ (closes issue #16216) Reported by: atis Patches:
+ 20091110__atis_work.diff.txt uploaded by tilghman (license 14)
+ Tested by: atis ........
+
+2009-11-10 20:09 +0000 [r229285] Joshua Colp <jcolp@digium.com>
+
+ * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) |
+ 15 lines Merged revisions 229281 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8
+ lines Remove broken support for direct transcoding between G.726
+ RFC3551 and G.726 AAL2. On some systems the translation core
+ would actually consider g726aal2 -> g726 -> signed linear to be a
+ quicker path then g726aal2 -> signed linear which exposed this
+ problem. (closes issue #15504) Reported by: globalnetinc ........
+ ................
+
+2009-11-10 17:52 +0000 [r229232] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 229168 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600
+ (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009)
+ | 9 lines don't crash on log message in solaris AST-2009-006
+ (closes issue #16206) Reported by: bklang Tested by: bklang
+ ........ ................
+
+2009-11-10 17:39 +0000 [r229231] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229228 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov
+ 2009) | 18 lines Merged revisions 229191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov
+ 2009) | 11 lines Document ExternalIVR event tag collision
+ ExternalIVR uses the D tag for two different event types. This
+ documents that behavior and how to differentiate between the two
+ cases. Also includes a minor spelling fix and clarification
+ (closes issue #16211) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
+ (license 903) ........ ................
+
+2009-11-10 15:47 +0000 [r229101] Matthew Nicholson <mnicholson@digium.com>
+
+ * UPGRADE-1.6.txt, main/editline/makelist.in, UPGRADE.txt: Reset
+ props that were accidently deleted in 229088.
+
+2009-11-10 15:28 +0000 [r229094] David Vossel <dvossel@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 229093 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r229093 | dvossel | 2009-11-10 09:27:45 -0600 (Tue, 10 Nov 2009)
+ | 11 lines fixes pgsql double free of threadstorage A thread
+ storage variable was being freed incorrectly, which resulted in a
+ double free if two queries were made in the same thread. (closes
+ issue #16011) Reported by: cristiandimache Patches:
+ issue16011.diff uploaded by dvossel (license 671) ........
+
+2009-11-10 15:16 +0000 [r229088] Matthew Nicholson <mnicholson@digium.com>
+
+ * UPGRADE-1.6.txt, main/editline/makelist.in, channels/chan_sip.c,
+ UPGRADE.txt: Reverted revision 202007. (closes issue #16175)
+ Reported by: paul-tg Tested by: paul-tg
+
+2009-11-10 11:25 +0000 [r229078] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10
+ Nov 2009) | 20 lines Schema file additions * Added
+ AsteriskDialplan, AsteriskAccount and AsteriskMailbox
+ objectClasses to allow standalone dialplan, account and mailbox
+ entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage,
+ AstAccountTransport, AstAccountPromiscRedir, -
+ AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
+ - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed
+ redundant IPaddr (there's already IPAddress) - Gives more
+ configuration Flags for SIP-Users available (tested) - Allows to
+ create Asterisk Attributes in defined Asterisk ObjectClasses
+ without extensibleObject (which really should be the last
+ resort); gives also additional possibilities for LDAP-filter
+ (closes issue #15874) Reported by: Medozas Patches:
+ asterisk.ldap-schema.patch uploaded by Medozas (license 41)
+ Tested by: Medozas, suretec ........
+
+2009-11-09 22:59 +0000 [r229017] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 229015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009)
+ | 8 lines Don't crash when bridge->tech_pvt == NULL This is a
+ similar solution to what is in place for chan_agent (closes issue
+ #16003) Reported by: atis Tested by: twilson ........
+
+2009-11-09 22:17 +0000 [r229012] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes segfault when transferring a queue
+ caller In sip_hangup we attempted to lock p->owner after we set
+ it to NULL. Thanks to fhackenberger for reporting the issue and
+ submitting a patch. (closes issue #15848) Reported by:
+ fhackenberger Patches: digium_bug_0015848 uploaded by
+ fhackenberger (license 592) Tested by: fhackenberger, lmadsen,
+ TomS, shin-shoryuken, dvossel
+
+2009-11-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc5
+
+2009-11-09 15:40 +0000 [r228900] Leif Madsen <lmadsen@digium.com>
+
+ * main/channel.c: Merged revisions 228897 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009)
+ | 14 lines Merged revisions 228896 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
+ | 6 lines Update WARNING message. Update a WARNING message to
+ give a suggested fix when encountered. (closes issue #16198)
+ Reported by: atis Tested by: atis ........ ................
+
+2009-11-09 14:48 +0000 [r228859] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600
+ (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov
+ 2009) | 8 lines Perform limited bounds checking when destroying
+ ast_mutex_t structures to make sure we don't try to use negative
+ indices. (closes issue #15588) Reported by: zerohalo Patches:
+ 20090820__issue15588.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........ ................
+
+2009-11-06 22:37 +0000 [r228694] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 228693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009)
+ | 16 lines Merged revisions 228692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
+ | 9 lines fixes audiohook write crash occuring in chan_spy
+ whisper mode. After writing to the audiohook list in ast_write(),
+ frames were being freed incorrectly. Under certain conditions
+ this resulted in a double free crash. (closes issue #16133)
+ Reported by: wetwired ........ ................
+
+2009-11-06 20:26 +0000 [r228649] Matthew Nicholson <mnicholson@digium.com>
+
+ * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600
+ (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
+ 2009) | 8 lines Properly handle '=' while decoding base64
+ messages and null terminate strings returned from BASE64_DECODE.
+ (closes issue #15271) Reported by: chappell Patches:
+ base64_fix.patch uploaded by chappell (license 8) Tested by:
+ kobaz ........ ................
+
+2009-11-06 18:43 +0000 [r228551] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) |
+ 11 lines Merged revisions 228547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4
+ lines Don't overwrite caller ID name on a trunk with the
+ configured fullname when using users.conf (issue ABE-1989)
+ ........ ................
+
+2009-11-06 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc4
+
+2009-11-06 17:54 +0000 [r228504] Joshua Colp <jcolp@digium.com>
+
+ * doc/tex/localchannel.tex, /: Merged revisions 228499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2
+ lines Fix the localchannel.tex file. ........
+
+2009-11-06 17:24 +0000 [r228421-228447] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_ilbc.c, /: Merged revisions 228441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 |
+ dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines
+ Fixes merging issue from 1.4, frame data is held in data.ptr in
+ trunk ........
+
+ * codecs/codec_ilbc.c, /: Merged revisions 228420 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009)
+ | 19 lines Merged revisions 228418 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
+ | 13 lines fixes segfault in iLBC For reasons not yet known, it
+ appears possible for an ast_frame to have a datalen greater than
+ zero while the actual data is NULL during Packet Loss
+ Concealment. Most codecs don't support PLC so this doesn't affect
+ them. This patch catches the malformed frame and prevents the
+ crash from occuring. Additional efforts to determine why it is
+ possible for a frame to look like this are still being
+ investigated. (issue #16979) ........ ................
+
+2009-11-06 16:44 +0000 [r228413] Joshua Colp <jcolp@digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) |
+ 14 lines Merged revisions 228409 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
+ lines Fix a bug caused by a partially invalid frame (from the
+ jitterbuffer) passing through the Asterisk core. (closes issue
+ #15560) Reported by: jvandal (closes issue #15709) Reported by:
+ covici ........ ................
+
+2009-11-06 15:43 +0000 [r228269-228340] David Vossel <dvossel@digium.com>
+
+ * /, main/astfd.c: Merged revisions 228339 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009)
+ | 12 lines Merged revisions 228338 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
+ | 5 lines fixes crash in astfd.c (closes issue #15981) Reported
+ by: slavon ........ ................
+
+ * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06
+ Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c
+ (closes issue #15394) Reported by: boroda Patches:
+ bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790)
+ Tested by: dbrooks, boroda ........
+
+2009-11-05 22:13 +0000 [r228198] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 |
+ tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines
+ Yet another error message in the dialplan (thanks,
+ rmudgett/russellb) ........
+
+2009-11-05 21:27 +0000 [r228195] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 |
+ jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
+ Fix the fix for chanspy option o In 224178, I assumed the
+ uploaded patch was correct as it had received positive feedback.
+ The flags were being checked in the incorrect location. Upon
+ testing the fix this time it was also found that the flags from
+ the dialplan weren't being copied to the
+ chanspy_translation_helper. (closes issue #16167) Reported by:
+ marhbere ........
+
+2009-11-05 21:27 +0000 [r228194] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 |
+ tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines
+ MEETME_INFO should not return a literal error message to the
+ dialplan. (closes issue #15450) Reported by: JimVanM Patches:
+ meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested
+ by: JimVanM ........
+
+2009-11-05 19:42 +0000 [r228148] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600
+ (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009)
+ | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to
+ chan_misdn connection. Patch submitted by gknispel_proformatique,
+ tested by francesco_r. "I have many crash since i have upgraded
+ to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out
+ an ast_frame. (closes issue #16041) Reported by: francesco_r
+ ........ ................
+
+2009-11-05 19:20 +0000 [r228093] Jason Parker <jparker@digium.com>
+
+ * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228080 | qwell | 2009-11-05 13:16:29 -0600
+ (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) |
+ 8 lines Fix crash on VPB exception when no hardware is present.
+ (closes issue #14970) Reported by: tzafrir Patches:
+ vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+ markwaters ........ ................
+
+2009-11-05 17:14 +0000 [r228017] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_externalivr.c, /: Merged revisions 228015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009)
+ | 4 lines Don't crash if no arguments are passed. (closes issue
+ #16119) Reported by: thedavidfactor ........
+
+2009-11-04 23:53 +0000 [r227947] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009)
+ | 21 lines Merged revisions 227944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
+ | 14 lines Fix incorrect filename comparsion after monitor file
+ change The logic to detect if a requested file is indeed a
+ different file from the current file was incorrect. The main
+ issue being confusion of the use of filename_base which was
+ previously set without pathing information and then compared to
+ another full path. Robust file comparison logic has been added to
+ properly check if two files are the same even if symlinks are
+ used. (closes issue #15313) Reported by: caspy Patches:
+ 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
+ 325) but mostly tilghman's work ........ ................
+
+2009-11-04 21:09 +0000 [r227760-227831] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov
+ 2009) | 17 lines Merged revisions 227827 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
+ 2009) | 10 lines This patch modifies the Dial application to
+ monitor the calling channel for hangups while playing back
+ announcements. (closes issue #16005) Reported by: falves11
+ Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson, falves11 Review:
+ https://reviewboard.asterisk.org/r/407/ ........ ................
+
+ * channels/chan_sip.c: Modify the SDP parsing code to parse session
+ and media level items separately. With the new code, media level
+ proprieties should no longer be confused with session level
+ proprieties. This change also reorganizes some of the SDP parsing
+ code which should make it easier to manage in the future. (closes
+ issue #14994) Reported by: frawd
+
+2009-11-04 19:28 +0000 [r227733-227748] Joshua Colp <jcolp@digium.com>
+
+ * /, static-http/prototype.js: Merged revisions 227739 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed,
+ 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5
+ lines Fix a security issue where it may be possible for someone
+ to execute a cross-site AJAX request exploit. (AST-2009-009)
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) |
+ 12 lines Merged revisions 227700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
+ lines Fix a security issue where sending a REGISTER with a
+ differing username in the From URI and Authorization header would
+ reveal whether it was valid or not. (AST-2009-008) ........
+ ................
+
+2009-11-03 20:01 +0000 [r227375] Jason Parker <jparker@digium.com>
+
+ * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) |
+ 9 lines Fix some build issues on Solaris. (closes issue #14517)
+ (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded
+ by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell
+ ........
+
+2009-11-03 19:49 +0000 [r227364-227371] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_controlplayback.c, /: Merged revisions 227368 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03
+ Nov 2009) | 8 lines Change warning message to debug message.
+ app_controlplayback outputs a warning, when in fact it is normal.
+ (closes issue #16071) Reported by: atis Patches:
+ controlplayback_warning.patch uploaded by atis (license 242)
+ ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 227361 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03
+ Nov 2009) | 11 lines Additional fixes to the
+ extensions.conf.sample file. Update the extensions.conf.sample
+ [stdexten] context so that we use the variable instead of
+ requiring it to be passed explicitly. Also updated uses of the
+ [stdexten] context throughout. (closes issue #15858) Reported by:
+ pprindeville Patches: stdexten-context-update.txt uploaded by
+ lmadsen (license 10) Tested by: pprindeville ........
+
+2009-11-03 18:15 +0000 [r227280] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
+ | 4 lines Make sure the outgoing flag is cleared if a new channel
+ fails to get created for outgoing calls. This is the relevant
+ portion of asterisk/trunk -r226648 ........
+
+2009-11-03 17:14 +0000 [r227239] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 |
+ dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines
+ user.conf entries in SIP were not having their peer type set.
+ (closes issue #16120) Reported by: jsmith ........
+
+2009-11-03 15:40 +0000 [r227170] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) |
+ 12 lines Merged revisions 227166 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
+ lines Fix a bug where an RPID header could be generated with a
+ blank username in the URI. (closes issue #15909) Reported by:
+ kobaz ........ ................
+
+2009-11-03 15:25 +0000 [r227165] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 227162 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03
+ Nov 2009) | 7 lines Update extensions.conf.sample file to fix
+ incorrect extensions. (closes issue #15857) Reported by:
+ pprindeville Patches: stdexten.patch#2 uploaded by pprindeville
+ (license 347) Tested by: pprindeville ........
+
+2009-11-03 13:51 +0000 [r227156] Olle Johansson <oej@edvina.net>
+
+ * Makefile, /, channels/chan_sip.c: Merged revisions 227091 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis,
+ 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
+ lines Use proper response code when violating Contact ACL's.
+ https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
+ quick review. (EDVX-003) ........ ................
+
+2009-11-02 21:06 +0000 [r226978] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_sip.c: SIP channel name uniqueness SIP channel
+ names were supposed to be unique by way of a name suffix derived
+ from the pointer to the channel's private data. Uniqueness was
+ preserved on 32-bit systems, but not on 64-bit systems. This
+ patch, as suggested by kpfleming, replaces this suffix with a
+ simple incremented unsigned int. (closes issue #15152) Reported
+ by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
+
+2009-11-02 18:12 +0000 [r226893] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) |
+ 18 lines Merged revisions 226889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
+ 11 lines Fix a bug where the recorded privacy introduction file
+ would not get removed if the caller hung up while the called
+ party had not yet answered. This was fixed by introducing an
+ argument to the 'n' option which, when enabled, removes the
+ introduction file under all scenarios. This was done to preserve
+ the behavior that has existed for quite some time. (closes issue
+ #14674) Reported by: ulogic Patches: bug14674.patch uploaded by
+ jpeeler (license 325) ........ ................
+
+2009-11-02 17:17 +0000 [r226815] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600
+ (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
+ | 8 lines Don't allow two separate instances of safe_asterisk
+ when restarting from the init script. (closes issue #14562)
+ Reported by: davidw Patches: Initially
+ 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
+ Modified to 20091030__Issue14562_diff.txt uploaded by davidw
+ (license 780) Tested by: davidw ........ ................
+
+2009-10-29 18:18 +0000 [r226540] Joshua Colp <jcolp@digium.com>
+
+ * doc/tex/localchannel.tex, channels/chan_local.c, /: Merged
+ revisions 226532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) |
+ 13 lines Merged revisions 226531 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
+ lines Add an option to enabling passing music on hold start and
+ stop requests through instead of acting on them in chan_local.
+ (closes issue #14709) Reported by: dimas ........
+ ................
+
+2009-10-28 21:32 +0000 [r226486] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * build_tools/get_documentation, /: remove empty awk pattern (//)
+ Solaris 10 nawk doesn't like the empty pattern such as '//' for
+ 'always'. Just remove that. No pattern at all always matches.
+ Merged revisions 226453 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk
+
+2009-10-28 20:13 +0000 [r226379-226385] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Merged revisions 226384 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500
+ (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009)
+ | 9 lines Update documentation in sip.conf.sample. Update the
+ documentation in sip.conf.sample in order to make it more clear
+ that directmedia/canreinvite do not cause Asterisk to ignore
+ reINVITEs. It is only used to stop Asterisk from generating a
+ reINVITE, but does not stop it from accepting them if necessary.
+ (closes issue #15644) Reported by: lmadsen ........
+ ................
+
+ * doc/tex/channelvariables.tex: Merged revisions 226378 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500
+ (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
+ | 7 lines Update CALLINGSUBADDR channel variable documentation.
+ (closes issue #15734) Reported by: alecdavis Patches:
+ channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis ........ ................
+
+2009-10-28 18:06 +0000 [r226170-226308] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 226305 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500
+ (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28
+ Oct 2009) | 2 lines Fix documentation (pointed out by
+ TheDavidFactor on #-dev) ........ ................
+
+ * main/manager.c, /: Merged revisions 226159 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009)
+ | 14 lines Merged revisions 226138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
+ | 7 lines Manager output is not always NULL-terminated, so force
+ a NULL at the end of the filestream. (closes issue #15495)
+ Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
+ by tilghman (license 14) Tested by: pdf ........ ................
+
+2009-10-27 17:12 +0000 [r226101] Terry Wilson <twilson@digium.com>
+
+ * res/res_http_post.c, /: Merged revisions 226099 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 |
+ twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines
+ Don't prepend the URI prefix to the post directory ........
+
+2009-10-27 00:16 +0000 [r226055] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configure, configure.ac: detect ARM Linux EABI OSARCH as
+ linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
+ if host_os is linux-gnueabi * When checking if we are Linux,
+ check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
+ the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
+ sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
+ tested for the value of 'linux-gnu' in one or two places in the
+ tree. This patch also fixes the check libcap to check for $OSARCH
+ rather than $host_os . See also:
+ http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
+ svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4
+ Merged revisions 226018 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk
+
+2009-10-26 19:42 +0000 [r225914] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 |
+ jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines
+ ACL check not present for verifying SIP INVITEs The ACL check in
+ check_peer_ok was missing and has now been restored. The missing
+ check allowed for calls to be made on prohibited networks where
+ an ACL was defined in sip.conf and the allowguest option was set
+ to off. See the AST security advisory below for more information.
+ Merge code associated with AST-2009-007. (closes issue #16091)
+ Reported by: thom4fun ........
+
+2009-10-26 15:56 +0000 [r225871] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Backport audio handling loop fixes from trunk
+ version of app_fax. This backport resolves some issues handling
+ audio frames during FAX processing, and ensures that the FAX
+ application doesn't accidentally get notified of a T.38
+ switchover at the end of a successful FAX. (closes issue #16127)
+
+2009-10-23 14:46 +0000 [r225651] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 |
+ dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines
+ Fixes an iterator memory leak and uninitialized memory ........
+
+2009-10-23 14:08 +0000 [r225585] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 225582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct
+ 2009) | 17 lines Merged revisions 225581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
+ 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
+ every build. For some reason the menuselect.makeopts file was
+ listed as PHONY in the Makefile, resulting in 'make' needing to
+ rebuild it for every build. This then resulted in the embedded
+ module rules being rebuilt on every build, which can be slow and
+ is unnecessary. This patch fixes the problem by properly allowing
+ 'make' to know when the menuselect.makeopts file needs to be
+ rebuilt (defining the proper dependencies). ........
+ ................
+
+2009-10-22 22:24 +0000 [r225516] Leif Madsen <lmadsen@digium.com>
+
+ * README, /: Merged revisions 225515 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225515 |
+ lmadsen | 2009-10-22 17:24:03 -0500 (Thu, 22 Oct 2009) | 8 lines
+ Update README documentation. Update the README documentation to
+ correctly describe which CLI command you should use when
+ attempting to get help from the CLI. (closes issue #16064)
+ Reported by: thedavidfactor Patches: readme.patch uploaded by
+ thedavidfactor (license 903) ........
+
+2009-10-22 21:55 +0000 [r225489] David Vossel <dvossel@digium.com>
+
+ * apps/app_externalivr.c, include/asterisk/tcptls.h, main/tcptls.c,
+ /, channels/chan_sip.c: Merged revisions 225445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 |
+ dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines
+ SIP TCP/TLS: move client connection setup/write into tcp helper
+ thread, various related locking/memory fixes. What this patch
+ fixes 1.Moves sip TCP/TLS connection setup into the TCP helper
+ thread: Connection setup takes awhile and before this it was
+ being done while holding the monitor lock. 2.Moves TCP/TLS
+ writing to the TCP helper thread: Through the use of a packet
+ queue and an alert pipe, the TCP helper thread can now be woken
+ up to write data as well as read data. 3.Locking error: sip_xmit
+ returned an XMIT_ERROR without giving up the tcptls_session lock.
+ This lock has been completely removed from sip_xmit and placed in
+ the new sip_tcptls_write() function. 4.Memory leak: When creating
+ a tcptls_client the tls_cfg was alloced but never freed unless
+ the tcptls_session failed to start. Now the session_args for a
+ sip client are an ao2 object which frees the tls_cfg on
+ destruction. 5.Pointer to stack variable: During
+ sip_prepare_socket the creation of a client's
+ ast_tcptls_session_args was done on the stack and stored as a
+ pointer in the newly created tcptls_session. Depending on the
+ events that followed, there was a slight possibility that pointer
+ could have been accessed after the stack returned. Given the new
+ changes, it is always accessed after the stack returns which is
+ why I found it. Notable code changes 1.I broke tcptls.c's
+ ast_tcptls_client_start() function into two functions. One for
+ creating and allocating the new tcptls_session, and a separate
+ one for starting and handling the new connection. This allowed me
+ to create the tcptls_session, launch the helper thread, and then
+ establish the connection within the helper thread. 2.Writes to a
+ tcptls_session are now done within the helper thread. This is
+ done by using an alert pipe to wake up the thread if new data
+ needs to be sent. The thread's sip_threadinfo object contains the
+ alert pipe as well as the packet queue. 3.Since the threadinfo
+ object contains the alert pipe, it must now be accessed outside
+ of the helper thread for every write (queuing of a packet). For
+ easy lookup, I moved the threadinfo objects from a linked list to
+ an ao2_container. (closes issue #13136) Reported by: pabelanger
+ Tested by: dvossel, whys (closes issue #15894) Reported by:
+ dvossel Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/380/ ........
+
+2009-10-22 21:54 +0000 [r225488] Leif Madsen <lmadsen@digium.com>
+
+ * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions
+ 225485 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009)
+ | 19 lines Merged revisions 225484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
+ | 11 lines Clean valgrind output by suppressing false errors.
+ Update valgrind.txt documentation and add valgrind.supp file in
+ order to allow those who are creating valgrind output to have
+ less false errors in the logfile. (closes issue #16007) Reported
+ by: atis Patches: valgrind.txt.diff uploaded by atis (license
+ 242) asterisk2.supp uploaded by atis (license 242) Tested by:
+ atis, amorsen ........ ................
+
+2009-10-22 17:14 +0000 [r225363] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
+ Merged revisions 225360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009)
+ | 11 lines Merged revisions 225105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
+ | 4 lines Fix documentation for ast_softhangup() and correct the
+ misuse thereof. (closes issue #16103) Reported by: majorbloodnok
+ ........ ................
+
+2009-10-21 22:00 +0000 [r225035-225308] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 225307 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500
+ (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009)
+ | 13 lines IAX2: VNAK loop caused by signaling frames with no
+ destination call number It is possible for the PBX thread to
+ queue up signaling frames before a destination call number is
+ received. This can result in signaling frames being sent out with
+ no destination call number. Since recent versions of Asterisk
+ require accurate destination callnumbers for all Full Frames,
+ this can cause a VNAK loop to occur. To resolve this no signaling
+ frames are sent until a destination callnumber is received, and
+ destination call numbers are now only required for iax_pvt
+ matching when the frame is an ACK. Review:
+ https://reviewboard.asterisk.org/r/413/ ........ ................
+
+ * configs/sip.conf.sample, channels/chan_iax2.c,
+ configs/iax.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 225033 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009)
+ | 27 lines Merged revisions 225032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
+ | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
+ id removes '(', ' ', ')', non-trailing '.', and '-' from the
+ string. This means values such as 555.5555 and test-test result
+ in 555555 and testtest. There are instances, such as Skype
+ integration, where a specific value is passed via caller id that
+ must be preserved unmodified. This patch makes the shrinking of
+ caller id optional in chan_sip and chan_iax in order to support
+ such cases. By default this option is on to preserve previous
+ expected behavior. (closes issue #15940) Reported by: dimas
+ Patches: v2-15940.patch uploaded by dimas (license 88)
+ 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/408/ ........ ................
+
+2009-10-20 22:11 +0000 [r224859] Tilghman Lesher <tlesher@digium.com>
+
+ * main/audiohook.c, funcs/func_speex.c, /: Merged revisions 224856
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500
+ (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
+ | 5 lines Pay attention to the return value of the manipulate
+ function. While this looks like an optimization, it prevents a
+ crash from occurring when used with certain audiohook callbacks
+ (diagnosed with SVN trunk, backported to 1.4 to keep the source
+ consistent across versions). ........ ................
+
+2009-10-20 17:50 +0000 [r224777] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 224774 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) |
+ 12 lines Merged revisions 224773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
+ lines Add support for relaying early media in the features
+ attended transfer option. (closes issue #14828) Reported by:
+ licedey ........ ................
+
+2009-10-20 00:00 +0000 [r224674] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp.c, /: Merged revisions 224671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct
+ 2009) | 14 lines Merged revisions 224670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct
+ 2009) | 7 lines Correct timestamp calculations when RTP sample
+ rates over 8kHz are used. While testing some endpoints that
+ support 16kHz and 32kHz sample rates, some log messages were
+ generated due to calc_rxstamp() computing timestamps in a way
+ that produced odd results, so this patch sanitizes the result of
+ the computations. ........ ................
+
+2009-10-19 19:54 +0000 [r224571] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) |
+ 12 lines Merged revisions 224565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
+ lines Do not attempt early media bridging (ie: direct RTP setup)
+ if options are enabled that should prevent it. (closes issue
+ #14763) Reported by: cupotka ........ ................
+
+2009-10-19 19:41 +0000 [r224563] Kevin P. Fleming <kpfleming@digium.com>
+
+ * formats/format_siren14.c, /: Merged revisions 224562 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r224562 | kpfleming | 2009-10-19 14:40:26 -0500 (Mon, 19 Oct
+ 2009) | 1 line Remove useless debugging message. ........
+
+2009-10-19 00:13 +0000 [r224447-224451] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009)
+ | 3 lines Allow ODBC storage to be queried with multiple
+ mailboxes, and remove multiple goto's. This corrects an issue
+ reported on the -users list. ........
+
+ * configs/res_odbc.conf.sample, /: Merged revisions 224446 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18
+ Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias
+ for "autocommit", but instead controls the default disposition of
+ uncommitted transactions. ........
+
+2009-10-17 01:58 +0000 [r224334] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500
+ (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
+ | 13 lines Fix stale caller id data from being reported in AMI
+ NewChannel event The problem here is that chan_dahdi is designed
+ in such a way to set certain values in the dahdi_pvt only once.
+ One of those such values is the configured caller id data in
+ chan_dahdi.conf. For PRI, the configured caller id data could be
+ overwritten during a call. Instead of saving the data and
+ restoring, it was decided that for all non-analog channels it was
+ simply best to not set the configured caller id in the first
+ place and also clear it at the end of the call. (closes issue
+ #15883) Reported by: jsmith ........ ................
+
+2009-10-16 20:58 +0000 [r224264] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500
+ (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
+ | 18 lines Never released PRI channels when using Busy() or
+ Congestion() dialplan apps. When the Busy() or Congestion()
+ application is used towards ISDN (an ISDN progress is sent), the
+ responding ISDN Disconnect or Release may contain the ISDN cause
+ user busy or one of the congestion causes. In chan_dahdi.c these
+ causes will only set the needbusy or needcongestion flags and not
+ activate the softhangup procedure. Unfortunately only the latter
+ can interrupt the endless wait loop of Busy()/Congestion().
+ Result: PRI channels staying in state busy for the rest of
+ asterisk life or until the other end times out and forces the
+ call to clear. (in issue 0014292) Reported by: tomaso Patches:
+ disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
+ patch is unrelated to the issue.) ........ ................
+
+2009-10-15 15:58 +0000 [r224181] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 |
+ jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines
+ Readd removed ability to allow listening to one side of the call
+ in app_chanspy (Option o) (closes issue #15675) Reported by:
+ john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks
+ (license 790) Tested by: jgutierrez on users list:
+ http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
+ ........
+
+2009-10-12 23:55 +0000 [r223835] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009)
+ | 15 lines Merged revisions 223804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
+ | 8 lines Ensure ringing continues for branched calls after
+ progress is received While waiting for an answer, don't send
+ progress for branched calls for which ringing was sent. (closes
+ issue #15028) Reported by: fnordian ........ ................
+
+2009-10-12 21:01 +0000 [r223757] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009)
+ | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2
+ options SWP-151 ........
+
+2009-10-12 14:37 +0000 [r223655] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12
+ Oct 2009) | 13 lines Remove automatic switching from T.38 to
+ voice mode in chan_sip. chan_sip has some code to automatically
+ switch from T.38 mode to voice mode when a voice frame is written
+ to the channel while it is in T.38 mode; this was intended to
+ handle the situation when a FAX transmission has ended and the
+ channel is not yet hung up, but is causing problems at the
+ beginning of FAX sessions as well when there are still voice
+ frames 'in flight' at the time the T.38 negotiation completes.
+ This patch removes the automatic switchover, and changes app_fax
+ to explicitly switch off T.38 mode when the FAX transmission
+ process ends. (closes issue #16025) Reported by: jamicque
+ ........
+
+2009-10-11 17:32 +0000 [r223490] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 223487 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009)
+ | 17 lines Merged revisions 223485-223486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009)
+ | 6 lines Don't use data outside of its scope. The purpose of
+ this code was to have a hangup frame put on the list of deferred
+ frames. However, the code that read the hangup frame was outside
+ of the scope of where the hangup frame was declared. ........
+ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009)
+ | 2 lines Remove some unnecessary code. ........ ................
+
+2009-10-09 23:12 +0000 [r223406] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation
+ of PRIREDIRECTIONREASON set by chan_sip. This commit is the
+ simplest way to solve a problem that has already been solved in
+ trunk with the "COLP/CONP and Redirecting party information into
+ Asterisk" commit. In trunk the redirection reason is translated
+ into a generic redirect reason. I would have had to do the same
+ fix except chan_sip never reads PRIREDIRECTREASON. So both
+ chan_dahdi and chan_h323 have been modified to interpret the one
+ different redirect reason of "no-answer" properly and set the
+ ISDN reason code 2 of "no reply". (closes issue #15033) Reported
+ by: steinwej
+
+2009-10-09 21:01 +0000 [r223333] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 |
+ kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10
+ lines Initiate T.38 switchover when acting as called party,
+ regardless of FAX direction. SendFAX() and ReceiveFAX() can be
+ given options to indicate whether they should act as the calling
+ or called party; this mode should be used to decide whether to
+ initiate a switchover to T.38, not the direction that the FAX
+ transfer will take place. (closes issue #16039) Reported by:
+ jamicque ........
+
+2009-10-09 18:53 +0000 [r223286] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 223273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct
+ 2009) | 14 lines Merged revisions 223225 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct
+ 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING
+ when originating calls. (closes issue #15104) Reported by:
+ nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
+ (license 96) Tested by: nblasgen, mnicholson ........
+ ................
+
+2009-10-09 18:29 +0000 [r223257] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct
+ 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri,
+ 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c
+ ........ ................
+
+2009-10-09 17:55 +0000 [r223208] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009)
+ | 16 lines Merged revisions 223205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009)
+ | 10 lines fixes sip registration using authuser in user.conf
+ (closes issue #14954) Reported by: tornblad Tested by:
+ mmichelson, tornblad, dvossel ........ ................
+
+2009-10-09 17:27 +0000 [r223173] Matthew Nicholson <mnicholson@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct
+ 2009) | 8 lines Don't close the sqlite database when reloading.
+ Only close the database when unloading. (closes issue #15953)
+ Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by
+ frawd (license 610) Tested by: frawd ........
+
+2009-10-09 17:09 +0000 [r223089-223133] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 |
+ dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines
+ 'auth=' did not parse md5 secret correctly (closes issue #15949)
+ Reported by: ebroad Patches: authparsefix.patch uploaded by
+ ebroad (license 878) 15949_trunk.diff uploaded by dvossel
+ (license 671) Tested by: ebroad ........
+
+ * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 |
+ dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines
+ p->peerauth is always empty in transmit_register() When using
+ callbackextension or specifing the peer name in a registration
+ string, the peer's specific auth settings set by the "auth="
+ strings within the peer definition are not used by the
+ registration. Thanks to ebroad for reporting the issue and
+ providing the patch. (closes issue #15955) Reported by: ebroad
+ Patches: regauthfix.patch uploaded by ebroad (license 878)
+ ........
+
+2009-10-08 20:00 +0000 [r222883] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c,
+ /, main/file.c: Merged revisions 222880 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009)
+ | 51 lines Merged revisions 222878 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009)
+ | 44 lines Make filestream frame handling safer by isolating
+ frames before returning them. This patch is related to a number
+ of issues on the bug tracker that show crashes related to freeing
+ frames that came from a filestream. A number of fixes have been
+ made over time while trying to figure out these problems, but
+ there re still people seeing the crash. (Note that some of these
+ bug reports include information about other problems. I am
+ specifically addressing the filestream frame crash here.) I'm
+ still not clear on what the exact problem is. However, what is
+ _very_ clear is that we have seen quite a few problems over time
+ related to unexpected behavior when we try to use embedded frames
+ as an optimization. In some cases, this optimization doesn't
+ really provide much due to improvements made in other areas. In
+ this case, the patch modifies filestream handling such that the
+ embedded frame will not be returned. ast_frisolate() is used to
+ ensure that we end up with a completely mallocd frame. In
+ reality, though, we will not actually have to malloc every time.
+ For filestreams, the frame will almost always be allocated and
+ freed in the same thread. That means that the thread local frame
+ cache will be used. So, going this route doesn't hurt. With this
+ patch in place, some people have reported success in not seeing
+ the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609)
+ Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt
+ uploaded by russell (license 2) Tested by: aragon, russell
+ (closes issue #15817) Reported by: zerohalo Tested by: zerohalo
+ (closes issue #15845) Reported by: marhbere Review:
+ https://reviewboard.asterisk.org/r/386/ ........ ................
+
+2009-10-08 19:41 +0000 [r222874] David Vossel <dvossel@digium.com>
+
+ * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions
+ 222873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 |
+ dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines
+ fixes an ast_netsock_list memory leak. ABE-1998 Review:
+ https://reviewboard.asterisk.org/r/395/ ........
+
+2009-10-08 16:51 +0000 [r222695-222802] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500
+ (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009)
+ | 12 lines Fix memory leak if chan_misdn config parameter is
+ repeated. Memory leak when the same config option is set more
+ than once in an misdn.conf section. Why must this be considered?
+ Templates! Defining a template with default port options and
+ later adding to or overriding some of them. Patches:
+ memleak-misdn.patch JIRA ABE-1998 ........ ................
+
+ * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500
+ (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009)
+ | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf:
+ astdtmf must be set to "yes". With "no", buffer loss does not
+ occur. The translated frame "f2" when passing through
+ ast_dsp_process() is not freed whenever it is not used further in
+ process_ast_dsp(). Then in the end it is never ever freed.
+ Patches: translate.patch JIRA ABE-1993 ........ ................
+
+2009-10-07 18:06 +0000 [r222549] Jason Parker <jparker@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 222548 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct
+ 2009) | 5 lines Remove 'keepstats' queue option from sample
+ config, as it's no longer used.
+ https://reviewboard.asterisk.org/r/115/ (closes issue #15820)
+ Reported by: kshumard ........
+
+2009-10-07 18:00 +0000 [r222547] Sean Bright <sean@malleable.com>
+
+ * funcs/func_strings.c: Fix merge error.
+
+2009-10-07 17:45 +0000 [r222544] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009)
+ | 14 lines Merged revisions 222542 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009)
+ | 8 lines crash on transfer handle_invite_replaces() attempts to
+ uplock a pvt's owner channel without first verifing that it
+ exists. (issue #16027) ........ ................
+
+2009-10-06 23:59 +0000 [r222354-222466] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500
+ (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009)
+ | 8 lines Add missing unlock(s) in dahdi_read (two cases in
+ trunk, and 1.6.2) (closes issue #15683) Reported by: alecdavis
+ ........ ................
+
+ * channels/chan_dahdi.c: Fix potential crash when entire span
+ request is received. The variable index used in this scenario for
+ accessing the dahdi_pvts was wrong and was most likely copied
+ from the several other places it is used correctly. (closes issue
+ #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch
+ uploaded by tsearle (license 373)
+
+ * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009)
+ | 9 lines Fix 222298 (crash during destruction of second channel
+ when variable set with setvar). I mistakenly reasoned that setvar
+ would be used on all channels. Since it can be set per channel,
+ give each dahdi channel a copy of the variable. (related to
+ #15899) ........
+
+2009-10-06 19:41 +0000 [r222311] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c, res/res_config_pgsql.c, /: Merged revisions
+ 222309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 |
+ tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10
+ lines Change schema query to involve the use of an optional
+ schema parameter. This change is done in such a way as to allow
+ the driver to continue to function with older databases which
+ don't have these features. (closes issue #16000) Reported by:
+ jamicque Patches: 20091002__issue16000.diff.txt uploaded by
+ tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+
+2009-10-06 19:27 +0000 [r222304] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009)
+ | 9 lines Fix crash during destruction of second channel when
+ variable set with setvar. The setvar line in chan_dahdi.conf is
+ shared among all the channels, so make sure to only free the
+ resources only when the last channel is destroyed. (closes issue
+ #15899) Reported by: tzafrir ........
+
+2009-10-06 19:22 +0000 [r222289] Tilghman Lesher <tlesher@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 222273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 |
+ tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines
+ When we call a gosub routine, the variables should be scoped to
+ avoid contaminating the caller. This affected the ~~EXTEN~~ hack,
+ where a subroutine might have changed the value before it was
+ used in the caller. Patch by myself, tested by ebroad on
+ #asterisk ........
+
+2009-10-06 Leif Madsen <lmadsen@digium.com>
+
+ * Released Asterisk 1.6.2.0-rc3
+
+2009-10-06 01:39 +0000 [r222113-222187] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c,
+ res/res_clialiases.c, /, channels/chan_sip.c,
+ funcs/func_dialgroup.c, include/asterisk/astobj2.h,
+ res/res_phoneprov.c: Merged revisions 222176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct
+ 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05
+ Oct 2009) | 20 lines Fix ao2_iterator API to hold references to
+ containers being iterated. See Mantis issue for details of what
+ prompted this change. Additional notes: This patch changes the
+ ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
+ instead of a macro, with a name that fits our naming policy;
+ also, it is now necessary to call ao2_iterator_destroy() on any
+ iterator that has been created. Currently this only releases the
+ reference to the container being iterated, but in the future this
+ could also release other resources used by the iterator, if the
+ iterator implementation changes to use additional resources.
+ (closes issue #15987) Reported by: kpfleming Review:
+ https://reviewboard.asterisk.org/r/383/ ........ ................
+
+ * configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c,
+ configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 222110
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05
+ Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be
+ supportable via configuration option. Many T.38 endpoints
+ incorrectly send the maximum IFP frame size they can accept as
+ the T38FaxMaxDatagram value in their SDP, when in fact this value
+ is supposed to be the maximum UDPTL payload size (datagram size)
+ they can accept. If the value they supply is small enough (a
+ commonly supplied value is '72'), T.38 UDPTL transmissions will
+ likely fail completely because the UDPTL packets will not have
+ enough room for a primary IFP frame and the redundancy used for
+ error correction. If this occurs, the Asterisk UDPTL stack will
+ emit log messages warning that data loss may occur, and that the
+ value may need to be overridden. This patch extends the
+ 't38pt_udptl' configuration option in sip.conf to allow the
+ administrator to override the value supplied by the remote
+ endpoint and supply a value that allows T.38 FAX transmissions to
+ be successful with that endpoint. In addition, in any SIP call
+ where the override takes effect, a debug message will be printed
+ to that effect. This patch also removes the T38FaxMaxDatagram
+ configuration option from udptl.conf.sample, since it has not
+ actually had any effect for a number of releases. In addition,
+ this patch cleans up the T.38 documentation in sip.conf.sample
+ (which incorrectly documented that T.38 support was passthrough
+ only). (issue #15586) Reported by: globalnetinc ........
+
+2009-10-02 17:35 +0000 [r222032] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500
+ (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
+ Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
+ memcpy. ........ ................
+
+2009-10-02 17:01 +0000 [r221923-221974] Tilghman Lesher <tlesher@digium.com>
+
+ * main/astobj2.c, /: Merged revisions 221971 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009)
+ | 9 lines Merged revisions 221970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009)
+ | 2 lines Ensure the result of the hash function is positive.
+ Negative array offsets suck. ........ ................
+
+ * /, main/logger.c: Merged revisions 221920 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 |
+ tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines
+ Initialize a variable that we check immediately upon startup.
+ (closes issue #15973) Reported by: atis ........
+
+2009-10-02 01:35 +0000 [r221879] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
+ Merged revisions 221844 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009)
+ | 33 lines Merged revisions 221769 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
+ | 26 lines Occasionally losing use of B channels in chan_misdn. I
+ have not been able to reproduce the problem of losing channels.
+ However, I have seen in the code a reentrancy problem that might
+ give these symptoms. The reentrancy patch does several things: 1)
+ Guards B channel and B channel structure allocation. 2) Makes the
+ B channel structure find routines more precise in locating
+ records. 3) Never leave a B channel allocated if we received
+ cause 44. The last item may cause temporary outgoing call
+ problems, but they should clear when the line becomes idle.
+ (closes issue #15490) Reported by: slutec18 Patches:
+ issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
+ Reported by: FabienToune Patches:
+ issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: FabienToune, rmudgett, slutec18 ........
+ ................
+
+2009-10-02 00:07 +0000 [r221744-221780] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions
+ 221777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009)
+ | 9 lines Merged revisions 221776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
+ | 2 lines Fix a bunch of off-by-one errors ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 |
+ tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines
+ Revision 220906 (a merge from 1.4) was not merged correctly,
+ causing a problem with non-dynamic peers. ........
+
+2009-10-01 19:35 +0000 [r221698] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 |
+ dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines
+ outbound tls connections were not defaulting to port 5061 (closes
+ issue #15854) Reported by: dvossel Patches:
+ sip_port_config_trunk.diff uploaded by dvossel (license 671)
+ Tested by: dvossel ........
+
+2009-10-01 16:57 +0000 [r221660] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221554,221589 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu,
+ 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE
+ constructs when it's just TRUE or FALSE. ................ r221589
+ | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9
+ lines Merged revisions 221588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
+ 2009) | 2 lines Use unsigned ints for portinuri flags. ........
+ ................
+
+2009-10-01 16:25 +0000 [r221622] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged
+ revisions 221592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 |
+ kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12
+ lines Remove ability to control T.38 FAX error correction from
+ udptl.conf. chan_sip has had the ability to control T.38 FAX
+ error correction mode on a per-peer (or global) basis for a
+ couple of releases now, which is where it should have been all
+ along. This patch removes the ability to configure it in
+ udptl.conf, but issues a warning if the user tries to do, telling
+ them to look at sip.conf.sample for how to configure it now. For
+ any SIP peers that are T.38 enabled in sip.conf, there is already
+ a default for FEC error correction even if the user does not
+ specify any mode, so this change will not turn off error
+ correction by default, it will have the same default value that
+ has been in the udptl.conf sample file. ........
+
+2009-09-30 23:07 +0000 [r221477-221485] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 |
+ mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2
+ lines Cleaned up merge from r221432 ........
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 221432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep
+ 2009) | 17 lines Merged revisions 221360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
+ 2009) | 10 lines Fix SRV lookup and Request-URI generation in
+ chan_sip. This patch adds a new field "portinuri" to the sip
+ dialog struct and the sip peer struct. That field is used during
+ RURI generation to determine if the port should be included in
+ the RURI. It is also used in some places to determine if an SRV
+ lookup should occur. (closes issue #14418) Reported by: klaus3000
+ Tested by: klaus3000, mnicholson Review:
+ https://reviewboard.asterisk.org/r/369/ ........ ................
+
+2009-09-30 21:46 +0000 [r221371-221472] Matthias Nick <mnick@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 221436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 |
+ mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines
+ Prevents from division by zero ........
+
+ * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
+ revisions 221368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) |
+ 23 lines Merged revisions 221153,221157,221303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
+ 2 lines check bounds - prevents for buffer overflow ........
+ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
+ 8 lines added a new dialplan function 'CSV_QUOTE' and changed the
+ cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
+ Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
+ mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
+ 30 Sep 2009) | 2 lines changed the prototype definition of
+ csv_quote ........ ................
+
+2009-09-30 19:15 +0000 [r221304] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c,
+ include/asterisk/rtp.h: Merged revisions 221266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009)
+ | 32 lines Merged revisions 221086 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
+ | 25 lines Change the SSRC by default when our media stream
+ changes Be default, change SSRC when doing an audio stream
+ changes Asterisk doesn't honor marker bit when reinvited to
+ already-bridged RTP streams,resulting in far-end stack discarding
+ packets with "old" timestamps that areactually part of a new
+ stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
+ a reinvite, unless the 'constantssrc' is set to true in sip.conf.
+ The original issue reported to Digium support detailed the
+ following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
+ Application Server Call comes in fromITSP, Asterisk dials the app
+ server which sends a re-invite back toAsterisk--not to negotiate
+ to send media directly to the ITSP, but to indicatethat it's
+ changing the stream it's sending to Asterisk. The app
+ servergenerates a new SSRC, sequence numbers, timestamps, and
+ sets the marker bit on the new stream. Asterisk passes through
+ the teimstamp of the new stream, butdoes not reset the SSRC,
+ sequence numbers, or set the marker bit. When the timestamp on
+ the new stream is older than the timestamp on the originalstream,
+ the ITSP (which doesn't know there has been any change) discards
+ the newframes because it thinks they are too old. This patch
+ addresses this by changing the SSRC on a stream update unless
+ constantssrc=true is set in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/374/ ........ ................
+
+2009-09-30 16:57 +0000 [r221204] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 221201 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009)
+ | 14 lines Merged revisions 221200 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
+ | 7 lines Avoid a potential NULL dereference. (closes issue
+ #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
+ uploaded by tilghman (license 14) Tested by: kobaz ........
+ ................
+
+2009-09-30 14:57 +0000 [r221089] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep
+ 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a()
+ option. We require box numbers, not names as the documentation
+ implies. (issue #14740) Reported by: pj Patches:
+ __20090729-app_voicemail-documentation.patch uploaded by lmadsen
+ (license 10) Tested by: seanbright, lmadsen ........
+
+2009-09-30 04:41 +0000 [r221027-221047] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_lock.c: Recorded merge of revisions 221044 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29
+ Sep 2009) | 8 lines Allow locks to be inherited through a
+ masquerade without causing starvation. (closes issue #14859)
+ Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
+ by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
+ uploaded by tilghman (license 14) Tested by: atis, tilghman
+ ........
+
+ * include/asterisk/smdi.h, include/asterisk/optional_api.h
+ (removed), apps/app_voicemail.c, include/asterisk/agi.h,
+ include/asterisk/monitor.h: Remove optional_api from 1.6.2
+ branch, since it is not currently working. This is a blocking
+ issue for the 1.6.2 release. (closes issue #15914) Reported by:
+ mbeckwell Branch:
+ http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162
+ Tested by: mbeckwell
+
+ * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009)
+ | 16 lines Merged revisions 220873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
+ | 9 lines Reduce CPU usage related to building a peer merely for
+ devicestates. This fixes a 100% CPU problem in the SIP driver,
+ found by profiling the driver while the problem was occurring.
+ (closes issue #14309) Reported by: pkempgen Patches:
+ 20090924__issue14309.diff.txt uploaded by tilghman (license 14)
+ Tested by: pkempgen, vrban ........ ................
+
+2009-09-29 20:24 +0000 [r220905-220934] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
+ spyee is masqueraded and chanspy_ds_chan_fixup() is called with
+ the channel locked. (closes issue #15965) Reported by: atis
+ Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: atis
+
+ * /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep
+ 2009) | 5 lines Fix options 'm' and 's'. They were swapped in the
+ code. Also document the fact that app_confbridge does not
+ automatically answer the channel. (closes issue #15964) Reported
+ by: shrift ........
+
+2009-09-29 17:06 +0000 [r220836] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009)
+ | 12 lines Make deletion of temporary greetings work properly
+ with IMAP_STORAGE When imapgreetings was set to yes, the message
+ was being deleted but wasn't actually being expunged. When
+ imapgreetings was set to no, the file based message was not being
+ deleted at all. All good now! (closes issue #14949) Reported by:
+ noahisaac Patches: vm_tempgreeting_removal.patch uploaded by
+ noahisaac (license 748), modified by me ........
+
+2009-09-28 19:13 +0000 [r220725] Sean Bright <sean@malleable.com>
+
+ * /, Makefile.rules: Merged revisions 220721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep
+ 2009) | 10 lines Merged revisions 220717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
+ 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
+ explicitly pass -O0 to the compiler so we override any default
+ optimization levels for a particular install. ........
+ ................
+
+2009-09-28 19:11 +0000 [r220722] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 |
+ jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines
+ Fix building of registration entry in build_peer when using
+ callbackextension Check for remotesecret option was
+ unintentionally always true, which therefore caused the secret
+ option to never be used. Thanks to dvossel for pointing out the
+ exact fix. (closes issue #15943) Reported by: tpsast ........
+
+2009-09-27 20:45 +0000 [r220632] Michiel van Baak <michiel@vanbaak.info>
+
+ * funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009)
+ | 3 lines add name argument for the CALLERID dialplan function to
+ the xml documentation. Pointed out to me on IRC by snuff-home.
+ Thanks ........
+
+2009-09-26 15:12 +0000 [r220589] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009)
+ | 2 lines Allow AES to compile, when OpenSSL is not present.
+ ........
+
+2009-09-24 20:38 +0000 [r220369] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 220365 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 |
+ dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines
+ fixes tcptls_session memory leak caused by ref count error
+ (closes issue #15939) Reported by: dvossel Review:
+ https://reviewboard.asterisk.org/r/375/ ........
+
+2009-09-24 19:42 +0000 [r220292] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged
+ revisions 220289 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009)
+ | 13 lines Merged revisions 220288 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
+ | 6 lines Implicitly sending a progress signal breaks some
+ applications. Call Progress() in your dialplan if you explicitly
+ want progress to be sent. (Reverts change 216430, closes issue
+ #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
+ list
+ http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
+ ........ ................
+
+2009-09-24 18:22 +0000 [r220103-220221] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 220217 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep
+ 2009) | 9 lines Merged revisions 220213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
+ 2009) | 1 line Resolve parallel build warnings. Reported by Klaus
+ Darilion on the asterisk-dev mailing list. ........
+ ................
+
+ * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400
+ (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu,
+ 24 Sep 2009) | 2 lines Remove the remaining bashisms in the
+ Makefile/mkpkgconfig ........ ................
+
+2009-09-24 08:43 +0000 [r220031] Michiel van Baak <michiel@vanbaak.info>
+
+ * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200
+ (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009)
+ | 7 lines mkpkgconfig does not need bash so make it use /bin/sh
+ This fixes building on all systems that don't have bash at
+ /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
+ #asterisk-dev ........ ................
+
+2009-09-24 07:45 +0000 [r219989] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_directory.c, /: Merged revisions 219987 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009)
+ | 8 lines Fix two possible crashes, one only in 1.6.1 and one in
+ 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg
+ Patches: 20090914__issue15739.diff.txt uploaded by tilghman
+ (license 14) 20090922__issue15739.diff.txt uploaded by tilghman
+ (license 14) Tested by: DLNoah, jeffg ........
+
+2009-09-22 21:48 +0000 [r219821] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500
+ (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009)
+ | 10 lines When IMAP variables were changed during a reload,
+ Voicemail did not use the new values. This change introduces a
+ configuration version variable, which ensures that connections
+ with the old values are not reused but are allowed to expire
+ normally. (closes issue #15934) Reported by: viniciusfontes
+ Patches: 20090922__issue15934.diff.txt uploaded by tilghman
+ (license 14) Tested by: viniciusfontes ........ ................
+
+2009-09-21 17:01 +0000 [r219722] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500
+ (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
+ Sep 2009) | 3 lines Reverting merge 219520. This change was not
+ necessary. ........ ................
+
+2009-09-20 18:21 +0000 [r219669] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/file.c: Merged revisions 219654 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009)
+ | 15 lines Merged revisions 219653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
+ | 8 lines Really stop the stream, when ast_closestream() is
+ called. (closes issue #15129) Reported by: bmh Patches:
+ 20090918__issue15129.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/372/ ........
+ ................
+
+2009-09-19 03:14 +0000 [r219590] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219587 | russell | 2009-09-18 21:59:52 -0500
+ (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009)
+ | 6 lines Make sure the iax_pvt exists before dereferencing it.
+ This fixes the latest crash posted on issue 15609. (issue #15609)
+ ........ ................
+
+2009-09-18 23:21 +0000 [r219452-219521] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500
+ (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009)
+ | 9 lines iax2 frame double free The iax frame's retrans sched id
+ was written over right before iax2_frame_free was called. In
+ iax2_frame_free that retrans id is used to delete the sched item.
+ By writing over the retrans field before the sched item could be
+ deleted, it was possible for a retransmit to occur on a freed
+ frame. ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009)
+ | 20 lines Merged revisions 219450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
+ | 14 lines via-header branches not updated correctly on INVITE
+ INVITE requests must always contain a new unique branch id. When
+ a new branch id is created for an INVITE, the dialog's
+ invite_branch variable must be updated so CANCEL requests use the
+ correct branch id. (closes issue #15262) Reported by: maniax
+ Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
+ (license 608) invite_new_branch_trunk.diff uploaded by dvossel
+ (license 671) Tested by: maniax, dvossel ........
+ ................
+
+2009-09-18 13:57 +0000 [r219415] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009)
+ | 6 lines Missing value setting line for maxsecs/maxmessage
+ (closes issue #15696) Reported by: fhackenberger Patches:
+ maxsecs.patch uploaded by fhackenberger (license 592) ........
+
+2009-09-17 22:38 +0000 [r219376] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 |
+ dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines
+ fixes deadlock when performing directed pickup w Invite/replaces
+ (closes issue #15340) Reported by: lmsteffan Patches:
+ deadlock.patch uploaded by lmsteffan (license 779) Tested by:
+ lmsteffan ........
+
+2009-09-17 22:37 +0000 [r219370] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep
+ 2009) | 12 lines Merged revisions 219320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
+ 2009) | 6 lines Send a 100 Trying response when we detect a
+ spiral. This was problematic during spiral tests at SIPit...
+ along with some other things as well. ........ ................
+
+2009-09-17 22:06 +0000 [r219307] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009)
+ | 27 lines Merged revisions 219303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
+ | 21 lines INVITE w/Replaces deadlock fix This patch cleans up
+ the locking logic in chan_sip.c's handle_invite_replaces()
+ function as well as making use of ast_do_masquerade() rather than
+ forcing the masquerade on an ast_read(). The code had several
+ redundant unlocks that would result in 'freed more times than
+ we've locked!' errors. I cleaned these up as well as moving all
+ the unlock logic to the end of the function. This patch should
+ also resolve the issue people were having with the replacecall
+ channel never being unlocked with one legged calls. (closes issue
+ #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
+ uploaded by dvossel (license 671) Tested by: irroot, dvossel
+ Review: https://reviewboard.asterisk.org/r/371/ ........
+ ................
+
+2009-09-17 19:58 +0000 [r219267] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 |
+ file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines
+ Ensure no spaces exist before "refresher=" when doing the
+ comparison. ........
+
+2009-09-17 Leif Madsen <lmadsen@digium.com>
+
+ * Released Asterisk 1.6.2.0-rc2
+
+2009-09-17 15:38 +0000 [r219194] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h,
+ include/asterisk/channel.h: Merged revisions 219139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500
+ (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
+ 2009) | 10 lines Prevent a potential race condition and crash
+ when hanging up a channel by removing the channel from the
+ channel list before begining channel tear down. This fix may
+ potentially cause problems with CDR backends that access the
+ channel a CDR is associated with via the channel list. This fix
+ makes the channel unavabile at the time when the CDR backend is
+ invoked. This has been documented in include/asterisk/cdr.h.
+ (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/362/ ........
+ ................
+
+2009-09-16 23:52 +0000 [r219063] Tilghman Lesher <tlesher@digium.com>
+
+ * main/config.c, configs/extensions.conf.sample, /: Merged
+ revisions 219061 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009)
+ | 15 lines Merged revisions 219023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
+ | 8 lines Properly deal with quotes in the arguments of '#exec'
+ includes. (closes issue #15583) Reported by: pkempgen Patches:
+ 20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+ 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+ 169) Tested by: pkempgen ........ ................
+
+2009-09-16 19:40 +0000 [r218938] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Merged revisions 218868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009)
+ | 20 lines Merged revisions 218867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
+ | 13 lines Fixes CID pattern matching behavior to mirror that of
+ extension pattern matching. Pattern matching for extensions uses
+ a type of scoring system, giving values for specificity to each
+ character in the pattern. Unfortunately, this is done character
+ by character, in order. This does lead to some less specific
+ patterns being first in line for matching, but it will usually
+ get the job done. This patch merely brings CID matching to the
+ same level as extension matching. This patch does not attempt to
+ tackle the problem shared by extension matching. (closes issue
+ #14708) Reported by: klaus3000 ........ ................
+
+2009-09-16 19:29 +0000 [r218937] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 |
+ mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12
+ lines Reverse order of args to fread. This way, we don't always
+ write a null byte into byte 1 of the buffer (closes issue #15905)
+ Reported by: ebroad Patches: freadfix.patch uploaded by ebroad
+ (license 878) Tested by: ebroad ........
+
+2009-09-16 19:25 +0000 [r218934] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 |
+ file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On
+ TCP and TLS connections do not attempt to stop retransmission of
+ the packet internally. This was preventing responses from being
+ properly processed because the packet was not being found causing
+ handle_response to return prematurely. ........
+
+2009-09-16 13:38 +0000 [r218802] Russell Bryant <russell@digium.com>
+
+ * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
+ revisions 218799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009)
+ | 16 lines Merged revisions 218798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
+ | 9 lines Remove the IAXy firmware from Asterisk. The firmware
+ can now be found on downloads.digium.com, where the rest of our
+ binary downloads live. This was the last part of our Asterisk
+ tarballs that was considered non-free by Debian. :-) (closes
+ issue #15838) Reported by: paravoid ........ ................
+
+2009-09-15 22:46 +0000 [r218733] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500
+ (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009)
+ | 6 lines If the user enters the same password as before, don't
+ signal an error when the change does nothing. (closes issue
+ #15492) Reported by: cbbs70a Patches:
+ 20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2009-09-15 19:24 +0000 [r218688] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 |
+ dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines
+ upward bound checking for port string to int conversion ........
+
+2009-09-15 16:18 +0000 [r218590] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep
+ 2009) | 15 lines Merged revisions 218578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
+ 2009) | 8 lines Send request contact header field with response
+ to registrer queries instead of the address of record. (closes
+ issue #14438) Reported by: ravindrad Patches: regquerypatch
+ uploaded by ravindrad (license 684) Tested by: ravindrad ........
+ ................
+
+2009-09-15 16:06 +0000 [r218582] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_followme.c, /: Merged revisions 218579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009)
+ | 16 lines Merged revisions 218577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
+ | 9 lines Ensure FollowMe sets language in channels it creates.
+ Also, not in the original bug report, but related fields are
+ accountcode and musicclass, and the inheritance of datastores.
+ (closes issue #15372) Reported by: Romik Patches:
+ 20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+ Tested by: cervajs ........ ................
+
+2009-09-15 15:59 +0000 [r218576] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 218430 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500
+ (Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
+ | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
+ crash in do_monitor. After talking to rmudgett about some of his
+ recent iflist locking changes, it was determined that the only
+ place that would destroy a channel without being explicitly to do
+ so was in handle_init_event. The loop to walk the interface list
+ has been modified to wait to destroy the channel until the
+ dahdi_pvt of the channel to be destroyed is no longer needed.
+ (closes issue #15378) Reported by: samy ........ ................
+
+2009-09-15 15:42 +0000 [r218507-218575] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 |
+ mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4
+ lines Use a better method of ensuring null-termination of the
+ buffer while reading the SDP when using TCP. ........
+
+ * /, channels/chan_sip.c: Merged revisions 218499,218504 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue,
+ 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent
+ over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53
+ -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP
+ socket is null-terminated. ........
+
+2009-09-15 15:05 +0000 [r218503] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 218500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep
+ 2009) | 9 lines Merged revisions 218497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
+ 2009) | 1 line Use proper hostname for downloading sound files.
+ ........ ................
+
+2009-09-14 19:49 +0000 [r218364] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile, apps/app_voicemail.c, /,
+ configs/voicemail.conf.sample: Merged revisions 218361 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500
+ (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
+ | 4 lines Don't say "Please try again" if we don't give the user
+ another chance to try again. (issue #15055, SWP-129) Reported by:
+ jthurman ........ ................
+
+2009-09-14 18:18 +0000 [r218300] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 218295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 |
+ file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do
+ not attempt to add a parking extension if an error occurred while
+ reading the configuration. ........
+
+2009-09-14 15:20 +0000 [r218238] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 218224 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500
+ (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
+ 2009) | 8 lines Ensure we don't pickup ourselves when doing
+ pickup by exten. (closes issue #15100) Reported by: lmsteffan
+ Patches: (modified) pickup.patch uploaded by lmsteffan (license
+ 779) ........ ................
+
+2009-09-13 22:12 +0000 [r218219] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0
+ that annoys gcc This memset doesn't write beyond the end of the
+ buffer. (tmpbuf has size of 4). Merged revisions 218184 via
+ svnmerge from http://svn.digium.com/svn/asterisk/trunk
+
+2009-09-13 05:59 +0000 [r218151] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 218150 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1
+ line get rid of mfcr2 monitor thread condition, is problematic
+ ........
+
+2009-09-11 06:00 +0000 [r217926-218055] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 218050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 |
+ tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines
+ Check the origination priority for more matches, not the current
+ priority. Found by Pavel Troller on the -dev list. ........
+
+ * apps/app_queue.c, /: Merged revisions 217990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009)
+ | 10 lines Merged revisions 217989 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
+ | 3 lines Don't ring another channel, if there's not enough time
+ for a queue member to answer. (Fixes AST-228) ........
+ ................
+
+ * channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /,
+ channels/chan_sip.c: Merged revisions 217916 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 |
+ tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines
+ Make calltoken support work with realtime users and peers.
+ ........
+
+2009-09-10 21:21 +0000 [r217821] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 217807 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500
+ (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009)
+ | 22 lines IAX2 encryption regression The IAX2 Call Token
+ security patch inadvertently broke the use of encryption due to
+ the reorganization of code in the socket_process() function. When
+ encryption is used, an incoming full frame must first be
+ decrypted before the information elements can be parsed. The
+ security release mistakenly moved IE parsing before decryption in
+ order to process the new Call Token IE. To resolve this,
+ decryption of full frames is once again done before looking into
+ the frame. This involves searching for an existing callno,
+ checking the pvt to see if encryption is turned on, and
+ decrypting the packet before the internal fields of the full
+ frame are accessed. (closes issue #15834) Reported by: karesmakro
+ Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
+ (license 671) Tested by: dvossel, karesmakro Review:
+ https://reviewboard.asterisk.org/r/355/ ........ ................
+
+2009-09-10 19:56 +0000 [r217739] mnick <mnick@localhost>:
+
+ * res/res_musiconhold.c, /: Merged revisions 217730 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) |
+ 17 lines Sets the correct musicclass after an announcement
+ (closes issue #15279) Reported by: mbeckwell Patches: patch.txt
+ uploaded by mnick (license ) Tested by: mnick (closes issue
+ #15832) Reported by: mbeckwell Patches: patch.txt uploaded by
+ mnick (license 874) Tested by: mnick ........
+
+2009-09-10 18:40 +0000 [r217665] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 216805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 |
+ oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
+ Since it's possible to have more than 999 calls, I'm changing the
+ call counter roof to something higher. ........
+
+2009-09-10 18:19 +0000 [r217647] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 217638 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 |
+ tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines
+ Verify support for wide ODBC character types before using them.
+ (closes issue #15870) Reported by: nic_bellamy ........
+
+2009-09-10 15:14 +0000 [r217632] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 217524 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1
+ line ast_log replaced for ast_verbose in MFCR2 event
+ notifications ........
+
+2009-09-10 12:09 +0000 [r217594] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 |
+ oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines
+ Include ActionID in all events that are responsed to AMI Action
+ SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy
+ Patches: manager_SIPshowregistry_actionid.patch uploaded by nic
+ bellamy (license 299) ........
+
+2009-09-09 20:37 +0000 [r217519] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc
+ 4.4 has more strict rules for aliasing. It doesn't like a struct
+ sockaddr_in pointer pointing to a struct sockaddr. So we make it
+ a union. Merged revisions 217445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk
+
+2009-09-09 10:58 +0000 [r217369] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 |
+ oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not
+ having any TLS session to write to is a serious XMIT_ERROR.
+ ........
+
+2009-09-08 22:20 +0000 [r217299] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 |
+ seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4
+ lines Fix compilation of app_meetme. Reported by ebroad in
+ #asterisk-bugs ........
+
+2009-09-08 20:33 +0000 [r217217] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009)
+ | 14 lines Merged revisions 217156 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
+ | 7 lines When MOH is playing on the channel, announcements sent
+ through the conference are not heard. (closes issue #14588)
+ Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
+ tilghman ........ ................
+
+2009-09-08 16:39 +0000 [r217077] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 217074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 |
+ kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9
+ lines Ensure that the default autoconf CFLAGS are not used. A
+ recent change to the configure script that allows the user to
+ specify CFLAGS and/or LDFLAGS to the script had the unfortunate
+ side effect of letting autoconf's default CFLAGS (-g -O2) feed in
+ to the rest of the build system, thereby overriding the
+ DONT_OPTIMIZE setting in menuselect. That problem is now
+ corrected. ........
+
+2009-09-08 15:36 +0000 [r217036] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_limit.c: Merged revisions 217033 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 |
+ tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines
+ Remove what appears to be an unnecessary define. (closes issue
+ #15851) Reported by: tzafrir ........
+
+2009-09-08 14:27 +0000 [r216994] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 |
+ dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
+ caller id number empty parse_uri was not being given the correct
+ scheme's, as a result, uri parsing did not parse the username
+ correctly. One of the side effects of this is an empty caller id.
+ (closes issue #15839) Reported by: ebroad Patches:
+ blank_cidv2.patch uploaded by ebroad (license 878)
+ parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
+ ebroad, dvossel ........
+
+2009-09-07 16:43 +0000 [r216647-216845] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 |
+ oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
+ Make sure we reset global_exclude_static at channel reload
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 |
+ oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If
+ there is no session timer in the INVITE, set it to default value
+ (not unset minimum = -1) Patch by oej closes issue #15621
+ Reported by: fnordian Tested by: atis ........
+
+ * CHANGES, UPGRADE.txt: Add docs
+
+ * configs/sip.conf.sample, apps/app_playback.c, main/pbx.c, /,
+ channels/chan_sip.c, apps/app_disa.c: Merged revisions 216438 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre,
+ 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
+ lines Make apps send PROGRESS control frame for early media and
+ fix too early media issue in SIP The issue at hand is that some
+ legacy (dying) PBX systems send empty media frames on PRI links
+ *before* any call progress. The SIP channel receives these frames
+ and by default signals 183 Session progress and starts sending
+ media. This will cause phones to play silence and ignore the
+ later 180 ringing message. A bad user experience. The fix is
+ twofold: - We discovered that asterisk apps that support early
+ media ("noanswer") did not send any PROGRESS frame to indicate
+ early media. Fixed. - We introduce a setting in chan_sip so that
+ users can disable any relay of media frames before the outbound
+ channel actually indicates any sort of call progress. In 1.4,
+ 1.6.0 and 1.6.1, this will be disabled for backward
+ compatibility. In later versions of Asterisk, this will be
+ enabled. We don't assume that it will change your Asterisk phone
+ experience - only for the better. We encourage third-party
+ application developers to make sure that if they have
+ applications that wants to send early media, add a PROGRESS
+ control frame transmission to make sure that all channel drivers
+ actually will start sending early media. This has not been the
+ default in Asterisk previous to this patch, so if you got
+ inspiration from our code, you need to update accordingly. Sorry
+ for the trouble and thanks for your support. This code has been
+ running for a few months in a large scale installation (over 250
+ servers with PRI and/or BRI links to old PBX systems). That's no
+ proof that this is an excellent patch, but, well, it's tested :-)
+ ........ ................
+
+2009-09-04 19:42 +0000 [r216598] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 |
+ dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines
+ sip peer matching by address only with TCP/TLS This patch removes
+ the contact header matching logic and adds logic to match all
+ tcp/tls connections by ip only Review:
+ https://reviewboard.asterisk.org/r/354/ ........
+
+2009-09-04 19:32 +0000 [r216597] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep
+ 2009) | 1 line Use ast_free() instead of free(). ........
+
+2009-09-04 17:53 +0000 [r216550-216553] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009)
+ | 2 lines Fix trunk breakage. ........
+
+ * UPGRADE-1.6.txt, main/pbx.c, /: Merged revisions 216547 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04
+ Sep 2009) | 3 lines Enable turning off the application delimiter
+ warning with the 'dontwarn' option. Suggested on the -dev list,
+ and implemented in an alternate way by me. ........
+
+2009-09-04 15:11 +0000 [r216469-216509] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, main/utils.c: Merged revisions 216506 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009)
+ | 9 lines Merged revisions 216435 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make asterisk compile under devmode with DEBUG_THREADS
+ enabled on OpenBSD ........ ................
+
+ * /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make sure canlog is set so we can compile with
+ DEBUG_THREADS enabled on OpenBSD ........
+
+2009-09-04 13:57 +0000 [r216267-216436] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 |
+ russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines
+ Do not treat every SIP peer as if they were configured with
+ insecure=port. There was a problem in the function responsible
+ for doing peer matching by IP address and port number such that
+ during the second pass for checking for a peer configured with
+ insecure=port, it would end up treating every peer as if it had
+ been configured that way. These changes fix the logic in the peer
+ IP and port comparison callback to handle insecure=port checking
+ properly. This problem was introduced when SIP peers were
+ converted to astobj2. Many thanks to dvossel for noticing this
+ while working on another peer matching issue. ........
+
+ * doc/IAX2-security.txt (added), /: Merged revisions 216264 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216264 | russell | 2009-09-04 05:48:44 -0500
+ (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216263 | russell | 2009-09-04 05:48:00 -0500
+ (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
+ Sep 2009) | 2 lines Add a plain text version of the IAX2 security
+ document. ........ ................ ................
+
+2009-09-04 06:14 +0000 [r216225] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/astobj2.c, /: Merged revisions 216222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 |
+ mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
+ make sure 'start' is always initialized. Makes asterisk compile
+ with --enable-dev-mode ........
+
+2009-09-03 19:44 +0000 [r216014-216099] Russell Bryant <russell@digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009)
+ | 16 lines Merged revisions 216085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216085 | russell | 2009-09-03 14:36:46 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
+ ........ ................ ................
+
+ * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216009 | russell | 2009-09-03 13:45:54 -0500
+ (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216008 | russell | 2009-09-03 13:44:58 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add IAX2 security document related to
+ AST-2009-006. ........ ................ ................
+
+2009-09-03 18:42 +0000 [r216007] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, channels/iax2-parser.c, main/astobj2.c,
+ configs/iax.conf.sample, include/asterisk/acl.h,
+ channels/iax2-parser.h, /, include/asterisk/astobj2.h,
+ channels/iax2.h, main/acl.c: Merged revisions 215955 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009)
+ | 6 lines Merge code associated with AST-2009-006 (closes issue
+ #12912) Reported by: rathaus Tested by: tilghman, russell,
+ dvossel, dbrooks ........
+
+2009-09-03 14:21 +0000 [r215887-215929] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 215891 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 |
+ oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add
+ known internal IP address when autodomain=yes (closes issue
+ #14573) Reported by: pj Patches: sip-internip-autodomain1.diff
+ uploaded by mnicholson (license 96) modified by oej Tested by: pj
+ ........
+
+ * main/rtp.c, channels/chan_sip.c: Fix bad reports in "sip show
+ channelstats". Not directly mergeable in svn trunk, needs more
+ tests, therefore committed directly to 1.6.2. (closes issue
+ #15819) Reported by: klaus3000 Patches:
+ asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded
+ by klaus3000 (license 65) Tested by: klaus3000, oej
+
+2009-09-03 06:02 +0000 [r215841] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/manager_1_1.txt, /: Merged revisions 215838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215838 |
+ mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines
+ Document that SIPshowpeer and SKINNYshowline now include the
+ configured parkinglot in their response. Prodded by snuff-work on
+ #asterisk-dev IRC channel ........
+
+2009-09-03 03:44 +0000 [r215802] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 |
+ tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines
+ Default the callback extension to "s". This is a regression.
+ (closes issue #15764) Reported by: elguero Change-type: bugfix
+ ........
+
+2009-09-03 00:34 +0000 [r215795] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215758 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009)
+ | 25 lines Merged revisions 215682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009)
+ | 18 lines Re-send non-100 provisional responses to prevent
+ cancellation From section 13.3.1.1 of RFC 3261: If the UAS
+ desires an extended period of time to answer the INVITE, it will
+ need to ask for an "extension" in order to prevent proxies from
+ canceling the transaction. A proxy has the option of canceling a
+ transaction when there is a gap of 3 minutes between responses in
+ a transaction. To prevent cancellation, the UAS MUST send a
+ non-100 provisional response at every minute, to handle the
+ possibility of lost provisional responses. (closes issue #11157)
+ Reported by: rjain Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/315/ ........ ................
+
+2009-09-02 21:46 +0000 [r215683] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 |
+ dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines
+ port string to int conversion using sscanf There are several
+ instances where a port is parsed from a uri or some other source
+ and converted to an int value using atoi(), if for some reason
+ the port string is empty, then a standard port is used. This
+ logic is used over and over, so I created a function to handle it
+ in a safer way using sscanf(). ........
+
+2009-09-02 21:37 +0000 [r215647-215680] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_sip.c, channels/chan_skinny.c: Merged revisions
+ 215665 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 |
+ mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines
+ add Parkinglot info to sip show peer <foo> and skinny show line
+ <foo> If we had this from the start, debugging the 'parking not
+ using configured parkinglot' bug would have been easier. ........
+
+ * /, main/features.c: Merged revisions 215622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215622 |
+ mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines
+ - lock channel before looking for a channel variable - Init the
+ parkings list member of struct parkinglot. Thanks Sean for the
+ explanation why this should be here. ........
+
+2009-09-02 18:52 +0000 [r215569-215570] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/Makefile, main/app.c: Merged revisions 215567 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r215567 | tilghman | 2009-09-02 13:37:25 -0500 (Wed, 02
+ Sep 2009) | 9 lines Close up to the soft open file limit (same on
+ Linux, but varies drastically on OS X). Also, a Makefile fix for
+ Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches:
+ 20090901__issue14542.diff.txt uploaded by tilghman (license 14)
+ Tested by: jtodd, tilghman Change-type: bugfix ........
+
+ * /, channels/chan_sip.c: Merged revisions 215222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 |
+ tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines
+ Fix register such that lines with a transport string, but without
+ an authuser, parse correctly. (AST-228) ........
+
+2009-09-02 17:35 +0000 [r215523] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215522 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 |
+ dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines
+ SIP uri parsing cleanup Now, the scheme passed to parse_uri can
+ either be a single scheme, or a list of schemes ',' delimited.
+ This gets rid of the whole problem of having to create two
+ buffers and calling parse_uri twice to check for separate
+ schemes. Review: https://reviewboard.asterisk.org/r/343/ ........
+
+2009-09-02 16:35 +0000 [r215512] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_skinny.c: Merged revisions 215479 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009)
+ | 3 lines like in chan_sip's sip_new skinny should copy the
+ configured parkinglot from a line to the newly created channel.
+ This makes callparking honor the configured parkinglot for skinny
+ lines as well. ........
+
+2009-09-02 16:09 +0000 [r215467] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 |
+ dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines
+ SIP support for keep-alive event keep-alive events are used by
+ Sipura/Linksys for NAT keepalive. There currently don't appear to
+ be any problems with NAT, but everytime a keep-alive event is
+ received, Asterisk responds with a "489 Bad event". This error
+ may indicate to a user that NAT problems exist just because this
+ even is not supported. Now, rather than respond with an error,
+ the packet is consumed and a "200 ok" is sent just to indicate we
+ received the packet. (issue #15084) Patches:
+ chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
+ ........
+
+2009-09-02 16:07 +0000 [r215465] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_sip.c: Merged revisions 215462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 |
+ mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12
+ lines Honor configured parkinglot when parking and retrieving
+ parked calls Thank oej for pointing out the fact that sip_new did
+ not copy parkinglot from the peer into the newly created channel.
+ (closes issue #15538) Reported by: gracedman Patches:
+ 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak
+ (license 7) With mod by me to also fix callparking as well (this
+ uploaded patch only fixed retrieving a parked call) Tested by:
+ gracedman, mvanbaak ........
+
+2009-09-02 01:49 +0000 [r215376] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * /, apps/app_softhangup.c: Merged revisions 215338 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500
+ (Tue, 01 Sep 2009) | 18 lines Merged revisions 215270 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009)
+ | 12 lines Use strrchr() so SoftHangup will correctly truncate
+ multi-hyphen channel names In general channel names are in the
+ form Foo/Bar-Z, but the channel name could have multiple hyphens
+ and look like Foo/B-a-r-Z. Use strrchr to truncate the channel
+ name at the last hyphen. (closes issue #15810) Reported by:
+ dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard
+ (license 733) ........ ................
+
+2009-09-01 20:00 +0000 [r215165] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c, /: Merged revisions 215161 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215161 |
+ kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3
+ lines Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS
+ frames are properly decoded. ........
+
+2009-08-31 16:22 +0000 [r214822-214960] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 214945 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500
+ (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009)
+ | 7 lines Also unlock the "other" channel, when returning, due to
+ glare. (closes issue #15787) Reported by: tim_ringenbach Patches:
+ chan_local.diff uploaded by tim ringenbach (license 540) Tested
+ by: tim_ringenbach ........ ................
+
+ * Makefile, /: Merged revisions 214898 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214898 |
+ tilghman | 2009-08-30 17:10:35 -0500 (Sun, 30 Aug 2009) | 2 lines
+ Force Darwin on ppc platforms to compile with a target level that
+ supports aliasing. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c: Merged revisions 214819 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 |
+ tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines
+ If lua is detected with the lua5.1 prefix (or not), adjust the
+ include path accordingly. Based upon feedback to a release
+ announcement on the -users list. See
+ http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html
+ ........
+
+2009-08-29 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.0-rc1 released.
+
+2009-08-28 20:17 +0000 [r214707] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 214702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009)
+ | 15 lines Merged revisions 214701 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009)
+ | 8 lines Modify comment to be a bit more accurate. We have kept
+ this comment around long enough, that it's pretty clear that
+ we're keeping the code, because changing the code would require a
+ pretty fundamental architectural shift. We've also taken
+ criticism in some quarters, because it was believed that it was
+ referring to the code being nasty. No, the code isn't nasty, just
+ the operation itself is rather odd. Fixed for eternity (probably
+ not). ........ ................
+
+2009-08-28 20:05 +0000 [r214700] Kevin P. Fleming <kpfleming@digium.com>
+
+ * makeopts.in, Makefile, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 214696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 |
+ kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9
+ lines Ensure that CFLAGS and/or LDFLAGS provided to configure
+ script are preserved. Cross-compilation environments want to
+ provide 'defaults' for compiler and linker options, and
+ frequently do this by specifying CFLAGS and LDFLAGS in the
+ environment or as command-line arguments to the configure script.
+ This patch modifies the configure script and Makefile to preserve
+ these settings and ensure they are used in the build process.
+ ........
+
+2009-08-28 18:43 +0000 [r214653] Mark Michelson <mmichelson@digium.com>
+
+ * /, include/asterisk/sched.h: Merged revisions 214650 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug
+ 2009) | 3 lines Fix some incorrect documentation of sched_thread
+ functions. ........
+
+2009-08-27 21:49 +0000 [r214202-214521] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/libcurl.m4 (added), /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 214518 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009)
+ | 14 lines Merged revisions 214517 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009)
+ | 7 lines Use autoconf to detect libcurl, as this enables
+ cross-compilation checks, something we didn't allow before.
+ (closes issue #15714) Reported by: pprindeville Patches:
+ 20090813__issue15714.diff.txt uploaded by tilghman (license 14)
+ Tested by: pprindeville ........ ................
+
+ * main/manager.c, /: Merged revisions 214514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214514 |
+ tilghman | 2009-08-27 16:26:37 -0500 (Thu, 27 Aug 2009) | 7 lines
+ Ensure that we check for the special value
+ CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by:
+ a_villacis Patches:
+ asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch
+ uploaded by a villacis (license 660) (Plus a few of my own, to
+ catch the remaining places within manager.c where it could have
+ been a problem) ........
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 214466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009)
+ | 9 lines Merged revisions 214436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009)
+ | 2 lines One more build system change, to make the descriptions
+ look better, if we have better information. ........
+ ................
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 214360 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500
+ (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009)
+ | 3 lines Make autoheader descriptions render correctly in our
+ autoconfig.h file. (Figured out while working with issue #14906)
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 214199 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 |
+ tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines
+ Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue
+ #15362) Reported by: klaus3000 Patches:
+ chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license
+ 65) ........
+
+2009-08-26 16:39 +0000 [r214196] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 214195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009)
+ | 25 lines Merged revisions 214194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009)
+ | 19 lines ast_write() ignores ast_audiohook_write() results In
+ ast_write(), if a channel has a list of audiohooks, those lists
+ are written to and the resulting frame is what ast_write() should
+ continue with. The problem was the returned audiohook frame was
+ not being handled at all, and the original frame passed into it
+ did not contain the mixed audio, so essentially audio was being
+ lost. One result of this was chan_spy's whisper mode no longer
+ worked. To complicate the issue, frames passed into ast_write may
+ either be a single frame, or a list of frames. So, as the list of
+ frames is processed in the audiohook_write, the returned frames
+ had to be added to a new list. (closes issue #15660) Reported by:
+ corruptor Tested by: dvossel ........ ................
+
+2009-08-25 22:43 +0000 [r213903-214155] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 214152 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 |
+ tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines
+ Not all versions of gnu-linux use glibc, which contains iconv.
+ Some (especially embedded systems) don't have iconv at all.
+ (closes issue #15169) Reported by: pprindeville ........
+
+ * /, main/say.c: Merged revisions 214071 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009)
+ | 17 lines Merged revisions 214068-214069 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009)
+ | 6 lines Fix pronunciation of German dates. (closes issue
+ #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
+ by Benjamin Kluck (license 803) ........ r214069 | tilghman |
+ 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should
+ always compile before committing... ........ ................
+
+ * /, pbx/pbx_dundi.c: Merged revisions 213975 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213975 |
+ tilghman | 2009-08-25 01:51:12 -0500 (Tue, 25 Aug 2009) | 6 lines
+ DUNDILOOKUP function in 1.6 should use comma delimiters. (closes
+ issue #15322) Reported by: chappell Patches:
+ dundilookup-0015322.patch uploaded by chappell (license 8)
+ ........
+
+ * main/pbx.c, /: Merged revisions 213971 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009)
+ | 14 lines Merged revisions 213970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009)
+ | 7 lines Improve error message by informing user exactly which
+ function is missing a parethesis. (closes issue #15242) Reported
+ by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
+ dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
+ loloski (license 68) ........ ................
+
+ * Makefile, /: Merged revisions 213904 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213904 |
+ tilghman | 2009-08-24 21:54:07 -0500 (Mon, 24 Aug 2009) | 6 lines
+ The DTD should be installed in the same path as the rest of the
+ XML documentation. (closes issue #15344) Reported by: tzafrir
+ Patches: makefile_appdocs_dtd.diff uploaded by tzafrir (license
+ 46) ........
+
+ * Makefile, /: Merged revisions 213900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r213900 | tilghman | 2009-08-24 21:41:17 -0500 (Mon, 24 Aug 2009)
+ | 11 lines Merged revisions 213899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009)
+ | 4 lines Use the default runlevels for Debian derivatives,
+ instead of making up our own. (closes issue #14730) Reported by:
+ pkempgen ........ ................
+
+2009-08-24 16:49 +0000 [r213836] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 213833 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009)
+ | 14 lines Fix storage of greetings when using IMAP_STORAGE The
+ store macro was not getting called preventing storage of IMAP
+ greetings at all. This has been corrected along with fixing
+ checking if the imapgreetings option is turned on to store the
+ greeting in IMAP. Lastly, the attachment filename was incorrectly
+ using the full path instead of just the basename, which was
+ causing problems with retrieval of the greeting. (closes issue
+ #14950) Reported by: noahisaac (closes issue #15729) Reported by:
+ lmadsen ........
+
+2009-08-24 04:54 +0000 [r213791] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 213790 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213790 | moy | 2009-08-24 00:46:28 -0400 (Mon, 24 Aug 2009) | 1
+ line improve handling of openr2_chan_disconnect_call API failure,
+ unlikely, but happened on openr2 library bug ........
+
+2009-08-21 22:54 +0000 [r213739] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 213738 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 |
+ tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines
+ Clarifying comments in sip_register, and removing a dead section
+ ........
+
+2009-08-21 22:23 +0000 [r213721] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 213716 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 |
+ dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines
+ Register request line contains wrong address when user domain and
+ register host differ (closes issue #15539) Reported by:
+ Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by
+ Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel
+ (license 671) Tested by: Nick_Lewis, dvossel ........
+
+2009-08-21 21:44 +0000 [r213698] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 213697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug
+ 2009) | 12 lines Ensure that realtime mailboxes properly report
+ status on subscription. This patch modifies app_voicemail's
+ response to mailbox status subscriptions (via the internal event
+ system) to ensure that a subscription triggers an explicit poll
+ of the mailbox, so the subscriber can get an immediate cached
+ event with that status. Previously, the cache was only populated
+ with the status of non-realtime mailboxes. (closes issue #15717)
+ Reported by: natmlt ........
+
+2009-08-21 21:12 +0000 [r213636] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 213635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 |
+ dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines
+ fixes sip register parsing when user@domain is used (issue
+ #15008) (issue #15672) ........
+
+2009-08-21 16:55 +0000 [r213563] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk.h, /: Merged revisions 213560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009)
+ | 14 lines Merged revisions 213559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009)
+ | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files.
+ (closes issue #15698) Reported by: slavon Patches:
+ 20090817__issue15698.diff.txt uploaded by tilghman (license 14)
+ Tested by: slavon, tilghman ........ ................
+
+2009-08-21 16:06 +0000 [r213497] Jason Parker <jparker@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 213494 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r213494 | qwell | 2009-08-21 11:04:21 -0500
+ (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) |
+ 5 lines Clarify queues.conf comments to specify that variables
+ should be set in the dialplan. (closes issue #15755) Reported by:
+ trendboy ........ ................
+
+2009-08-21 04:25 +0000 [r213475-213481] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 213454 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213454 | moy | 2009-08-21 00:09:26 -0400 (Fri, 21 Aug 2009) | 1
+ line increment the mfcr2 monitor count when clearing the call
+ request ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 213216 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213216 | moy | 2009-08-19 23:26:59 -0400 (Wed, 19 Aug 2009) | 1
+ line fixed bug caused by calling ast_request without calling
+ ast_call on an R2 channel, ie, CHANISAVAIL ........
+
+2009-08-21 03:53 +0000 [r213453] Terry Wilson <twilson@digium.com>
+
+ * main/loader.c, /: Merged revisions 213450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 |
+ twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines
+ Make LOAD_ORDER actually work ........
+
+2009-08-20 21:50 +0000 [r213413] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 213404 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009)
+ | 12 lines Fix greeting retrieval from IMAP Properly check for
+ the current voicemail state and if it doesn't exist, create it.
+ (closes issue #14597) Reported by: wtca Patches: 14597_v2.patch
+ uploaded by mmichelson (license 60) Tested by: jpeeler ........
+
+2009-08-20 20:37 +0000 [r213350] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 213327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 |
+ mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7
+ lines Fix a crash by checking the proper pointer for validity
+ before deferencing it. (closes issue #15751) Reported by: atis
+ Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license
+ 242) ........
+
+2009-08-19 22:41 +0000 [r213182] Jason Parker <jparker@digium.com>
+
+ * main/alaw.c, main/ulaw.c, /: Merged revisions 213179 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) |
+ 5 lines Fix compile when certain G711 menuselect options are
+ enabled. (closes issue #15697) Reported by: slavon ........
+
+2009-08-19 21:25 +0000 [r213128] David Vossel <dvossel@digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 213113 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500
+ (Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009)
+ | 8 lines Fixes memory leak caused by incorrectly freeing
+ mixmonitor (closes issue #15699) Reported by: edantie Patches:
+ mixmonitor.patch uploaded by edantie (license 862) ........
+ ................
+
+2009-08-19 21:22 +0000 [r213095-213117] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 213098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 |
+ tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines
+ Better parsing for the "register" line Allows characters that are
+ otherwise used as delimiters to be used within certain fields
+ (like the secret). (closes issue #15008, closes issue #15672)
+ Reported by: tilghman Patches: 20090818__issue15008.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen, tilghman
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 213093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 |
+ tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines
+ If we have realtime caching enabled, 'sip reload' must purge
+ users/peers, even if the config files haven't changed. (closes
+ issue #12869) Reported by: bcnit Patches:
+ 20090819__issue12869__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: lasko ........
+
+2009-08-19 15:35 +0000 [r213047] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 213046 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213046 |
+ russell | 2009-08-19 10:32:18 -0500 (Wed, 19 Aug 2009) | 4 lines
+ Don't blow up on a NULL cdr. Reported in #asterisk-dev. ........
+
+2009-08-18 20:34 +0000 [r212931-212944] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Merged revisions 212939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212939 |
+ kpfleming | 2009-08-18 15:33:34 -0500 (Tue, 18 Aug 2009) | 1 line
+ Remove some accidentally-committed properties. ........
+
+ * sounds/Makefile, doc/tex/asterisk.tex, CREDITS, /,
+ UPGRADE-1.4.txt, sounds/sounds.xml, build_tools/prep_tarball:
+ Merged revisions 212922 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212922 |
+ kpfleming | 2009-08-18 15:29:37 -0500 (Tue, 18 Aug 2009) | 6
+ lines Convert this branch to Opsound music-on-hold. For more
+ details:
+ http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
+ ........
+
+2009-08-18 19:28 +0000 [r212866] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/extconfig.conf.sample: Merged revisions 212857 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18
+ Aug 2009) | 4 lines Make the default extconfig.conf match entries
+ with the sample res_mysql.conf. This eliminates a future source
+ of possible confusion with the configuration of 1.6.1 and higher.
+ ........
+
+2009-08-18 16:56 +0000 [r212769] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, /: Merged revisions 212758 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500
+ (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18
+ Aug 2009) | 1 line Removed some deadwood and added some doxygen
+ comments. ........ ................
+
+2009-08-18 16:41 +0000 [r212767] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /: Merged revisions 212764 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r212764 | seanbright | 2009-08-18 12:38:36 -0400 (Tue, 18 Aug
+ 2009) | 18 lines Merged revisions 212763 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug
+ 2009) | 11 lines Delay the creation of temporary files until we
+ have a valid manager command to handle. Without this patch,
+ asterisk creates a temporary file before determining if the
+ specified command is valid. If invalid, we weren't properly
+ cleaning up the file. (closes issue #15730) Reported by: zmehmood
+ Patches: M15730.diff uploaded by junky (license 177) Tested by:
+ zmehmood ........ ................
+
+2009-08-17 20:01 +0000 [r212631] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 212627 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009)
+ | 4 lines Check the return value of opendir(3), or we may crash.
+ (closes issue #15720) Reported by: tobias_e ........
+
+2009-08-17 18:56 +0000 [r212580-212584] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_agent.c: Merged revisions 212581 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug
+ 2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in
+ chan_agent. (closes issue #15668) Reported by: davidw ........
+
+ * main/logger.c: Merged revisions 212574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212574 |
+ seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8
+ lines Correct the return value check for ast_safe_system. The
+ logic here was reversed as ast_safe_system returns -1 on error
+ and not on success. Fix suggested by reporter. (closes issue
+ #15667) Reported by: loic ........
+
+2009-08-17 16:52 +0000 [r212509] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 212506 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500
+ (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009)
+ | 12 lines Fix segfault when reloading chan_misdn. If more ports
+ were specified than configured in misdn.conf a reload would crash
+ asterisk. The problem was the unconfigured port was using data
+ from the previously configured port. When the data for an
+ unconfigured port was freed a crash would result from the double
+ free. (closes issue #12113) Reported by: agupta Patches:
+ bug12113.patch uploaded by jpeeler (license 325) ........
+ ................
+
+2009-08-17 15:51 +0000 [r212434] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 212431 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500
+ (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix
+ uninitialized variable causing random MWI indications. (closes
+ issue #15727) Reported by: doda Patches: dahdi_changes.patch
+ uploaded by doda (license 853) ........ r212430 | rmudgett |
+ 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix
+ uninitialized variable. ........ ................
+
+2009-08-14 17:37 +0000 [r212250] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_curl.c, /: Merged revisions 212249 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212249 |
+ tilghman | 2009-08-14 12:36:40 -0500 (Fri, 14 Aug 2009) | 2 lines
+ Add SSL_VERIFYPEER, as requested on the -users list ........
+
+2009-08-13 15:47 +0000 [r212116] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 212113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 |
+ kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3
+ lines Ensure that T38FaxVersion is put into outgoing SDP in the
+ proper case. ........
+
+2009-08-13 13:56 +0000 [r212070] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 212067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 |
+ file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines
+ Check an actual populated variable when seeing if we need to do
+ video or not. ........
+
+2009-08-13 11:47 +0000 [r212030] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, /: Merged revisions 212027 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r212027 | ghenry | 2009-08-13 12:37:12 +0100 (Thu, 13
+ Aug 2009) | 6 lines Fixed typo (closes issue #15710) Reported by:
+ suretec ........
+
+2009-08-12 23:16 +0000 [r211951-211959] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 211957 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug
+ 2009) | 17 lines Merged revisions 211953 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug
+ 2009) | 10 lines This patch adds additional checking when
+ generating queue log TRANSFER events. The additional checks
+ prevent generation of false TRANSFER events in certain
+ situations. (closes issue #14536) Reported by: aragon Patches:
+ queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, mnicholson ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 211876 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 |
+ mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11
+ lines Make asterisk handle 423 Interval Too Short messages
+ better. This change uses separate values for the acceptable
+ minimum expiry provided by the 423 error and the expiry value
+ stored in the configuration file. Previously, the value pulled
+ from the configuration file would be overwritten. (closes issue
+ #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff
+ uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch
+ uploaded by Nick (license 657) Tested by: mnicholson ........
+
+2009-08-12 16:21 +0000 [r211785] Gavin Henry <ghenry@suretecsystems.com>
+
+ * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, /: Merged revisions 211767 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r211767 | ghenry | 2009-08-12 17:00:46 +0100 (Wed, 12
+ Aug 2009) | 33 lines Added three new attributes and applied a
+ patch to res_config_ldap.c attributetype (
+ AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC
+ 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR
+ caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
+ attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC
+ 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR
+ caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
+ attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent'
+ DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch
+ SUBSTR caseIgnoreSubstringsMatch SYNTAX
+ 1.3.6.1.4.1.1466.115.121.1.15) and patch
+ fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725)
+ Reported by: macogeek Patches:
+ fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license
+ 863) Tested by: suretec ........
+
+2009-08-10 19:51 +0000 [r211580-211585] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
+ (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
+ Aug 2009) | 1 line Conversion specifiers, not format specifiers
+ ........ ................
+
+ * apps/app_queue.c, apps/app_talkdetect.c, agi/eagi-sphinx-test.c,
+ res/res_config_curl.c, channels/chan_usbradio.c,
+ channels/chan_misdn.c, res/snmp/agent.c, apps/app_sms.c,
+ apps/app_verbose.c, apps/app_stack.c, apps/app_mixmonitor.c,
+ main/asterisk.c, main/dsp.c, main/timing.c,
+ doc/CODING-GUIDELINES, funcs/func_speex.c, main/frame.c,
+ utils/muted.c, apps/app_meetme.c, apps/app_alarmreceiver.c,
+ cdr/cdr_pgsql.c, res/res_musiconhold.c, channels/chan_iax2.c,
+ apps/app_followme.c, main/enum.c, main/indications.c,
+ res/res_config_sqlite.c, channels/misdn_config.c, utils/frame.c,
+ main/cli.c, pbx/pbx_loopback.c, channels/chan_phone.c,
+ funcs/func_enum.c, res/res_smdi.c, channels/chan_skinny.c,
+ funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c,
+ res/res_config_ldap.c, apps/app_adsiprog.c,
+ funcs/func_dialplan.c, main/pbx.c, main/dnsmgr.c,
+ funcs/func_sprintf.c, funcs/func_timeout.c, channels/chan_sip.c,
+ apps/app_privacy.c, res/res_limit.c, apps/app_waitforsilence.c,
+ codecs/codec_speex.c, agi/eagi-test.c, apps/app_morsecode.c,
+ funcs/func_cut.c, channels/chan_oss.c, main/netsock.c,
+ apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
+ pbx/pbx_dundi.c, utils/extconf.c, pbx/pbx_config.c,
+ apps/app_chanspy.c, res/res_odbc.c, apps/app_voicemail.c,
+ apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, /,
+ apps/app_record.c, main/utils.c, cdr/cdr_adaptive_odbc.c,
+ res/res_http_post.c, main/config.c, res/ael/pval.c, main/cdr.c,
+ main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
+ main/manager.c, apps/app_setcallerid.c, apps/app_osplookup.c,
+ main/features.c, main/http.c, channels/xpmr/xpmr.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, res/res_config_pgsql.c,
+ channels/chan_agent.c, funcs/func_math.c, apps/app_waituntil.c,
+ apps/app_disa.c, main/acl.c, apps/app_originate.c,
+ channels/iax2-provision.c: AST-2009-005
+
+2009-08-10 14:15 +0000 [r211350] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
+ file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
+ retrieval of the port used for the video stream when adding SDP
+ to a SIP message. (closes issue #15121) Reported by: jsmith
+ ........
+
+2009-08-09 15:43 +0000 [r211235-211278] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/astfd.c: Merged revisions 211275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
+ | 9 lines Merged revisions 211274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
+ | 2 lines Small oops. Clear the flags which have been checked.
+ ........ ................
+
+ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
+ tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
+ Check for NULL frame, before dereferencing pointer. (closes issue
+ #15617) Reported by: rain ........
+
+2009-08-07 20:18 +0000 [r211122] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
+ | 11 lines Recorded merge of revisions 211112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
+ | 4 lines Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936) ........ ................
+
+2009-08-07 18:20 +0000 [r211051] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 211040 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
+ | 21 lines Merged revisions 211038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
+ | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
+ not the membername. This is a partial revert of revision 82590,
+ which was an attempted cleanup, but in reality, it broke
+ QUEUE_MEMBER_LIST, which has always been intended as a method by
+ which component interfaces could be queried from the queue.
+ Membername isn't useful here, because that field cannot be used
+ to obtain further information about the member. See the
+ documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+ QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+ member argument for further justification. (closes issue #15664)
+ Reported by: rain Patches: app_queue-queue_member_list.diff
+ uploaded by rain (license 327) ........ ................
+
+2009-08-07 13:10 +0000 [r210995] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /: Merged revisions 210992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
+ kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
+ lines Workaround broken T.38 endpoints that offer tiny
+ MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
+ the maximum IFP size that should be sent to them, rather than the
+ maximum packet payload size. If such an endpoint also requests
+ UDPRedundancy as the error correction mode, we'll end up
+ calculating a tiny maximum IFP size, so small as to be unusable.
+ This patch sets a lower bound on what we'll consider the remote's
+ maximum IFP size to be, assuming that endpoints that do this
+ really can accept larger packets than they've offered to accept.
+ (closes issue #15649) Reported by: dazza76 ........
+
+2009-08-06 21:47 +0000 [r210911-210917] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 210914 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
+ | 14 lines Merged revisions 210913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
+ | 7 lines Because channel information can be accessed outside of
+ the channel thread, we must lock the channel prior to modifying
+ it. (closes issue #15397) Reported by: caspy Patches:
+ 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy ........ ................
+
+ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
+ revisions 210908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
+ tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
+ Allow Gosub to recognize quote delimiters without consuming them.
+ (closes issue #15557) Reported by: rain Patches:
+ 20090723__issue15557.diff.txt uploaded by tilghman (license 14)
+ Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
+ ........
+
+2009-08-06 17:49 +0000 [r210820] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
+ file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
+ Accept additional T.38 reinvites after an initial one has been
+ handled. Discussion of this subject has yielded that it is not
+ actually acceptable to change T.38 parameters after the initial
+ reinvite but declining is harsh and can cause the fax to fail
+ when it may be possible to allow it to continue. This patch
+ changes things so that additional T.38 reinvites are accepted but
+ parameter changes ignored. This gives the fax a fighting chance.
+ (closes issue #15610) Reported by: huangtx2009 ........
+
+2009-08-05 20:43 +0000 [r210686] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
+ (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
+ | 14 lines Dialplan starts execution before the channel setup is
+ complete. * Issue 15655: For the case where dialing is complete
+ for an incoming call, dahdi_new() was asked to start the PBX and
+ then the code set more channel variables. If the dialplan hungup
+ before these channel variables got set, asterisk would likely
+ crash. * Fixed potential for overlap incoming call to erroneously
+ set channel variables as global dialplan variables if the
+ ast_channel structure failed to get allocated. * Added missing
+ set of CALLINGSUBADDR in the dialing is complete case. (closes
+ issue #15655) Reported by: alecdavis ........ ................
+
+2009-08-05 18:56 +0000 [r210565-210566] Leif Madsen <lmadsen@digium.com>
+
+ * /: Merged revisions 210564 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009)
+ | 19 lines Merged revisions 210563 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+ | 11 lines Update imapstorage.txt documentation. Updated the
+ imapstorage.txt documentation to reflect that issues with
+ c-client versions older than 2007 seem to cause crashing issues
+ that are not seen with more recent versions. Documentation has
+ been updated to reflect this. (closes issue #14496) Reported by:
+ vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks ........ ................
+
+ * doc/tex/imapstorage.tex: Merged revisions 210564 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500
+ (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+ | 11 lines Update imapstorage.txt documentation. Updated the
+ imapstorage.txt documentation to reflect that issues with
+ c-client versions older than 2007 seem to cause crashing issues
+ that are not seen with more recent versions. Documentation has
+ been updated to reflect this. (closes issue #14496) Reported by:
+ vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks ........ ................
+
+2009-08-04 14:55 +0000 [r210191-210241] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 210238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug
+ 2009) | 16 lines Merged revisions 210237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
+ 2009) | 10 lines Eliminate spurious compiler warnings from system
+ headers on *BSD platforms. Ensure that system headers located in
+ /usr/local/include are actually treated as system headers by the
+ compiler, and not as local headers which are subject to warnings
+ from the -Wundef compiler option and others. (closes issue
+ #15606) Reported by: mvanbaak ........ ................
+
+ * configs/sip.conf.sample, configs/skinny.conf.sample, main/rtp.c,
+ channels/chan_mgcp.c, doc/chan_sip-perf-testing.txt,
+ contrib/scripts/realtime_pgsql.sql, /, channels/chan_sip.c,
+ channels/chan_skinny.c, configs/mgcp.conf.sample,
+ doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt,
+ configs/res_ldap.conf.sample: Merged revisions 210190 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03
+ Aug 2009) | 11 lines Rename 'canreinvite' option to
+ 'directmedia', with backwards compatibility. It is clear from
+ multiple mailing list, forum, wiki and other sorts of posts that
+ users don't really understand the effects that the 'canreinvite'
+ config option actually has, and that in some cases they think
+ that setting it to 'no' will actually cause various other
+ features (T.38, MOH, etc.) to not work properly, when in fact
+ this is not the case. This patch changes the proper name of the
+ option to what it should have been from the beginning
+ ('directmedia'), but preserves backwards compatibility for
+ existing configurations. ........
+
+2009-08-01 11:33 +0000 [r209837-209906] Russell Bryant <russell@digium.com>
+
+ * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209887 | russell | 2009-08-01 06:29:25 -0500
+ (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
+ | 5 lines Resolve a valgrind warning about a read from
+ uninitialized memory. (issue #15396) Reported by: aragon ........
+ ................
+
+ * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209839 | russell | 2009-08-01 06:02:07 -0500
+ (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009)
+ | 13 lines Modify how Playtones() is used in Milliwatt() to
+ resolve gain issue. When Milliwatt() was changed internally to
+ use Playtones() so that the proper tone was used, it introduced a
+ drop in gain in the output signal. So, use the playtones API
+ directly and specify a volume argument such that the output
+ matches the gain of the original Milliwatt() code. (closes issue
+ #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff
+ uploaded by russell (license 2) Tested by: rue_mohr ........
+ ................
+
+ * /, main/event.c: Merged revisions 209835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 |
+ russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines
+ Fix ast_event_queue_and_cache() to actually do the cache() part.
+ (closes issue #15624) Reported by: ffossard Tested by: russell
+ ........
+
+2009-08-01 01:34 +0000 [r209816] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx/pbx_config.c, channels/misdn/isdn_lib.c, utils/frame.c,
+ main/pbx.c, /, main/Makefile, channels/misdn/ie.c: Merged
+ revisions 209760-209761 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul
+ 2009) | 13 lines Merged revisions 209759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
+ 2009) | 7 lines Minor changes inspired by testing with latest
+ GCC. The latest GCC (what will become 4.5.x) has a few new
+ warnings, that in these cases found some either downright buggy
+ code, or at least seriously poorly designed code that could be
+ improved. ........ ................ r209761 | kpfleming |
+ 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert
+ accidental Makefile change. ................
+
+2009-07-31 22:01 +0000 [r209715] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 209711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 |
+ russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines
+ Fix some places where ast_event_type was used instead of
+ ast_event_ie_type. ........
+
+2009-07-30 18:51 +0000 [r209594] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_console.c, include/asterisk/abstract_jb.h,
+ apps/app_forkcdr.c, channels/chan_dahdi.c,
+ contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
+ codecs/lpc10/pitsyn.c: Merged revisions 209554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
+ dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
+ Fixes numerous spelling errors. Patch submitted by alecdavis.
+ (closes issue #15595) Reported by: alecdavis ........
+
+2009-07-30 14:40 +0000 [r209518] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 |
+ mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8
+ lines Fix a crash that can result if text codecs are allowed but
+ textsupport is disabled. (closes issue #15596) Reported by:
+ fabled Patches: sip-red.patch uploaded by fabled (license 448)
+ ........
+
+2009-07-28 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-beta4
+
+2009-07-28 00:19 +0000 [r209328] Tilghman Lesher <tlesher@digium.com>
+
+ * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
+ | 9 lines Merged revisions 209315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
+ | 2 lines Publish French extra sounds ........ ................
+
+2009-07-27 21:44 +0000 [r209265-209282] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
+ kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
+ lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
+ messages about T.38 negotiation in debug level 1 messages, clean
+ up some looping logic, and correct an improper use of ast_free()
+ for freeing an ast_frame. ........
+
+ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
+ kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
+ lines Make T.38 switchover in ReceiveFAX synchronous. In receive
+ mode, if the channel that ReceiveFAX is running on supports T.38,
+ we should *always* attempt to switch T.38, rather than listening
+ for an incoming CNG tone and only triggering on that. The channel
+ may be using a low-bitrate codec that distorts the CNG tone, the
+ sending FAX endpoint may not send CNG at all, or there could be a
+ variety of other reasons that we don't detect it, but in all
+ those cases if T.38 is available we certainly want to use it.
+ ........
+
+2009-07-27 20:58 +0000 [r209238] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Merged revisions 209235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 |
+ mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5
+ lines Gracefully handle malformed RTP text packets. AST-2009-004
+ ........
+
+2009-07-27 20:33 +0000 [r209234] David Brooks <dbrooks@digium.com>
+
+ * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c,
+ channels/chan_vpb.cc, res/res_smdi.c, /,
+ include/asterisk/module.h, main/features.c, res/res_agi.c: Merged
+ revisions 209098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
+ dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
+ Fixing typos. Replaces "recieved" with "received" and "initilize"
+ with "initialize" (closes issue #15571) Reported by: alecdavis
+ ........
+
+2009-07-27 20:23 +0000 [r209135-209222] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 209197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
+ 2009) | 9 lines Honor channel's music class when using realtime
+ music on hold. (closes issue #15051) Reported by: alexh Patches:
+ 15051.patch uploaded by mmichelson (license 60) Tested by: alexh
+ ........
+
+ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+ 209132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
+ 2009) | 24 lines Merged revisions 209131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+ 2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+ if desired. There are some VoIP providers out there that will not
+ accept SDP offers with odd numbered UDPTL ports. While it is my
+ personal opinion that these VoIP providers are misinterpreting
+ RFC 2327, it really is not a big deal to play along with their
+ silly little games. Of course, since restricting UDPTL ports to
+ only even numbers reduces the range of available ports by half,
+ so the option to use only even port numbers is off by default. A
+ user can enable the behavior by setting use_even_ports=yes in
+ udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+ 15182.patch uploaded by mmichelson (license 60) Tested by:
+ CGMChris ........ ................
+
+2009-07-27 16:07 +0000 [r209063] David Brooks <dbrooks@digium.com>
+
+ * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing
+ typos "recieved" with "received". From issue #15360, forgot to
+ apply to trunk and other branches.
+
+2009-07-27 15:40 +0000 [r209059] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 209056 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
+ kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
+ lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
+ underscore-variants to sub-makes. During the recent Makefile
+ improvements I made, it seemed the 'make' was automatically
+ carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
+ I removed the explict export of them. However, there are some
+ circumstances where make does this, and some where it does not,
+ so I've brought them back to ensure they are always exported. I
+ also removed an extraneous double setting of _ASTLDFLAGS on *BSD
+ platforms. ........
+
+2009-07-27 01:23 +0000 [r208927] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, main/translate.c: Merged revisions
+ 208924 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009)
+ | 9 lines Merged revisions 208923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
+ | 2 lines Fix logic errors from 208746 ........ ................
+
+2009-07-26 14:07 +0000 [r208889] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208886 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26
+ Jul 2009) | 2 lines add OpenBSD to the install_prereq script
+ ........
+
+2009-07-25 12:31 +0000 [r208816-208853] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208848 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25
+ Jul 2009) | 2 lines libxml2-dev is needed as well by default.
+ ........
+
+ * main/cli.c, /, configs/cli_aliases.conf.sample: Merged revisions
+ 208813 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 |
+ mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10
+ lines add default alias reload to run module reload. Requiring
+ 'module reload' to reload everything, including core etc makes
+ russell very unhappy. The default configuration already loads the
+ 'friendly' aliases template. Added 'reload=module reload' to that
+ template. Also removed the comment in main/cli.c that reload
+ should come back. ........
+
+2009-07-25 06:26 +0000 [r208755] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, channels/chan_skinny.c,
+ main/translate.c: Merged revisions 208749 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009)
+ | 13 lines Merged revisions 208746 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
+ | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
+ trivial changes, but I did not know of any other way to fix the
+ "dereferencing type-punned pointer will break strict-aliasing
+ rules" error without creating a tmp variable in chan_skinny.
+ ........ ................
+
+2009-07-24 21:13 +0000 [r208695-208710] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 208709 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208709 |
+ russell | 2009-07-24 16:12:43 -0500 (Fri, 24 Jul 2009) | 2 lines
+ Remove trailing whitespace. ........
+
+ * main/cli.c, /: Merged revisions 208706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208706 |
+ russell | 2009-07-24 15:54:37 -0500 (Fri, 24 Jul 2009) | 6 lines
+ Note that "reload" needs to be added back. I keep getting annoyed
+ at having to type "module reload" to reload everything, so I'm
+ adding a note that we need to add "reload" back. "module reload"
+ doesn't really make sense as the command to reload everything,
+ including the core. ........
+
+ * main/cli.c, /: Merged revisions 208693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208693 |
+ russell | 2009-07-24 15:25:23 -0500 (Fri, 24 Jul 2009) | 2 lines
+ Don't log a warning for something that does not affect operation.
+ ........
+
+2009-07-24 19:42 +0000 [r208664] Mark Michelson <mmichelson@digium.com>
+
+ * /: Fixing trunk-blocked property.
+
+2009-07-24 18:56 +0000 [r208596] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009)
+ | 14 lines Merged revisions 208592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
+ | 7 lines Do not log an ERROR if autoservice_stop() returns -1.
+ This does not indicate an error. A return of -1 just means that
+ the channel has been hung up. (reported in #asterisk-dev)
+ ........ ................
+
+2009-07-24 18:32 +0000 [r208591] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul
+ 2009) | 16 lines Merged revisions 208587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
+ 2009) | 10 lines Only send a BYE when hanging up a channel that
+ is up. For cases where Asterisk sends an INVITE and receives a
+ non 2XX final response, Asterisk would follow the INVITE
+ transaction by immediately sending a BYE, which was unnecessary.
+ (closes issue #14575) Reported by: chris-mac ........
+ ................
+
+2009-07-24 15:06 +0000 [r208551] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 208548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 |
+ kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8
+ lines Resolve a T.38 negotiation issue left over from the
+ udptl-updates merge. The udptl-updates branch that was merged
+ yesterday failed to properly send back T.38 SDP responses with
+ the correct error correction mode, if the incoming SDP from the
+ other end caused us to change error correction modes. This patch
+ corrects that situation. ........
+
+2009-07-24 14:39 +0000 [r208545] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208542 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24
+ Jul 2009) | 13 lines use aptitude for debian based systems The
+ function to check wether we need to install packages was using
+ dpkg-query which was gives wrong output on Debian 5 Also, the
+ apt-get has been replaced with aptitude because aptitude is now
+ the preferred way to handle packages on Debian (closes issue
+ #15570) Reported by: mvanbaak Patches:
+ 2009072400_installprereq-aptitude.diff uploaded by mvanbaak
+ (license 7) ........
+
+2009-07-23 22:31 +0000 [r208501] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, main/rtp.c, main/channel.c,
+ main/udptl.c, main/frame.c, /, channels/chan_sip.c,
+ apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h: Merged
+ revisions 208464 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 |
+ kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46
+ lines Rework of T.38 negotiation and UDPTL API to address
+ interoperability problems Over the past couple of months, a
+ number of issues with Asterisk negotiating (and successfully
+ completing) T.38 sessions with various endpoints have been found.
+ This patch attempts to address many of them, primarily focused
+ around ensuring that the endpoints' MaxDatagram size is honored,
+ and in addition by ensuring that T.38 session parameter
+ negotiation is performed correctly according to the ITU T.38
+ Recommendation. The major changes here are: 1) T.38 applications
+ in Asterisk (app_fax) only generate/receive IFP packets, they do
+ not ever work with UDPTL packets. As a result of this, they
+ cannot be allowed to generate packets that would overflow the
+ other endpoints' MaxDatagram size after the UDPTL stack adds any
+ error correction information. With this patch, the application is
+ told the maximum *IFP* size it can generate, based on a
+ calculation using the far end MaxDatagram size and the active
+ error correction mode on the T.38 session. The same is true for
+ sending *our* MaxDatagram size to the remote endpoint; it is
+ computed from the value that the application says it can accept
+ (for a single IFP packet) combined with the active error
+ correction mode. 2) All treatment of T.38 session parameters as
+ 'capabilities' in chan_sip has been removed; these parameters are
+ not at all like audio/video stream capabilities. There are strict
+ rules to follow for computing an answer to a T.38 offer, and
+ chan_sip now follows those rules, using the desired parameters
+ from the application (or channel) that wants to accept the T.38
+ negotiation. 3) chan_sip now stores and forwards
+ ast_control_t38_parameters structures for tracking 'our' and
+ 'their' T.38 session parameters; this greatly simplifies
+ negotiation, especially for pass-through calls. 4) Since T.38
+ negotiation without specifying parameters or receiving the final
+ negotiated parameters is not very worthwhile, the AST_CONTROL_T38
+ control frame has been removed. A note has been added to
+ UPGRADE.txt about this removal, since any out-of-tree
+ applications that use it will no longer function properly until
+ they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
+ https://reviewboard.asterisk.org/r/310/ ........
+
+2009-07-23 19:36 +0000 [r208391] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul
+ 2009) | 24 lines Merged revisions 208386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
+ 2009) | 17 lines Fix a problem where a 491 response could be sent
+ out of dialog. This generalizes the fix for issue 13849. The
+ initial fix corrected the problem that Asterisk would reply with
+ a 491 if a reinvite were received from an endpoint and we had not
+ yet received an ACK from that endpoint for the initial INVITE it
+ had sent us. This expansion also allows Asterisk to appropriately
+ handle an INVITE with authorization credentials if Asterisk had
+ not received an ACK from the previous transaction in which
+ Asterisk had responded to an unauthorized INVITE with a 407.
+ (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
+ uploaded by mmichelson (license 60) Tested by: klaus3000 ........
+ ................
+
+2009-07-23 19:25 +0000 [r208387] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500
+ (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009)
+ | 6 lines Only set the priindication setting when not performing
+ a reload (closes issue #14696) Reported by: fdecher ........
+ ................
+
+2009-07-23 16:30 +0000 [r208266-208320] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul
+ 2009) | 9 lines Merged revisions 208312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
+ 2009) | 3 lines Remove inaccurate XXX comment. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul
+ 2009) | 15 lines Merged revisions 208262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
+ 2009) | 8 lines Properly handle 183 responses which do not
+ contain an SDP. (closes issue #15442) Reported by: ffloimair
+ Patches: 15442.patch uploaded by mmichelson (license 60) Tested
+ by: tkarl, ffloimair ........ ................
+
+2009-07-22 21:46 +0000 [r208116] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 |
+ qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines
+ Restore an int declaration on PPC platforms. This x is one crafty
+ little bugger... It was used for 2 different things (one of which
+ was only done on PPC) in 1.4. One of the uses were removed in
+ trunk, and with it went the declaration. (closes issue #14038)
+ Reported by: ffloimair ........
+
+2009-07-22 16:52 +0000 [r207949-208053] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_realtime.c: Merged revisions 208052 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208052 |
+ tilghman | 2009-07-22 11:49:42 -0500 (Wed, 22 Jul 2009) | 7 lines
+ Clarify documentation on 'realtime update2' to show more than one
+ condition. (closes issue #15357) Reported by: snuffy Patches:
+ bug_fix_doc_update2.diff uploaded by snuffy (license 35)
+ (slightly modified by me) ........
+
+ * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500
+ (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009)
+ | 8 lines Force an error if a blank is passed to QUOTE (because
+ the documentation states the argument is not optional). This
+ change makes URIENCODE and QUOTE behave similarly, since the
+ documentation states that the argument is not optional, for both.
+ (closes issue #15439) Reported by: pkempgen Patches:
+ 20090706__issue15439.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2009-07-21 22:23 +0000 [r207930] Russell Bryant <russell@digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 207925 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r207925 | russell | 2009-07-21 17:22:18 -0500 (Tue, 21 Jul 2009)
+ | 4 lines Note that we use tabs instead of spaces for
+ indentation. I'm surprised this was never actually in here...
+ ........
+
+2009-07-21 20:30 +0000 [r207785-207862] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500
+ (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
+ | 9 lines Wait for wink before dialing when using E&M wink
+ signaling There was already code for other signaling types in
+ dahdi_handle_event to handle dialing if a dial operation dial
+ string was present. Simply add SIG_EMWINK to the list. (closes
+ issue #14434) Reported by: araasch ........ ................
+
+ * channels/chan_dahdi.c: Revert r207638, this approach could
+ potentially block for an unacceptable amount of time.
+
+2009-07-21 14:32 +0000 [r207727] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 207723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul
+ 2009) | 11 lines Merged revisions 207714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
+ 2009) | 5 lines Document default timeout for AMI originations.
+ AST-224 ........ ................
+
+2009-07-21 13:56 +0000 [r207685] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile,
+ codecs/Makefile, utils/Makefile, funcs/Makefile,
+ codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile,
+ codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules,
+ pbx/Makefile, res/Makefile: Merged revisions 207680 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500
+ (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
+ 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
+ honored. This commit changes the build system so that
+ user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
+ the compiler/linker *after* all flags provided by the build
+ system itself, so that the user can effectively override the
+ build system's flags if desired. In addition, ASTCFLAGS and
+ ASTLDFLAGS can now be provided *either* in the environment before
+ running 'make', or as variable assignments on the 'make' command
+ line. As a result, the use of COPTS and LDOPTS is no longer
+ necessary, so they are no longer documented, but are still
+ supported so as not to break existing build systems that supply
+ them when building Asterisk. ........ ................
+
+2009-07-21 04:51 +0000 [r207638] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Wait for wink before dialing when using
+ E&M wink signaling This patch adds a new dahdi_wait function to
+ specifically wait for the wink event. If the wink is not
+ eventually received the channel is hung up. (closes issue #14434)
+ Reported by: araasch Patches: emwinkmod uploaded by araasch
+ (license 693)
+
+2009-07-20 22:14 +0000 [r207523] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul
+ 2009) | 39 lines Merged revisions 207423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
+ 2009) | 33 lines Answer video SDP offers properly when
+ videosupport is not enabled. Copied from Review board: In issue
+ 12434, the reporter describes a situation in which audio and
+ video is offered on the call, but because videosupport is
+ disabled in sip.conf, Asterisk gives no response at all to the
+ video offer. According to RFC 3264, all media offers should have
+ a corresponding answer. For offers we do not intend to actually
+ reply to with meaningful values, we should still reply with the
+ port for the media stream set to 0. In this patch, we take note
+ of what types of media have been offered and save the information
+ on the sip_pvt. The SDP in the response will take into account
+ whether media was offered. If we are not otherwise going to
+ answer a media offer, we will insert an appropriate m= line with
+ the port set to 0. It is important to note that this patch is
+ pretty much a bandage being applied to a broken bone. The patch
+ *only* helps for situations where video is offered but
+ videosupport is disabled and when udptl_pt is disabled but T.38
+ is offered. Asterisk is not guaranteed to respond to every media
+ offer. Notable cases are when multiple streams of the same type
+ are offered. The 2 media stream limit is still present with this
+ patch, too. In trunk and the 1.6.X branches, things will be a bit
+ different since Asterisk also supports text in SDPs as well.
+ (closes issue #12434) Reported by: mnnojd Review:
+ https://reviewboard.asterisk.org/r/311 Review:
+ https://reviewboard.asterisk.org/r/313 ........ ................
+
+2009-07-20 16:41 +0000 [r207364] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 207361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009)
+ | 16 lines Merged revisions 207360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
+ | 9 lines Only do the chan->fdno check in ast_read() in a
+ developer build. I changed this check to only happen in a
+ dev-mode build. I also added a comment explaining what is going
+ on. I also made it so that detection of this situation does not
+ affect ast_read() operation. (closes issue #14723) Reported by:
+ seadweller ........ ................
+
+2009-07-18 04:19 +0000 [r207327] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 207317 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17 Jul 2009)
+ | 3 lines Flag field in wrong position. Reported by "Hoggins!" on
+ asterisk-dev list. ........
+
+2009-07-18 03:50 +0000 [r207315] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Merged
+ revisions 145293,158010 from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 to make
+ merging easier. These changes are already on trunk.
+ ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500
+ (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c
+ channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk
+ to make merging easier later. ........ r145200 | rmudgett |
+ 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines *
+ Miscellaneous formatting changes to make v1.4 and trunk more
+ merge compatible in the mISDN area. channels/chan_misdn.c *
+ Eliminated redundant code in cb_events() EVENT_SETUP ........
+ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008)
+ | 9 lines improved helptext of misdn_set_opt. ........ r142181 |
+ rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
+ Cleaned up comment ........ r138738 | rmudgett | 2008-08-18
+ 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
+ channels/chan_misdn.c * Made bearer2str() use
+ allowed_bearers_array[] * Made use the causes.h defines instead
+ of hardcoded numbers. * Made use Asterisk presentation indicator
+ values if either of the mISDN presentation or screen options are
+ negative. * Updated the misdn_set_opt application option
+ descriptions. * Renamed the awkward Caller ID presentation
+ misdn_set_opt application option value not_screened to
+ restricted. Deprecated the not_screened option value.
+ channels/misdn/isdn_lib.c * Made use the causes.h defines instead
+ of hardcoded numbers. * Fixed some spelling errors and typos. *
+ Added all defined facility code strings to fac2str().
+ channels/misdn/isdn_lib.h * Added doxygen comments to struct
+ misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
+ comments to struct misdn_stack. channels/misdn_config.c
+ configs/misdn.conf.sample * Updated the mISDN presentation and
+ screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex)
+ * Updated the misdn_set_opt application option descriptions. *
+ Fixed some spelling errors and typos. ................ r158010 |
+ rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
+ Merged revision 157977 from
+ https://origsvn.digium.com/svn/asterisk/team/group/issue8824
+ ........ Fixes JIRA ABE-1726 The dial extension could be empty if
+ you are using MISDN_KEYPAD to control ISDN provider features.
+ ................
+
+2009-07-17 22:31 +0000 [r207226-207257] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17
+ Jul 2009) | 2 lines Add flag here, too (as requested by jsmith)
+ ........
+
+ * /, doc/tex/odbcstorage.tex, UPGRADE.txt: Merged revisions 207224
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17
+ Jul 2009) | 2 lines Document the "flag" field in the
+ voicemessages table. ........
+
+2009-07-17 19:40 +0000 [r207104-207159] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500
+ (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009)
+ | 7 lines Fix format specifier to print out an unsigned long
+ long. Yep, it's even ifdefed out code. But it made it to the RR
+ list... (closes issue #14726) Reported by: lmadsen ........
+ ................
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17
+ Jul 2009) | 2 lines Update some missing allowed options for
+ overlapdial ........
+
+2009-07-17 17:52 +0000 [r206869-207030] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 |
+ dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines
+ sip option flags handled incorrectly (closes issue #15376)
+ Reported by: Takehiko Ooshima Tested by: dvossel,
+ Takehiko_Ooshima ........
+
+ * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009)
+ | 20 lines Merged revisions 206938 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009)
+ | 14 lines SIP incorrect From: header information when callpres
+ is prohib Some ITSP make use of the "Anonymous" display name to
+ detect a requirement to withhold caller id across the PSTN. This
+ does not work if the display name is "Unknown". (closes issue
+ #14465) Reported by: Nick_Lewis Patches:
+ chan_sip.c-callerpres.patch uploaded by Nick (license 657)
+ chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license
+ 671) Tested by: Nick_Lewis, dvossel ........ ................
+
+ * /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009)
+ | 6 lines TIMEOUT(absolute) returned negative value. (closes
+ issue #15513) Reported by: ys ........
+
+ * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500
+ (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009)
+ | 6 lines error in iax.conf related IP-based access control
+ (closes issue #15518) Reported by: pkempgen ........
+ ................
+
+ * /, main/callerid.c: Merged revisions 206868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009)
+ | 14 lines Merged revisions 206867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009)
+ | 8 lines avoid segfault caused by user error If the CALLERPRES()
+ dialplan function is set to nothing, a segfault occurs. This is
+ user error to begin with, but I'd rather see a cli warning
+ message than have Asterisk crash on me. ........ ................
+
+2009-07-16 16:53 +0000 [r206811] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500
+ (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009)
+ | 6 lines Fix a memory leak. (closes issue #15517) Reported by:
+ adomjan Patches: func_realtime.c-ast_variable_destroy.diff
+ uploaded by adomjan (license 487) ........ ................
+
+2009-07-15 22:04 +0000 [r206770] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 |
+ dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines
+ Session timer were not activated if Supported header field in
+ INVITE had both "timer" and other options. (closes issue #15403)
+ Reported by: makoto Patches: sip-session-timer.patch uploaded by
+ makoto (license ........
+
+2009-07-15 21:50 +0000 [r206765] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
+ Merged revisions 206707 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009)
+ | 33 lines Merged revisions 206706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500
+ (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... Fixed chan_misdn crash because mISDNuser library is
+ not thread safe. With Asterisk the mISDNuser library is driven by
+ two threads concurrently: 1.
+ channels/misdn/isdn_lib.c::manager_event_handler() 2.
+ channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls
+ into the library are done concurrently and recursively from
+ isdn_lib.c. Both threads can fiddle with the master/child
+ layer3_proc_t lists. One thread may traverse the list when the
+ other interrupts it and then removes the list element which the
+ first thread was currently handling. This is exactly what caused
+ the crash. About 60 calls were needed to a Gigaset CX475 before
+ it occurred once. This patch adds locking when calling into the
+ mISDNuser library. This also fixes some cb_log calls with wrong
+ port parameter. JIRA ABE-1913 Patches: misdn-locking.patch
+ (Modified with mostly cosmetic changes) ..........
+ ................ ................
+
+2009-07-15 20:20 +0000 [r206703] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 |
+ dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines
+ callerid(num) is wrong when username is missing A domain only sip
+ uri <sip:123.123.123.123> would return 123.123.123.123 as callid
+ num. Now, if the username is missing from a uri, the callerid num
+ field is left empty. (closes issue #15476) Reported by: viraptor
+ ........
+
+2009-07-15 16:04 +0000 [r206639] Sean Bright <sean@malleable.com>
+
+ * codecs/codec_dahdi.c, /: Merged revisions 206636 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400
+ (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed,
+ 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we
+ are asking for it. ........ ................
+
+2009-07-14 20:26 +0000 [r206598] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c, contrib/scripts/meetme.sql: Merged
+ revisions 206567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206567 |
+ tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines
+ Document all meetme realtime fields, and in the process, make
+ some field lengths more consistent. (closes issue #15493)
+ Reported by: lasko Patches: meetme.diff uploaded by lasko
+ (license 833) ........
+
+2009-07-14 19:49 +0000 [r206565] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500
+ (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009)
+ | 28 lines Fixes several call transfer issues with chan_misdn. *
+ issue #14355 - Crash if attempt to transfer a call to an
+ application. Masquerade the other pair of the four asterisk
+ channels involved in the two calls. The held call already must be
+ a bridged call (not an applicaton) or it would have been
+ rejected. * issue #14692 - Held calls are not automatically
+ cleared after transfer. Allow the core to initate disconnect of
+ held calls to the ISDN port. This also fixes a similar case where
+ the party on hold hangs up before being transferred or taken off
+ hold. * JIRA ABE-1903 - Orphaned held calls left in
+ music-on-hold. Do not simply block passing the hangup event on
+ held calls to asterisk core. * Fixed to allow held calls to be
+ transferred to ringing calls. Previously, held calls could only
+ be transferred to connected calls. * Eliminated unused call
+ states to simplify hangup code. * Eliminated most uses of
+ "holded" because it is not a word. (closes issue #14355) (closes
+ issue #14692) Reported by: sodom Patches:
+ misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett ........ ................
+
+2009-07-14 14:59 +0000 [r206389] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 206386 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206386 | russell | 2009-07-14 09:51:44 -0500
+ (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206385 | russell | 2009-07-14 09:48:00 -0500
+ (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009)
+ | 6 lines Ensure apathetic replies are sent out on the proper
+ socket. chan_iax2 supports multiple address bindings. The
+ send_apathetic_reply() function did not attempt to send its
+ response on the same socket that the incoming message came in on.
+ ........ ................ ................
+
+2009-07-14 01:59 +0000 [r206373] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+ revisions 206341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009)
+ | 11 lines Merged revisions 206284 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009)
+ | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911
+ ........ ................
+
+2009-07-13 23:27 +0000 [r206281] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 |
+ dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines
+ dns lookup of peername rather than peer's host in
+ transmit_register() (closes issue #15052) Reported by: fsantulli
+ Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by
+ fsantulli (license 818) Tested by: fsantulli ........
+
+2009-07-13 16:24 +0000 [r206187] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009)
+ | 2 lines Remove reference to non-existent help file ........
+
+2009-07-10 21:46 +0000 [r205986] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 |
+ dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines
+ SIP register not using peer's outbound proxy If callbackextension
+ is defined for a peer it successfully causes a registration to
+ occur, but the registration ignores the outboundproxy settings
+ for the peer. This patch allows the peer to be passed to
+ obproxy_get() in transmit_register(). (closes issue #14344)
+ Reported by: Nick_Lewis Patches:
+ callbackextension_peer_trunk.diff uploaded by dvossel (license
+ 671) Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/294/ ........
+
+2009-07-10 18:45 +0000 [r205942] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /: Merged revisions 205939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 |
+ kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line
+ Update comments about the level of T.38 support in Asterisk.
+ ........
+
+2009-07-10 17:54 +0000 [r205882] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul
+ 2009) | 30 lines Merged revisions 205877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500
+ (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500
+ (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........ ................ ................
+ ................
+
+2009-07-10 16:47 +0000 [r205841] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009)
+ | 37 lines Merged revisions 205804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009)
+ | 31 lines SIP registration auth loop caused by stale nonce If an
+ endpoint sends two registration requests in a very short period
+ of time with the same nonce, both receive 401 responses from
+ Asterisk, each with a different nonce (the second 401 containing
+ the current nonce and the first one being stale). If the endpoint
+ responds to the first 401, it does not match the current nonce so
+ Asterisk sends a third 401 with a newly generated nonce (which
+ updates the current nonce)... Now if the endpoint responds to the
+ second 401, it does not match the current nonce either and
+ Asterisk sends a fourth 401 with a newly generated nonce... This
+ loop goes on and on. There appears to be a simple fix for this.
+ If the nonce from the request does not match our nonce, but is a
+ good response to a previous nonce, instead of sending a 401 with
+ a newly generated nonce, use the current one instead. This breaks
+ the loop as the nonce is not updated until a response is
+ received. Additional logic has been added to make sure no nonce
+ can be responded to twice though. (closes issue #15102) Reported
+ by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license
+ 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by:
+ Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........
+ ................
+
+2009-07-10 16:01 +0000 [r205781] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 205780 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205780 |
+ kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11
+ lines Eliminate extraneous LOG_DEBUG messages generated by
+ app_fax. The transmit_audio() and transmit_t38() functions in
+ app_fax have processing loops that are supposed to wait for
+ frames to arrive on the channel and then handle them, but they
+ also have short timeouts so that the loops can have watchdog
+ timers and do other required processing. This commit changes the
+ loops to not actually call ast_read() and attempt to process the
+ returned frame unless a frame actually arrived, eliminating
+ hundreds of LOG_DEBUG messages and slightly improving
+ performance. ........
+
+2009-07-10 15:58 +0000 [r205779] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul
+ 2009) | 16 lines Merged revisions 205775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........ ................
+
+2009-07-10 15:36 +0000 [r205773] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 |
+ kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12
+ lines Fix some remaining T.38 negotiation problems in app_fax.
+ Revision 205696 did not quite fix all the issues with the T.38
+ negotiation changes and app_fax; this patch corrects them, along
+ with a couple of other minor issues. (closes issue #15480)
+ Reported by: dimas Patches: test2-15480.patch uploaded by dimas
+ (license 88) ........
+
+2009-07-09 23:56 +0000 [r205731] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009)
+ | 21 lines No audio on calls from Asterisk to various ISDN
+ devices until DTMF sent by caller. Add missing clearing of the
+ dialing flag when the ISDN call is CONNECTED. (i.e. When libpri
+ generates the event PRI_EVENT_ANSWER.) (closes issue #15420)
+ Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt
+ uploaded by alecdavis (license 585) Tested by: scottbmilne,
+ alecdavis (closes issue #15416) Reported by: avinoash (closes
+ issue #15389) Reported by: alecdavis This patch should also fix
+ the following issue: (issue #15205) Reported by: vinsik ........
+
+2009-07-09 21:27 +0000 [r205699] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c:
+ Merged revisions 205696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 |
+ kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16
+ lines Repair ability of SendFAX/ReceiveFAX to respond to T.38
+ switchover. Recent changes in T.38 negotiation in Asterisk caused
+ these applications to not respond when the other endpoint
+ initiated a switchover to T.38; this resulted in the T.38
+ switchover failing, and the FAX attempt to be made using an audio
+ connection, instead of T.38 (which would usually cause the FAX to
+ fail completely). This patch corrects this problem, and the
+ applications will now correctly respond to the T.38 switchover
+ request. In addition, the response will include the appopriate
+ T.38 session parameters based on what the other end offered and
+ what our end is capable of. (closes issue #14849) Reported by:
+ afosorio ........
+
+2009-07-09 16:19 +0000 [r205595-205603] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500
+ (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09
+ Jul 2009) | 2 lines Changing ast_samp2tv to not use floating
+ point. ........ ................
+
+ * channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /:
+ Merged revisions 205479 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009)
+ | 16 lines Merged revisions 205471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009)
+ | 10 lines Fixes 8khz assumptions Many calculations assume 8khz
+ is the codec rate. This is not always the case. This patch only
+ addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there
+ are other areas that make this assumption as well. Review:
+ https://reviewboard.asterisk.org/r/306/ ........ ................
+
+2009-07-09 08:34 +0000 [r205535] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, main/ssl.c: Merged revisions 205532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 |
+ mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines
+ pthread_self returns a pthread_t which is not an unsigned int on
+ all pthread implementations. Casting it to an unsigned int fixes
+ compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit
+ ........
+
+2009-07-08 22:15 +0000 [r205411-205413] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/pbx.h, include/asterisk/devicestate.h,
+ main/pbx.c, /, main/devicestate.c: Merged revisions 205412 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500
+ (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009)
+ | 6 lines moving ast_devstate_to_extenstate to pbx.c from
+ devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This
+ change fixes a compile time error with chan_vpb as well. ........
+ ................
+
+ * /, main/devicestate.c: Merged revisions 205410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205410 |
+ dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines
+ missing comma in devstatestring array ........
+
+2009-07-08 19:28 +0000 [r205353] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 205350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul
+ 2009) | 20 lines Merged revisions 205349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul
+ 2009) | 14 lines Prevent phantom calls to queue members. If a
+ caller were to hang up while a periodic announcement or position
+ were being said, the return value for those functions would
+ incorrectly indicate that the caller was still in the queue. With
+ these changes, the problem does not occur. (closes issue #14631)
+ Reported by: latinsud Patches: queue_announce_ghost_call2.diff
+ uploaded by latinsud (license 745) (with small modification from
+ me) ........ ................
+
+2009-07-08 18:22 +0000 [r205302] Jason Parker <jparker@digium.com>
+
+ * config.guess, config.sub, /: Merged revisions 205291 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205291 | qwell | 2009-07-08 13:19:46 -0500
+ (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul
+ 2009) | 1 line Update config.guess and config.sub from the
+ savannah.gnu.org git repo. ........ ................
+
+2009-07-08 18:18 +0000 [r205287] David Brooks <dbrooks@digium.com>
+
+ * /, main/features.c: Merged revisions 205254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 |
+ dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines
+ Fixes Park() argument handling Park() was not respecting the
+ arguments passed to it. Any extension/context/priority given to
+ it was being ignored. This patch remedies this. (closes issue
+ #15380) Reported by: DLNoah ........
+
+2009-07-08 17:00 +0000 [r205223] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: oops, fixing build
+
+2009-07-08 16:55 +0000 [r205217] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500
+ (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009)
+ | 10 lines ast_samp2tv needs floating point for 16khz audio In
+ ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The
+ .5 is currently stripped off because we don't calculate using
+ floating points. This causes madness with 16khz audio. (issue
+ ABE-1899) Review: https://reviewboard.asterisk.org/r/305/
+ ........ ................
+
+2009-07-08 16:30 +0000 [r205207] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: Merged revisions 205196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009)
+ | 9 lines Merged revisions 205188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009)
+ | 2 lines Add redirection warnings for the invalid language codes
+ previously removed. ........ ................
+
+2009-07-08 15:57 +0000 [r205148-205154] Russell Bryant <russell@digium.com>
+
+ * /, main/ssl.c: Merged revisions 205151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 |
+ russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines
+ Use tabs instead of spaces for indentation. ........
+
+ * include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c,
+ /, main/Makefile, res/res_crypto.c, main/ssl.c (added): Merged
+ revisions 205120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 |
+ russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
+ Move OpenSSL initialization to a single place, make library usage
+ thread-safe. While doing some reading about OpenSSL, I noticed a
+ couple of things that needed to be improved with our usage of
+ OpenSSL. 1) We had initialization of the library done in multiple
+ modules. This has now been moved to a core function that gets
+ executed during Asterisk startup. We already link OpenSSL into
+ the core for TCP/TLS functionality, so this was the most logical
+ place to do it. 2) OpenSSL is not thread-safe by default.
+ However, making it thread safe is very easy. We just have to
+ provide a couple of callbacks. One callback returns a thread ID.
+ The other handles locking. For more information, start with the
+ "Is OpenSSL thread-safe?" question on the FAQ page of
+ openssl.org. ........
+
+2009-07-06 13:41 +0000 [r204951] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, /: Merged revisions 204948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 |
+ kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7
+ lines Improve handling of AST_CONTROL_T38 and
+ AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This
+ change allows applications that request T.38 negotiation on a
+ channel that does not support it to get the proper indication
+ that it is not supported, rather than thinking that negotiation
+ was started when it was not. ........
+
+2009-07-02 22:06 +0000 [r204838] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500
+ (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009)
+ | 10 lines Removed confusing warning message "Got Busy in
+ Connected State" If an incoming mISDN call is answered with the
+ Answer application and a subsequent Dial gets a busy endpoint
+ then it is valid for that already connected channel to get the
+ busy indication. Asterisk will play the busy tones until the
+ dialplan plays something else or hangs up the call. (closes issue
+ #11974) Reported by: fvdb ........ ................
+
+2009-07-02 16:12 +0000 [r204711] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c: Merged revisions 204710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009)
+ | 21 lines Merged revisions 204681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009)
+ | 14 lines Improved mapping of extension states from combined
+ device states. This fixes a few issues with incorrect extension
+ states and adds a cli command, core show device2extenstate, to
+ display all possible state mappings. (closes issue #15413)
+ Reported by: legart Patches: exten_helper.diff uploaded by
+ dvossel (license 671) Tested by: dvossel, legart, amilcar Review:
+ https://reviewboard.asterisk.org/r/301/ ........ ................
+
+2009-06-30 21:30 +0000 [r204611] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500
+ (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009)
+ | 6 lines More incorrect language codes, plus ensuring that
+ regionalizations use the specified language, and not English for
+ grammar. (closes issue #15022) Reported by: greenfieldtech
+ Patches: 20090519__issue15022.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+2009-06-30 18:55 +0000 [r204478] Jason Parker <jparker@digium.com>
+
+ * /, main/say.c: Merged revisions 204475 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) |
+ 9 lines Merged revisions 204474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) |
+ 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a
+ comment typo in passing. ........ ................
+
+2009-06-30 18:44 +0000 [r204473] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge
+ of revisions 204470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009)
+ | 18 lines Recorded merge of revisions 204469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009)
+ | 11 lines "tw" is the language specification for Twi (from
+ Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier
+ Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__trunk.diff.txt uploaded by
+ tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt
+ uploaded by tilghman (license 14)
+ 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by
+ tilghman (license 14) Tested by: volivier ........
+ ................
+
+2009-06-30 17:22 +0000 [r204442] Russell Bryant <russell@digium.com>
+
+ * configs/res_config_sqlite.conf (removed),
+ configs/res_config_sqlite.conf.sample (added), /: Merged
+ revisions 204440 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 |
+ russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines
+ Rename res_config_sqlite.conf to res_config_sqlite.conf.sample
+ (missing .sample). ........
+
+2009-06-29 22:53 +0000 [r204250-204304] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun
+ 2009) | 15 lines Merged revisions 204300 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun
+ 2009) | 9 lines Add error message so that it is clear why a SIP
+ peer was not processed when a DNS lookup fails on a host or
+ outboundproxy. (closes issue #13432) Reported by: p_lindheimer
+ Patches: outboundproxy.patch uploaded by p (license 558) ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun
+ 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun
+ 2009) | 22 lines Fix a problem where chan_sip would ignore "old"
+ but valid responses. chan_sip has had a problem for quite a long
+ time that would manifest when Asterisk would send multiple SIP
+ responses on the same dialog before receiving a response. The
+ problem occurred because chan_sip only kept track of the highest
+ outgoing sequence number used on the dialog. If Asterisk sent two
+ requests out, and a response arrived for the first request sent,
+ then Asterisk would ignore the response. The result was that
+ Asterisk would continue retransmitting the requests and ignoring
+ the responses until the maximum number of retransmissions had
+ been reached. The fix here is to rearrange the code a bit so that
+ instead of simply comparing the sequence number of the response
+ to our latest outgoing sequence number, we walk our list of
+ outstanding packets and determine if there is a match. If there
+ is, we continue. If not, then we ignore the response. In doing
+ this, I found a few completely useless variables that I have now
+ removed. (closes issue #11231) Reported by: flefoll Review:
+ https://reviewboard.asterisk.org/r/298 ........ r204246 |
+ mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3
+ lines Fix build oops. ........ ................
+
+2009-06-27 09:55 +0000 [r203961] Russell Bryant <russell@digium.com>
+
+ * CHANGES, /: Merged revisions 203960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 |
+ russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines
+ Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt.
+ ........
+
+2009-06-27 01:24 +0000 [r203941] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500
+ (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009)
+ | 16 lines The ISDN CPE side should not exclusively pick B
+ channels normally. Before this patch, Asterisk unconditionally
+ picked B channels exclusively on the CPE side and normally
+ allowed alternative B channels on the network side. Now Asterisk
+ does the opposite. Reasons for the CPE side to normally not pick
+ B channels exclusively: * For CPE point-to-multipoint mode (i.e.
+ phone side), the CPE side does not have enough information to
+ exclusively pick B channels. (There may be other devices on the
+ line.) * Q.931 gives preference to the network side picking B
+ channels. * Some telcos require the CPE side to not pick B
+ channels exclusively. (closes issue #14383) Reported by:
+ mbrancaleoni ........ ................
+
+2009-06-26 22:14 +0000 [r203857] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500
+ (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009)
+ | 5 lines Make sure to recreate the dahdi pseudo channel after
+ dahdi restart (closes issue #14477) Reported by: timking ........
+ ................
+
+2009-06-26 21:27 +0000 [r203782-203828] Russell Bryant <russell@digium.com>
+
+ * /, main/file.c: Merged revisions 203802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009)
+ | 22 lines Merged revisions 203785 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009)
+ | 15 lines Don't fast forward past the end of a message. This is
+ nice change for users of the voicemail application. If someone
+ gets a little carried away with fast forwarding through a
+ message, they can easily get to the end and accidentally exit the
+ voicemail application by hitting the fast forward key during the
+ following prompt. This adds some safety by not allowing a fast
+ forward past the end of a message. (closes issue #14554) Reported
+ by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
+ 707) Tested by: lacoursj ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 |
+ russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines
+ Ensure the TCP read buffer is fully initialized before handling
+ each packet. (closes issue #14452) Reported by: umberto71
+ ........
+
+2009-06-26 20:18 +0000 [r203731] David Brooks <dbrooks@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009)
+ | 16 lines Fixing voicemail's error in checking max silence vs
+ min message length Max silence was represented in milliseconds,
+ yet vmminsecs (minmessage) was represented as seconds. Also, the
+ inequality was reversed. The warning, if triggered, was "Max
+ silence should be less than minmessage or you may get empty
+ messages", which should have been logged if max silence was
+ greater than minmessage, but the check was for less than. Also,
+ conforming if statement to coding guidelines. closes issue
+ #15331) Reported by: markd Review:
+ https://reviewboard.asterisk.org/r/293/ ........
+
+2009-06-26 19:49 +0000 [r203715] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c: Merged revisions 203702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 |
+ russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines
+ Make invalid hints report Unavailable instead of Idle. (closes
+ issue #14413) Reported by: pj ........
+
+2009-06-26 19:48 +0000 [r203712] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 203710 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009)
+ | 7 lines moving debug message from level 0 to 1. (closes issue
+ #15404) Reported by: leobrown Patches: iax_codec_debug.patch
+ uploaded by leobrown (license 541) ........
+
+2009-06-26 19:42 +0000 [r203709] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009)
+ | 16 lines Check if polarityonanswerdelay has elapsed before
+ setting a channel as answered after a polarity reversal.
+ Previously on a polarity switch event chan_dahdi would set the
+ channel immediately as answered. This would cause problems if a
+ polarity reversal occurred when the line was picked up as the
+ dial would not have yet occurred. Now if the polarity reversal
+ occurs before delay has elapsed after coming off hook or an
+ answer, it is ignored. Also, some refactoring was done in
+ _handle_event. (closes issue #13917) Reported by: alecdavis
+ Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by
+ alecdavis (license 585) Tested by: alecdavis ........
+
+2009-06-26 19:38 +0000 [r203705] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c,
+ main/channel.c, main/frame.c, /, channels/chan_sip.c,
+ apps/app_fax.c: Merged revisions 203699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 |
+ file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines
+ Improve T.38 negotiation by exchanging session parameters between
+ application and channel. ........
+
+2009-06-25 21:46 +0000 [r203445] David Vossel <dvossel@digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25
+ Jun 2009) | 4 lines fixes a few redundant conditions (issue
+ #15269) ........
+
+2009-06-25 21:21 +0000 [r203400] Terry Wilson <twilson@digium.com>
+
+ * main/cli.c, /: Merged revisions 203381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009)
+ | 11 lines Merged revisions 203380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009)
+ | 4 lines I didn't see that Mark already fixed the underlying
+ issue! Yay for removing useless code. ........ ................
+
+2009-06-25 21:08 +0000 [r203379] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 203376 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009)
+ | 16 lines Merged revisions 203375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009)
+ | 9 lines Fix a case where CDR answer time could be before the
+ start time involving parking. (closes issue #13794) Reported by:
+ davidw Patches: 13794.patch uploaded by murf (license 17)
+ 13794.patch.160 uploaded by murf (license 17) Tested by: murf,
+ dbrooks ........ ................
+
+2009-06-25 19:27 +0000 [r203276] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) |
+ 10 lines Unmute when we get a dtmfup (we muted on dtmfdown)
+ event. This would occasionally cause one-way audio when using
+ hardware DTMF detection. (closes issue #14761) Reported by:
+ tzafrir Patches: v1-14761.patch uploaded by dimas (license 88)
+ Tested by: tzafrir, dimas ........
+
+2009-06-25 16:08 +0000 [r203119] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009)
+ | 18 lines Merged revisions 203115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009)
+ | 11 lines Resolve a crash related to a T.38 reinvite race
+ condition. This change resolves a crash observed locally during
+ some T.38 testing. A call was set up using a call file, and when
+ the T.38 reinvite came in, the channel state was still
+ AST_STATE_DOWN. The reason is explained by a comment in the code
+ that previously lived in the handling of AST_STATE_RINGING. This
+ change modifies the logic to handle the same race condition for
+ any channel state that is not UP. (closes ABE-1895) ........
+ ................
+
+2009-06-24 21:27 +0000 [r203077] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500
+ (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009)
+ | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid
+ format is: pritimer=timer_name,timer_value * Fixed segfault if
+ the ',' is missing. * Completely check the range returned by
+ pri_timer2idx() to prevent possible access outside array bounds.
+ ........ ................
+
+2009-06-24 18:30 +0000 [r202970] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun
+ 2009) | 9 lines Merged revisions 202966 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun
+ 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding
+ the same thing in-line. ........ ................
+
+2009-06-24 18:11 +0000 [r202928] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 |
+ file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines
+ Ensure the default settings are applied for T.38 when we set it
+ up for a peer. ........
+
+2009-06-23 23:58 +0000 [r202842] Sean Bright <sean@malleable.com>
+
+ * doc/tex/cdrdriver.tex, /, doc/tex/billing.tex: Merged revisions
+ 202840-202841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202840 |
+ seanbright | 2009-06-23 19:53:45 -0400 (Tue, 23 Jun 2009) | 1
+ line Remove some trailing whitespace before making content
+ changes. ........ r202841 | seanbright | 2009-06-23 19:57:07
+ -0400 (Tue, 23 Jun 2009) | 1 line Change some section names in
+ the CDR tex documentation. ........
+
+2009-06-23 22:47 +0000 [r202805] Russell Bryant <russell@digium.com>
+
+ * doc/tex/cdrdriver.tex, /: Merged revisions 202804 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202804 | russell | 2009-06-23 17:47:26 -0500 (Tue, 23 Jun 2009)
+ | 2 lines Clean up section hierarchy for the CDR chapter.
+ ........
+
+2009-06-23 22:12 +0000 [r202765] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) |
+ 1 line I could have sworn I committed this patch ages ago, but...
+ bug fix with setting NAI properly on linksets in certain
+ situations. ........
+
+2009-06-23 16:33 +0000 [r202673] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009)
+ | 18 lines Merged revisions 202671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
+ | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
+ non-standard port and transport (closes issue #14659) Reported
+ by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
+ by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
+ by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
+ https://reviewboard.asterisk.org/r/288/ ........ ................
+
+2009-06-22 20:19 +0000 [r202495-202511] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 202497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009)
+ | 11 lines Merged revisions 202496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
+ | 4 lines Report CallerID change during a masquerade. Reported
+ by: markster ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009)
+ | 9 lines Merged revisions 202414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
+ | 2 lines Make Polycom subscription type override check more
+ explicit. ........ ................
+
+2009-06-22 16:31 +0000 [r202473] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun
+ 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid
+ potential crashes during reload. Pointed out by Russell while
+ working on the CEL branch. ........
+
+2009-06-22 15:37 +0000 [r202411] David Vossel <dvossel@digium.com>
+
+ * main/loader.c, /, include/asterisk/module.h: Merged revisions
+ 202410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 |
+ dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines
+ attempting to load running modules Modules placed in the priority
+ heap for loading were not properly removed from the linked list.
+ This resulted in some modules attempting to load twice. ........
+
+2009-06-22 15:17 +0000 [r202340-202346] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun
+ 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
+ 2009) | 26 lines Fix a situation in which Asterisk would not stop
+ retransmitting 487s. If a CANCEL were received by Asterisk, we
+ would send a 487 in response to the original INVITE and a 200 OK
+ for the CANCEL. If there were a network hiccup which caused the
+ 200 OK and the 487 to be lost, then the UA communicating with
+ Asterisk may try to retransmit its CANCEL. Asterisk's response to
+ this used to be to try sending another 487 to the canceled INVITE
+ and another 200 OK to the CANCEL. The problem here is that the
+ originally-sent 487 was sent "reliably" meaning that it will be
+ retransmitted until it is received properly. So when we receive
+ the second CANCEL it is likely that the first batch of 487s we
+ sent is still going strong and reaches the UA. The result was
+ that the second set of 487s would be retransmitted constantly
+ until the maximum number of retries had been reached. The fix for
+ this is that if we receive a second CANCEL for an INVITE, then we
+ cancel the retransmission of the first set of 487s and start a
+ second set. This causes the dialog to be terminated reasonably.
+ (closes issue #14584) Reported by: klaus3000 Patches:
+ 14584_v2.patch uploaded by mmichelson (license 60) Tested by:
+ klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
+ -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
+ left from previous commit. ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun
+ 2009) | 31 lines Merged revisions 202336 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
+ 2009) | 25 lines Fix a possible infinite loop in SDP parsing
+ during glare situation. There was a while loop in
+ get_ip_and_port_from_sdp which was controlled by a call to
+ get_sdp_iterate. The loop would exit either if what we were
+ searching for was found or if the return was NULL. The problem is
+ that get_sdp_iterate never returns NULL. This means that if what
+ we were searching for was not present, the loop would run
+ infinitely. This modification of the loop fixes the problem.
+ (closes issue #15213) Reported by: schmidts (closes issue #15349)
+ Reported by: samy (closes issue #14464) Reported by: pj (closes
+ issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
+ uploaded by mmichelson (license 60) Tested by: aragon ........
+ ................
+
+2009-06-21 16:16 +0000 [r202261-202265] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 |
+ russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines
+ Fix possibility of crashiness during reload in custom fields
+ handling. ........
+
+ * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 |
+ russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines
+ Standardize return values of load_config() so reload() doesn't
+ report an error on success. ........
+
+2009-06-20 19:14 +0000 [r202186] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 |
+ seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5
+ lines Fix version detection for API changes in spandsp. (closes
+ issue #15355) Reported by: deuffy ........
+
+2009-06-19 21:08 +0000 [r202007] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Added deadlock protection to
+ try_suggested_sip_codec in chan_sip.c. Review:
+ https://reviewboard.asterisk.org/r/287/
+
+2009-06-19 20:26 +0000 [r201995] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 201994 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500
+ (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009)
+ | 8 lines timestamp was being converted to host order as a short
+ rather than a long (closes issue #15361) Reported by: ffloimair
+ Patches: ts_issue.diff uploaded by dvossel (license 671) ........
+ ................
+
+2009-06-19 15:49 +0000 [r201785-201906] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c, /: Merged revisions 201904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009)
+ | 4 lines Fix 2 typos and add support for wide character types.
+ Reported by Benny Amorsen via the asterisk-users mailing list.
+ http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html
+ ........
+
+ * /, main/features.c: Merged revisions 201829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009)
+ | 13 lines Merged revisions 201828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009)
+ | 6 lines If the "h" extension fails, give it another chance in
+ main/pbx.c. If the "h" extension fails, give it another chance in
+ main/pbx.c, when it returns from the bridge code. Fixes an issue
+ where the "h" extension may occasionally not fire, when a Dial is
+ executed from a Macro. Debugged in #asterisk with user tompaw.
+ ........ ................
+
+ * /, apps/Makefile: Merged revisions 201783 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 |
+ tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines
+ One of the changes in 1.6.1 was to allow app_directory to use
+ functionality within app_voicemail for directory functions. It is
+ therefore no longer necessary for app_directory to be linked
+ against the ODBC libraries (and it never was necessary for
+ app_directory to be linked against IMAP, though it was). ........
+
+2009-06-18 16:44 +0000 [r201679] David Vossel <dvossel@digium.com>
+
+ * channels/misdn/isdn_lib.c, utils/conf2ael.c, main/ast_expr2.c,
+ utils/stereorize.c, main/ast_expr2f.c, res/ael/ael_lex.c,
+ utils/ael_main.c, utils/extconf.c, channels/xpmr/xpmr.c,
+ pbx/pbx_config.c, res/res_config_ldap.c, apps/app_rpt.c,
+ main/asterisk.c, codecs/gsm/src/gsm_destroy.c, /,
+ channels/h323/ast_h323.cxx: Merged revisions 201678 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009)
+ | 11 lines fixes some memory leaks and redundant conditions
+ (closes issue #15269) Reported by: contactmayankjain Patches:
+ patch.txt uploaded by contactmayankjain (license 740)
+ memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
+ Tested by: contactmayankjain, dvossel ........
+
+2009-06-18 15:40 +0000 [r201614] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 201610 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201610 | russell | 2009-06-18 10:27:10 -0500
+ (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009)
+ | 29 lines Fix memory corruption and leakage related reloads of
+ non files mode MoH classes. For Music on Hold classes that are
+ not files mode, meaning that we are executing an application that
+ will feed us audio data, we use a thread to monitor the external
+ application and read audio from it. This thread also makes use of
+ the MoH class object. In the MoH class destructor, we used
+ pthread_cancel() to ask the thread to exit. Unfortunately, the
+ code did not wait to ensure that the thread actually went away.
+ What needed to be done is a pthread_join() to ensure that the
+ thread fully cleans up before we proceed. By adding this one
+ line, we resolve two significant problems: 1) Since the thread
+ was never joined, it never fully goes away. So, on every reload
+ of non-files mode MoH, an unused thread was sticking around. 2)
+ There was a race condition here where the application monitoring
+ thread could still try to access the MoH class, even though the
+ thread executing the MoH reload has already destroyed it. (issue
+ #15109) Reported by: jvandal (issue #15123) Reported by:
+ axisinternet (issue #15195) Reported by: amorsen (issue AST-208)
+ ........ ................
+
+2009-06-18 15:23 +0000 [r201595] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201570 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 |
+ dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines
+ parsing extension correctly from sip register lines If a
+ transport type was specified, but no extension, parsing of the
+ extension would return whatever was after the transport rather
+ than defaulting to 's'. (closes issue #15111) Reported by: ffs
+ Patches: chan_sip.c_register-parser.patch uploaded by ffs
+ (license 730) Tested by: ffs, dvossel ........
+
+2009-06-17 21:33 +0000 [r201533] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 201531 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009)
+ | 7 lines Initialize additional variables, to prevent a possible
+ crash. (closes issue #15186) Reported by: ajohnson Patches:
+ 20090528__issue15186.diff.txt uploaded by tilghman (license 14)
+ Tested by: ajohnson ........
+
+2009-06-17 20:12 +0000 [r201461-201465] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 |
+ mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12
+ lines Fix problem with no audio due to ignoring the SDP. A recent
+ change to our SDP version comparison made audio not function on
+ some calls. This was because of a test wherein we were trying to
+ see if an unsigned value was less than 0. This is a dumb
+ comparison and arguably the compiler should have warned about it.
+ Alas, though, it slipped past. Now it's fixed by changing the
+ variable to be a signed type. Found by several developers. Tested
+ by mnicholson and dbrooks. ........
+
+ * main/channel.c, /: Merged revisions 201458 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun
+ 2009) | 15 lines Merged revisions 201450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Change the datastore traversal in
+ ast_do_masquerade to use a safe list traversal. It is possible
+ for datastore fixup functions to remove the datastore from the
+ list and free it. In particular, the queue_transfer_fixup in
+ app_queue does this. While I don't yet know of this causing any
+ crashes, it certainly could. Found while discussing a separate
+ issue with Brian Degenhardt. ........ ................
+
+2009-06-17 20:01 +0000 [r201447-201454] David Vossel <dvossel@digium.com>
+
+ * doc/datastores.txt, /: Merged revisions 201453 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 |
+ dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines
+ ast_channel_datastore_alloc is no longer used. updating
+ datastores.txt to reflect that. ........
+
+ * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500
+ (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009)
+ | 19 lines StopMixMonitor race condition (not giving up file
+ immediately) StopMixMonitor only indicates to the MixMonitor
+ thread to stop writing to the file. It does not guarantee that
+ the recording's file handle is available to the dialplan
+ immediately after execution. This results in a race condition. To
+ resolve this, the filestream pointer is placed in a datastore on
+ the channel. When StopMixMonitor is called, the datastore is
+ retrieved from the channel and the filestream is closed
+ immediately before returning to the dialplan. Documentation
+ indicating the use of StopMixMonitor to free files has been
+ updated as well. (closes issue #15259) Reported by: travisghansen
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/283/ ........ ................
+
+2009-06-17 19:49 +0000 [r201446] David Brooks <dbrooks@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009)
+ | 16 lines Merged revisions 201380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
+ | 9 lines Checks for NULL sip_pvt pointer in
+ chan_sip.c->acf_channel_read() Zombie channels could be passed,
+ and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
+ checking for NULL pointer. (closes issue #15330) Reported by:
+ okrief Tested by: dbrooks ........ ................
+
+2009-06-17 15:25 +0000 [r201360] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 |
+ dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines
+ SIP registry ref count error During a sip reload, the list of
+ sip_registry objects are supposed to be traversed, unlinked, and
+ destroyed, but destruction never takes place due to a ref
+ counting error. This causes a memory leak when registry items are
+ removed from sip.conf and reloaded. While the registries are
+ removed from the global list, they are not removed from the
+ scheduler. Because of this, SIP register attempts continue to be
+ sent out for the item even though it may no longer be in the
+ .conf. (closes issue #15295) Reported by: amorsen Review:
+ https://reviewboard.asterisk.org/r/282/ ........
+
+2009-06-17 12:06 +0000 [r201265] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 201262 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500
+ (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
+ to be appended is empty. When the list to be appended is empty,
+ and the list to be appended to is *not*, AST_LIST_APPEND_LIST
+ would actually cause the target list to become broken, and no
+ longer have a pointer to its last entry. This patch fixes the
+ problem. (reported by Stanislaw Pitucha on the asterisk-dev
+ mailing list) ........ ................
+
+2009-06-16 22:30 +0000 [r201224] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 |
+ dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines
+ fix issue with build_contact introduced by the "SIP trasnport
+ type issues" commit ........
+
+2009-06-16 19:47 +0000 [r200990-201097] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, apps/app_chanspy.c,
+ apps/app_mixmonitor.c, main/channel.c, main/autoservice.c,
+ main/frame.c, /, apps/app_meetme.c, main/slinfactory.c,
+ include/asterisk/linkedlists.h, main/file.c,
+ include/asterisk/channel.h: Merged revisions 201056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500
+ (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
+ 2009) | 11 lines Improve support for media paths that can
+ generate multiple frames at once. There are various media paths
+ in Asterisk (codec translators and UDPTL, primarily) that can
+ generate more than one frame to be generated when the application
+ calling them expects only a single frame. This patch addresses a
+ number of those cases, at least the primary ones to solve the
+ known problems. In addition it removes the broken TRACE_FRAMES
+ support, fixes a number of bugs in various frame-related API
+ functions, and cleans up various code paths affected by these
+ changes. https://reviewboard.asterisk.org/r/175/ ........
+ ................
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
+ revisions 201090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201090 |
+ kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5
+ lines Another minor fix to compiler attribute checking.
+ Defaulting to 'static' for the function scope was bad... so
+ remove it. ........
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
+ revisions 200985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 |
+ kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7
+ lines Fix problems with new compiler attribute checking in
+ configure script. The last changes to ast_gcc_attribute.m4 caused
+ some problems checking for various attributes, because the scope
+ of the symbol the attribute is applied to can be important; this
+ patch allows the scope to be specified for the check. ........
+
+2009-06-16 16:28 +0000 [r200984] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 |
+ dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines
+ SIP transport type issues What this patch addresses: 1.
+ ast_sip_ouraddrfor() by default binds to the UDP address/port
+ reguardless if the sip->pvt is of type UDP or not. Now when no
+ remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
+ transport type, attempting to set the address and port to the
+ correct TCP/TLS bindings if necessary. 2. It is not necessary to
+ send the port number in the Contact header unless the port is
+ non-standard for the transport type. This patch fixes this and
+ removes the todo note. 3. In sip_alloc(), the default dialog
+ built always uses transport type UDP. Now sip_alloc() looks at
+ the sip_request (if present) and determines what transport type
+ to use by default. 4. When changing the transport type of a
+ sip_socket, the file descriptor must be set to -1 and in some
+ cases the tcptls_session's ref count must be decremented and set
+ to NULL. I've encountered several issues associated with this
+ process and have created a function, set_socket_transport(), to
+ handle the setting of the socket type. (closes issue #13865)
+ Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
+ Kristijan (license 753) 13865.patch uploaded by mmichelson
+ (license 60) tls_port_v5.patch uploaded by vrban (license 756)
+ transport_issues.diff uploaded by dvossel (license 671) Tested
+ by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
+ https://reviewboard.asterisk.org/r/278/ ........
+
+2009-06-16 16:05 +0000 [r200948] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009)
+ | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail
+ can only use one storage module at the moment. Because it's
+ unclear that selecting one of the storage modules in menuselect
+ will disable filesystem storage we now have a FILE_STORAGE option
+ that conflicts with the other modules. (closes issue #15333)
+ ........
+
+2009-06-16 12:55 +0000 [r200842] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_smdi.c, /: Merged revisions 200841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200841 |
+ eliel | 2009-06-16 08:32:00 -0400 (Tue, 16 Jun 2009) | 6 lines
+ Show the interface name on error, if it is not found. If the
+ smdiport specified is not found, show the interface name instead
+ of '(null)'. ........
+
+2009-06-16 02:41 +0000 [r200807] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 200799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200799 |
+ moy | 2009-06-15 21:24:30 -0500 (Mon, 15 Jun 2009) | 2 lines keep
+ backwards compatible chan_dahdi with older openr2 versions by not
+ using the new skip category feature unless supported ........
+
+2009-06-16 01:30 +0000 [r200690-200765] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15
+ Jun 2009) | 11 lines Ensure that configure-script testing for
+ compiler attributes actually works. The configure script tests
+ for compiler attributes didn't actually enable enough warnings or
+ provide a proper test harness to determine whether the compiler
+ supports the attribute in question or not; this caused gcc 4.1 to
+ report that it supports 'weakref', but it doesn't actually
+ support it in the way that is needed for our optional API
+ mechanism. The new configure script test will properly
+ distinguish between full support and partial support for this
+ attribute, among others. ........
+
+ * CHANGES, /: Merged revisions 200726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 |
+ kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6
+ lines Document the new automatic 'ignoresdpversion' behavior.
+ Asterisk will now automatically ignore incorrect incoming SDP
+ version numbers when necessary to complete a T.38 re-INVITE
+ operation. ........
+
+ * /, channels/chan_sip.c: Merged revisions 200689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200689 |
+ kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 11
+ lines Accept T.38 re-INVITE responses with invalid SDP versions.
+ This commit changes the 'incoming SDP version' check logic a bit
+ more; when 'ignoresdpversion' is *not* set for a peer, if we
+ initiate a re-INVITE to switch to T.38, we'll always accept the
+ peer's SDP response, even if they don't properly increment the
+ SDP version number as they should. If this situation occurs, a
+ warning message will be generated suggesting that the peer's
+ configuration be changed to include the 'ignoresdpversion'
+ configuration option (although ideally they'd fix their SIP
+ implementation to be RFC compliant). AST-221 ........
+
+2009-06-15 15:23 +0000 [r200517] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun
+ 2009) | 11 lines Merged revisions 200513 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
+ 2009) | 5 lines Add INFO to our allowed methods so that endpoints
+ know they may send it to us. AST-223 ........ ................
+
+2009-06-14 06:33 +0000 [r200512] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
+ build_tools/menuselect-deps.in: Merged revisions 200477 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r200477 | moy | 2009-06-14 01:13:48 -0500 (Sun, 14 Jun
+ 2009) | 3 lines added openr2 to menuselect-deps.in, recent commit
+ in menuselect made me realize this was never done but was working
+ anyways also added support for skip category request feature of
+ openr2 and updated chan_dahdi.conf.sample ........
+
+2009-06-12 19:08 +0000 [r200364] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 200361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun
+ 2009) | 16 lines Merged revisions 200360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
+ 2009) | 10 lines Suppress a warning message and give a better
+ return code when generating inband ringing after a call is
+ answered. (closes issue #15158) Reported by: madkins Patches:
+ 15158.patch uploaded by mmichelson (license 60) Tested by:
+ madkins ........ ................
+
+2009-06-12 02:20 +0000 [r200198-200255] Sean Bright <sean@malleable.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 200254 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r200254 | seanbright | 2009-06-11 22:20:19 -0400 (Thu,
+ 11 Jun 2009) | 5 lines Call chgrp instead of chown when setting
+ run directory group ownership. (issue #13153) Reported by:
+ pabelanger ........
+
+ * Makefile, /: Merged revisions 199781 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 |
+ seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2
+ lines Fix all of the parallel build warnings issued when running
+ make -j#. ........
+
+ * /: Undo block of revision 199782 (will be merging it momentarily)
+
+2009-06-11 21:35 +0000 [r200172] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null
+
+2009-06-11 21:18 +0000 [r200154] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 |
+ mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5
+ lines Fix a crash due to a potentially NULL p->options. Thanks to
+ mnicholson for pointing it out. ........
+
+2009-06-11 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-beta3
+
+2009-06-11 12:19 +0000 [r200051] Leif Madsen <lmadsen@digium.com>
+
+ * build_tools/make_version_h, /, build_tools/make_version_c: Merged
+ revisions 200039 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 |
+ lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
+ Fix path for .flavor and .version (issue #14737) Reported by:
+ davidw Patches: flavor.patch uploaded by davidw (license 780)
+ Tested by: davidw ........
+
+2009-06-10 20:37 +0000 [r199998] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Fixes the argument order in definition of
+ new_find_extension(). In the definition of new_find_extension(),
+ the arguments 'callerid' and 'label' were swapped. The prototype
+ declaration and all calls to the function are ordered 'callerid'
+ then 'label', but the function itself was ordered 'label' then
+ 'callerid'. (closes issue #15303) Reported by: JimDickenson
+
+2009-06-10 20:18 +0000 [r199966] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 |
+ mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6
+ lines Only try to use the invite_branch on outgoing INVITEs with
+ auth credentials. I have added a comment to the code to help ease
+ understanding of the logic here as well. ........
+
+2009-06-10 16:13 +0000 [r199860] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400
+ (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
+ 10 Jun 2009) | 2 lines __WORDSIZE is not available on all
+ platforms, so use sizeof(void *) instead. ........
+ ................
+
+2009-06-09 20:48 +0000 [r199744-199819] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 |
+ dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
+ CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command
+ only used UDP rather than copying the transport type from the
+ peer. (closes issue #15283) Reported by: jthurman Patches:
+ sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
+ Tested by: jthurman, dvossel ........
+
+ * main/loader.c, /, res/res_timing_pthread.c,
+ include/asterisk/module.h, res/res_timing_dahdi.c,
+ res/res_timing_timerfd.c: Merged revisions 199743 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009)
+ | 11 lines module load priority This patch adds the option to
+ give a module a load priority. The value represents the order in
+ which a module's load() function is initialized. The lower the
+ value, the higher the priority. The value is only checked if the
+ AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
+ flag is not set, the value will never be read and the module will
+ be given the lowest possible priority on load. Since some modules
+ are reliant on a timing interface, the timing modules have been
+ given a high load priorty. (closes issue #15191) Reported by:
+ alecdavis Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/262/ ........
+
+2009-06-08 19:39 +0000 [r199634] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400
+ (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
+ 2009) | 21 lines Increase the size of our thread stack on 64 bit
+ processors. We were setting the stack size for each thread to
+ 240KB regardless of architecture, which meant that in some
+ scenarios we actually had less available stack space on 64 bit
+ processors (pointers use 8 bytes instead of 4). So now we
+ calculate the stack size we reserve based on the platform's
+ __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
+ bit -> 1008KB (that's right, we're ready for 128 bit processors)
+ Patch typed by me but written by several members of
+ #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
+ issue #14932) Reported by: jpiszcz Patches:
+ 06052009_issue14932.patch uploaded by seanbright (license 71)
+ Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
+ 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
+ stack size calculation just introduced. ........ ................
+
+2009-06-08 17:42 +0000 [r199591] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Recorded merge of revisions 199588 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon,
+ 08 Jun 2009) | 9 lines Fix a deadlock that could occur when
+ setting rtp stats on SIP calls. (closes issue #15143) Reported
+ by: cristiandimache Patches: 15143.patch uploaded by mmichelson
+ (license 60) Tested by: cristiandimache ........
+
+2009-06-06 21:39 +0000 [r199369] Russell Bryant <russell@digium.com>
+
+ * Makefile, /: Merged revisions 199368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199368 |
+ russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines
+ Switch from "echo -n" to printf. On my mac, the -n was just
+ getting printed out. ........
+
+2009-06-05 21:25 +0000 [r199299] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, /, main/devicestate.c: Merged
+ revisions 199298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009)
+ | 21 lines Merged revisions 199297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
+ | 14 lines Fixes issue with hints giving unexpected results.
+ Hints with two or more devices that include ONHOLD gave
+ unexpected results. (closes issue #15057) Reported by:
+ p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
+ (license 671) pbx.c.1.4.patch uploaded by p (license 558)
+ devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
+ p_lindheimer, dvossel Review:
+ https://reviewboard.asterisk.org/r/254/ ........ ................
+
+2009-06-05 13:52 +0000 [r199230] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun
+ 2009) | 14 lines Correct "dahdi show channels" output when
+ specifying a group. Since a DAHDI channel may belong to multiple
+ groups, we need to use a bitwise and instead of equivalence to
+ determine whether to display the channel information. (closes
+ issue #15248) Reported by: gentian Patches: 15248.patch uploaded
+ by mmichelson (license 60) Tested by: gentian ........
+
+2009-06-04 19:15 +0000 [r199140] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 199139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500
+ (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
+ Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
+ ................
+
+2009-06-04 14:53 +0000 [r199054] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk/_private.h, main/asterisk.c, main/loader.c, /:
+ Merged revisions 199051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun
+ 2009) | 47 lines Merged revisions 199022 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
+ 2009) | 40 lines Safely handle AMI connections/reload requests
+ that occur during startup. During asterisk startup, a lock on the
+ list of modules is obtained by the primary thread while each
+ module is initialized. Issue 13778 pointed out a problem with
+ this approach, however. Because the AMI is loaded before other
+ modules, it is possible for a module reload to be issued by a
+ connected client (via Action: Command), causing a deadlock. The
+ resolution for 13778 was to move initialization of the manager to
+ happen after the other modules had already been lodaded. While
+ this fixed this particular issue, it caused a problem for users
+ (like FreePBX) who call AMI scripts via an #exec in a
+ configuration file (See issue 15189). The solution I have come up
+ with is to defer any reload requests that come in until after the
+ server is fully booted. When a call comes in to ast_module_reload
+ (from wherever) before we are fully booted, the request is added
+ to a queue of pending requests. Once we are done booting up, we
+ then execute these deferred requests in turn. Note that I have
+ tried to make this a bit more intelligent in that it will not
+ queue up more than 1 request for the same module to be reloaded,
+ and if a general reload request comes in ('module reload') the
+ queue is flushed and we only issue a single deferred reload for
+ the entire system. As for how this will impact existing
+ installations - Before 13778, a reload issued before module
+ initialization was completed would result in a deadlock. After
+ 13778, you simply couldn't connect to the manager during startup
+ (which causes problems with #exec-that-calls-AMI configuration
+ files). I believe this is a good general purpose solution that
+ won't negatively impact existing installations. (closes issue
+ #15189) (closes issue #13778) Reported by: p_lindheimer Patches:
+ 06032009_15189_deferred_reloads.diff uploaded by seanbright
+ (license 71) Tested by: p_lindheimer, seanbright Review:
+ https://reviewboard.asterisk.org/r/272/ ........ ................
+
+2009-06-03 15:24 +0000 [r198827-198886] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 198856 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 |
+ dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
+ Generic call forward api, ast_call_forward() The function
+ ast_call_forward() forwards a call to an extension specified in
+ an ast_channel's call_forward string. After an ast_channel is
+ called, if the channel's call_forward string is set this function
+ can be used to forward the call to a new channel and terminate
+ the original one. I have included this api call in both
+ channel.c's ast_request_and_dial() and feature.c's
+ feature_request_and_dial(). App_dial and app_queue already
+ contain call forward logic specific for their application and
+ options. (closes issue #13630) Reported by: festr Review:
+ https://reviewboard.asterisk.org/r/271/ ........
+
+ * channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009)
+ | 8 lines fixes issue with channels not going down after transfer
+ Iax2 currently does not support native bridging if the timeoutms
+ value is set. We check for that in iax2_bridge, but then set
+ timeoutms to 0 by default. If the timeoutms is not provided it is
+ set to -1. By setting timeoutms to 0 it is processed causing a
+ bridging retry loop. (closes issue #15216) Reported by: oxymoron
+ Tested by: dvossel ........
+
+2009-06-02 13:51 +0000 [r198794] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 198791 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 |
+ file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
+ Correct documentation for the register line, specifically where
+ the domain should be specified. (closes issue #14367) Reported
+ by: Nick_Lewis ........
+
+2009-06-01 21:04 +0000 [r198730] Russell Bryant <russell@digium.com>
+
+ * channels/iax2-parser.c, /: Merged revisions 198729 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009)
+ | 2 lines Tell the IAX2 parser about more control frame types.
+ ........
+
+2009-06-01 18:44 +0000 [r198629] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/scripts/meetme.sql: Merged revisions 198626 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01
+ Jun 2009) | 2 lines Add information for new meetme realtime
+ fields ........
+
+2009-05-31 17:53 +0000 [r198471] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 198470 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009)
+ | 2 lines Fix documentation for FIELDQTY. ........
+
+2009-05-31 01:48 +0000 [r198440] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) |
+ 11 lines Avoid a crash when res_timing_dahdi is unloaded but
+ wasn't properly loaded. if dahdi_test_timer() fails,
+ timing_funcs_handle remains NULL causing a crash when calling
+ ast_unregister_timing_interface() with a NULL pointer. (closes
+ issue #15234) Reported by: eliel Patches: timing_dahdi1.diff
+ uploaded by eliel (license 64) ........
+
+2009-05-31 01:21 +0000 [r198436] Russell Bryant <russell@digium.com>
+
+ * res/res_smdi.c, /: Merged revisions 198312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009)
+ | 12 lines Merged revisions 198311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009)
+ | 5 lines Fix a crash that occurred when MWI SMDI messages
+ expired. (closes issue #14561) Reported by: cmoss28 ........
+ ................
+
+2009-05-30 20:22 +0000 [r198297-198397] Sean Bright <sean.bright@gmail.com>
+
+ * res/res_jabber.c, /: Merged revisions 198375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 |
+ seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13
+ lines Properly terminate the receive buffer before sending to
+ iksemel. aji_io_recv takes the maximum number of bytes to read
+ (instead of the total buffer size), so we have to subtract 1 from
+ our buffer size. Without this, when we receive packets that are
+ larger than our buffer, iksemel will choke and things get wonky.
+ (closes issue #15232) Reported by: lp0 Patches:
+ 05302009_res_jabber.c.patch uploaded by seanbright (license 71)
+ Tested by: seanbright, lp0 ........
+
+ * res/res_jabber.c, /: Merged revisions 198371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May
+ 2009) | 19 lines Merged revisions 198370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May
+ 2009) | 12 lines Properly terminate AMI JabberSend response
+ messages. The response message (either Error or Success) needs an
+ extra trailing \r\n after the fields to inform the client that
+ the message is complete. (closes issue #14876) Reported by: srt
+ Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
+ (license 71) asterisk_14876.patch uploaded by srt (license 378)
+ trunk-14876-2.diff uploaded by phsultan (license 73) ........
+ ................
+
+ * apps/app_dial.c, /: Merged revisions 198285 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May
+ 2009) | 15 lines Merged revisions 198251 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May
+ 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we
+ treat a missing one. (closes issue #15056) Reported by:
+ p_lindheimer Patches: 05292009_bug15056.diff uploaded by
+ seanbright (license 71) Tested by: p_lindheimer ........
+ ................
+
+2009-05-30 02:35 +0000 [r198250] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 |
+ file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines
+ When removing all packets from a dialog we also need to free the
+ data if present. ........
+
+2009-05-29 23:05 +0000 [r198148-198188] Russell Bryant <russell@digium.com>
+
+ * /, configs/modules.conf.sample: Merged revisions 198186 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29
+ May 2009) | 2 lines Suggesting that only a single timing module
+ be loaded is no longer necessary. ........
+
+ * /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009)
+ | 2 lines Improve handling of trying to ACK too many timer
+ expirations. ........
+
+ * /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009)
+ | 38 lines Resolve issues with choppy sound when using
+ res_timing_pthread. The situation that caused this problem was
+ when continuous mode was being turned on and off while a rate was
+ set for a timing interface. A very easy way to replicate this bug
+ was to do a Playback() from behind a Local channel. In this
+ scenario, a rate gets set on the channel for doing file playback.
+ At the same time, continuous mode gets turned on and off about
+ every 20 ms as frames get queued on to the PBX side channel from
+ the other side of the Local channel. Essentially, this module
+ treated continuous mode and a set rate as mutually exclusive
+ states for the timer to be in. When I dug deep enough, I observed
+ the following pattern: 1) Set timer to tick every 20 ms. 2) Wait
+ almost 20 ms ... 3) Continuous mode gets turned on for a queued
+ up frame 4) Continuous mode gets turned off 5) The timer goes
+ back to its tick per 20 ms. state but starts counting at 0 ms. 6)
+ Goto step 2. Sometimes, res_timing_pthread would make it 20 ms
+ and produce a timer tick, but not most of the time. This is what
+ produced the choppy sound (or sometimes no sound at all). Now,
+ the module treats continuous mode and a set rate as completely
+ independent timer modes. They can be enabled and disabled
+ independently of each other and things work as expected. (closes
+ issue #14412) Reported by: dome Patches: issue14412.diff.txt
+ uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt
+ uploaded by russell (license 2) Tested by: DennisD, russell
+ ........
+
+2009-05-29 19:26 +0000 [r198111] Eliel C. Sardanons <eliels@gmail.com>
+
+ * CREDITS, /: Merged revisions 198083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198083 |
+ eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines
+ Apply anti-spam obfuscation to an email address. ........
+
+2009-05-29 19:14 +0000 [r198075] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged
+ revisions 198072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May
+ 2009) | 21 lines Merged revisions 198068 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May
+ 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as
+ the default CDR disposition. This change also involves the
+ addition of an AST_CDR_FLAG_ORIGINATED flag that is used on
+ originated channels to distinguish: them from dialed channels.
+ (closes issue #12946) Reported by: meral Patches: null-cdr2.diff
+ uploaded by mnicholson (license 96) Tested by: mnicholson,
+ dbrooks (closes issue #15122) Reported by: sum Tested by: sum
+ ........ ................
+
+2009-05-29 18:40 +0000 [r198066] Joshua Colp <jcolp@digium.com>
+
+ * /, main/file.c: Merged revisions 198064 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 |
+ file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix
+ a memory leak of the write buffer when writing a file. ........
+
+2009-05-29 18:18 +0000 [r198008] Sean Bright <sean.bright@gmail.com>
+
+ * Makefile, /: Merged revisions 198000 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May
+ 2009) | 15 lines Merged revisions 197998 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May
+ 2009) | 8 lines Fix 'make config' target for Slackware. There was
+ a missing semi-colon after the echo statement in the Makefile
+ that was causing problems for some users. Fix suggested by
+ reporter. (closes issue #15225) Reported by: pdavis ........
+ ................
+
+2009-05-29 16:29 +0000 [r197994] Russell Bryant <russell@digium.com>
+
+ * /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009)
+ | 2 lines Trim trailing whitespace so that I can work on this bug
+ without it bothering me. :-) ........
+
+2009-05-28 23:54 +0000 [r197894] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 197828 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009)
+ | 8 lines Update documentation in MixMonitor. Updated the
+ MixMonitor documentation for the 'b' option so that it is more
+ obvious that you must not optimize away the Local channel when
+ using this option. (closes issue #14829) Reported by: licedey
+ Tested by: mmichelson, licedey, lmadsen ........
+
+2009-05-28 18:50 +0000 [r197703] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2
+ lines Fix a bug where the trunkmtu setting was not set to the
+ default value of 1240 on load but was on reload. ........
+
+2009-05-28 16:15 +0000 [r197625] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) |
+ 19 lines Merged revisions 197562 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) |
+ 13 lines 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 15:44 +0000 [r197548-197619] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
+ Merged revisions 197606 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May
+ 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu,
+ 28 May 2009) | 16 lines 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 ........ ................
+
+ * main/audiohook.c, apps/app_chanspy.c, /,
+ include/asterisk/audiohook.h: Merged revisions 197543 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500
+ (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May
+ 2009) | 21 lines 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 14:56 +0000 [r197471-197542] Joshua Colp <jcolp@digium.com>
+
+ * /, main/utils.c: Merged revisions 197538 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 |
+ file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix
+ a bug in stringfields where it did not actually free the pools of
+ memory. (closes issue #15074) Reported by: pj ........
+
+ * /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) |
+ 15 lines Merged revisions 197466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8
+ lines 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
+ ........ ................
+
+2009-05-28 11:40 +0000 [r197441] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, doc/ldap.txt,
+ configs/res_ldap.conf.sample: issue #15155 and issue #15156 from
+ trunk
+
+2009-05-27 23:49 +0000 [r197375] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/xml.c: Merged revisions 197374 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r197374 |
+ tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines
+ Revert commit 192032. This define is needed on Mac OS X. ........
+
+2009-05-27 22:23 +0000 [r197336] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/agi.h, /: Merged revisions 197335 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May
+ 2009) | 3 lines Ensure that this header includes xmldoc.h, since
+ it depends on it. ........
+
+2009-05-27 20:11 +0000 [r197263] Sean Bright <sean.bright@gmail.com>
+
+ * Makefile, /: Merged revisions 197260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 |
+ seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6
+ lines 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 19:30 +0000 [r197247] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_cut.c: Recorded merge of revisions 197209 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500
+ (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009)
+ | 5 lines 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 17:28 +0000 [r197176] Jeff Peeler <jpeeler@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 16:12 +0000 [r196950-197092] Sean Bright <sean.bright@gmail.com>
+
+ * configs/smdi.conf.sample, configs/extensions.conf.sample,
+ configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /,
+ configs/vpb.conf.sample: Merged revisions 197089 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May
+ 2009) | 6 lines Fix references to /etc/dahdi/system.conf and
+ /etc/asterisk/chan_dahdi.conf in the sample configuration files.
+ (closes issue #15207) Reported by: seandarcy ........
+
+ * /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May
+ 2009) | 9 lines Display an error message when chan_alsa fails to
+ load due to a missing or inaccessible configuration file. Before
+ this change, when chan_alsa failed to load due to a missing or
+ inaccessible configuration file, no message would be displayed.
+ With this change, when chan_alsa fails to load due to a missing
+ or inaccessible configuration file, a message will be displayed.
+ (closes issue #14760) Reported by: Nick_Lewis Patches:
+ chan_alsa.c-confload.patch uploaded by Nick (license 657)
+ ........
+
+ * main/xmldoc.c, /: Merged revisions 196948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196948 |
+ seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8
+ lines Reset the terminal to the correct fg/bg after XML
+ documenation is rendered. (closes issue #15200) Reported by:
+ ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright
+ (license 71) Tested by: ajohnson ........
+
+ * main/manager.c, /: Merged revisions 196945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196945 |
+ seanbright | 2009-05-26 18:38:05 -0400 (Tue, 26 May 2009) | 13
+ lines Add ActionID to CoreShowChannel event. There is
+ inconsistency in how we handle manager responses that are lists
+ of items and, unfortunately, third parties have come to rely on
+ ActionID being on every event within those lists instead of just
+ keeping track of the ActionID for the current response. This
+ change makes CoreShowChannels include the ActionID with each
+ CoreShowChannel event generated as a result of it being called.
+ (closes issue #15001) Reported by: sum Patches:
+ patchactionid2.patch uploaded by sum (license 766) ........
+
+2009-05-26 22:44 +0000 [r196870-196949] Russell Bryant <russell@digium.com>
+
+ * /, autoconf/ast_check_osptk.m4 (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 196946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 |
+ russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines
+ Update configure script to check for OSP toolkit 3.5.0. (closes
+ issue #14988) Reported by: tzafrir Patches: configure.ac.diff
+ uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded
+ by homesick (license 91) ........
+
+ * /, res/res_convert.c: Merged revisions 196843 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009)
+ | 16 lines Merged revisions 196826 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009)
+ | 9 lines 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 16:39 +0000 [r196793] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_queue.c, /: Merged revisions 196792 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196792 |
+ seanbright | 2009-05-26 12:38:54 -0400 (Tue, 26 May 2009) | 2
+ lines Add a missing unref for queues in handle_statechange.
+ ........
+
+2009-05-26 13:47 +0000 [r196661-196724] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 |
+ file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix
+ a bug where the sip unregister CLI command did not completely
+ unregister the peer. (closes issue #15118) Reported by: alecdavis
+ Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis
+ (license 585) ........
+
+ * contrib/scripts/safe_asterisk, /: Merged revisions 196658 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue,
+ 26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7
+ lines Remove some bash specific stuff from safe_asterisk. (closes
+ issue #10812) Reported by: paravoid Patches:
+ safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
+ ........ ................
+
+2009-05-23 05:29 +0000 [r196487] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 196456 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r196456 | moy | 2009-05-22 23:27:47 -0500 (Fri, 22 May 2009) | 1
+ line set MFCR2_CATEGORY just when starting the pbx ........
+
+2009-05-22 21:59 +0000 [r196452] David Vossel <dvossel@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 196416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 |
+ dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
+ SIP set outbound transport type from Registration In sip.conf the
+ transport option allows for the configuration of what transport
+ types (udp, tcp, and tls) a peer will accept, but only the first
+ type listed was used for outbound connections. This patch changes
+ this. Now the default transport type is only used until the peer
+ registers. When registration takes place the transport type is
+ parsed out of the Contact header. If the Contact header's
+ transport type is equal to one that the peer supports, the peer's
+ default transport type for outbound connections is set to match
+ the Contact header's type. If the Contact header's transport type
+ is not present, then the peer's default transport type is set to
+ match the one the peer registered with. When a peer unregisters
+ or the registration expires, the default transport type for that
+ peer is reset. (closes issue #12282) Reported by: rjain Patches:
+ reg_patch_1.diff uploaded by dvossel (license 671) Tested by:
+ dvossel (closes issue #14727) Reported by: pj Patches:
+ reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj,
+ dvossel Review: https://reviewboard.asterisk.org/r/249/ ........
+
+2009-05-22 19:48 +0000 [r196378] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, apps/app_minivm.c: Merged revisions 196377 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196377 |
+ eliel | 2009-05-22 15:38:33 -0400 (Fri, 22 May 2009) | 11 lines
+ Unregister every registered application by MiniVM. The MinivmMWI
+ application was not being unregistered on unload and we were not
+ able to load again the module or reload it. (closes issue #15174)
+ Reported by: junky Patches: unregister_minivm_mwi.diff uploaded
+ by junky (license 177) ........
+
+2009-05-22 13:59 +0000 [r196120] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 196117 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri,
+ 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5
+ lines 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 19:15 +0000 [r196000] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 195995 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500
+ (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009)
+ | 14 lines 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 15:57 +0000 [r195883] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195882
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500
+ (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May
+ 2009) | 13 lines 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 23:31 +0000 [r195842] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 195839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195839 |
+ tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines
+ If a variable had a blank value upon the initial setting, then it
+ would do nothing. Identified by Dmitry Andrianov via private
+ email, fixed by me. ........
+
+2009-05-20 17:35 +0000 [r195639-195707] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 195698 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) |
+ 12 lines Merged revisions 195688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5
+ lines 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: Merged revisions 195636 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) |
+ 12 lines Merged revisions 195635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5
+ lines Fix a bug where the MeetMe option 'D' did not actually
+ prompt for the pin. (closes issue #15050) Reported by: pmhaddad
+ ........ ................
+
+2009-05-19 20:19 +0000 [r195531] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 195521 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r195521 | tilghman | 2009-05-19 15:16:01 -0500
+ (Tue, 19 May 2009) | 14 lines Merged revisions 195520 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009)
+ | 7 lines 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 14:49 +0000 [r195452] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 195449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) |
+ 14 lines Merged revisions 195448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7
+ lines 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 21:25 +0000 [r195405] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/manager.c, /: Merged revisions 195369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195369 |
+ eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines
+ Fix the CLI command 'manager show command' documentation and
+ functionality. The CLI command 'manager show command' supports
+ passing multiple action names in the same line, but it was not
+ allowing that because of a incorrect check in the argumentes
+ counter. Also the documentation was updated to show that this
+ usage of the command is possible. ........
+
+2009-05-18 20:55 +0000 [r195359-195373] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, include/asterisk/smdi.h, res/res_monitor.c,
+ apps/app_voicemail.c, res/res_smdi.c, /,
+ include/asterisk/monitor.h: Merged revisions 195370 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r195370 | tilghman | 2009-05-18 15:52:33 -0500
+ (Mon, 18 May 2009) | 15 lines Recorded merge of revisions 195366
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009)
+ | 8 lines 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) ........ ................
+
+ * main/asterisk.c, /: Merged revisions 195320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195320 |
+ tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines
+ Move the spawn of astcanary down, until after the call to
+ daemon(3). This avoids possible conflicts with the internal
+ implementation of daemon(3). (closes issue #15093) Reported by:
+ tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by
+ tilghman (license 14) Tested by: tzafrir ........
+
+2009-05-18 19:01 +0000 [r195319] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_externalivr.c, /: Merged revisions 195316 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May
+ 2009) | 18 lines Fix externalivr's setvariable command so that it
+ properly sets multiple variables. The command had a for loop that
+ was guaranteed to only execute once since the continuation
+ operation of the loop would set the input buffer NULL. I rewrote
+ the loop so that its operation was more obvious, and it would set
+ multiple variables correctly. I also reduced stack space required
+ for the function, constified the input string, and modified the
+ function so that it would not modify the input string while I was
+ at it. (closes issue #15114) Reported by: chris-mac Patches:
+ 15114.patch uploaded by mmichelson (license 60) Tested by:
+ chris-mac ........
+
+2009-05-18 15:57 +0000 [r195212] Joshua Colp <jcolp@digium.com>
+
+ * main/frame.c, /: Merged revisions 195207 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195207 | file | 2009-05-18 12:53:26 -0300 (Mon, 18 May 2009) |
+ 14 lines Merged revisions 195206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7
+ lines 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) ........ ................
+
+2009-05-18 14:54 +0000 [r195164] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_dial.c, main/pbx.c, /, apps/app_macro.c: Merged
+ revisions 195162 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195162 |
+ eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines
+ Warn about the use of the application WaitExten() within a
+ Macro(). Update applications documentation to warn the user about
+ the use of the WaitExten() application within a Macro().
+ Recommend the use of Read() instead. (closes issue #14444)
+ Reported by: ewieling ........
+
+2009-05-18 14:00 +0000 [r195099] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 195096 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) |
+ 12 lines Merged revisions 195095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5
+ lines 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 13:50 +0000 [r195093-195094] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, main/xml.c: Merged revisions 195075 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195075 |
+ eliel | 2009-05-18 09:30:34 -0400 (Mon, 18 May 2009) | 3 lines Do
+ not avoid loading the XML documentation if not XInclude
+ substitution is done. ........
+
+ * doc/appdocsxml.dtd, Makefile, /, main/xml.c: Merged revisions
+ 194982 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194982 |
+ eliel | 2009-05-16 16:01:22 -0400 (Sat, 16 May 2009) | 20 lines
+ Allow to include sections of other parts of the xml
+ documentation. Avoid duplicating xml documentation by allowing to
+ include other parts of the xml documentation using XInclude.
+ Example: <xi:include
+ xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
+ (Insert this line to include the synopsis of the CHANNEL function
+ xml documentation). It is also possible to include documentation
+ from other files in the 'documentation/' directory using the
+ href="" attribute inside a xinclude element. (closes issue
+ #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling
+ ........
+
+2009-05-18 13:39 +0000 [r195092] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 195089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 |
+ file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix
+ a bug where specifying an empty outboundproxy would cause packets
+ to get sent to ourself. (closes issue #15106) Reported by:
+ timeshell ........
+
+2009-05-18 13:14 +0000 [r195024] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 195021 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195021 | russell | 2009-05-18 07:59:11 -0500 (Mon, 18 May 2009)
+ | 12 lines Recorded merge of revisions 195020 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009)
+ | 5 lines Don't try to unlock a bogus channel. (closes issue
+ #15144) Reported by: cristiandimache ........ ................
+
+2009-05-16 18:43 +0000 [r194946] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/pbx.c, /: Merged revisions 194945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194945 |
+ eliel | 2009-05-16 14:32:11 -0400 (Sat, 16 May 2009) | 8 lines
+ Fix a missing unlock in case of error, and a missing free().
+ Always free the allocated memory for a string field, because we
+ are always using it (not only when xmldocs are enabled). Also if
+ there is an error allocating memory for the string field remember
+ to unlock the list of registered applications, before returning.
+ ........
+
+2009-05-15 22:48 +0000 [r194836-194877] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 194874 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r194874 | dvossel | 2009-05-15 17:44:44 -0500
+ (Fri, 15 May 2009) | 23 lines Merged revisions 194873 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009)
+ | 17 lines 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 ........ ................
+
+ * channels/chan_iax2.c, channels/iax2-parser.c,
+ channels/iax2-parser.h, /, channels/iax2.h: Merged revisions
+ 194833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009)
+ | 24 lines Merged revisions 194557,194685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009)
+ | 10 lines 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/ ........ r194685 |
+ dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
+ Update to previous IAX2 "Ghost" Channels patch. Fixed some
+ comments made on reviewboard for the previous patch. (issue
+ #14207) ........ ................
+
+2009-05-15 18:44 +0000 [r194717-194768] Russell Bryant <russell@digium.com>
+
+ * configs/logger.conf.sample, /: Merged revisions 194765 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r194765 | russell | 2009-05-15 13:43:42 -0500
+ (Fri, 15 May 2009) | 10 lines Merged revisions 194764 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009)
+ | 2 lines Fix some spelling fail. ........ ................
+
+ * /, codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Merged
+ revisions 194722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194722 |
+ russell | 2009-05-15 12:59:08 -0500 (Fri, 15 May 2009) | 4 lines
+ Shuttle some bits around to address some gain issues with G.722.
+ (closes AST-209) ........
+
+ * codecs/Makefile, codecs/g722/Makefile (removed), /: Merged
+ revisions 194718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194718 |
+ russell | 2009-05-15 12:37:12 -0500 (Fri, 15 May 2009) | 2 lines
+ Further simplify codec_g722 build. ........
+
+ * codecs/Makefile, /: Merged revisions 194714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194714 |
+ russell | 2009-05-15 12:24:39 -0500 (Fri, 15 May 2009) | 2 lines
+ Actually force running make for g722. ........
+
+2009-05-15 13:47 +0000 [r194650] Michiel van Baak <michiel@vanbaak.info>
+
+ * CREDITS, /: Merged revisions 194649 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194649 |
+ mvanbaak | 2009-05-15 15:43:24 +0200 (Fri, 15 May 2009) | 2 lines
+ add eliel ........
+
+2009-05-15 13:42 +0000 [r194648] Eliel C. Sardanons <eliels@gmail.com>
+
+ * doc/appdocsxml.dtd, main/xmldoc.c, /: Merged revisions 194635 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r194635 | eliel | 2009-05-15 09:23:37 -0400 (Fri, 15 May
+ 2009) | 16 lines Allow to specify an enumlist inside an enum. It
+ was not possible to use an enumlist inside an enum: <enumlist>
+ <enum name="aa"> <enumlist> ... </enumlist> </enum> </enumlist>
+ Now we will be able to insert as many levels as we want. (closes
+ issue #15112) Reported by: lmadsen ........
+
+2009-05-14 22:31 +0000 [r194545] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Merged revisions 194520 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194520 | kpfleming | 2009-05-14 17:26:02 -0500 (Thu, 14 May
+ 2009) | 9 lines Merged revisions 194509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May
+ 2009) | 1 line Update URL to Reviewboard ........
+ ................
+
+2009-05-14 22:23 +0000 [r194510] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 194496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May
+ 2009) | 30 lines Merged revisions 194484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May
+ 2009) | 24 lines 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 ........ ................
+
+2009-05-14 17:07 +0000 [r194437] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 194434 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194434 |
+ file | 2009-05-14 14:05:33 -0300 (Thu, 14 May 2009) | 7 lines Fix
+ a bug where the 'T' option to Meetme did not work. (closes issue
+ #15031) Reported by: Stochastic (closes issue #13801) Reported
+ by: justdave ........
+
+2009-05-14 16:23 +0000 [r194431] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 194430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194430 |
+ tilghman | 2009-05-14 11:22:14 -0500 (Thu, 14 May 2009) | 7 lines
+ If the timing ended on a zero, then we would loop forever.
+ (closes issue #14983) Reported by: teox Patches:
+ 20090513__issue14983.diff.txt uploaded by tilghman (license 14)
+ Tested by: teox ........
+
+2009-05-13 13:42 +0000 [r194213] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 194209 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) |
+ 18 lines Merged revisions 194208 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) |
+ 11 lines 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 00:54 +0000 [r194141] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 194138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194138 | tilghman | 2009-05-12 19:52:49 -0500 (Tue, 12 May 2009)
+ | 14 lines Merged revisions 194137 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009)
+ | 7 lines 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 22:48 +0000 [r194059] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 194057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194057 | mnicholson | 2009-05-12 17:32:13 -0500 (Tue, 12 May
+ 2009) | 22 lines Merged revisions 194028 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May
+ 2009) | 16 lines 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 20:51 +0000 [r193962] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 193954 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 |
+ mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18
+ lines Update spiral support in trunk and 1.6.X to match what is
+ in 1.4. In 1.4, a SIP spiral is treated the same way as a call
+ forward. This works much better than what is currently in trunk
+ and 1.6.X. The code in trunk and 1.6.X did not create a new call
+ to the recipient of the spiral, instead trying to continue the
+ same call. In addition to just being plain wrong, this also had
+ the side effect of only being able to spiral calls to other SIP
+ channels. With this in place, as long as call forwards are
+ honored, SIP spirals will work properly. This means that it will
+ work for outbound calls made by the Queue, Dial, and Page
+ applications. For originated calls and spool calls, however, the
+ spiral will not work properly until a generic call forward
+ mechanism is introduced into Asterisk. (relates to issue #13630)
+ ........
+
+2009-05-12 20:42 +0000 [r193823-193959] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 193956 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193956 | tilghman | 2009-05-12 15:40:22 -0500
+ (Tue, 12 May 2009) | 13 lines Merged revisions 193955 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009)
+ | 6 lines 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) ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 193870 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r193870 | tilghman | 2009-05-12 12:29:33 -0500 (Tue, 12 May 2009)
+ | 2 lines Convert a THREADSTORAGE object into a simple malloc'd
+ object (as suggested by Russell on -dev) ........
+
+ * apps/app_voicemail.c, /: Recorded merge of revisions 193756 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193756 | tilghman | 2009-05-11 17:50:47 -0500
+ (Mon, 11 May 2009) | 25 lines Recorded merge of revisions 193755
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009)
+ | 18 lines 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 22:12 +0000 [r193719] Russell Bryant <russell@digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 193718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r193718 | russell | 2009-05-11 17:04:40 -0500 (Mon, 11 May 2009)
+ | 12 lines Fix some timer state corruption. In res_timer_timerfd,
+ handle the case that set_rate gets called while a timer is still
+ in continuous mode. In this case, we want to remember the
+ configured rate, but not actually set it until continuous mode
+ has been disabled. Thanks to dvossel for finding and helping to
+ debug the problem. (closes issue #15080) Reported by: dvossel
+ Tested by: dvossel ........
+
+2009-05-11 19:17 +0000 [r193617] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 193614 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500
+ (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009)
+ | 12 lines 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 18:59 +0000 [r193612] Leif Madsen <lmadsen@digium.com>
+
+ * /, funcs/func_channel.c: Update CHANNEL(transfercapabilities)
+ documentation. (closes issue #15073) Reported by: pkempgen
+ Patches: 20090511__issue15073__trunk.diff.txt uploaded by
+ tilghman (license 14)
+
+2009-05-10 17:08 +0000 [r193503] Joshua Colp <jcolp@digium.com>
+
+ * main/bridging.c, /: Merged revisions 193502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193502 |
+ file | 2009-05-10 14:07:46 -0300 (Sun, 10 May 2009) | 2 lines Fix
+ a bug where receiving a control frame of subclass -1 would cause
+ certain channels to get hung up. ........
+
+2009-05-09 11:33 +0000 [r193462] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h, /: Merged revisions 193461 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r193461 | russell | 2009-05-09 06:33:09 -0500 (Sat, 09 May 2009)
+ | 2 lines Minor documentation update for ast_event_queue().
+ ........
+
+2009-05-08 20:52 +0000 [r193390] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 193387 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 |
+ dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines
+ TCP not matching valid peer. find_peer() does not find a valid
+ peer when using pvt->recv as the sockaddr_in argument. Because of
+ the way TCP works, the port number in pvt->recv is not what we're
+ looking for at all. There is currently only one place that
+ find_peer searches for a peer using the sockaddr_in argument. If
+ the peer is not found after using pvt->recv (works for UDP since
+ the port number will be correct), a temp sockaddr_in struct is
+ made using the Contact header in the sip_request. This has the
+ correct port number in it. Review:
+ http://reviewboard.digium.com/r/236/ ........
+
+2009-05-08 19:51 +0000 [r193350] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 193349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193349 |
+ mmichelson | 2009-05-08 14:50:44 -0500 (Fri, 08 May 2009) | 12
+ lines Reset the members' call counts when resetting queue
+ statistics. This helps to prevent odd scenarios where a queue
+ will claim to have taken 0 calls, but the members appear to have
+ taken a non-zero amount. (closes issue #15068) Reported by: sum
+ Patches: patchreset.patch uploaded by sum (license 766) Tested
+ by: sum ........
+
+2009-05-08 15:36 +0000 [r193336] Sean Bright <sean.bright@gmail.com>
+
+ * funcs/func_devstate.c, /: Merged revisions 193274 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r193274 | seanbright | 2009-05-08 11:18:40 -0400 (Fri, 08 May
+ 2009) | 2 lines Fix the spelling of UNAVAILABLE in func_devstate
+ CLI completion. ........
+
+2009-05-08 14:55 +0000 [r193266] David Vossel <dvossel@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 193263 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500
+ (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009)
+ | 9 lines "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 14:12 +0000 [r193197] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/logger.conf.sample, /, main/logger.c: Merged revisions
+ 193194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May
+ 2009) | 13 lines Merged revisions 193193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May
+ 2009) | 7 lines 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 23:44 +0000 [r193123] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 193120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009)
+ | 26 lines Merged revisions 193119 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009)
+ | 19 lines 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 22:51 +0000 [r193080] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 193077 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500
+ (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009)
+ | 5 lines 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 17:53 +0000 [r192936-193008] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_odbc.c: Merged revisions 193006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193006 |
+ tilghman | 2009-05-07 12:51:13 -0500 (Thu, 07 May 2009) | 7 lines
+ Second result should not contain data from the first result.
+ (closes issue #15039) Reported by: jims Patches:
+ 20090506__issue15039.diff.txt uploaded by tilghman (license 14)
+ Tested by: jims ........
+
+ * channels/chan_unistim.c, /: Merged revisions 192938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009)
+ | 6 lines Send DTMF frame before playing back audio. (closes
+ issue #14858) Reported by: barryf Patches:
+ 20090507__bug14858.diff.txt uploaded by tilghman (license 14)
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 192933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009)
+ | 17 lines Merged revisions 192932 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009)
+ | 10 lines 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 ........ ................
+
+2009-05-07 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-beta2
+
+2009-05-06 22:20 +0000 [r192874] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 192861 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009)
+ | 17 lines Merged revisions 192858 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009)
+ | 10 lines 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 17:57 +0000 [r192813] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 190946 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) |
+ 1 line Make sure that we do not clear the down flag on the BRI
+ during PTMP link transients. Also refix SS7 audio that the early
+ media patch broke. ........
+
+2009-05-06 17:41 +0000 [r192637-192810] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 192808 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) |
+ 10 lines Fix a bug where a timer would be created but not
+ acknowledged. This scenario crept up if chan_iax2 was loaded with
+ no configuration file present. It would create a timer and tell
+ it to go at an interval but the thread that normally acknowledges
+ it would not be created because no configuration file was
+ present. The timer will now be closed if no configuration file is
+ present. (closes issue #15014) Reported by: madkins ........
+
+ * res/res_clialiases.c, /: Merged revisions 192736 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192736 | file | 2009-05-06 13:09:27 -0300 (Wed, 06 May 2009) | 4
+ lines Make the code that prevents an infinite loop from happening
+ into a case insensitive check. (thanks eliel) ........
+
+ * res/res_clialiases.c, /: Merged revisions 192700 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192700 | file | 2009-05-06 11:35:47 -0300 (Wed, 06 May 2009) | 5
+ lines Fix an infinite loop with tab completion of CLI aliases
+ that reference themselves. (closes issue #15020) Reported by:
+ junky ........
+
+ * /, channels/chan_sip.c: Merged revisions 192634 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) |
+ 14 lines Merged revisions 192633 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7
+ lines 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 20:02 +0000 [r192528] Sean Bright <sean.bright@gmail.com>
+
+ * /, static-http/astman.js: Merged revisions 192525 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r192525 | seanbright | 2009-05-05 15:57:49 -0400
+ (Tue, 05 May 2009) | 18 lines Merged revisions 192524 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May
+ 2009) | 11 lines 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 18:27 +0000 [r192402-192480] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 192462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) |
+ 15 lines Merged revisions 192454 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8
+ lines 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, /: Merged revisions 192430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192430 | file | 2009-05-05 14:46:51 -0300 (Tue, 05 May 2009) |
+ 12 lines Merged revisions 192429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5
+ lines Fix a bug where the followme application would continue
+ trying numbers after the caller hung up. (closes issue #13624)
+ Reported by: sgenyuk ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 192387 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 |
+ file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines
+ Fix a bug with setting t38pt_udptl at the user or peer level. If
+ an incoming call authenticated as a user or peer and t38pt_udptl
+ was not set to yes in general then no UDPTL session would be
+ present and any T38 related things would fail. This commit
+ changes it so that if after authenticating T38 is enabled but no
+ UDPTL session is present one will be created. (issue AST-215)
+ ........
+
+2009-05-05 13:43 +0000 [r192298-192360] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astobj2.c, include/asterisk/stringfields.h, /, main/utils.c:
+ Merged revisions 192357 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 |
+ kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5
+ lines Correct some flaws in the memory accounting code for
+ stringfields and ao2 objects Under some conditions, the memory
+ allocation for stringfields and ao2 objects would not have
+ supplied valid file/function names for MALLOC_DEBUG tracking, so
+ this commit corrects that. ........
+
+ * main/astobj2.c, main/datastore.c, main/channel.c, /,
+ include/asterisk/astobj2.h, include/asterisk/datastore.h,
+ include/asterisk/channel.h: Merged revisions 192318 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May
+ 2009) | 5 lines Properly account for memory allocated for
+ channels and datastores As in previous commits, when channels are
+ allocated (with ast_channel_alloc) or datastores are allocated
+ (with ast_datastore_alloc) properly account for the memory being
+ owned by the caller, instead of the allocator function itself.
+ ........
+
+ * include/asterisk/stringfields.h, /, main/utils.c: Merged
+ revisions 192279 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 |
+ kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5
+ lines Ensure that string pools allocated to hold stringfields are
+ properly accounted in MALLOC_DEBUG mode This commit modifies the
+ stringfield pool allocator to remember the 'owner' of the
+ stringfield manager the pool is being allocated for, and ensures
+ that pools allocated in the future when fields are populated are
+ owned by that file/function. ........
+
+2009-05-04 22:48 +0000 [r192217] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 192214 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r192214 | dvossel | 2009-05-04 17:44:51 -0500
+ (Mon, 04 May 2009) | 17 lines Merged revisions 192213 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009)
+ | 11 lines 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 ........ ................
+
+2009-05-04 19:34 +0000 [r192175] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 192059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192059 |
+ kpfleming | 2009-05-04 18:24:16 +0200 (Mon, 04 May 2009) | 5
+ lines Ensure that astobj2 memory allocations are properly
+ accounted for when MALLOC_DEBUG is used This commit ensures that
+ all astobj2 allocated objects are properly accounted for in
+ MALLOC_DEBUG mode by passing down the file/function/line
+ information from the module/function that actually called the
+ astobj2 allocation function. ........
+
+2009-05-04 19:31 +0000 [r192135-192173] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, res/res_agi.c: Merged revisions 192171 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192171 | tilghman | 2009-05-04 14:29:13 -0500 (Mon, 04 May 2009)
+ | 8 lines Restore 'asyncagi break' command to 1.6.1 and higher.
+ (closes issue #14985) Reported by: nikkk Patches:
+ 20090428__bug14985.diff.txt uploaded by tilghman (license 14)
+ 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license
+ 14) Tested by: nikkk ........
+
+ * autoconf/ast_ext_tool_check.m4, /: Merged revisions 192132 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04
+ May 2009) | 6 lines Pass libraries in LIBS, not LDFLAGS. (closes
+ issue #14671) Reported by: Chainsaw Patches:
+ asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by
+ Chainsaw (license 723) ........
+
+2009-05-04 17:45 +0000 [r192097] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_forkcdr.c, /: Merged revisions 192096 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192096 |
+ lmadsen | 2009-05-04 13:42:56 -0400 (Mon, 04 May 2009) | 4 lines
+ Commit documentation changes related to issue #14801. (issue
+ #14801) ........
+
+2009-05-04 15:54 +0000 [r192033] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, main/xml.c: Merged revisions 192032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192032 |
+ eliel | 2009-05-04 11:35:35 -0400 (Mon, 04 May 2009) | 3 lines Do
+ not re-define _POSIX_C_SOURCE if it was already defined. ........
+
+2009-05-04 10:01 +0000 [r191958] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configs/modules.conf.sample: Merged revisions 191955 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04
+ May 2009) | 8 lines Ensure that by default only one console
+ channel driver is loaded This configuration file was changed to
+ ensure that only one console channel driver (chan_oss) is loaded
+ by default, but the change would only work if chan_console was
+ not built. Now it will work as expected; if chan_alsa or
+ chan_console are built and installed, they will not be loaded
+ unless explicity requested. ........
+
+2009-05-03 14:06 +0000 [r191885] Russell Bryant <russell@digium.com>
+
+ * Makefile, /: Merged revisions 191884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191884 |
+ russell | 2009-05-03 09:05:10 -0500 (Sun, 03 May 2009) | 2 lines
+ Remove unnecessary compiler flag ........
+
+2009-05-02 18:48 +0000 [r191779] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/logger.c: Merged revisions 191775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191775 |
+ kpfleming | 2009-05-02 20:39:48 +0200 (Sat, 02 May 2009) | 5
+ lines Fix an error in queue_log file rotation optimization code
+ This code was copy-and-pasted without properly changing
+ references to event_rotate into queue_rotate, so under some
+ conditions the log rotation would rotate queue_log even though it
+ was not necessary. ........
+
+2009-05-02 15:52 +0000 [r191703] Sean Bright <sean.bright@gmail.com>
+
+ * main/asterisk.c, /: Merged revisions 191700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191700 |
+ seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1
+ line Update copyright year to 2009 ........
+
+2009-05-01 20:02 +0000 [r191554-191563] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 191560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009)
+ | 13 lines Merged revisions 191559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009)
+ | 6 lines 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) ........ ................
+
+ * channels/chan_iax2.c, /: Merged revisions 191494 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009)
+ | 4 lines Set debug message back to DEBUG level. (closes issue
+ #15007) Reported by: hulber ........
+
+2009-05-01 18:20 +0000 [r191508] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 191489 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009)
+ | 15 lines Merged revisions 191488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009)
+ | 9 lines 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. ........ ................
+
+2009-04-30 17:46 +0000 [r191224-191370] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Merged revisions 191367 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 |
+ tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines
+ Detect eaccess (or euidaccess) before using it. Reported by
+ Andrew Lindh via the -dev list. ........
+
+ * main/asterisk.c, /: Merged revisions 191283 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191283 |
+ tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11
+ lines Change working directory to / under certain conditions. If
+ backgrounding and no core will be produced, then changing the
+ directory won't break anything; likewise, if the CWD isn't
+ accessible by the current user, then a core wasn't possible
+ anyway. (closes issue #14831) Reported by: chris-mac Patches:
+ 20090428__bug14831.diff.txt uploaded by tilghman (license 14)
+ 20090430__bug14831.diff.txt uploaded by tilghman (license 14)
+ Tested by: chris-mac ........
+
+ * /, channels/h323/ast_h323.cxx, channels/chan_h323.c: Merged
+ revisions 191219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 |
+ tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines
+ Make H.323 compile with FDLEAK detection code enabled ........
+
+2009-04-29 18:40 +0000 [r191139] David Brooks <dbrooks@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 191136 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191136 |
+ dbrooks | 2009-04-29 13:32:58 -0500 (Wed, 29 Apr 2009) | 3 lines
+ Removing crufty code that is no longer necessary. Code cleanup.
+ ........
+
+2009-04-29 08:59 +0000 [r190994] Russell Bryant <russell@digium.com>
+
+ * main/indications.c, /: Merged revisions 190993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190993 |
+ russell | 2009-04-29 03:58:39 -0500 (Wed, 29 Apr 2009) | 7 lines
+ Log an error message if indications.conf is not found. (closes
+ issue #14990) Reported by: tzafrir Patches: indications_err.diff
+ uploaded by tzafrir (license 46) ........
+
+2009-04-29 06:38 +0000 [r190985] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c, /: Merged revisions 190830 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r190830 | transnexus | 2009-04-28 17:10:42 +0800 (Tue, 28 Apr
+ 2009) | 2 lines Updated for OSP Toolkit 3.5. ........
+
+2009-04-28 17:33 +0000 [r190907] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/tex/cdrdriver.tex, /: Merged revisions 190904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r190904 | tilghman | 2009-04-28 12:31:43 -0500 (Tue, 28 Apr 2009)
+ | 2 lines UniqueID column has a maximum size of 150 ........
+
+2009-04-28 14:17 +0000 [r190732-190869] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 190865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190865 |
+ kpfleming | 2009-04-28 09:15:47 -0500 (Tue, 28 Apr 2009) | 5
+ lines Build XML documention from *only* the source files that
+ have docs in them Change the build process so that
+ doc/core-en_US.xml is dependent solely on the source files that
+ have documentation in them, not on all source files. ........
+
+ * /, Makefile.rules: Merged revisions 190861 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190861 |
+ kpfleming | 2009-04-28 09:12:09 -0500 (Tue, 28 Apr 2009) | 5
+ lines Remove Makefile rules for bison and flex sources We never,
+ ever want these files to processed automatically, because we
+ store the output files in Subversion and users should never need
+ to rebuild them. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in: Merged revisions
+ 190725 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr
+ 2009) | 13 lines Merged revisions 190721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr
+ 2009) | 7 lines 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 19:36 +0000 [r190729] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 190726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190726 |
+ tilghman | 2009-04-27 14:34:48 -0500 (Mon, 27 Apr 2009) | 4 lines
+ Don't warn on pipe in the System call. (closes issue #14979)
+ Reported by: pj ........
+
+2009-04-27 19:15 +0000 [r190666] Russell Bryant <russell@digium.com>
+
+ * res/res_smdi.c, /: Merged revisions 190663 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009)
+ | 22 lines Merged revisions 190661-190662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009)
+ | 9 lines 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) ........ r190662 | russell | 2009-04-27 14:03:59 -0500
+ (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661. ........
+ ................
+
+2009-04-27 16:28 +0000 [r190625] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 190622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190622 |
+ mmichelson | 2009-04-27 11:26:14 -0500 (Mon, 27 Apr 2009) | 3
+ lines Update warning message to not have pipes and contain all
+ options. ........
+
+2009-04-23 21:23 +0000 [r190383] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 190371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ ........
+
+2009-04-23 20:44 +0000 [r190355] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 190352 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190352 |
+ tilghman | 2009-04-23 15:42:11 -0500 (Thu, 23 Apr 2009) | 7 lines
+ Labels are sometimes (most of the time?) NULL for extensions.
+ (closes issue #14895) Reported by: chris-mac Patches:
+ 20090423__bug14895__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen ........
+
+2009-04-23 19:18 +0000 [r190297] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 190287 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r190287 | file | 2009-04-23 16:15:30 -0300 (Thu,
+ 23 Apr 2009) | 13 lines Merged revisions 190286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6
+ lines 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 17:47 +0000 [r190253] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 190250 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190250 |
+ mmichelson | 2009-04-23 12:45:35 -0500 (Thu, 23 Apr 2009) | 9
+ lines Fix reversed behavior of leavewhenempty option in
+ queues.conf. (closes issue #14650) Reported by: alecdavis
+ Patches: 14650.patch uploaded by mmichelson (license 60) Tested
+ by: mmichelson, lmadsen ........
+
+2009-04-22 21:43 +0000 [r190096] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Merged revisions 190093 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r190093 | tilghman | 2009-04-22 16:38:15 -0500
+ (Wed, 22 Apr 2009) | 14 lines Merged revisions 190092 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009)
+ | 7 lines 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 21:18 +0000 [r189997-190066] Jeff Peeler <jpeeler@digium.com>
+
+ * main/cli.c, funcs/func_groupcount.c, /, main/app.c,
+ include/asterisk/channel.h: Merged revisions 190057 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009)
+ | 9 lines Fix building of chan_h323 with gcc-3.3 There seems to
+ be a bug with old versions of g++ that doesn't allow a structure
+ member to use the name list. Rename list member to group_list in
+ ast_group_info and change the few places it is used. (closes
+ issue #14790) Reported by: stuarth ........
+
+ * channels/h323/chan_h323.h, /, channels/h323/ast_h323.cxx,
+ channels/chan_h323.c: Merged revisions 189993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189993 |
+ jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines
+ Make chan_h323 respect packetization settings and fix small
+ reload issue. 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. Also, an
+ unrelated config reload issue has been fixed in the case of the
+ config file not changing. (closes issue #12415) Reported by: pj
+ Patches: 2009012200_h323packetization.diff.txt uploaded by
+ mvanbaak (license 7), modified by me ........
+
+2009-04-22 18:01 +0000 [r189986] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 189951 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189951 |
+ russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines
+ Fix call parking callback. Pipes -> Commas. ........
+
+2009-04-22 16:04 +0000 [r189816-189914] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_unistim.c, /: Merged revisions 189911 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009)
+ | 7 lines Do not continue to receive DTMF, when the channel is
+ hungup and about to be destroyed. (closes issue #14858) Reported
+ by: barryf Patches: 20090421__bug14858.diff.txt uploaded by
+ tilghman (license 14) Tested by: barryf ........
+
+ * /, configure, configure.ac: Merged revisions 189813 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r189813 | tilghman | 2009-04-22 01:33:08 -0500 (Wed, 22 Apr 2009)
+ | 3 lines Detect liblua on SuSE, and add libm for linking for
+ Fedora. (Reported via the -dev list, Subject: Compiling Asterisk
+ with LUA) ........
+
+2009-04-21 20:45 +0000 [r189775] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189771 |
+ dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines
+ Fixes segfault when switching UDP to TCP in sip.conf after
+ reload. If transport in sip.conf is switched from UDP to TCP,
+ Asterisk segfaults right after issuing a sip reload. The problem
+ is the socket type is changed to TCP but the fd may still be
+ present for UDP. Later, when the TCP session should be created or
+ set using an existing one, it isn't because the old file
+ descriptor is still present. Now every time transport is changed
+ during a sip.conf reload, the file descriptor is set to -1,
+ signifying it must be created or found. (closes issue #14727)
+ Reported by: pj Tested by: dvossel Review:
+ http://reviewboard.digium.com/r/229/ ........
+
+2009-04-20 22:11 +0000 [r189540] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c, /: Merged revisions 189539 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r189539 | tilghman | 2009-04-20 17:10:25 -0500 (Mon, 20 Apr 2009)
+ | 3 lines Use nanosleep instead of poll. This is not just because
+ mmichelson suggested it, but also because Mac OS X puked on my
+ poll(). ........
+
+2009-04-20 21:41 +0000 [r189536] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 189495,189516 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r189495 | twilson | 2009-04-20 16:24:34 -0500
+ (Mon, 20 Apr 2009) | 9 lines Merged revisions 189463 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20
+ Apr 2009) | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL
+ ........ ................ r189516 | twilson | 2009-04-20 16:29:29
+ -0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189465 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009)
+ | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is
+ set ........ ................
+
+2009-04-20 21:36 +0000 [r189533] Sean Bright <sean.bright@gmail.com>
+
+ * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189464 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r189464 | seanbright | 2009-04-20 17:09:59 -0400
+ (Mon, 20 Apr 2009) | 20 lines Merged revisions 189462 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr
+ 2009) | 13 lines 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@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 17:11 +0000 [r189353] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 |
+ file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
+ Fix a bug with non-UDP connections that caused dialogs to not get
+ freed. This issue crept up because of a reference count issue on
+ non-UDP based dialogs. The dialog reference count was increased
+ when transmitting a packet reliably but never decreased. This
+ caused the dialog structure to hang around despite being unlinked
+ from the dialogs container. (closes issue #14919) Reported by:
+ vrban ........
+
+2009-04-20 14:07 +0000 [r189281] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 189278 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr
+ 2009) | 18 lines Merged revisions 189277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
+ 2009) | 12 lines 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 01:42 +0000 [r189207-189208] David Vossel <dvossel@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500
+ (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009)
+ | 12 lines 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/ ........ ................
+
+ * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500
+ (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009)
+ | 12 lines 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 21:56 +0000 [r189140] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+ revisions 189137 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009)
+ | 17 lines Merged revisions 188833,189134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
+ | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
+ Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
+ rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
+ Modifed/added some debug messages. JIRA ABE-1835 ........
+ ................
+
+2009-04-17 20:21 +0000 [r189105] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 |
+ mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13
+ lines Prevent a crash when SIP blonde transferring an unbridged
+ call. If one attempts to use the attended transfer button on a
+ SIP phone to transfer an unbridged call (such as a call to an
+ IVR) but hangs up while the target of the transfer is still
+ ringing, we need to not crash. The problem was that ast_hangup
+ was called from outside the channel thread. AST-211 ........
+
+2009-04-17 19:47 +0000 [r189081] Sean Bright <sean.bright@gmail.com>
+
+ * main/asterisk.c, /: Merged revisions 189077 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 |
+ seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1
+ line Fix copy/paste error with 'transmit silence' flag. ........
+
+2009-04-17 17:31 +0000 [r189068] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 189010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr
+ 2009) | 12 lines Merged revisions 189009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
+ 2009) | 5 lines Make Busy() application set the CDR disposition
+ to BUSY. (closes issue #14306) Reported by: cristiandimache
+ ........ ................
+
+2009-04-17 14:50 +0000 [r188941-188950] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) |
+ 22 lines Merged revisions 188946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
+ 15 lines 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, /: Merged revisions 188938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri,
+ 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4
+ lines Fix a situation where the DAHDI channel private structure
+ lock was not unlocked when it should have been. (issue AST-210)
+ ........ ................
+
+2009-04-16 22:05 +0000 [r188777-188839] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009)
+ | 14 lines Merged revisions 188835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
+ | 7 lines 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 ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500
+ (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009)
+ | 4 lines Umask should not be exported into global namespace.
+ (closes issue #14912) Reported by: jcapp ........
+ ................
+
+2009-04-15 20:20 +0000 [r188474-188598] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/file.c: Merged revisions 188585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr
+ 2009) | 13 lines Merged revisions 188582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
+ 2009) | 7 lines Update ast_readvideo_callback to match
+ ast_readaudio_callback. This fixes potential refcount errors that
+ may occur on ast_filestreams. AST-208 ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 |
+ mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3
+ lines Fix a couple of queue member reference leaks. ........
+
+2009-04-14 17:46 +0000 [r188259-188416] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 188413 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 |
+ file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix
+ an incorrect clock rate when sending T140 text. (closes issue
+ #14029) Reported by: epicac ........
+
+ * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 |
+ file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix
+ a bug with the change I made yesterday to outbound proxy support.
+ Per discussion with oej on IRC we need the actual IP address, not
+ the outbound proxy IP address, in the sa field. Upon further
+ inspection this should make the behaviour of all other uses of
+ the outbound proxy in the code. ........
+
+2009-04-14 05:47 +0000 [r188209-188213] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 188210 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 |
+ tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
+ As suggested by Russell, warn users when their dialplan arguments
+ contain pipes, but not commas. ........
+
+ * /, utils/smsq.c: Merged revisions 188206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 |
+ tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
+ Application delimiter is ',', not '|'. (closes issue #14881)
+ Reported by: stegro Patches: smsq.patch uploaded by stegro
+ (license 752) ........
+
+2009-04-13 19:33 +0000 [r188105] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 188102 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr
+ 2009) | 5 lines Fix another crash related to cached realtime
+ music on hold. This was another off-by-one problem caused by
+ moh_register. ........
+
+2009-04-13 16:34 +0000 [r188070] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 |
+ file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
+ Fix a bug where using an outbound proxy would cause the local
+ address to be 127.0.0.1. Copy the outbound proxy IP address into
+ the SIP dialog structure as the IP address we will be sending to.
+ This has to be done because the logic that determines what local
+ IP address to use in the SIP messages is not aware of an outbound
+ proxy being in place. It only knows what IP address we are
+ sending to. (closes issue #12006) Reported by: mnicholson
+ ........
+
+2009-04-13 14:20 +0000 [r188039] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 |
+ mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6
+ lines Set all queue variables on both the caller and member
+ channels. This allows for the variables to be accessed if a
+ member macro is run. Thanks to Grigoriy Puzankin for bringing
+ this up on the -dev list. ........
+
+2009-04-10 20:28 +0000 [r187916] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/Makefile, /: Merged revisions 187906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 |
+ jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
+ 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 Patches:
+ issue_11966.patch uploaded by kpfleming (license 421) Tested by:
+ jpeeler ........
+
+2009-04-10 17:31 +0000 [r187769] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/sip-friends.sql,
+ contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500
+ (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10
+ Apr 2009) | 2 lines Add lastms column to the contributed table
+ designs ........ ................
+
+2009-04-10 16:54 +0000 [r187724] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, build_tools/embed_modules.xml: Merged revisions 187721 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10
+ Apr 2009) | 5 lines clean up some patterns for files to remove
+ add embedding support for bridge and test modules ........
+
+2009-04-10 16:05 +0000 [r187679] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 |
+ tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines
+ Ensure pvt is not NULL before dereferencing it. (closes issue
+ #14784) Reported by: pj ........
+
+2009-04-10 16:01 +0000 [r187677] Russell Bryant <russell@digium.com>
+
+ * tests/test_sched.c, tests/test_heap.c, /: Merged revisions 187675
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r187675 | russell | 2009-04-10 11:00:29 -0500 (Fri, 10
+ Apr 2009) | 2 lines Disable test modules by default. ........
+
+2009-04-10 03:57 +0000 [r187601] Tilghman Lesher <tlesher@digium.com>
+
+ * main/audiohook.c, main/bridging.c, main/channel.c, main/pbx.c,
+ main/manager.c, /, include/asterisk/linkedlists.h,
+ main/features.c, main/http.c, main/app.c,
+ include/asterisk/lock.h: Merged revisions 187599 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009)
+ | 2 lines Modify headers and macros, according to Russell's
+ suggestions on the -dev list ........
+
+2009-04-09 21:09 +0000 [r187564] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merge revision 187488 from trunk.
+
+2009-04-09 19:29 +0000 [r187531-187546] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c, /: Merged revisions 186379 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 |
+ dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 6 lines
+ audio_audiohook_write_list() did not correctly update sample size
+ after ast_translate. audio_audiohook_write_list() did not take
+ into account that the sample size may change after translation
+ depending on if the original frame is is 8khz or 16khz. the
+ sample size is now updated after translating to reflect this
+ possibility. This caused the audio on the receiving end to sound
+ terrible. Thanks to jcolp and mmichelson for helping me work this
+ out. (issue AST-197) ........
+
+ * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009)
+ | 16 lines Merged revisions 185845 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
+ | 10 lines Fixes issue with dropped calles due to re-Invite glare
+ and re-Invites never executing after a 491 Acknowledgement for
+ 491 responses were never being processed because it didn't match
+ our pending invite's seqno. Since the ACK was never processed,
+ the 491 frame would continue to be retransmitted until eventually
+ the call was dropped due to max retries. Now during a pending
+ invite, if we receive another invite, we send an 491 and hold on
+ to that glare invite's seqno in the "glareinvite" variable for
+ that sip_pvt struct. When ACK's are received, we first check to
+ see if it is in response to our pending invite, if not we check
+ to see if it is in response to a glare invite. In this case, it
+ is in response to the glare invite and must be dealt with or the
+ call is dropped. I've changed the wait time for resending the
+ re-Invite after receving a 491 response to comply with RFC 3261.
+ Before this patch the scheduled re-Invite would only change a
+ flag indicating that the re-Invite should be sent out, now it
+ actually sends it out as well. (closes issue #12013) Reported by:
+ alx Review: http://reviewboard.digium.com/r/213/ ........
+ ................
+
+2009-04-09 19:15 +0000 [r187496] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 187421,187424 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu,
+ 09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using
+ cached realtime moh. The moh_register function links an mohclass
+ and then immediately unrefs the class since the container now has
+ a reference. The problem with using realtime music on hold is
+ that the class is allocated, registered, and started in one fell
+ swoop. The refcounting logic resulted in the count being off by
+ one. The same problem did not happen when using a static config
+ because the allocation and registration of an mohclass is a
+ separate operation from starting moh. This also did not affect
+ non-cached realtime moh because the classes are not registered at
+ all. I also have modified res_musiconhold to use the _t_ variants
+ of the ao2_ functions so that more info can be gleaned when
+ attempting to trace the refcounts. I found this to be incredibly
+ helpful for debugging this issue and there's no good reason to
+ remove it. (closes issue #14661) Reported by: sum ........
+ r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr
+ 2009) | 3 lines Use safe macro practices even though they really
+ aren't necessary. ........
+
+2009-04-09 18:55 +0000 [r187051-187487] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /, include/asterisk/linkedlists.h,
+ include/asterisk/lock.h: Merged revisions 187483 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500
+ (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009)
+ | 8 lines Race condition between ast_cli_command() and 'module
+ unload' could cause a deadlock. Add lock timeouts to avoid this
+ potential deadlock. (closes issue #14705) Reported by: jamessan
+ Patches: 20090320__bug14705.diff.txt uploaded by tilghman
+ (license 14) Tested by: jamessan ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 |
+ tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines
+ Allow '/' in username portion of register; this is a regression.
+ (closes issue #14668) Reported by: Netview ........
+
+ * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions
+ 187363 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009)
+ | 10 lines Merged revisions 187362 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009)
+ | 3 lines Permit zero-length text messages in SIP. (Related to an
+ issue posted to the -users list, subject "AEL2, BASE64_DECODE and
+ hexadecimal") ........ ................
+
+ * main/asterisk.c, agi/Makefile, build_tools/cflags.xml,
+ utils/Makefile, include/asterisk.h, /, main/Makefile,
+ main/file.c, main/astfd.c (added): Merged revisions 187302 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500
+ (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009)
+ | 3 lines Add debugging mode for diagnosing file descriptor
+ leaks. (Related to issue #14625) ........ r187301 | tilghman |
+ 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops,
+ missed this file in the last commit. ........ ................
+
+ * /, funcs/func_odbc.c: Merged revisions 187050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187050 |
+ tilghman | 2009-04-08 12:08:43 -0500 (Wed, 08 Apr 2009) | 7 lines
+ If the first column is empty, output a delimiter anyway. (closes
+ issue #14848) Reported by: john8675309 Patches:
+ 20090408__bug14848.diff.txt uploaded by tilghman (license 14)
+ Tested by: john8675309 ........
+
+2009-04-08 16:54 +0000 [r186988-187049] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 187046 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500
+ (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr
+ 2009) | 10 lines Fix a small logical error when loading moh
+ classes. We were unconditionally incrementing the number of
+ mohclasses registered. However, we should actually only increment
+ if the call to moh_register was successful. While this probably
+ has never caused problems, I noticed it and decided to fix it
+ anyway. ........ ................
+
+ * main/channel.c, /: Merged revisions 186985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr
+ 2009) | 30 lines Merged revisions 186984 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr
+ 2009) | 24 lines Make a couple of changes with regards to a new
+ message printed in ast_read(). "ast_read() called with no
+ recorded file descriptor" is a new message added after a bug was
+ discovered. Unfortunately, it seems there are a bunch of places
+ that potentially make such calls to ast_read() and trigger this
+ error message to be displayed. This commit does two things to
+ help to make this message appear less. First, the message has
+ been downgraded to a debug level message if dev mode is not
+ enabled. The message means a lot more to developers than it does
+ to end users, and so developers should take an effort to be sure
+ to call ast_read only when a channel is ready to be read from.
+ However, since this doesn't actually cause an error in operation
+ and is not something a user can easily fix, we should not spam
+ their console with these messages. Second, the message has been
+ moved to after the check for any pending masquerades. ast_read()
+ being called with no recorded file descriptor should not
+ interfere with a masquerade taking place. This could be seen as a
+ simple way of resolving issue #14723. However, I still want to
+ try to clear out the existing ways of triggering this message,
+ since I feel that would be a better resolution for the issue.
+ ........ ................
+
+2009-04-08 12:39 +0000 [r186929] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 186928 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186928 |
+ russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines
+ Update some comments and resolve potential memory corruption in
+ chan_sip. While browsing chan_sip the other day, I noticed this
+ dangerous code in dialog_needdestroy(). This function is an
+ ao2_callback. It is absolutely _not_ okay to unlock the container
+ from within this function. It's also not clear why it was useful.
+ Given that it could cause memory corruption, I have removed it.
+ There was also a TODO comment left describing a potential
+ implementation of an improvement to the needdestroy handling. I'm
+ not convinced that what was described is the best choice here, so
+ I have briefly described the way that this function is used today
+ that could be improved. ........
+
+2009-04-08 05:08 +0000 [r186901] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 |
+ tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines
+ Add lastms to the require API call. ........
+
+2009-04-08 00:10 +0000 [r186836-186845] Mark Michelson <mmichelson@digium.com>
+
+ * formats/format_wav_gsm.c, /, formats/format_wav.c: Merged
+ revisions 186842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr
+ 2009) | 14 lines Merged revisions 186841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr
+ 2009) | 8 lines Fix a few typos of the word "frequency." (closes
+ issue #14842) Reported by: jvandal Patches: frequency-typo.diff
+ uploaded by jvandal (license 413) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 |
+ mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7
+ lines Fix bad merge from fix for issue 13867. (closes issue
+ #14686) Reported by: davidw ........
+
+ * main/channel.c, /: Merged revisions 186833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr
+ 2009) | 15 lines Merged revisions 186832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr
+ 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a
+ p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
+ warning sounds will not be properly played to either party of the
+ bridge. (closes issue #14845) Reported by: adomjan ........
+ ................
+
+2009-04-07 22:33 +0000 [r186807] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_macro.c: Merged revisions 186799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009)
+ | 10 lines Merged revisions 186775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009)
+ | 3 lines Fix Macro documentation to match current (and intended)
+ behavior. (See -dev mailing list) ........ ................
+
+2009-04-07 20:59 +0000 [r186723] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 186720 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr
+ 2009) | 12 lines Merged revisions 186719 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr
+ 2009) | 6 lines Ensure that \r\n is printed after the ActionID in
+ an OriginateResponse. (closes issue #14847) Reported by: kobaz
+ ........ ................
+
+2009-04-03 20:21 +0000 [r186469] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500
+ (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr
+ 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not
+ properly switch formats when requested Don't offer
+ AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
+ provide a slight performance benefit, the translation core in
+ Asterisk has some flaws when a channel driver offers multiple raw
+ formats. this fix is much simpler than fixing the translation
+ core to solve that issue (although that will be done later).
+ ........ ................
+
+2009-04-03 20:05 +0000 [r186449] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
+ revisions 186444,186447 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009)
+ | 14 lines Merged revisions 186415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009)
+ | 7 lines Distinguish in a sent email between simple sends and
+ forwards. (closes issue #11678) Reported by: jamessan Patches:
+ 20090330__bug11678.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman, lmadsen ........ ................ r186447 |
+ tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines
+ Merged revisions 186445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009)
+ | 2 lines Found a conflict in the last commit, due to multiple
+ targets ........ ................
+
+2009-04-03 15:56 +0000 [r186324] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/crypto.h, /: Merged revisions 186321 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri,
+ 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
+ lines Fix a problem with the crypto variable definitions not
+ actually being defined properly. (closes issue #14804) Reported
+ by: jvandal ........ ................
+
+2009-04-03 15:19 +0000 [r186302] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c, /: Merged revisions 186297 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r186297 | tilghman | 2009-04-03 10:18:28 -0500 (Fri, 03 Apr 2009)
+ | 4 lines Compatibility fix for glibc 2.4 (Closes issue #14820)
+ Reported by: phsultan ........
+
+2009-04-03 14:34 +0000 [r186289] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr
+ 2009) | 20 lines Fix the ability to retrieve voicemail messages
+ from IMAP. A recent change made interactive vm_states no longer
+ get added to the list of vm_states and instead get stored in
+ thread-local storage. In trunk and all the 1.6.X branches, the
+ problem is that when we search for messages in a voicemail box,
+ we would attempt to update the appropriate vm_state struct by
+ directly searching in the list of vm_states instead of using the
+ get_vm_state_by_imap_user function. This meant we could not find
+ the interactive vm_state that we wanted. (closes issue #14685)
+ Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
+ (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........
+
+2009-04-03 02:11 +0000 [r186233] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009)
+ | 29 lines Merged revisions 186229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
+ | 21 lines Fix a memory leak in cdr_radius. I came across this
+ while doing some testing of my ast_channel_ao2 branch. After
+ running a test overnight that generated over 5 million calls,
+ Asterisk had taken up about 1 GB of my system memory. So, I
+ re-ran the test with MALLOC_DEBUG turned on. However, it showed
+ no leaks in Asterisk during the test, even though Asterisk was
+ still consuming it somehow. Instead, I turned to valgrind, which
+ when run with --leak-check=full, told me exactly where the leak
+ came from, which was from allocations inside the radiusclient-ng
+ library. This explains why MALLOC_DEBUG did not report it. After
+ a bit of analysis, I found that we were leaking a little bit of
+ memory every time a CDR record was passed to cdr_radius. I don't
+ actually have a radius server set up to receive CDR records.
+ However, I always have my development systems compile and install
+ all modules. In addition to making sure there are not build
+ errors across modules, always loading modules helps find bugs
+ like this, too, so it is strongly recommend for all developers.
+ ........ ................
+
+2009-04-02 22:00 +0000 [r186178] Mark Michelson <mmichelson@digium.com>
+
+ * configs/features.conf.sample, /: Merged revisions 186175 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500
+ (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
+ 2009) | 5 lines Fix instructions in one-step parking comment to
+ make more sense. Changed a capital K to a lowercase k. ........
+ ................
+
+2009-04-02 17:28 +0000 [r186111] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500
+ (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
+ Apr 2009) | 3 lines ensure that the buffer passed to
+ DAHDI_SET_BUFINFO is fully initialized ........ ................
+
+2009-04-02 17:14 +0000 [r186022-186063] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 186060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009)
+ | 16 lines Merged revisions 186059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
+ (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
+ Apr 2009) | 2 lines Fix for AST-2009-003 ........
+ ................ ................
+
+ * main/strings.c, /: Merged revisions 186021 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186021 |
+ tilghman | 2009-04-02 10:14:22 -0500 (Thu, 02 Apr 2009) | 7 lines
+ Missed a common case for needing to extend the buffer. (closes
+ issue #14716) Reported by: sum Patches:
+ 20090402__bug14716.diff.txt uploaded by tilghman (license 14)
+ Tested by: sum ........
+
+2009-04-02 13:54 +0000 [r185957] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500
+ (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr
+ 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
+ DAHDI_GET_PARAMS ioctls were recently corrected to show that they
+ do, in fact, read data from userspace as part of their work. due
+ to this fix, valgrind now reports a number of cases where
+ chan_dahdi passed an uninitialized (or partially) buffer to these
+ ioctls, which could lead to unexpected behavior. this patch
+ corrects chan_dahdi to ensure that buffers passed to these ioctls
+ are always fully initialized. ........ ................
+
+2009-04-01 22:44 +0000 [r185947] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/pbx.h, include/asterisk/strings.h,
+ main/taskprocessor.c, res/res_odbc.c,
+ include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c,
+ main/manager.c, /, main/tdd.c, include/asterisk/astobj2.h,
+ main/ast_expr2f.c: Merged revisions 185912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185912 |
+ tilghman | 2009-04-01 15:13:28 -0500 (Wed, 01 Apr 2009) | 4 lines
+ Merge changes from str_substitution that are unrelated to that
+ branch. Included is a small bugfix to an ast_str helper, but most
+ of these changes are simply doxygen fixes. ........
+
+2009-04-01 13:51 +0000 [r185775] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 185772 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009)
+ | 14 lines Merged revisions 185771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
+ | 6 lines Fix a case where DTMF could bypass audiohooks. This
+ change fixes a situation where an audiohook that wants DTMF would
+ not actually get it. This is in the code path where we end DTMF
+ digit length emulation while handling a NULL frame. ........
+ ................
+
+2009-03-31 22:38 +0000 [r185667] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils, /: Merged revisions 185664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 |
+ kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line
+ ignore copied (generated) file ........
+
+2009-03-31 22:13 +0000 [r185472-185605] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185604 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185604 |
+ mmichelson | 2009-03-31 17:12:52 -0500 (Tue, 31 Mar 2009) | 3
+ lines Fix trunk's compilation. ........
+
+ * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar
+ 2009) | 12 lines Merged revisions 185599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
+ 2009) | 6 lines Fix crash that would occur if an empty member was
+ specified in queues.conf. (closes issue #14796) Reported by: pida
+ ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500
+ (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar
+ 2009) | 8 lines Fix Russian voicemail intro to say the word
+ "messages" properly. (closes issue #14736) Reported by: chappell
+ Patches: voicemail_no_messages.diff uploaded by chappell (license
+ 8) ........ ................
+
+2009-03-31 17:51 +0000 [r185428] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_gtalk.c, /: Merged revisions 185363 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500
+ (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009)
+ | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains
+ extra whitespaces To drill into the xmpp to find the capabilities
+ between channels, chan_gtalk calls iks_child() and iks_next().
+ iks_child() and iks_next() are functions in the iksemel xml
+ parsing library that traverse xml nodes. The bug here is that
+ both iks_child() and iks_next() will return the next iks_struct
+ node *regardless* of type. chan_gtalk expects the next node to be
+ of type IKS_TAG, which in most cases, it is, but in this case (a
+ call being made from the Empathy IM client), there exists
+ iks_struct nodes which are not IKS_TAG data (they are extraneous
+ whitespaces), and chan_gtalk doesn't handle that case, so
+ capabilities don't match, and a call cannot be made.
+ iks_first_tag() and iks_next_tag(), on the other hand, will not
+ return the very next iks_struct, but will check to see if the
+ next iks_struct is of type IKS_TAG. If it isn't, it will be
+ skipped, and the next struct of type IKS_TAG it finds will be
+ returned. This assures that chan_gtalk will find the iks_struct
+ it is looking for. This fix simply changes all calls to
+ iks_child() and iks_next() to become calls to iks_first_tag() and
+ iks_next_tag(), which resolves the capability matching. The
+ following is a payload listing from Empathy, which, due to the
+ extraneous whitespace, will not be parsed correctly by iksemel:
+ <iq from='dbrooksjab@235-22-24-10/Telepathy'
+ to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
+ <session xmlns='http://www.google.com/session'
+ initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
+ id='1837267342'> <description
+ xmlns='http://www.google.com/session/phone'> <payload-type
+ clockrate='16000' name='speex' id='96'/> <payload-type
+ clockrate='8000' name='PCMA' id='8'/> <payload-type
+ clockrate='8000' name='PCMU' id='0'/> <payload-type
+ clockrate='90000' name='MPA' id='97'/> <payload-type
+ clockrate='16000' name='SIREN' id='98'/> <payload-type
+ clockrate='8000' name='telephone-event' id='99'/> </description>
+ </session> </iq> Review: http://reviewboard.digium.com/r/181/
+ ........ ................
+
+2009-03-31 14:59 +0000 [r185264] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 |
+ russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines
+ Don't free() an astobj2 object. (closes issue #14672) Reported
+ by: makoto ........
+
+2009-03-31 14:11 +0000 [r185200] Joshua Colp <jcolp@digium.com>
+
+ * main/audiohook.c, /: Merged revisions 185197 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) |
+ 15 lines Merged revisions 185196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
+ lines Fix crash when moving audiohooks between channels. Handle
+ the scenario where we are called to move audiohooks between
+ channels and the source channel does not actually have any on it.
+ (closes issue #14734) Reported by: corruptor ........
+ ................
+
+2009-03-30 20:52 +0000 [r185128-185129] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged
+ revisions 185123 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009)
+ | 9 lines Merged revisions 185121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
+ | 1 line Update the channel allocation method documentation.
+ ........ ................
+
+ * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500
+ (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
+ | 19 lines Make chan_misdn BRI TE side normally defer channel
+ selection to the NT side. Channel allocation collisions are not
+ handled by chan_misdn very well. This patch simply avoids the
+ problem for BRI only. For PRI, allocation collisions are still
+ possible but less likely since there are simply more channels
+ available and each end could use a different allocation strategy.
+ misdn.conf options available: te_choose_channel - Use to force
+ the TE side to allocate channels. method - Specify the channel
+ allocation strategy. (closes issue #13488) Reported by:
+ Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
+ Tested by: crich, siepkes, festr ........ ................
+
+2009-03-30 16:52 +0000 [r185089] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar
+ 2009) | 45 lines Merged revisions 185031 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
+ 2009) | 39 lines Fix queue weight behavior so that calls in
+ low-weight queues are not inappropriately blocked. (This is
+ copied and pasted from the review request I made for this patch)
+ Asterisk has some odd behavior when queue weights are used. The
+ current logic used when potentially calling a queue member is: If
+ the member we are going to call is part of another queue and
+ _that other queue has any callers in it_ and has a higher weight
+ than the queue we are calling from, then don't try to contact
+ that member. The issue here is what I have marked with
+ underscores. If the higher-weighted queue has any callers in it
+ at all, then the queue member will be unreachable from the
+ lower-weighted queue. This has the potential to be really really
+ bad if using a queue strategy, such as leastrecent or
+ fewestcalls, with the potential to call the same member
+ repeatedly. The fix proposed by garychen on issue 13220 is very
+ simple and, as far as I can see, works well for this situation.
+ With this set of changes, the logic used becomes: If the member
+ we are going to call is part of another queue, the other queue
+ has a higher weight than the queue we are calling from, and the
+ higher weight queue has at least as many callers as available
+ members, then do not try to contact the queue member. If the
+ higher weighted queue has fewer callers than available members,
+ then there is no reason to deny the call to this member since the
+ other queue can afford to spare a member. Since the fix involved
+ writing a generic function for determining the number of
+ available members in the queue, I also modified the is_our_turn
+ function to make use of the new num_available_members function to
+ determine if it is our turn to try calling a member. There is one
+ small behavior change. Before writing this patch, if you had
+ autofill disabled, then if you were the head caller in a queue,
+ you would automatically be told that it was your turn to try
+ calling a member. This did not take into account whether there
+ were actually any queue members available to take the call. Now
+ we actually make sure there is at least one member available to
+ take the call if autofill is disabled. (closes issue #13220)
+ Reported by: garychen Review:
+ http://reviewboard.digium.com/r/202/ ........ ................
+
+2009-03-30 14:43 +0000 [r184951] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) |
+ 21 lines Merged revisions 184947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
+ 14 lines Improve our handling of T38 in the initial INVITE from a
+ device. We now answer with matching media streams to what is
+ requested. If an INVITE is received with both a T38 and RTP media
+ stream this means we answer with both. For any outgoing calls
+ created as a result of this inbound one no T38 is requested in
+ the initial INVITE. Instead if we start receiving udptl packets
+ we trigger a reinvite on the outbound side. (closes issue #12437)
+ Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
+ Review: http://reviewboard.digium.com/r/208/ ........
+ ................
+
+2009-03-30 13:57 +0000 [r184913] Russell Bryant <russell@digium.com>
+
+ * channels/h323/Makefile.in, /: Merged revisions 184910 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30
+ Mar 2009) | 4 lines Fix build error when chan_h323 is not being
+ built. (reported by cai1982 in #asterisk-dev) ........
+
+2009-03-29 05:56 +0000 [r184839-184846] Russell Bryant <russell@digium.com>
+
+ * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009)
+ | 13 lines Merged revisions 184842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
+ | 5 lines Ensure targs variable is fully initialized. (closes
+ issue #14758) Reported by: tim_ringenbach ........
+ ................
+
+ * channels/Makefile, /: Merged revisions 184838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 |
+ russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines
+ Simplify chan_h323 build to not require a second run of "make".
+ (closes issue #14715) Reported by: jthurman Patches:
+ h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license
+ 614) Tested by: tzafrir, russell ........
+
+2009-03-27 19:21 +0000 [r184779] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, main/timing.c, main/channel.c, /,
+ bridges/bridge_softmix.c, include/asterisk/timing.h,
+ include/asterisk/channel.h: Merged revisions 184762 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar
+ 2009) | 12 lines Improve timing interface to remember which
+ provider provided a timer The ability to load/unload timing
+ interfaces is nice, but it means that when a timer is allocated,
+ it may come from provider A, but later provider B becomes the
+ 'preferred' provider. If this happens, all timer API calls on the
+ timer that was provided by provider A will actually be handed to
+ provider B, which will say WTF and return an error. This patch
+ changes the timer API to include a pointer to the provider of the
+ timer handle so that future operations on the timer will be
+ forwarded to the proper provider. (closes issue #14697) Reported
+ by: moy Review: http://reviewboard.digium.com/r/211/ ........
+
+2009-03-27 18:12 +0000 [r184707-184729] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27
+ Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure
+ we use the best RNG available. ........
+
+ * apps/app_queue.c, apps/app_voicemail.c, main/cli.c,
+ include/asterisk/app.h, /, apps/app_dumpchan.c, main/app.c:
+ Merged revisions 184693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184693 |
+ russell | 2009-03-27 11:21:10 -0500 (Fri, 27 Mar 2009) | 2 lines
+ Change global_app_buf to ast_str_thread_global_buf. ........
+
+2009-03-27 15:58 +0000 [r184650-184678] Joshua Colp <jcolp@digium.com>
+
+ * /, bridges/bridge_softmix.c: Merged revisions 184677 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184677 | file | 2009-03-27 12:57:28 -0300 (Fri, 27 Mar 2009) | 7
+ lines Fix a potential timer leak in bridge_softmix. It is
+ possible for a bridge to be created without actually being used.
+ In that scenario a timing file descriptor would be opened and not
+ closed. To fix this the timing file descriptor is now closed in
+ the destroy callback, not the thread function. ........
+
+ * /, res/res_agi.c: Merged revisions 184673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 |
+ file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix
+ speech structure leak in the AGI speech recognition integration.
+ The AGI dialplan applications did not destroy the speech
+ structure automatically if it was not destroyed by the running
+ AGI script. They will now do this. (issue LUMENVOX-15) ........
+
+ * /, bridges/bridge_softmix.c: Merged revisions 184639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184639 | file | 2009-03-27 11:18:40 -0300 (Fri, 27 Mar 2009) | 2
+ lines Remove a cast that is not needed. ........
+
+2009-03-27 14:09 +0000 [r184632] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /,
+ res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions
+ 184630 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 |
+ russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines
+ Change g_eid to ast_eid_default. ........
+
+2009-03-27 13:59 +0000 [r184612-184629] Joshua Colp <jcolp@digium.com>
+
+ * /, bridges/bridge_softmix.c: Merged revisions 184628 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184628 | file | 2009-03-27 10:57:29 -0300 (Fri, 27 Mar 2009) | 6
+ lines Fix a potential race condition when creating a software
+ based mixing bridge. It was possible for no timer to become
+ available between creating the bridge and starting it. We now
+ open a timer when creating it and keep it open until the bridge
+ is destroyed. ........
+
+ * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) |
+ 16 lines Merged revisions 184565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
+ lines Fix an issue where nat=yes would not always take effect for
+ the RTP session on outgoing calls. If calls were placed using an
+ IP address or hostname the global nat setting was copied over but
+ was not set on the RTP session itself. This caused the RTP stack
+ to not perform symmetric RTP actions. (closes issue #14546)
+ Reported by: acunningham ........ ................
+
+2009-03-27 02:35 +0000 [r184514-184552] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009)
+ | 20 lines Fix some issues with rwlock corruption that caused
+ deadlock like symptoms. When dvossel and I were doing some load
+ testing last week, we noticed that we could make Asterisk trunk
+ lock up instantly when we started generating a bunch of calls.
+ The backtraces of locked threads were bizarre, and many were
+ stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a
+ number of places where a backtrace would be loaded into an
+ invalid index of the backtrace array. It's an off by one error,
+ which ends up writing over the rwlock itself. 2) Ensure that in
+ the array of held locks, we NULL out an index once it is not
+ being used so that it's not confusing when analyzing its
+ contents. 3) Remove a bunch of logging referring to an rwlock
+ operating being done with "deep reentrancy". It is normal for
+ _many_ threads to hold a read lock on an rwlock. ........
+
+ * /, main/file.c: Merged revisions 184515 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 |
+ russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines
+ Don't act surprised if we get a -1 indication. ........
+
+ * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26
+ Mar 2009) | 2 lines Pass more useful information through to lock
+ tracking when DEBUG_THREADS is on. ........
+
+2009-03-26 22:19 +0000 [r184454] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 184448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar
+ 2009) | 9 lines Merged revisions 184447 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
+ 2009) | 3 lines use new, improved 8kHz prompts ........
+ ................
+
+2009-03-25 22:15 +0000 [r184343-184346] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 184344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 |
+ russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines
+ Remove unneeded AST_LIST_ENTRY() and comment on the purpose of
+ ast_event_ref. ........
+
+ * include/asterisk/_private.h, channels/chan_iax2.c,
+ channels/chan_dahdi.c, include/asterisk/event.h,
+ apps/app_minivm.c, res/ais/evt.c, main/event.c,
+ include/asterisk/strings.h, main/asterisk.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c,
+ channels/chan_unistim.c, include/asterisk/devicestate.h, /,
+ channels/chan_sip.c, main/devicestate.c: Merged revisions 184339
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25
+ Mar 2009) | 35 lines Improve performance of the ast_event cache
+ functionality. This code comes from
+ svn/asterisk/team/russell/event_performance/. Here is a summary
+ of the changes that have been made, in order of both invasiveness
+ and performance impact, from smallest to largest. 1) Asterisk
+ 1.6.1 introduces some additional logic to be able to handle
+ distributed device state. This functionality comes at a cost. One
+ relatively minor change in this patch is that the extra
+ processing required for distributed device state is now
+ completely bypassed if it's not needed. 2) One of the things that
+ I noticed when profiling this code was that a _lot_ of time was
+ spent doing string comparisons. I changed the way strings are
+ represented in an event to include a hash value at the front. So,
+ before doing a string comparison, we do an integer comparison on
+ the hash. 3) Finally, the code that handles the event cache has
+ been re-written. I tried to do this in a such a way that it had
+ minimal impact on the API. I did have to change one API call,
+ though - ast_event_queue_and_cache(). However, the way it works
+ now is nicer, IMO. Each type of event that can be cached (MWI,
+ device state) has its own hash table and rules for hashing and
+ comparing objects. This by far made the biggest impact on
+ performance. For additional details regarding this code and how
+ it was tested, please see the review request. (closes issue
+ #14738) Reported by: russell Review:
+ http://reviewboard.digium.com/r/205/ ........
+
+2009-03-25 19:27 +0000 [r184266-184283] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 |
+ file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix
+ issue with a T38 reinvite being sent even if not configured to do
+ so. If we receive a T38 request negotiate control frame we should
+ only attempt to do so if the option is enabled on the dialog.
+ ........
+
+ * main/bridging.c, /: Merged revisions 183652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183652 |
+ file | 2009-03-22 18:00:28 -0300 (Sun, 22 Mar 2009) | 4 lines Fix
+ a minor logic flaw with the bridge generic thread. We only want
+ to move the channel pointers that are actually present. ........
+
+2009-03-25 15:33 +0000 [r184256] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/asterisk.c, /: Merged revisions 184220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) |
+ 19 lines Merged revisions 184188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
+ 13 lines Avoid destroying the CLI line when moving the cursor
+ backward and trying to autocomplete. When moving the cursor
+ backward and pressing TAB to autocomplete, a NULL is put in the
+ line and we are loosing what we have already wrote after the
+ actual cursor position. (closes issue #14373) Reported by: eliel
+ Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
+ by: lmadsen ........ ................
+
+2009-03-25 14:40 +0000 [r184150-184221] Russell Bryant <russell@digium.com>
+
+ * main/timing.c, /: Merged revisions 184219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184219 |
+ russell | 2009-03-25 09:33:32 -0500 (Wed, 25 Mar 2009) | 2 lines
+ Include poll-compat.h ........
+
+ * main/timing.c, /: Merged revisions 184151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184151 |
+ russell | 2009-03-24 21:03:13 -0500 (Tue, 24 Mar 2009) | 2 lines
+ Change poll() to ast_poll(). ........
+
+ * utils/Makefile, /, include/asterisk/compat.h: Merged revisions
+ 184147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 |
+ russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines
+ Fix build issues on Mac OSX. (closes issue #14714) Reported by:
+ ygor ........
+
+2009-03-24 22:42 +0000 [r184082] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar
+ 2009) | 15 lines Merged revisions 184078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
+ 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
+ The 'digit' variable is guaranteed to be non-NULL, so the if
+ statement could never evaluate true. Changing to ast_strlen_zero
+ makes the logic correct. This was found while reviewing
+ ast_channel_ao2 code review. ........ ................
+
+2009-03-24 22:02 +0000 [r184041-184044] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 184043 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184043 |
+ russell | 2009-03-24 17:00:58 -0500 (Tue, 24 Mar 2009) | 2 lines
+ Put siren7 and siren14 in ast_best_codec() just so they're in
+ there somewhere. ........
+
+ * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009)
+ | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low
+ and =medium The default codec configuration for chan_iax2 is
+ bandwidth=low. I noticed slin16 being negotiated as the codec in
+ some test calls, but that no longer happens after this change.
+ ........
+
+2009-03-24 15:29 +0000 [r183868-183917] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 183914 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500
+ (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
+ | 3 lines Additionally note that the operator option needs an 'o'
+ extension. (Related to issue #14731) ........ ................
+
+ * /, main/http.c: Merged revisions 183865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 |
+ tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines
+ Allow browsers to cache images and other static content. (This is
+ a regression over 1.4) ........
+
+2009-03-23 19:00 +0000 [r183769] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar
+ 2009) | 13 lines Merged revisions 183700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
+ 2009) | 7 lines Fix a memory leak in res_monitor.c The only way
+ that this leak would occur is if Monitor were started using the
+ Manager interface and no File: header were given. Discovered
+ while reviewing the ast_channel_ao2 review request. ........
+ ................
+
+2009-03-23 18:12 +0000 [r183704] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009)
+ | 7 lines Fixes a documentation error introduced during the CLI
+ cleanup at AstriDevCon 2008. (closes issue #14655) Reported by:
+ ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728)
+ Tested by: lmadsen ........
+
+2009-03-20 17:09 +0000 [r183564] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183560 | russell | 2009-03-20 12:00:58 -0500
+ (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009)
+ | 2 lines Fix a crash in IAX2 registration handling found during
+ load testing with dvossel. ........ ................
+
+2009-03-20 12:19 +0000 [r183519] Eliel C. Sardanons <eliels@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183511 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) |
+ 2 lines Remove duplicate <description> inside the xml
+ documentation. ........
+
+2009-03-19 19:20 +0000 [r183337] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500
+ (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009)
+ | 8 lines Delay signalling progress until a PRI channel really
+ signals progress. (closes issue #13034) Reported by: klaus3000
+ Patches: 20090316__bug13034.diff.txt uploaded by tilghman
+ (license 14) patch_trunk_183progress_klaus3000.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000 ........
+ ................
+
+2009-03-19 18:20 +0000 [r183263] Russell Bryant <russell@digium.com>
+
+ * main/loader.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Merged revisions 183242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009)
+ | 10 lines Merged revisions 183241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009)
+ | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving
+ like expected. ........ ................
+
+2009-03-19 18:12 +0000 [r183247] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 183244 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 |
+ mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16
+ lines Fix a memory leak associated with queues. For every attempt
+ that app_queue made to place an outbound call to a queue member,
+ we would allocate a queue_end_bridge structure. When the bridge
+ for the call had completed, we would free the structure.
+ Unfortunately not all call attempts actually end up bridged to a
+ member, so we need to be more selective of when to allocate the
+ structure. With this change, the allocation occurs in an area
+ where we can guarantee that the call will be bridged. (closes
+ issue #14680) Reported by: caspy Patches: 14680.patch uploaded by
+ mmichelson (license 60) Tested by: caspy ........
+
+2009-03-19 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-beta1
+
+2009-03-19 16:11 +0000 [r183122] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar
+ 2009) | 20 lines Merged revisions 183115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
+ 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
+ would erroneously report the device as "in use." A user was
+ having an issue where if an outgoing SIP call was canceled, the
+ SIP device would remain in use if we had not received any
+ response to the initial INVITE we sent out. The SIP device would
+ remain in use until the autocongestion timer was exhausted. I
+ tracked down the cause of this to be the section of code I am
+ removing here. I asked several people what the purpose of this
+ code was meant to be, but no one could give me any sort of answer
+ as to why this was here. The person who was having this issue has
+ been using this patch for several months and it has stopped the
+ problems they have had. AST-196 ........ ................
+
+2009-03-19 15:45 +0000 [r183068-183111] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 |
+ file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
+ Improve our triggering of a T38 switchover internally when
+ triggered by a received reinvite. Previously we reached across
+ the channel bridge to get the other party's SIP dialog structure
+ in order to trigger an outgoing reinvite. This is extremely
+ dangerous to do and only works if bridged to another SIP channel.
+ This patch changes this to use the T38 control frame method of
+ requesting a switchover. This change also causes the SIP channel
+ driver to propogate back whether the switchover worked or not
+ instead of blindly accepting the incoming T38 reinvite. Review:
+ http://reviewboard.digium.com/r/200/ ........
+
+ * main/channel.c, /: Merged revisions 183057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 |
+ file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix
+ an issue where a T38 control frame would get dropped. If two
+ channels were bridged together using a generic bridge the T38
+ control frame would get passed up instead of being indicated on
+ the other channel. ........
+
+2009-03-18 21:19 +0000 [r183031] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18
+ Mar 2009) | 4 lines Add some code removed by mistake from commit
+ 182722 that works around a file descriptor leak in versions of
+ PWLib prior to 1.12.0. ........
+
+2009-03-18 14:39 +0000 [r182947] Russell Bryant <russell@digium.com>
+
+ * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c,
+ configure, apps/app_mp3.c, res/res_agi.c,
+ include/asterisk/poll-compat.h, channels/chan_alsa.c,
+ main/asterisk.c, apps/app_nbscat.c, /, main/Makefile,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/io.h, main/utils.c, include/asterisk/channel.h:
+ Merged revisions 182847 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009)
+ | 52 lines Merged revisions 182810 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
+ | 44 lines Fix cases where the internal poll() was not being used
+ when it needed to be. We have seen a number of problems caused by
+ poll() not working properly on Mac OSX. If you search around,
+ you'll find a number of references to using select() instead of
+ poll() to work around these issues. In Asterisk, we've had poll.c
+ which implements poll() using select() internally. However, we
+ were still getting reports of problems. vadim investigated a bit
+ and realized that at least on his system, even though we were
+ compiling in poll.o, the system poll() was still being used. So,
+ the primary purpose of this patch is to ensure that we're using
+ the internal poll() when we want it to be used. The changes are:
+ 1) Remove logic for when internal poll should be used from the
+ Makefile. Instead, put it in the configure script. The logic in
+ the configure script is the same as it was in the Makefile.
+ Ideally, we would have a functionality test for the problem, but
+ that's not actually possible, since we would have to be able to
+ run an application on the _target_ system to test poll()
+ behavior. 2) Always include poll.o in the build, but it will be
+ empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
+ throughout the source tree to ast_poll(). I feel that it is good
+ practice to give the API call a new name when we are changing its
+ behavior and not using the system version directly in all cases.
+ So, normally, ast_poll() is just redefined to poll(). On systems
+ where AST_POLL_COMPAT is defined, ast_poll() is redefined to
+ ast_internal_poll(). 4) Change poll() in main/poll.c to be
+ ast_internal_poll(). It's worth noting that any code that still
+ uses poll() directly will work fine (if they worked fine before).
+ So, for example, out of tree modules that are using poll() will
+ not stop working or anything. However, for modules to work
+ properly on Mac OSX, ast_poll() needs to be used. (closes issue
+ #13404) Reported by: agalbraith Tested by: russell, vadim
+ http://reviewboard.digium.com/r/198/ ........ ................
+
+2009-03-17 20:53 +0000 [r182725] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /,
+ channels/h323/ast_h323.cxx, configure,
+ autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h,
+ channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions
+ 182722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 |
+ jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
+ Allow H.323 Plus library to be used in addition to the OpenH323
+ library Chan_h323 can now be compiled against both the previously
+ supported versions of OpenH323 as well as the current H.323 Plus
+ (version 1.20.2). The configure script has been modified to look
+ in the default install location of h323 to hopefully help avoid
+ using the environment variables OPENH323DIR and PWLIBDIR. Also,
+ the CLI command "h323 show version" has been added which
+ indicates which version of h323 is in use. (closes issue #11261)
+ Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
+ uploaded by jthurman (license 614) ........
+
+2009-03-17 16:46 +0000 [r182592] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 182553 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 |
+ russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines
+ Tweak the handling of the frame list inside of ast_answer(). This
+ does not change any behavior, but moves the frames from the local
+ frame list back to the channel read queue using an O(n) algorithm
+ instead of O(n^2). ........
+
+2009-03-17 15:01 +0000 [r182528-182534] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, /: Merged revisions 182530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 |
+ kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2
+ lines correct logic flaw in ast_answer() changes in r182525
+ ........
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 182525 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 |
+ kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11
+ lines Improve behavior of ast_answer() to not lose incoming
+ frames ast_answer(), when supplied a delay before returning to
+ the caller, use ast_safe_sleep() to implement the delay.
+ Unfortunately during this time any incoming frames are discarded,
+ which is problematic for T.38 re-INVITES and other sorts of
+ channel operations. When a delay is not passed to ast_answer(),
+ it still delays for up to 500 milliseconds, waiting for media to
+ arrive. Again, though, it discards any control frames, or
+ non-voice media frames. This patch rectifies this situation, by
+ storing all incoming frames during the delay period on a list,
+ and then requeuing them onto the channel before returning to the
+ caller. http://reviewboard.digium.com/r/196/ ........
+
+2009-03-17 05:54 +0000 [r182453] Tilghman Lesher <tlesher@digium.com>
+
+ * main/db.c, /: Merged revisions 182450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009)
+ | 14 lines Merged revisions 182449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
+ | 7 lines Fix race in astdb The underlying db1 implementation
+ does not fully isolate the pages retrieved from astdb, so the
+ lock protecting accesses needs to be extended until the copy from
+ the shared memory structure is done. (closes issue #14682)
+ Reported by: makoto ........ ................
+
+2009-03-17 02:02 +0000 [r182409] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 182408 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009)
+ | 8 lines OPENR2 uses an incorrect string value if the extension
+ delimiter is not present. * Fixed OPENR2 using an incorrect
+ string value if the extension delimiter is not present in the
+ Dial() function. This was fixed for SS7 and PRI in trunk
+ -r172400. * Made OPENR2 stripmsd behavior the same as the SS7,
+ PRI, and others. * Removed trailing whitespace that appeared with
+ OPENR2. ........
+
+2009-03-16 20:51 +0000 [r182360-182361] Russell Bryant <russell@digium.com>
+
+ * /: svnmerge init
+
+ * / (added): Create a branch for 1.6.2
+
+2009-03-16 20:35 +0000 [r182355] Russell Bryant <russell@digium.com>
+
+ * CREDITS, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ CHANGES, makeopts.in: Add MFC/R2 support for chan_dahdi. This
+ commit introduces official support for R2 signaling in
+ chan_dahdi. The modifications to chan_dahdi, and the supporting
+ library, LibOpenR2, were both written by Moises Silva. Many users
+ are using this code, or a variant of it, in Asterisk 1.2, 1.4 and
+ 1.6 in Brazil, México and Argentina. An unknown number of users
+ (but at least 1) are using it in each of the following countries:
+ Colombia, Nepal, Thailand, Venezuela, Perú, and probably others.
+ To use this code, LibOpenR2 must be installed from
+ http://www.libopenr2.org/. Information about configuration can be
+ found in configs/chan_dahdi.conf.sample. The code committed is
+ the most up to date version, which was being maintained in
+ svn/asterisk/team/moy/mfcr2/. I would also like to include a
+ Thank You to the many others that tested this code beyond those
+ listed in this commit message. These are the names that I could
+ find in the mantis issue. (closes issue #12509) Reported by: moy
+ Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested
+ by: moy, korihor, viniciusfontes, Skarmeth, loloski,
+ asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare,
+ ecarruda, rtorresduque, PTorres, ychen Review:
+ http://reviewboard.digium.com/r/40/
+
+2009-03-16 17:49 +0000 [r182282] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 182281 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16
+ Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32
+ byte random padding at the beginning of an encrypted IAX2 frame
+ turns out to not be all that random at all. This patch calls
+ ast_random to fill the padding buffer with random data. The
+ padding is randomized at the beginning of every encrypted call
+ and for every encrypted retransmit frame. Review:
+ http://reviewboard.digium.com/r/193/ ........
+
+2009-03-16 17:33 +0000 [r182211-182278] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c: Fix an off-by-one error in the FILE() function,
+ and extend FILE()'s length parameter to work like variable
+ substitution. Previously, FILE() returned one less character than
+ specified, due to the terminating NULL. Both the offset and
+ length parameters now behave identically to the way variable
+ substitution offsets and lengths also work. (closes issue #14670)
+ Reported by: BMC
+
+ * channels/chan_local.c, /: Merged revisions 182208 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16
+ Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak
+ of a local pvt structure. (closes issue #14656) Reported by:
+ caspy Patches: 20090313__bug14656__2.diff.txt uploaded by
+ tilghman (license 14) Tested by: caspy ........
+
+2009-03-16 13:58 +0000 [r182171] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Fix a memory leak in the ast_answer /
+ __ast_answer API call. For a channel that is not yet answered
+ this API call will wait until a voice frame is received on the
+ channel before returning. It does this by waiting for frames on
+ the channel and reading them in. The frames read in were not
+ freed when they should have been.
+
+2009-03-13 21:26 +0000 [r182029-182121] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Change faulty comparison used when announcing
+ average hold minutes and seconds (closes issue #14227) Reported
+ by: caspy
+
+ * main/features.c: Remove ast_ prefix from functions which are not
+ public.
+
+ * /, main/features.c: Merged revisions 181990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar
+ 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and
+ peer when interpreting DTMF. Dynamic features defined in the
+ applicationmap section of features.conf allow one to specify
+ whether the caller, callee, or both have the ability to use the
+ feature. The documentation in the features.conf.sample file could
+ be interpreted to mean that one only needs to set the
+ DYNAMIC_FEATURES channel variable on the calling channel in order
+ to allow for the callee to be able to use the features which he
+ should have permission to use. However, the DYNAMIC_FEATURES
+ variable would only be read from the channel of the participant
+ that pressed the DTMF sequence to activate the feature. The
+ result of this was that the callee was unable to use dynamic
+ features unless the dialplan writer had taken measures to be sure
+ that the DYNAMIC_FEATURES variable was set on the callee's
+ channel. This commit changes the behavior of
+ ast_feature_interpret to concatenate the values of
+ DYNAMIC_FEATURES from both parties involved in the bridge. The
+ features themselves determine who has permission to use them, so
+ there is no reason to believe that one side of the bridge could
+ gain the ability to perform an action that they should not have
+ the ability to perform. Kevin Fleming pointed out on the
+ asterisk-users list that the typical way that this was worked
+ around in the past was by setting _DYNAMIC_FEATURES on the
+ calling channel so that the value would be inherited by the
+ called channel. While this works, the documentation alone is not
+ enough to figure out why this is necessary for the callee to be
+ able to use dynamic features. In this particular case, changing
+ the code to match the documentation is safe, easy, and will
+ generally make things easier for people for future installations.
+ This bug was originally reported on the asterisk-users list by
+ David Ruggles. (closes issue #14657) Reported by: mmichelson
+ Patches: 14657.patch uploaded by mmichelson (license 60) ........
+
+2009-03-13 17:25 +0000 [r182022] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix an issue with requesting a T38 reinvite
+ before the call is answered. The code responsible for sending the
+ T38 reinvite did not check if an INVITE was already being
+ handled. This caused things to get confused and the call to fail.
+ The code now defers sending the T38 reinvite until the current
+ INVITE is done being handled. (issue AST-191)
+
+2009-03-13 16:55 +0000 [r181985] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: improve a bit of suboptimal code
+
+2009-03-13 01:26 +0000 [r181899] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Merged revisions 181898 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Just
+ recording the v1.4 change in trunk since it originally came from
+ here. ........ r181898 | rmudgett | 2009-03-12 20:19:29 -0500
+ (Thu, 12 Mar 2009) | 4 lines Use the correct branch integrated
+ property when generating the version string. Copied the
+ make_version file from Asterisk trunk. ........
+
+2009-03-12 21:43 +0000 [r181769-181846] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Run the macro on the queue member's channel
+ when he answers, not the caller's channel.
+
+ * /, channels/chan_sip.c: Merged revisions 181768 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar
+ 2009) | 22 lines Properly send a 487 on an INVITE we have not
+ responded to if we receive a BYE. If we receive an INVITE from an
+ endpoint and then later receive a BYE from that same endpoint
+ before we have sent a final response for the INVITE, then we need
+ to respond to the INVITE with a 487. There was logic in the code
+ prior to this commit which seemed to exist solely to handle this
+ situation, but there was one condition in an if statement which
+ was incorrect. The only way we would send a 487 was if the
+ sip_pvt had no owner channel. This made no sense since we created
+ the owner channel when we received the INVITE, meaning that the
+ majority of the time we would never send the 487. The 487 being
+ sent should not rely on whether we have created a channel. Its
+ delivery should be dependent on the current state of the initial
+ INVITE transaction. With this commit, that logic is now correctly
+ in place. (closes issue #14149) Reported by: legranjl Patches:
+ 14149.patch uploaded by mmichelson (license 60) Tested by:
+ legranjl ........
+
+2009-03-12 17:32 +0000 [r181731] Tilghman Lesher <tlesher@digium.com>
+
+ * main/translate.c: Adjust translation table column widths based
+ upon the translation times. Previously, only 5 columns were
+ displayed, and if a translation time exceeded 99,999 useconds, it
+ would be displayed as 0, instead of its actual time. (closes
+ issue #14532) Reported by: pj Patches:
+ 20090311__bug14532.diff.txt uploaded by tilghman (license 14)
+ Tested by: pj
+
+2009-03-12 16:56 +0000 [r181612-181665] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 181664 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar
+ 2009) | 2 lines Fix incorrect usage of strncasecmp... I really
+ meant to use strcasecmp. ........
+
+ * /, res/res_musiconhold.c: Merged revisions 181659-181660 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8
+ lines Fix another scenario where depending on configuration the
+ stream would not get read. For custom commands we don't know
+ whether the audio is coming from a stream or not so we are going
+ to have to read the data despite no channels. (closes issue
+ #14416) Reported by: caspy ........ r181660 | file | 2009-03-12
+ 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in
+ previous commit. ........
+
+ * /, res/res_musiconhold.c: Merged revisions 181655 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar
+ 2009) | 10 lines Fix issue with streaming MOH failing if nobody
+ is listening. When a music class is setup to actually provide
+ music on hold from a stream we need to constantly read audio from
+ it since it will constantly be providing audio. This is now done
+ despite there being no channels listening to it. (closes issue
+ #14416) Reported by: caspy ........
+
+ * apps/app_dial.c: Fix crash when sleep and retries argument was
+ not given to RetryDial application. (closes issue #14647)
+ Reported by: sherpya
+
+2009-03-12 01:33 +0000 [r181542-181577] Richard Mudgett <rmudgett@digium.com>
+
+ * build_tools/make_version: Whitespace chages.
+
+ * build_tools/make_version: Use the correct branch integrated
+ property when generating the version string
+
+2009-03-11 23:14 +0000 [r181499] Michiel van Baak <michiel@vanbaak.info>
+
+ * configs/sip.conf.sample: Provide correct hint to debug SIP
+ trouble in the default config (closes issue #14646) Reported by:
+ strk
+
+2009-03-11 22:25 +0000 [r181465] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Make handling of the BRIDGE_PLAY_SOUND variable
+ thread-safe.
+
+2009-03-11 22:20 +0000 [r181444] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 181436 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar
+ 2009) | 4 lines Allow prefix to set localstatedir (when used and
+ different from the default). This is similar to the /etc change
+ that was made for the non-FreeBSD case. ........
+
+2009-03-11 22:14 +0000 [r181424-181428] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Make handling of the BRIDGEPVTCALLID variable
+ thread-safe.
+
+ * main/channel.c, /: Merged revisions 181423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009)
+ | 9 lines Make code that updates BRIDGEPEER variable thread-safe.
+ It is not safe to read the name field of an ast_channel without
+ the channel locked. This patch fixes some places in channel.c
+ where this was being done, and lead to crashes related to
+ masquerades. (closes issue #14623) Reported by: guillecabeza
+ ........
+
+2009-03-11 17:34 +0000 [r181371] David Vossel <dvossel@digium.com>
+
+ * channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions
+ 181340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009)
+ | 11 lines encrypted IAX2 during packet loss causes decryption to
+ fail on retransmitted frames If an iax channel is encrypted, and
+ a retransmit frame is sent, that packet's iseqno is updated while
+ it is encrypted. This causes the entire frame to be corrupted.
+ When the corrupted frame is sent, the other side decrypts it and
+ sends a VNAK back because the decrypted frame doesn't make any
+ sense. When we get the VNAK, we look through the sent queue and
+ send the same corrupted frame causing a loop. To fix this,
+ encrypted frames requiring retransmission are decrypted, updated,
+ then re-encrypted. Since key-rotation may change the key held by
+ the pvt struct, the keys used for encryption/decryption are held
+ within the iax_frame to guarantee they remain correct. (closes
+ issue #14607) Reported by: stevenla Tested by: dvossel Review:
+ http://reviewboard.digium.com/r/192/ ........
+
+2009-03-11 17:26 +0000 [r181345] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181328 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) |
+ 14 lines Fix issue where an attended transfer could not be
+ completed under a rare scenario. When completing an attended
+ transfer chan_sip does a check to make sure the extension in the
+ URI portion of the Refer-To header is a local valid extension. We
+ don't actually need to check this since we know for sure the
+ other channel is already up and talking to the extension. Some
+ devices do not put the extension in the Refer-To header either,
+ which can cause the extension check to fail. We now no longer do
+ this check if it is an attended transfer. (closes issue #14628)
+ Reported by: sverre Patches: 14628.diff uploaded by file (license
+ 11) ........
+
+2009-03-11 17:04 +0000 [r181301] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/astobj2.h: Turn off malloc debugging of astobj2,
+ since it apparently doesn't work too well during startup.
+
+2009-03-11 16:40 +0000 [r181296] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9
+ lines Fix a problem with inband DTMF detection on outgoing SIP
+ calls when dtmfmode=auto. When dtmfmode was set to auto the
+ inband DTMF detector was not setup on outgoing SIP calls. This
+ caused inband DTMF detection to fail. The inband DTMF detector is
+ now setup for both dtmfmode inband and auto. (closes issue
+ #13713) Reported by: makoto ........
+
+2009-03-11 16:19 +0000 [r181292] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: Replace contents of this doc with a
+ pointer to its new home
+
+2009-03-11 14:28 +0000 [r181244] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix segfault when dialing a typo'd queue If
+ trying to dial a non-existent queue, there would be a segfault
+ when attempting to access q->weight, even though q was NULL. This
+ problem was introduced during the queue-reset merge and thus only
+ affects trunk. (closes issue #14643) Reported by: alecdavis
+
+2009-03-11 13:44 +0000 [r181210] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_confbridge.c: Don't play the "you are about to be placed
+ into the conference" and "the leader has left the conference"
+ sounds if the quiet option is enabled. (reported by Vadim Lebedev
+ on the asterisk-dev list)
+
+2009-03-11 04:06 +0000 [r181135] Jeff Peeler <jpeeler@digium.com>
+
+ * utils/Makefile, include/asterisk/utils.h,
+ include/asterisk/astmm.h, channels/chan_sip.c,
+ channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c,
+ pbx/pbx_config.c: Fix malloc debug macros to work properly with
+ h323. The main problem here was that cstdlib was undefining free
+ thereby causing the proper debug macros to not be used.
+ ast_h323.cxx has been changed to call ast_free instead to avoid
+ the issue. A few other issues were addressed: - There were a few
+ instances of functions improperly passing ast_free instead of
+ ast_free_ptr. - Some clean up was done to avoid the debug macros
+ intentionally being redefined. (copied below from Kevin's commit,
+ appreciate the help) - disable astmm.h from doing anything when
+ STANDALONE is defined, which is used by the tools in the utils/
+ directory that use parts of Asterisk header files in hackish
+ ways; also ensure that utils/extconf.c and utils/conf2ael.c are
+ compiled with STANDALONE defined. (closes issue #13593) Reported
+ by: pj
+
+2009-03-11 02:25 +0000 [r181099] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: tabs to spaces
+
+2009-03-11 00:49 +0000 [r181032-181033] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add missing comment that quotes RFC 3891
+
+ * /, channels/chan_sip.c: Merged revisions 181029,181031 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar
+ 2009) | 9 lines Fix incorrect tag checking on transfers when
+ pedantic=yes is enabled. (closes issue #14611) Reported by:
+ klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt
+ uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
+ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar
+ 2009) | 3 lines Remove unused variables. ........
+
+2009-03-11 00:29 +0000 [r181027-181028] Tilghman Lesher <tlesher@digium.com>
+
+ * main/strings.c, main/hashtab.c, include/asterisk/astobj2.h,
+ main/heap.c, include/asterisk/strings.h,
+ include/asterisk/hashtab.h, main/astobj2.c,
+ include/asterisk/heap.h: Add MALLOC_DEBUG to various utility
+ APIs, so that memory leaks can be tracked back to their source.
+ (related to issue #14636)
+
+ * main/pbx.c: Spacing changes only
+
+2009-03-10 22:03 +0000 [r180944] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac, autoconf/ast_prog_sed.m4,
+ autoconf/ast_check_gnu_make.m4: Merged revisions 180941 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) |
+ 1 line Make things happier when using autoconf 2.62+ ........
+
+2009-03-10 22:03 +0000 [r180935-180942] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: Add some notes on getting in
+ contact with the dev community
+
+ * doc/google-soc2009-ideas.txt: Remove difficulty and language
+ specifiers
+
+ * doc/google-soc2009-ideas.txt: Expand upon documentation of
+ manager event project
+
+2009-03-10 21:15 +0000 [r180898] Michiel van Baak <michiel@vanbaak.info>
+
+ * CHANGES: list the move of the astvarrundir from /var/run to
+ /var/run/asterisk (actually its $(localstatedir)/run/asterisk
+ Makes setups with asterisk as non-root easier to manage because
+ you can setup permissions on this dir instead of touching a file
+ and setting permissions on that. Files that come to mind are
+ asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell
+
+2009-03-10 19:36 +0000 [r180859-180862] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: add more projects
+
+ * doc/google-soc2009-ideas.txt: add more project ideas
+
+2009-03-10 14:40 +0000 [r180800] Joshua Colp <jcolp@digium.com>
+
+ * main/manager.c: Reset the thread local string buffer when
+ handling the UserEvent action. (closes issue #14593) Reported by:
+ JimDickenson
+
+2009-03-09 22:00 +0000 [r180750] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: Add current mentors list, and first
+ pass on a project list broken out of "PineMango" I will work on
+ adding projects that have been sent to be via email tomorrow.
+
+2009-03-09 20:58 +0000 [r180719] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/rtp.h, include/asterisk/extconf.h,
+ main/devicestate.c, include/asterisk/tcptls.h, main/enum.c,
+ include/asterisk/callerid.h, include/asterisk/doxyref.h,
+ include/asterisk/event.h, include/asterisk/audiohook.h,
+ include/asterisk/dsp.h, include/asterisk/timing.h,
+ include/asterisk/udptl.h, include/asterisk/dlinkedlists.h,
+ include/asterisk/utils.h, include/asterisk/devicestate.h,
+ include/asterisk/taskprocessor.h, include/asterisk/enum.h,
+ include/asterisk/astobj2.h, include/asterisk/config.h,
+ include/asterisk/channel.h, include/asterisk/manager.h,
+ include/asterisk/heap.h, include/asterisk/logger.h,
+ include/asterisk/http.h, include/asterisk/res_odbc.h,
+ include/asterisk/app.h, main/tcptls.c,
+ include/asterisk/linkedlists.h, include/asterisk/sched.h,
+ include/asterisk/datastore.h, include/asterisk/lock.h,
+ include/asterisk/pbx.h, include/asterisk/dnsmgr.h: Add Doxygen
+ documentation for API changes from 1.6.0 to 1.6.1 Copied from my
+ review board description: This is a continuation of the API
+ changes documentation started for describing changes between
+ releases. Most of the API changes were pretty simple needing only
+ to be brought to attention via the new "Asterisk API Changes"
+ list. However, if you see anything that needs further explanation
+ feel free to supplement what is there. The current method of
+ documenting is to add (in the header file): \version <ver number>
+ <description of changes> and then to add the function to the
+ change list in doxyref.h on the AstAPIChanges page. I also made
+ sure all the functions that were newly added were tagged with
+ \since 1.6.1. I think this is a good habit to start both for the
+ historical aspect as well as for the future ability to easily add
+ a "New Asterisk API" page. Review:
+ http://reviewboard.digium.com/r/190/
+
+2009-03-09 14:14 +0000 [r180684] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt (added): Add skeleton for GSoC ideas
+ list
+
+2009-03-07 15:36 +0000 [r180641] Russell Bryant <russell@digium.com>
+
+ * contrib/asterisk-ng-doxygen: Make some minor updates to the
+ doxygen configuration - add bridges directory to be processed -
+ add some res/ subdirs - alphabetize subdirs - use consistent
+ indentation
+
+2009-03-06 18:25 +0000 [r180579] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 180567 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri,
+ 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when
+ IMAP storage is enabled. ........
+
+2009-03-06 17:26 +0000 [r180534] David Vossel <dvossel@digium.com>
+
+ * /, main/enum.c: Merged revisions 180532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009)
+ | 9 lines Fix handling of backreferences for ENUM lookups enum.c
+ did not handle regex backtraces correctly. The '\1' in the regex
+ is a backreference that requires a pattern match to be inserted.
+ The way the code used to work is that it would find the
+ backreference and insert the entire input string minus the '+'.
+ This is incorrect. The regexec() function takes in a variable
+ called pmatch which is an array of structs containing the start
+ and end indexes for each backreference substring. The original
+ code actually passed the pmatch array pointer into regexec but
+ never did anything with it. Now when a backtrace is found, the
+ backtrace number is looked up in the pmatch array and the correct
+ substring is inserted. (closes issue #14576) Reported by:
+ chris-mac Review: http://reviewboard.digium.com/r/187/ ........
+
+2009-03-05 23:26 +0000 [r180383-180465] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 180464 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu,
+ 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when
+ identical mailbox names were defined in separate contexts. There
+ was a fix put in a while back so that an X-Asterisk-VM-Context
+ message header was added to stored IMAP voicemails. This would
+ allow for us to differentiate if the same mailbox name was used
+ in multiple contexts. The problem still left was that not all
+ places where messages were retrieved actually attempted to use
+ this header for information when retrieving messages. This commit
+ fixes that so that MWI and message retrieval from VoiceMailMain
+ work as expected. (closes issue #13853) Reported by: vicks1
+ Patches: 13853_v2.patch uploaded by mmichelson (license 60)
+ Tested by: lmadsen ........
+
+ * /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged
+ revisions 180380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar
+ 2009) | 25 lines Fix broken mailbox parsing when searchcontexts
+ option is enabled. When using the searchcontexts option in
+ voicemail.conf, the code made the assumption that all mailbox
+ names defined were unique across all contexts. However, the code
+ did nothing to actually enforce this assumption, nor did it do
+ anything to alert a user that he may have created an ambiguity in
+ his voicemail.conf file by defining the same mailbox name in
+ multiple contexts. With this change, we now will issue a nice
+ long warning if searchcontexts is on and we encounter the same
+ mailbox name in multiple contexts and ignore any duplicates after
+ the first box. Whether searchcontexts is enabled or not, if we
+ come across a duplicate mailbox in the same context, then we will
+ issue a warning and ignore the duplicated mailbox. I have also
+ added a small note to voicemail.conf.sample in the explanation
+ for searchcontexts explaining that you cannot define the same
+ mailbox in multiple contexts if you have enabled the option.
+ (closes issue #14599) Reported by: lmadsen Patches: 14599.patch
+ uploaded by mmichelson (license 60) (with slight modification)
+ Tested by: lmadsen ........
+
+2009-03-05 19:05 +0000 [r180382] Michiel van Baak <michiel@vanbaak.info>
+
+ * Makefile: Make sure we terminate the first s| command so we can
+ actually produce correct files.
+
+2009-03-05 18:29 +0000 [r180373] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c, /, include/asterisk/frame.h, main/rtp.c: Merged
+ revisions 180372 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar
+ 2009) | 9 lines Fix problems when RTP packet frame size is
+ changed During some code analysis, I found that calling
+ ast_rtp_codec_setpref() on an ast_rtp session does not work as
+ expected; it does not adjust the smoother that may on the RTP
+ session, in fact it summarily drops it, even if it has data in
+ it, even if the current format's framing size has not changed.
+ This is not good. This patch changes this behavior, so that if
+ the packetization size for the current format changes, any
+ existing smoother is safely updated to use the new size, and if
+ no smoother was present, one is created. A new API call for
+ smoothers, ast_smoother_reconfigure(), was required to implement
+ these changes. Review: http://reviewboard.digium.com/r/184/
+ ........
+
+2009-03-05 18:18 +0000 [r180369] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_bridge.c (added), main/Makefile,
+ bridges/bridge_simple.c, bridges/bridge_softmix.c,
+ include/asterisk/channel.h, bridges/bridge_multiplexed.c,
+ CHANGES, Makefile, include/asterisk/bridging_technology.h
+ (added), bridges (added), bridges/bridge_builtin_features.c,
+ include/asterisk/bridging_features.h (added),
+ include/asterisk/bridging.h (added), apps/app_confbridge.c
+ (added), main/bridging.c (added), bridges/Makefile: Merge phase 1
+ support for the new bridging architecture. This commit brings in
+ the bridging core, bridging technologies, and the ConfBridge
+ application. For usage information on the ConfBridge application
+ please see the output of "core show application ConfBridge" from
+ the CLI. For API documentation please see the doxygen page
+ describing the architecture and the documentation for each API
+ call. Review: http://reviewboard.digium.com/r/93/
+
+2009-03-05 06:21 +0000 [r180304-180334] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/editors/asterisk.vim: Also highlight the preamble and
+ postamble
+
+ * contrib/editors/ael.vim (added), contrib/editors/asterisk.vim
+ (added), contrib/editors (added), contrib/editors/asteriskvm.vim
+ (added): Add syntax coloring files for Vim, including a new one
+ for AEL
+
+2009-03-04 21:01 +0000 [r180261] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Resolve object matching issues related to
+ the removal of the sip_user object. Previously, chan_sip had both
+ sip_peer and sip_user objects in memory. A patch went in to
+ remove sip_user to simplify the code, since everything could be
+ done with just sip_peer. This patch resolves some regressions
+ found that were introduced by those changes. This code comes from
+ svn/asterisk/team/group/sip-object-matching/. Here is a list of
+ the changes that have been made: 1) When doing a match by name
+ with the find_peer() function, make it much easier to specify
+ which objects should be matched by having a parameter that
+ specifies exactly which object types should be considered. Also,
+ update find_by_name() to handle this parameter. Finally, update
+ all code to use the new option values. 2) When looking up an
+ object for an outbound request by name, consider peers only.
+ (create_addr()) 3) Only match peers on an incoming registration
+ request. 4) When doing authentication (except for SUBSCRIBE),
+ look up users by name, instead of all objects by name. 5) When
+ doing authentication (except for SUBSCRIBE), after looking for a
+ user by name, look for a peer by IP address, instead of all
+ objects by IP address. 6) When handling the SIP qualify CLI
+ command or manager action, look for a peer by name, instead of
+ any object by name. 7) When handling the SIP unregister CLI
+ command, look for a peer by name, instead of any object by name.
+ 9) In sip_do_debug_peer(), search for a peer by name, instead of
+ any object by name. 9) When handling the SIPPEER() dialplan
+ function, search for a peer by name, instead of any object by
+ name. 10) In the following session timer related functions,
+ st_get_se(), st_get_refresher(), and st_get_mode(), when looking
+ for an object for a given sip_pvt using pvt->peername, look for a
+ peer by name, instead of any object by name. 11) Fix build_peer()
+ to properly handle the case where separate type=peer and
+ type=user entries were specified in sip.conf. (closes issue
+ #14505) Reported by: lmadsen Review:
+ http://reviewboard.digium.com/r/172/
+
+2009-03-04 20:48 +0000 [r180259] Tilghman Lesher <tlesher@digium.com>
+
+ * main/aescrypt.c, main/abstract_jb.c, main/acl.c, main/app.c,
+ main/alaw.c: Spacing changes only
+
+2009-03-04 19:24 +0000 [r180195] Joshua Colp <jcolp@digium.com>
+
+ * /, main/callerid.c: Merged revisions 180194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4
+ lines Look for the number in a callerid string starting from the
+ end. This way a value using <> can exist in the name portion.
+ (issue #AST-194) ........
+
+2009-03-04 17:03 +0000 [r180155] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Allow for "magic"
+ pickups to work when we wish to ignore the context When the
+ subscription context for a call pickup subscription differs from
+ the context of the call pickup target, there's not an easy way to
+ divine what context should be used for the pickup. The way to
+ work around this is to use PICKUPMARK as the context for the
+ pickup. This has been documented in the sip.conf.sample file
+ (ABE-1708) closes issue #14567 submitted by: alecdavis
+
+2009-03-04 14:39 +0000 [r180120] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Remove duplicate 'k' and 'K' Dial options.
+ (closes issue #14601) Reported by: alecdavis Patches:
+ app_dial.optionk.diff.txt uploaded by alecdavis (license 585)
+
+2009-03-03 23:35 +0000 [r180079] Steve Murphy <murf@digium.com>
+
+ * utils/Makefile: My bad! left check_expr2 in the ALL_UTILS list by
+ mistake. Already done to 1.6.x
+
+2009-03-03 23:21 +0000 [r180032] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, include/asterisk/app.h, apps/app_read.c,
+ main/app.c: app_read does not break from prompt loop with user
+ terminated empty string In app.c, ast_app_getdata is called to
+ stream the prompts and receive DTMF input. If ast_app_getdata()
+ receives an empty string caused by the user inputing the end of
+ string character, in this case '#', it should break from the
+ prompt loop and return to app_read, but instead it cycles through
+ all the prompts. I've added a return value for this special case
+ in ast_readstring() which uses an enum I've delcared in apps.h.
+ This enum is now used as a return value for ast_app_getdata().
+ (closes issue #14279) Reported by: Marquis Patches:
+ fix_app_read.patch uploaded by Marquis (license 32)
+ read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested
+ by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/
+
+2009-03-03 22:49 +0000 [r180007] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions
+ 180006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar
+ 2009) | 17 lines Clarify some documentation of queues.conf.sample
+ It had always been possible to explicitly specify a "blank" value
+ for a sound file in queues.conf and have no sound played back.
+ The problem with this is that it would result in some ugly CLI
+ warnings from file.c. This commit introduces a check when playing
+ a file in app_queue to see if the name of the file is zero-length
+ and return early if that is the case. Also, the ability to
+ specify the blank sound files in queues.conf is now mentioned
+ more clearly in queues.conf.sample (closes issue #14227) Reported
+ by: caspy ........
+
+2009-03-03 22:12 +0000 [r179973] Steve Murphy <murf@digium.com>
+
+ * utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h,
+ main/ast_expr2.y, main/ast_expr2f.c, main/ast_expr2.fl,
+ main/ast_expr2.c: Merged revisions 179807 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some
+ work to do to port these changes to trunk; the check_expr stuff
+ hasn't been updated here for quite some time, it appears. I added
+ some more tests to the check_expr2 suite. I had to play around
+ with the makefile a bit, etc. I added STANDALONE2 #ifdefs to
+ ast_expr2.y so as not to conflict structure with aelparse.
+ ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar
+ 2009) | 19 lines These changes allow AEL to better check ${}
+ constructs within $[...], that are concatenated with text. I
+ modified and added rules in ast_expr2.fl to better handle the
+ concatenations. I added some default routines to ast_expr2.y so
+ the standalone would compile. It also looks like I haven't run
+ this thru bison since 2.1, so it's good to get this updated. The
+ Makefile has comments added now for check_expr2 and check_expr to
+ explain what they are for, and how to run them. The testexpr2s
+ stuff has been removed, in favor of check_expr2. expr2.testinput
+ has been updated to include the two expressions that inspired
+ these changes (from mcnobody on #asterisk this morning) The
+ regression has been run and all looks well. ........
+
+2009-03-03 22:01 +0000 [r179972] David Vossel <dvossel@digium.com>
+
+ * apps/app_meetme.c: app_meetme not setting filename and fileformat
+ correctly for realtime When app_meetme finds a realtime
+ conference, it doesn't get the filename and fileformat correctly
+ when 'r' is set. Now app_meetme first checks to see if fileformat
+ and filename are declared in the db, if they're not it checks the
+ .conf file, if its not declared there either it then uses
+ defaults. (closes issue #14545) Reported by: dalbaech Patches:
+ app_meetme-realtime5.patch uploaded by dvossel (license 671)
+ Realtime_Conference_Record_workaround.txt uploaded by dalbaech
+ (license 705) Tested by: dvossel, dalbaech Review:
+ http://reviewboard.digium.com/r/180/
+
+2009-03-03 20:59 +0000 [r179937] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_timing_dahdi.c, doc/timing.txt (added): Add documentation
+ for timing modules used in Asterisk This document specifies the
+ timing modules available in Asterisk beginning with Asterisk
+ 1.6.1. The document goes into detail about the differences
+ between each and gives a general overview of what timing is used
+ for in Asterisk. There is also a section which can be used to
+ help customize your setup or to troubleshoot timing issues you
+ may have. I also added messages to the DAHDI timing test used in
+ res_timing_dahdi.c that points to this new documentation if
+ people experience problems. Big thanks to all who contributed
+ comments on this. (closes issue #14490) Reported by: mmichelson
+ Patches: timing.txt uploaded by mmichelson (license 60) Review:
+ http://reviewboard.digium.com/r/164/
+
+2009-03-03 20:02 +0000 [r179903] Brian Degenhardt <bmd@digium.com>
+
+ * apps/app_directed_pickup.c: fix a leaked channel lock (and future
+ deadlock) when we try to pick up our own channel
+
+2009-03-03 18:28 +0000 [r179841] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 179840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9
+ lines Do not assume that the bridge_cdr is still attached to the
+ channel when the 'h' exten is finished executing. It is possible
+ for a masquerade operation to occur when the 'h' exten is
+ operating. This operation moves the CDR records around causing
+ the bridge_cdr to no longer exist on the channel where it is
+ expected to. We can not safely modify it afterwards because of
+ this, so don't even try. (closes issue #14564) Reported by: meric
+ ........
+
+2009-03-03 17:03 +0000 [r179745] Mark Michelson <mmichelson@digium.com>
+
+ * pbx/pbx_spool.c: Convert pbx_spool to use string fields instead
+ of statically-sized buffers. In tests run after making this
+ conversion, I noticed an approximate 85% reduction in memory
+ usage for call file processing. Review:
+ http://reviewboard.digium.com/r/168/
+
+2009-03-03 16:47 +0000 [r179742] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 179741 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009)
+ | 6 lines Ensure chan->fdno always gets reset to -1 after
+ handling a channel fd event. Since setting fdno to -1 had to be
+ moved, a couple of other code paths that do process an fd event
+ return early and do not pass through the code path where it was
+ moved to. So, set it to -1 in a few other places, too. ........
+
+2009-03-03 15:13 +0000 [r179675] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Please prefix default values with DEFAULT
+
+2009-03-03 14:40 +0000 [r179672] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 179671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3
+ lines Move where fdno is set to the default value to *after* the
+ read callback of the channel driver is called. We have to do this
+ as the underlying channel driver may need the fdno value to
+ determine what to read. ........
+
+2009-03-03 13:54 +0000 [r179609] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 179608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009)
+ | 9 lines Make it easier to detect an improper call to
+ ast_read(). When you call ast_waitfor() on a channel, the index
+ into the channel fds array that holds the file descriptor that
+ poll() determines has input available is stored in fdno. This
+ patch clears out this value after a call to ast_read() and also
+ reports errors if ast_read() is called without an fdno set. From
+ a discussion on the asterisk-dev list. ........
+
+2009-03-03 00:01 +0000 [r179537] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 179536 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009)
+ | 15 lines Fix bridging regression from commit 176701 This fixes
+ a bad regression where the bridge would exit after an attended
+ transfer was made. The problem was due to nexteventts getting set
+ after the masquerade which caused the bridge to return
+ AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
+ tim_ringenbach ........
+
+2009-03-02 23:36 +0000 [r179533] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 179532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009)
+ | 40 lines Move ast_waitfor() down to avoid the results of the
+ API call becoming stale. This call to ast_waitfor() was being
+ done way too soon in this section of code. Specifically, there
+ was code in between the call to waitfor and the code that uses
+ the result that puts the channel in autoservice. By putting the
+ channel in autoservice, the previous results of ast_waitfor()
+ become meaningless, as the autoservice thread will do it's own
+ ast_waitfor() and ast_read() on the channel. So, when we came
+ back out of autoservice and eventually hit the block of code that
+ calls ast_read() on the channel, there may not actually be any
+ input on the channel available. Even though the previous call to
+ ast_waitfor() in app_meetme said there was input, the autoservice
+ thread has since serviced the channel for some period of time.
+ This bug manifested itself while dvossel was doing some testing
+ of MeetMe in Asterisk trunk. He was using the timerfd timing
+ module. When the code hit ast_read() erroneously, it determined
+ that it must have been called because of input on the timer fd,
+ as chan->fdno was set to AST_TIMING_FD, since that was the cause
+ of the last legitimate call to ast_read() done by autoservice. In
+ this test, an IAX2 channel was calling into the MeetMe
+ conference. It was _much_ more likely to be seen with an IAX2
+ channel because of the way audio is handled. Every audio frame
+ that comes in results in a call to ast_queue_frame(), which then
+ uses ast_timer_enable_continuous() to notify the channel thread
+ that a frame is waiting to be handled. So, the chances of
+ ast_waitfor() indicating that a channel needs servicing due to a
+ timer event on an IAX2 event is very high. Finally, it is
+ interesting to note that if a different timing interface was
+ being used, this bug would probably not be noticed. When
+ ast_read() is called and erroneously thinks that there is a timer
+ event to handle, it calls the ast_timer_ack() function. The
+ pthread and dahdi timing modules handle the ack() function being
+ called when there is no event by simply ignoring it. In the case
+ of the timerfd module, it results in a read() on the timer fd
+ that will block forever, as there is no data to read. This caused
+ Asterisk to lock up very quickly. Thanks to dvossel and
+ mmichelson for the fun debugging session. :-) ........
+
+2009-03-02 23:10 +0000 [r179469] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/app.c: Merged revisions 179468 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009)
+ | 10 lines When ending a recording with silence detection,
+ remember to reduce the duration. The end of the recording is
+ correspondingly trimmed, but the duration was not trimmed by the
+ number of seconds trimmed, so the saved duration was necessarily
+ longer than the actual soundfile duration. (closes issue #14406)
+ Reported by: sasargen Patches: 20090226__bug14406.diff.txt
+ uploaded by tilghman (license 14) Tested by: sasargen ........
+
+2009-03-02 23:06 +0000 [r179462-179465] Russell Bryant <russell@digium.com>
+
+ * res/res_timing_timerfd.c: Fix a reference leak in
+ timerfd_set_rate(). (found during a debugging session with
+ dvossel and mmichelson.)
+
+ * main/channel.c, /: Merged revisions 179461 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009)
+ | 8 lines Ensure that only one thread is calling ast_settimeout()
+ on a channel at a time. For example, with an IAX2 channel, you
+ can have both the channel thread and the chan_iax2 processing
+ threads calling this function, and doing so twice at the same
+ time is a bad thing. (Found in a debugging session with dvossel
+ and mmichelson) ........
+
+2009-03-02 20:16 +0000 [r179396] Jason Parker <jparker@digium.com>
+
+ * /, main/editline/configure, main/editline/np/unvis.c,
+ main/editline/sys.h, main/editline/configure.in: Merged revisions
+ 179395 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) |
+ 1 line Remove several silly warnings in editline. One about a
+ broken preprocessor directive, and another about strlcpy/strlcat.
+ (closes issue #14264) Reported by: dimas ........
+
+2009-03-02 17:18 +0000 [r179361] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: Backport 1.6.0 fix to trunk (failsafe
+ if db is not loaded)
+
+2009-03-02 14:28 +0000 [r179291-179323] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Do not try to remove a registration
+ scheduled item if the scheduler context has already been
+ destroyed. (closes issue #14580) Reported by: alecdavis
+
+ * main/audiohook.c: Fix issue where changing the volume of both
+ directions of audio did not work. (closes issue #14574) Reported
+ by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK
+ (license 545)
+
+2009-03-01 23:25 +0000 [r179219-179254] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_speech_utils.c: Swap reversed timevals. This was pointed
+ out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
+
+ * channels/chan_sip.c: Properly free memory and remove scheduler
+ entries when a transmission failure occurs. Previously, only the
+ "data" field of the sip_pkt created during __sip_reliable_xmit
+ was freed when XMIT_ERROR was returned by __sip_xmit. When
+ retrans_pkt was called, this inevitably resulted in the reading
+ and writing of freed memory. XMIT_ERROR is a condition meaning
+ that we don't want to attempt resending the packet at all. The
+ proper action to take is to remove the scheduler entry we just
+ created, free the packet's data as well as the packet itself, and
+ unlink it from the list of packets on the sip_pvt structure.
+ (closes issue #14455) Reported by: Nick_Lewis Patches:
+ 14455.patch uploaded by mmichelson (license 60) Tested by:
+ Nick_Lewis
+
+2009-02-27 21:47 +0000 [r179164] Russell Bryant <russell@digium.com>
+
+ * res/res_ais.c, doc/distributed_devstate.txt,
+ configs/ais.conf.sample: Mark res_ais as experimental, as the
+ binary event format is subject to change.
+
+2009-02-27 21:32 +0000 [r179161] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: If config file is blank, don't load
+ module. (Closes issue #14563)
+
+2009-02-27 21:23 +0000 [r179154] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt: Add a note about the ordering of entries in sip.conf
+ in 1.6.1.
+
+2009-02-27 20:34 +0000 [r179122] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Add reload support to chan_skinny.
+ Special thanks goes to DEA who had to redo this patch twice
+ because we first put unload/load support in and later redid the
+ way we configure devices and lines. (closes issue #10297)
+ Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by
+ wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA
+ (license 3) With mods by me based on feedback from wedhorn and
+ Russell and seanbright Tested by: DEA, mvanbaak, pj Review:
+ http://reviewboard.digium.com/r/130/
+
+2009-02-27 19:04 +0000 [r179057] Jason Parker <jparker@digium.com>
+
+ * doc/tex/channelvariables.tex: Update documentation for DIALEDTIME
+ and ANSWEREDTIME variables. (closes issue #14566) Reported by:
+ klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
+ klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
+ klaus3000 (license 65)
+
+2009-02-27 15:51 +0000 [r179021] Russell Bryant <russell@digium.com>
+
+ * sounds/Makefile: Fix downloading SIREN7 and SIREN14 sound
+ packages. In passing, also fix downloading SLIN16 extra sound
+ packages. (closes issue #14565) Reported by: jtodd
+
+2009-02-27 03:45 +0000 [r178986] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c, configs/features.conf.sample: Merged
+ revisions 178956 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 In this
+ case, it's just a matter of reducing the default timeouts from
+ 2000 to 1000 msec, as the max def feature digit timeout is no
+ longer halved. ........ r178956 | murf | 2009-02-26 14:27:32
+ -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default
+ feature digit timeout to 1000 ms from the previous default of
+ 500. As per bug 14515, a dev discussion arrived at a "mediated
+ concensus" of a default feature digit timeout of 1.0 sec. Some
+ voted for 1300; ctooley thought 1500 for distracted phone users
+ in phone booths; kpfleming put his foot down at 1.0 sec. Users
+ who found the previous default max delay of 250 msec perfect, are
+ welcome to override the new default. Notice that I said that 250
+ msec was the default; wait a minute, you might say, the config
+ file said it was 500 msec!; well, because of the bug fix for
+ 14515, we found that 500 msec was actually enforcing a max of
+ 250. The bug fix would restore 500 msec, but we felt even that
+ was a bit tight for most users... 2000 msec was pushed earlier by
+ mmichelson, so that reduces to 1000 msec after the bug fix.
+ Enjoy! ........
+
+2009-02-26 18:41 +0000 [r178919] Tilghman Lesher <tlesher@digium.com>
+
+ * main/features.c, CHANGES, configs/features.conf.sample: Sound
+ confirmation of call pickup success. (closes issue #13826)
+ Reported by: azielke Patches: pickupsound2-trunk.patch uploaded
+ by azielke (license 548) __20081124_bug_13826_updated.patch
+ uploaded by lmadsen (license 10) Tested by: lmadsen
+
+2009-02-26 17:46 +0000 [r178871] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: IAX2 prune realtime, minor tweak to last
+ fix A return statement was missing which caused unexpected cli
+ output. issue #14479
+
+2009-02-26 17:45 +0000 [r178828-178870] Steve Murphy <murf@digium.com>
+
+ * apps/app_osplookup.c, apps/app_rpt.c: These small fixes prevent
+ compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to
+ break my dev-mode build. Not a problem in 1.6.x.
+
+ * /, main/features.c: Merged revisions 178804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) |
+ 28 lines This patch prevents the feature detection timeout from
+ being cut in half. Because the ast_channel_bridge() call will
+ return 0 and pass a frame pointer for both DTMF_BEGIN and
+ DTMF_END, the feature_timer field in hte config struct is getting
+ decremented twice, which effectively cuts the digittimeout in
+ half. I added conditions to the if statement to only let DTMF_END
+ frames to flow thru, which solved the problem. Also, when the
+ frame pointer is null, let control flow thru-- this usually
+ happens on timeouts. I added a comment to the code to explain
+ what's going on and why. Many thanks to sodom for reporting this
+ problem. Personnally, it always seemed like something was wrong
+ with the featuredigittimeout, but I never could quite decide
+ what... and was too busy to investigate. This bug forced the
+ issue, and now we know. Sodom had other issues in 14515, but I
+ couldn't reproduce them. If he still has problems, and wants to
+ get them solved, he is welcome to reopen 14515. (closes issue
+ #14515) Reported by: sodom Patches: 14515.patch uploaded by murf
+ (license 17) Tested by: murf, sodom ........
+
+2009-02-26 16:42 +0000 [r178801] Joshua Colp <jcolp@digium.com>
+
+ * main/file.c: Fix an issue where the timer for file playback would
+ not be stopped if DAHDI was not installed. (closes issue #14541)
+ Reported by: grant
+
+2009-02-26 15:50 +0000 [r178767] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: IAX2 prune realtime fix Iax2 prune realtime
+ had issues. If "iax2 prune realtime all" was called, it would
+ appear like the command was successful, but in reality nothing
+ happened. This is because the reload that was supposed to take
+ place checks the config files, sees no changes, and does nothing.
+ If there had been a change in the the config file, the realtime
+ users would have been marked for deletion and everything would
+ have been fine. Now prune_users() and prune_peers() are called
+ instead of reload_config() to prune all users/peers that are
+ realtime. These functions remove all users/peers with the
+ rtfriend and delme flags set. iax2_prune_realtime() also lacked
+ the code to properly delete a single friend. For example. if iax2
+ prune realtime <friend> was called, only the peer instance would
+ be removed. The user would still remain. (closes issue #14479)
+ Reported by: mousepad99 Review:
+ http://reviewboard.digium.com/r/176/
+
+2009-02-26 15:40 +0000 [r178764] Joshua Colp <jcolp@digium.com>
+
+ * main/indications.c: Ensure there is a valid tone part before
+ trying to play tones. (closes issue #14558) Reported by:
+ alecdavis
+
+2009-02-26 15:02 +0000 [r178733] Olle Johansson <oej@edvina.net>
+
+ * configs/res_snmp.conf.sample: Clarifications on the different
+ models and reference to further docs.
+
+2009-02-26 13:39 +0000 [r178703-178704] Kevin P. Fleming <kpfleming@digium.com>
+
+ * README: another minor commit to test post-commit script changes
+ (now testing post-revprop-change as well, third try)
+
+ * README: minor commit to test post-commit script changes
+
+2009-02-25 19:49 +0000 [r178573-178607] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c: Picky, picky buildbots
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/stdtime/localtime.c: Use notification when timezone files
+ change and re-scan then. (closes issue #14300) Reported by:
+ jamessan Patches: 20090127__bug14300.diff.txt uploaded by
+ tilghman (license 14) 20090224__bug14300.diff uploaded by
+ jamessan (license 246) Tested by: jamessan Review:
+ http://reviewboard.digium.com/r/136/
+
+ * res/res_odbc.c: Oops, wrong direction of command
+
+2009-02-25 12:45 +0000 [r178509] Russell Bryant <russell@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 178508 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009)
+ | 2 lines Update the copyright year for the main page of the
+ doxygen documentation. ........
+
+2009-02-24 23:27 +0000 [r178375-178446] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/extensions.conf.sample: Merged revisions 178445 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009)
+ | 5 lines Add section about the #exec command in configuration
+ files. (closes issue #14540) Reported by: jtodd Patch by: jtodd,
+ with additional notes by tilghman (license 14) ........
+
+ * main/asterisk.c: Apparently, a void cast doesn't override
+ warn_unused_result.
+
+ * main/asterisk.c: The 3 possible errors with pipe(2) are all
+ impossible in this situation.
+
+2009-02-24 20:39 +0000 [r178374] Russell Bryant <russell@digium.com>
+
+ * /, main/rtp.c: Merged revisions 178373 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009)
+ | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset
+ to 0 properly. (issue #14460) Reported by: moliveras Tested by:
+ russell ........
+
+2009-02-24 20:06 +0000 [r178303-178342] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/astcanary.c, main/asterisk.c: Use a SIGPIPE to kill the
+ process, instead of depending upon the astcanary process being
+ inherited by init.
+
+ * utils/astcanary.c: Cause astcanary to exit if Asterisk exits
+ abnormally and doesn't kill astcanary. Also, add some
+ documentation supporting the use of astcanary. (closes issue
+ #14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff
+ uploaded by KNK (license 545)
+
+2009-02-24 17:42 +0000 [r178300] David Vossel <dvossel@digium.com>
+
+ * doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Allows
+ manager command to see if IAX link is trunked and encrypted.
+ Displays what kind of encryption is enabled as well. Manager
+ command "iaxpeers" now shows if a link is trunked and encrypted.
+ Instead of encryption saying simply "yes" or "no", it now
+ displays what type of encryption is enabled and if keyrotation is
+ on or not. (closes issue #14427) Reported by: snuffy Patches:
+ iax_show_trunks.diff uploaded by snuffy (license 35)
+ 2009022200_iax2_show_trunkencryption.diff.txt uploaded by
+ mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review:
+ http://reviewboard.digium.com/r/173/
+
+2009-02-24 15:18 +0000 [r178213] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 178205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9
+ lines Skip check for extension when subscribing for MWI. Since
+ the remote side is not actually subscribing to a specific
+ extension when subscribing for MWI just skip the check to see if
+ the extension exists. They can't use it to specify the mailbox
+ either since we require configuration of that in sip.conf (closes
+ issue #14531) Reported by: festr ........
+
+2009-02-23 23:11 +0000 [r178142] Russell Bryant <russell@digium.com>
+
+ * /, main/rtp.c: Merged revisions 178141 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009)
+ | 14 lines Fix infinite DTMF when a BEGIN is received without an
+ END. This commit is related to rev 175124 of 1.4 where a previous
+ attempt was made to fix this problem. The problem with the
+ previous patch was that the inserted code needed to go _before_
+ setting the lastrxts to the current timestamp. Because those were
+ the same, the dtmfcount variable was never decremented, and so
+ the END was never sent. In passing, I removed the dtmfsamples
+ variable which was completed unused. I also removed a redundant
+ setting of the lastrxts variable. (closes issue #14460) Reported
+ by: moliveras ........
+
+2009-02-23 21:02 +0000 [r178107] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Permit emailsubject and emailbody to be set per mailbox. (closes
+ issue #14372) Reported by: fhackenberger Patches:
+ voicemail_individual_subject_and_body_1.6.1 uploaded by
+ fhackenberger (license 592) with additional fixes by Corydon76
+ (license 14)
+
+2009-02-23 18:23 +0000 [r178061] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: update the new manager commands in
+ chan_skinny to match chan_sip's headers. requested by oej.
+
+2009-02-23 17:59 +0000 [r178030] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Changes the way keyrotation is enabled by
+ default Key rotation was enabled by default by setting the global
+ encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this
+ is that if encryption is not enabled, and the encryption method
+ is set to anything except 0, the peer appears to have encryption
+ enabled when issuing a "iax2 show peers". Rather than have the
+ key rotation bit always set by default, it is now only set when
+ an encryption method is enabled. (closes issue #14523) Reported
+ by: mvanbaak
+
+2009-02-23 17:48 +0000 [r178027] Michiel van Baak <michiel@vanbaak.info>
+
+ * CHANGES: list the addition of the SKINNY manager actions in the
+ CHANGES file.
+
+2009-02-23 17:29 +0000 [r178022] Russell Bryant <russell@digium.com>
+
+ * tests/test_sched.c, main/sched.c: Fix a regression in scheduler
+ entry ordering, and add a regression test for it. (closes issue
+ #14522) Reported by: pj Tested by: russell
+
+2009-02-22 23:04 +0000 [r177988] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/manager_1_1.txt, channels/chan_skinny.c: Add a couple of
+ manager commands to chan_skinny Added: SKINNYdevices
+ SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521)
+ Reported by: mvanbaak Review:
+ http://reviewboard.digium.com/r/170/
+
+2009-02-21 15:59 +0000 [r177944] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: On update, test against the existence of
+ sipregs.
+
+2009-02-21 14:37 +0000 [r177913] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: add extra check for sysinfo/sysctl (closes issue
+ #14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff
+ uploaded by snuffy (license 35)
+
+2009-02-21 14:16 +0000 [r177884] Sean Bright <sean.bright@gmail.com>
+
+ * main/hashtab.c, include/asterisk/hashtab.h: Trailing whitespace,
+ minor coding guideline fixes, and start beefing up the hashtab
+ documentation a bit.
+
+2009-02-21 13:17 +0000 [r177855] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/indications.h: Fix build issues on Solaris and
+ OpenBSD. (closes issue #14512) Reported by: snuffy
+
+2009-02-21 13:13 +0000 [r177849-177852] Michiel van Baak <michiel@vanbaak.info>
+
+ * Makefile, contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.archlinux.asterisk,
+ contrib/scripts/safe_asterisk: set
+ ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path When
+ running asterisk as non-root and without this patch the pidfile
+ wants to go into /var/run/asterisk.pid. This directory is not
+ writable for the non-root user and changing permissions is not an
+ option. Putting it in /var/run/asterisk/asterisk.pid makes it
+ possible to set permissions on the /var/run/asterisk dir so
+ everything works as it should be. Patched committed is based on
+ pabelanger's patch. (closes issue #13153) Reported by: pabelanger
+ Patches: 2009012900_bug13153-nonrootscripts.diff.txt uploaded by
+ mvanbaak (license 7) Review: http://reviewboard.digium.com/r/139/
+
+ * channels/chan_sip.c: make chan_sip.c compile on OpenBSD again.
+
+2009-02-20 23:02 +0000 [r177732-177787] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 177786 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009)
+ | 9 lines Don't print the CR-NL combination when we aren't
+ outputting to the manager. An embedded CR-NL in a CLI command
+ screws up several AMI parsers that don't expect to see that
+ combination in the middle of output. (Closes issue #14305)
+ Reported by: martins Patch by: tilghman ........
+
+ * /, include/asterisk/threadstorage.h: Merged revisions 177701 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009)
+ | 3 lines This exception does not appear to still be true for
+ Solaris 10, and OpenSolaris definitely needs it to be removed.
+ Fixed for snuff-home on -dev channel. ........
+
+2009-02-20 20:29 +0000 [r177699] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * apps/app_fax.c: Make app_fax compatible with spandsp-0.0.6pre4
+ Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a
+ pages_transferred integer to indicate the number of pages
+ transferred (so far) during the fax session. The
+ spandsp-0.0.6pre4 release removed the pages_transferred integer
+ and replaced it with two different integers - pages_tx and
+ pages_rx. This revision uses the new integers for
+ spandsp-0.0.6pre4 while maintaining backwards compatibility for
+ previous spandsp releases.
+
+2009-02-20 17:29 +0000 [r177661-177664] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, main/app.c, apps/app_system.c: Allow
+ semicolons to be escaped, when passing arguments to the System
+ command. (closes issue #14231) Reported by: jcovert Patches:
+ 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14)
+ corrected_20090113__bug14231__2.diff.txt uploaded by jcovert
+ (license 551) Tested by: jcovert
+
+ * apps/app_voicemail.c: Oops, merge broke trunk
+
+2009-02-20 00:35 +0000 [r177624] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Set sip_request ast_str data to NULL so
+ ast_str_copy allocates space properly in copy_request (issue
+ #14478) Reported by: erik_dedecker
+
+2009-02-19 23:56 +0000 [r177595] Steve Murphy <murf@digium.com>
+
+ * /, main/Makefile, main/ast_expr2f.c: Merged revisions 177540 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was
+ already pretty 8-bit clean; but I'm still removing the --full
+ from the flex command so everything is uniform. ........ r177540
+ | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
+ This patch fixes a problem with 8-bit input to the ast_expr2
+ scanner. The real culprit was the --full argument to flex in the
+ Makefile! This causes a 7-bit scanner to be generated. I reviewed
+ the rules and found one rule where I needed to specifically
+ include 8-bit chars for a token. I tested against the text
+ supplied by ibercom, and all looks very well. This has been there
+ a surprisingly long time! (closes issue #14498) Reported by:
+ ibercom Patches: 14498.patch uploaded by murf (license 17) Tested
+ by: murf ........
+
+2009-02-19 22:33 +0000 [r177506-177537] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 177536 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19
+ Feb 2009) | 7 lines Fix up potential crashes, by reducing the
+ sharing between interactive and non-interactive threads. (closes
+ issue #14253) Reported by: Skavin Patches:
+ 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Skavin ........
+
+ * doc/database_transactions.txt (added): Document how to use
+ database transactions
+
+2009-02-19 16:45 +0000 [r177387] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/channel.h: Fix another merge error from 176708
+
+2009-02-19 16:38 +0000 [r177384] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_speech_utils.c: Merged revisions 177383 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb
+ 2009) | 3 lines If we are able to create a speech structure unset
+ the ERROR variable in case it was previously set. (issue
+ #LUMENVOX-13) ........
+
+2009-02-19 15:56 +0000 [r177356] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Fix mismerge from revision 176708 pointed out by
+ Kaloyan Kovachev on the asterisk-dev mailing list. Thanks!
+
+2009-02-19 00:26 +0000 [r177320] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h, funcs/func_odbc.c, CHANGES,
+ res/res_odbc.c, configs/res_odbc.conf.sample: ODBC transaction
+ support
+
+2009-02-19 00:08 +0000 [r177291] Joshua Colp <jcolp@digium.com>
+
+ * CHANGES: Update CHANGES file to include MWI subscription support
+ that was added some time ago.
+
+2009-02-18 23:51 +0000 [r177287] Tilghman Lesher <tlesher@digium.com>
+
+ * main/strings.c: Handle negative length and eliminate a condition
+ that is always true.
+
+2009-02-18 23:50 +0000 [r177286] Steve Murphy <murf@digium.com>
+
+ * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177225 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) |
+ 34 lines This patch fixes a regression of sorts that was
+ introduced in rev 24425. It basically fixes AST-190/ABE-1782.
+ What was wrong: the user has 6000 extensions in one context; and
+ then 6000 contexts, one per extension. The parser could only
+ handle about 4893 of the 6000 extens in the single context. This
+ was due to the regression I mentioned. To get rid of shift/reduce
+ conflicts, Luigi set up right-recursive lists for globals,
+ context elements, switch lists, and statements. Right recursive
+ lists got rid of the warnings, but instead, they use up a
+ tremendous amount of stack space when the lists are long. I saw
+ this a few years back, and resolved not to fix it until someone
+ complained. That day has arrived! After the changes were made, I
+ ran the regression test suite, and there were no problems. I took
+ the test case the user provided, and added 100,000 extensions to
+ the single context, that already had 6,000 extens in it. (I'll
+ see your 6, and raise you 100!) It takes a few minutes to read it
+ all in, check it and generate code for it, but no problems. So, I
+ think I can say that fundamentally, there are no longer any
+ limits on the number of items you can place in contexts,
+ statement blocks, switches, or globals, beyond your virt mem
+ constraints. ........
+
+2009-02-18 23:09 +0000 [r177229] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c: fix two very minor bugs: if anyone ever uses
+ SLINEAR16 as a format in RTP, ensure that the samples are
+ byte-swapped to network order if needed. also, when a smoother is
+ operating on a format that has a sample rate other than 8000
+ samples per second, use the proper sample rate for computing
+ delivery timestamps.
+
+2009-02-18 22:51 +0000 [r177226] David Vossel <dvossel@digium.com>
+
+ * main/features.c: Locking issue in action_bridge and bridge_exec
+ action_bridge() and bridge_exec() both search for the channels to
+ bridge to, and then immediately drop the lock. Instead, they
+ should hold the lock until the masquerade is complete. This will
+ guarantee the channel remains and prevent any other weirdness
+ from occurring. In action_bridge() some more weirdness comes into
+ play. Both channels are needlessly locked at the same time and
+ perform the exact same logic. It makes sense from a coding
+ organizational standpoint, but could cause a theoretical deadlock
+ so I split the code up. There is an issue associated with this,
+ but since its a rather complicated thing to reproduce I'm not
+ certain this alone will close it. issue# 14296 Review:
+ http://reviewboard.digium.com/r/167/
+
+2009-02-18 20:11 +0000 [r177162] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4,
+ channels/h323/cisco-h225.h, channels/h323/caps_h323.cxx,
+ channels/h323/ast_h323.cxx, channels/h323/ast_ptlib.h (added),
+ configure, channels/h323/compat_h323.h, configure.ac,
+ channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
+ channels/h323/ast_h323.h, channels/h323/chan_h323.h,
+ channels/h323/cisco-h225.cxx: Modify h323 to build against PTLib
+ as well as the older PWLib Several changes in PTLib have occurred
+ requiring build time detection. Changes accounted for include the
+ library name change, config option change, install location
+ change, and a boolean type change which is handled by
+ ast_ptlib.h. Also, the sed check has been modified to properly
+ work with autoconf >= 2.62. (closes issue #14224) Reported by:
+ bergolth Patches: asterisk-autoconf-sed.patch uploaded by
+ bergolth (license 661) asterisk-pwlib-v3.patch uploaded by
+ bergolth (license 661) Tested by: jpeeler
+
+2009-02-18 19:12 +0000 [r177101] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: Re-add 'o' option to MeetMe, reverting rev
+ 62297. Enabling this option by default proved to be a bad idea,
+ as the talker detection is not very reliable. So, make it
+ optional again, and off by default. (issue #13801) Reported by:
+ justdave
+
+2009-02-18 19:05 +0000 [r177098] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/config.h: Merged revisions 177096 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009)
+ | 2 lines Document the return value of the update method (as
+ requested on -dev list) ........
+
+2009-02-18 17:24 +0000 [r177035] Doug Bailey <dbailey@digium.com>
+
+ * main/utils.c: Fixed error where a check for an zero length,
+ terminated string was needed.
+
+2009-02-18 17:11 +0000 [r177005] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix ordering of output for a ChannelUpdate
+ manager event. (closes issue #14497) Reported by: vinsik Patches:
+ chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
+
+2009-02-18 16:09 +0000 [r176948] Doug Bailey <dbailey@digium.com>
+
+ * main/utils.c: Need to take into account the \0 terminator of the
+ old string to determine the amount available.
+
+2009-02-18 15:35 +0000 [r176943] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: This patch fixes merge_contexts_and_delete so it does
+ not deadlock when hints are present. Reason: when I re-engineered
+ the merge_and_delete func to reduce its lock time, I failed to
+ notice that the functions it calls still also do locking as
+ before. This leads to deadlocks on dialplan reloads, when there
+ are actually living, subscribed hints registered in the system.
+ While the reporter come across this problem while using AEL, I
+ might note that these deadlocks should also happen if
+ extensions.conf were used. Here I added these routines to pbx.c:
+ ast_add_extension_nolock add_pri_lockopt
+ ast_add_extension2_lockopt find_context add_hint_nolock All of
+ the above routines are static and restricted to be used only
+ within pbx.c, and more specifically within the
+ merge_contexts_and_delete routine. They are pretty much the same
+ as their counterparts except they don't lock contexts or hints.
+ Most of them now do the real work of their name-alike, with
+ optional locking via extra arguments, and are called by their
+ name-alike. The goal was to have the original functions so they
+ would behave exactly as before. Both PJ and I tested these fixes,
+ and the deadlocking problem is no longer encountered. (closes
+ issue #14357) Reported by: pj Patches: 14357.diff uploaded by
+ murf (license 17) Tested by: pj, murf
+
+2009-02-18 06:14 +0000 [r176901-176904] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/heap.h: Add example code for a heap traversal.
+
+ * main/pbx.c: Fix a number of incorrect uses of strncpy(). The big
+ problem here is that the 3rd argument provided in these uses of
+ strncpy() did not reserve a byte for the null terminator, leaving
+ the potential for writing one byte past the end of the buffer.
+ Aside from this, there were coding guidelines violations with
+ regards to spacing, as well as hard coded lengths being used
+ instead of sizeof().
+
+2009-02-18 02:55 +0000 [r176869] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * channels/chan_sip.c: T38 faxdetect should jump to the 'fax'
+ extension for incoming calls only The previous implementation of
+ T38 faxdetect resulted in both sides of the call jumping to a fax
+ extension when both sides had 't38pt_udptl=yes' and
+ 'faxdetect=yes' in sip.conf and a 'fax' extension in the current
+ context. This revision will jump to a 'fax' extension on incoming
+ calls only.
+
+2009-02-18 02:02 +0000 [r176841] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp.c: suppress smoothers for Siren codecs as well as Speex
+ and G.723.1
+
+2009-02-17 22:52 +0000 [r176771] Russell Bryant <russell@digium.com>
+
+ * apps/app_milliwatt.c: Remove a dependency that no longer exists.
+
+2009-02-17 22:28 +0000 [r176760] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice
+ with G723. This commit brings in the changes that were living out
+ on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder
+ branch. codec_dahdi.c now always uses signed linear as the simple
+ codec so that a soft g729 codec will not end up being preferred
+ to the hardware codec. There are also changes to allow
+ codec_dahdi.c to feed packets to the hardware in the native
+ sample size of the codec. This solves problems with choppy audio
+ when using G723.
+
+2009-02-17 22:08 +0000 [r176708] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 176701 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009)
+ | 17 lines Modify bridging to properly evaluate DTMF after first
+ warning is played The main problem is currently if the Dial flag
+ L is used with a warning sound, DTMF is not evaluated after the
+ first warning sound. To fix this, a flag has been added in
+ ast_generic_bridge for playing the warning which ensures that if
+ a scheduled warning is missed, multiple warrnings are not played
+ back (due to a feature evaluation or waiting for digits).
+ ast_channel_bridge was modified to store the nexteventts in the
+ ast_bridge_config structure as that information was lost every
+ time ast_channel_bridge was reentered, causing a hangup due to
+ incorrect time calculations. (closes issue #14315) Reported by:
+ tim_ringenbach Reviewed on reviewboard:
+ http://reviewboard.digium.com/r/163/ ........
+
+2009-02-17 22:02 +0000 [r176706] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_sched.c: Use constants from inttypes.h to clear up
+ 32-bit compilation errors
+
+2009-02-17 21:59 +0000 [r176705] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * channels/chan_sip.c: create a UDPTL structure in
+ create_addr_from_peer() if it does not already exist for T38 This
+ is required to create a UDPTL structure in
+ create_addr_from_peer() to handle the scenario where
+ 't38pt_udptl=yes' is not defined in the [general] section of
+ sip.conf but is defined the peer's context. I tested this patch
+ by enabling t38pt_udptl in the [general] section on one system
+ and only enabling t38pt_udptl in a peer's context on the system
+ sending a fax. Without the patch, the sending system will fail to
+ initiate T38 negotiation with the warning message, "No way to add
+ SDP without an UDPTL structure". When this patch is applied the
+ sending side will successfully initiate T38 negotiation.
+
+2009-02-17 21:40 +0000 [r176697] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/frame.h: Clear up documentation of
+ AST_FRIENDLY_OFFSET in frame.h
+
+2009-02-17 21:23 +0000 [r176669] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Recorded merge of revisions 176661 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009)
+ | 9 lines Backport change to 1.4: Prior to masquerade, move the
+ group definitions to the channel performing the masq, so that the
+ group count lingers past the bridge. (closes issue #14275)
+ Reported by: kowalma Patches: 20090216__bug14275.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: kowalma ........
+
+2009-02-17 21:22 +0000 [r176666] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
+ res/res_timing_timerfd.c, include/asterisk/timing.h,
+ main/timing.c: Update the timing API to have better support for
+ multiple timing interfaces. 1) Add module use count handling so
+ that timing modules can be unloaded. 2) Implement unload_module()
+ functions for the timing interface modules. 3) Allow multiple
+ timing modules to be loaded, and use the one with the highest
+ priority value. 4) Report which timing module is being use in the
+ "timing test" CLI command. (closes issue #14489) Reported by:
+ russell Review: http://reviewboard.digium.com/r/162/
+
+2009-02-17 21:14 +0000 [r176642] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: Prior to masquerade, move the group
+ definitions to the channel performing the masq, so that the group
+ count lingers past the bridge. (closes issue #14275) Reported by:
+ kowalma Patches: 20090216__bug14275.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: kowalma
+
+2009-02-17 21:04 +0000 [r176632-176639] Russell Bryant <russell@digium.com>
+
+ * tests/test_sched.c (added), main/sched.c: Significantly improve
+ scheduler performance under high load. This patch changes the
+ scheduler to use a max-heap to store pending scheduler entries
+ instead of a fully sorted doubly linked list. When the number of
+ entries in the scheduler gets large, this will perform much
+ better. For much more detailed information on this change, see
+ the review request. Review: http://reviewboard.digium.com/r/160/
+
+ * tests/test_heap.c (added): Add a test module for the heap
+ implementation. Review: http://reviewboard.digium.com/r/160/
+
+ * main/Makefile, main/heap.c (added), include/asterisk/heap.h
+ (added): Add an implementation of the heap data structure. A heap
+ is a convenient data structure for implementing a priority queue.
+ Code from svn/asterisk/team/russell/heap/. Review:
+ http://reviewboard.digium.com/r/160/
+
+2009-02-17 20:50 +0000 [r176631] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/config.h: Typo
+
+2009-02-17 20:41 +0000 [r176627] Russell Bryant <russell@digium.com>
+
+ * channels/chan_unistim.c, main/pbx.c, apps/app_read.c,
+ configs/indications.conf.sample, apps/app_playtones.c (added),
+ include/asterisk/indications.h, apps/app_readexten.c,
+ apps/app_disa.c, UPGRADE.txt, include/asterisk/channel.h,
+ include/asterisk/_private.h, main/indications.c, main/loader.c,
+ main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
+ funcs/func_channel.c, res/snmp/agent.c, main/app.c,
+ res/res_indications.c (removed), main/asterisk.c: Merge a large
+ set of updates to the Asterisk indications API. This patch
+ includes a number of changes to the indications API. The primary
+ motivation for this work was to improve stability. The object
+ management in this API was significantly flawed, and a number of
+ trivial situations could cause crashes. The changes included are:
+ 1) Remove the module res_indications. This included the critical
+ functionality that actually loaded the indications configuration.
+ I have seen many people have Asterisk problems because they
+ accidentally did not have an indications.conf present and loaded.
+ Now, this code is in the core, and Asterisk will fail to start
+ without indications configuration. There was one part of
+ res_indications, the dialplan applications, which did belong in a
+ module, and have been moved to a new module, app_playtones. 2)
+ Object management has been significantly changed. Tone zones are
+ now managed using astobj2, and it is no longer possible to crash
+ Asterisk by issuing a reload that destroys tone zones while they
+ are in use. 3) The API documentation has been filled out. 4) The
+ API has been updated to follow our naming conventions. 5) Various
+ bits of code throughout the tree have been updated to account for
+ the API update. 6) Configuration parsing has been mostly
+ re-written. 7) "Code cleanup" The code is from
+ svn/asterisk/team/russell/indications/. Review:
+ http://reviewboard.digium.com/r/149/
+
+2009-02-17 18:49 +0000 [r176592] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c, res/res_odbc.c: Add assertions in the quest to
+ track down a refcount leak. (closes issue #14485) Reported by:
+ davevg
+
+2009-02-17 17:33 +0000 [r176557] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, apps/app_queue.c: Fix a race condition that caused
+ device states to become incorrect for hints. The problem here is
+ that the hint processing code was subscribed to the wrong event
+ type. So, it started processing state for a hint too soon, before
+ the device state cache had been updated. Also, fix a similar bug
+ in app_queue, as it was also subscribed to the wrong event type.
+ (closes issue #14461) Reported by: alecdavis
+
+2009-02-17 17:28 +0000 [r176513-176556] Olle Johansson <oej@edvina.net>
+
+ * configs/extconfig.conf.sample: Typo
+
+ * main/config.c: If there are no realtime engines, there's no
+ reason to check for realtime families
+
+2009-02-17 14:39 +0000 [r176360-176501] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: In this version, we can combine the queries,
+ because we support dropping nonexistent columns.
+
+ * /, channels/chan_sip.c: Merged revisions 176426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009)
+ | 10 lines After a 'sip reload', qualifies for realtime peers
+ weren't immediately restarted, instead waiting until the next
+ registration. We're now caching the qualify across a
+ reload/restart and starting the qualify immediately upon loading
+ the peer. (closes issue #14196) Reported by: pdf Patches:
+ 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
+ Tested by: pdf ........
+
+ * main/strings.c: Might want to update the buffer pointer after a
+ realloc (or we crash) (closes issue #14485) Reported by: davevg
+
+2009-02-16 23:37 +0000 [r176356] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/sounds.xml: add support for Siren7 and Siren14 flavors of
+ prompts and music on hold
+
+2009-02-16 23:33 +0000 [r176355] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 176354 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16
+ Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not
+ being relayed correctly during bridging This should have been
+ committed with rev176247, but I missed it. srcupdate frames no
+ longer break out of the native bridge, but are not being sent to
+ the other call leg either. This fixs that. issue #13749 ........
+
+2009-02-16 23:14 +0000 [r176320] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_skinny.c: Use the correct list macros for deleting
+ an item from the middle of a list. (issue #13777) Reported by: pj
+ Patches: 20090203__bug13777.diff.txt uploaded by Corydon76
+ (license 14) Tested by: pj
+
+2009-02-16 21:45 +0000 [r176255] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/utils.c, include/asterisk/stringfields.h: Merged
+ revisions 176216 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb
+ 2009) | 3 lines fix a flaw in the ast_string_field_build() family
+ of API calls; these functions made no attempt to reuse the space
+ already allocated to a field, so every time the field was written
+ it would allocate new space, leading to what appeared to be a
+ memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46
+ -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the
+ last stringfields commit... don't mark additional space as
+ allocated if the string was built using already-allocated space
+ ........
+
+2009-02-16 21:40 +0000 [r176253] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 176249,176252 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon,
+ 16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it
+ to be nonblocking atomically Apparently on FreeBSD, attempting to
+ set the O_NONBLOCKING flag separately from opening the file was
+ causing an "inappropriate ioctl for device" error. While I cannot
+ fathom why this would be happening, I certainly am not opposed to
+ making the code a bit more compact/efficient if it also fixes a
+ bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
+ uploaded by ys (license 281) Tested by: ys ........ r176252 |
+ mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3
+ lines Remove unused variable and make dev-mode compilation happy
+ ........
+
+2009-02-16 21:30 +0000 [r176248] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Merged revisions 175597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 |
+ dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
+ Fixed iax2 key rotation backwards compatibility Turns key
+ rotation back on by default. Added bit into encryption IE to
+ indicate whether or not key rotation is supported or not. If it
+ is not supported then it is not enabled, which insures backwards
+ compatibility. This eliminates the need for the keyrotate option
+ in iax.conf, so it has been removed. ........
+
+2009-02-16 18:25 +0000 [r176174] Mark Michelson <mmichelson@digium.com>
+
+ * main/logger.c: Assist proper thread synchronization when stopping
+ the logger thread. I was finding that on my dev box, occasionally
+ attempting to "stop now" in trunk would cause Asterisk to hang. I
+ traced this to the fact that the logger thread was waiting on a
+ condition which had already been signalled. The logger thread
+ also need to be sure to check the value of the
+ close_logger_thread variable. The close_logger_thread variable is
+ only checked when the list of logmessages is empty. This allows
+ for the logger thread to print and free any pending messages
+ before exiting.
+
+2009-02-16 17:44 +0000 [r176138] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Can't set debug level 2 (intense
+ debugging) unless the syntax matches
+
+2009-02-16 17:09 +0000 [r176100] Russell Bryant <russell@digium.com>
+
+ * channels/chan_features.c (removed): Remove chan_features. Review:
+ http://reviewboard.digium.com/r/161/
+
+2009-02-16 15:36 +0000 [r176030] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 176029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9
+ lines Don't have the Via header stored as a stringfield as it can
+ change often during the lifetime of a dialog. This issue crept up
+ with subscriptions on the AA50. When an outgoing NOTIFY is sent a
+ new branch value is created and the Via header is changed to
+ reflect it. Since this was a stringfield a new spot in the pool
+ was used for the value while the old was left untouched/unused.
+ If the current pool was full a new pool was created. This would
+ cause memory usage to increase steadily. (issue #AA50-2332)
+ ........
+
+2009-02-16 02:54 +0000 [r175983] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Make the causes array static, and remove the type
+ name as it is not needed.
+
+2009-02-16 00:26 +0000 [r175952] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_unistim.c, /, channels/chan_sip.c,
+ include/asterisk/manager.h, doc/unistim.txt: Merged revisions
+ 175921 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009)
+ | 3 lines fix mis-spelling of the word registered. Reported by
+ De_Mon on #asterisk-dev. ........
+
+2009-02-15 21:27 +0000 [r175829-175882] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/sched.h, main/sched.c: Make ast_sched_report()
+ and ast_sched_dump() thread safe.
+
+ * channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: Fix
+ a number of problems with ast_sched_report(). 1) It had numerous
+ coding guidelines violations with regards to formatting. 2) It
+ allocated memory using ast_calloc() that was never freed. 3) It
+ didn't check for failure from the allocation. 4) It used
+ sprintf() and strcat() to build the result, doing zero checking
+ to prevent writing past the end of the provided buffer. The
+ function also lacks API documentation, but that has not been
+ addressed in this commit.
+
+2009-02-15 20:39 +0000 [r175783-175827] Olle Johansson <oej@edvina.net>
+
+ * formats/format_ilbc.c, /: Merged revisions 175825 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb
+ 2009) | 2 lines format_ilbc does not depend on codec libraries
+ and can therefore always be made. My mistake. Ursäkta! ........
+
+ * formats/format_ilbc.c, /: Merged revisions 175792 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb
+ 2009) | 2 lines Disable format_ilbc.so by default, like
+ codec_ilbc.so ........
+
+ * /, channels/chan_sip.c: Merged revisions 175777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2
+ lines Make sure that the debug line is not printed on debug level
+ 0 ........
+
+2009-02-13 20:57 +0000 [r175655-175663] Mark Michelson <mmichelson@digium.com>
+
+ * doc/manager_1_1.txt, CHANGES, apps/app_queue.c: Merge queue-reset
+ branch to Asterisk From a user point-of-view, this adds new CLI
+ commands and Manager Actions to better facilitate the reloading
+ of queues and the resetting of their statistics. The new CLI
+ commands are the "queue reload" and "queue reset stats" commands.
+ The new manager actions are the QueueReload and QueueReset
+ commands. Review: http://reviewboard.digium.com/r/115
+
+ * doc/manager_1_1.txt, apps/app_chanspy.c: Add manager events for
+ chanspy starting or stopping (closes issue #14469) Reported by:
+ caio1982 Patches: chanspy_events2.diff uploaded by caio1982
+ (license 22)
+
+2009-02-13 20:26 +0000 [r175623-175636] Russell Bryant <russell@digium.com>
+
+ * res/res_jabber.c: fix a few more XML documentation problems
+
+ * main/pbx.c: add missing </para>
+
+2009-02-13 20:11 +0000 [r175597] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, channels/iax2.h, channels/chan_iax2.c:
+ Fixed iax2 key rotation backwards compatibility Turns key
+ rotation back on by default. Added bit into encryption IE to
+ indicate whether or not key rotation is supported or not. If it
+ is not supported then it is not enabled, which insures backwards
+ compatibility. This eliminates the need for the keyrotate option
+ in iax.conf, so it has been removed. Review:
+ http://reviewboard.digium.com/r/159/
+
+2009-02-13 19:49 +0000 [r175591] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 175590 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri,
+ 13 Feb 2009) | 16 lines Fix a potential crash situation when
+ using IMAP voicemail If calling into VoiceMailMain when using
+ IMAP storage, it was possible to crash Asterisk by hanging up the
+ phone when prompted for a voicemail mailbox. This patch fixes the
+ issue. While it may appear that this patch is superficial, it
+ allows code execution to continue to the failure case just below
+ the IMAP_STORAGE code block where this patch has been applied
+ (closes issue #14473) Reported by: dwpaul Patches:
+ voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
+ 689) ........
+
+2009-02-13 16:41 +0000 [r175549] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_record.c: Add an option to keep the recorded file upon
+ hangup. (closes issue #14341) Reported by: fnordian
+
+2009-02-13 13:41 +0000 [r175508-175512] Kevin P. Fleming <kpfleming@digium.com>
+
+ * CHANGES: document G.722.1/.1C support
+
+ * main/frame.c, channels/chan_sip.c, include/asterisk/rtp.h,
+ channels/chan_h323.c, include/asterisk/frame.h,
+ formats/format_siren14.c (added), main/rtp.c,
+ formats/format_siren7.c (added): Add basic (passthrough,
+ playback, record) support for ITU G.722.1 and G.722.1C (also
+ known as Siren7 and Siren14) This patch adds passthrough, file
+ recording and file playback support for the codecs listed above,
+ with negotiation over SIP/SDP supported. Due to Asterisk's
+ current limitation of treating a codec/bitrate combination as a
+ unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are
+ supported. Along the way, some related work was done: 1) The
+ rtpPayloadType structure definition, used as a return result for
+ an API call in rtp.h, was moved from rtp.c to rtp.h so that the
+ API call was actually usable. The only previous used of the API
+ all was chan_h323.c, which had a duplicate of the structure
+ definition instead of doing it the right way. 2) The hardcoded
+ SDP sample rates for various codecs in chan_sip.c were removed,
+ in favor of storing these sample rates in rtp.c along with the
+ codec definitions there. A new API call was added to allow
+ retrieval of the sample rate for a given codec. 3) Some basic
+ 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip
+ *must* decline any media streams offered for these codecs that
+ are not at the bitrates that we support (otherwise Bad Things
+ (TM) would result). Review: http://reviewboard.digium.com/r/158/
+
+2009-02-13 04:22 +0000 [r175411-175475] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * CHANGES: add 'faxbuffers' configuration option information to
+ CHANGES
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
+ dynamic fax buffer configuration option to chan_dahdi.conf When
+ the 'faxdetect' configuration option is used, one may also want
+ to use the 'faxbuffers' configuration option in chan_dahdi.conf.
+ This option will dynamically use the configured 'faxbuffers'
+ buffer policy on a channel for the life of the call following the
+ detection of fax tones. The faxbuffers buffer policy will be
+ reverted during call teardown. An example use of 'faxbuffers' is
+ below. This example would switch to using 6 buffers with a full
+ buffer policy. faxbuffers=>6,full
+
+2009-02-12 21:41 +0000 [r175368] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Remove useless string copy, and make sscanf
+ safe again
+
+2009-02-12 21:27 +0000 [r175344] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
+ force encryption option to iax.conf This patch adds
+ forceencryption=yes as an iax.conf option. When force encryption
+ is enabled, no unencrypted connections are allowed. This insures
+ all connections are encrypted. This is a new feature, so CHANGES
+ and iax.conf.sample are updated as well. (closes issue #13285)
+ Reported by: sgofferj Tested by: russell Review:
+ http://reviewboard.digium.com/r/150/
+
+2009-02-12 21:25 +0000 [r175334] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c, /: Merged revisions 175311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009)
+ | 9 lines Fix crashes when receiving certain T.38 packets. Also,
+ increase the maximum size of T.38 packets and warn users when
+ they try to set the limits above those maximums. (closes issue
+ #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: schern ........
+
+2009-02-12 20:48 +0000 [r175298] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 175294 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009)
+ | 9 lines Fix ParkedCall event information for From field in the
+ case of a blind transfer If the parker information can not be
+ obtained from the peer, try and see if the BLINDTRANSFER channel
+ variable has been set. Previously, a blind transfer to the
+ ParkAndAnnounce app would return nothing for the From. Closes
+ AST-189 ........
+
+2009-02-12 20:45 +0000 [r175255-175295] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Avoid using ast_strdupa() in a loop.
+
+ * build_tools/cflags.xml: Don't enable something by default that
+ has a dependency on something _not_ enabled by default.
+ menuselect was not happy with this.
+
+2009-02-12 18:48 +0000 [r175250] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: correct warning message to not refer
+ specifically to DAHDI
+
+2009-02-12 18:00 +0000 [r175188] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 175187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009)
+ | 6 lines Fix crash in event of failed attempt to transfer to
+ parking The peer may not necessarily exist, such as in the case
+ of a transfer to ParkAndAnnounce. In this case don't try to play
+ a sound to it. ........
+
+2009-02-12 17:07 +0000 [r175127] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Setting key rotation to be off by default
+ Key rotation breaks compatibility between (trunk/1.6.1) and
+ (1.2/1.4/1.6.0). As a follow up to this, I am investigating
+ possible ways to allow key rotation to be on by default and not
+ affect the other branches, but for now it must be turned off.
+
+2009-02-12 16:57 +0000 [r175125] Russell Bryant <russell@digium.com>
+
+ * /, main/rtp.c: Merged revisions 175124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009)
+ | 27 lines Don't send DTMF for infinite time if we do not receive
+ an END event. I thought that this was going to end up being a
+ pretty gnarly fix, but it turns out that there was actually
+ already a configuration option in rtp.conf, dtmftimeout, that was
+ intended to handle this situation. However, in between Asterisk
+ 1.2 and Asterisk 1.4, the code that processed the option got
+ lost. So, this commit brings it back to life. The default timeout
+ is 3 seconds. However, it is worth noting that having this be
+ configurable at all is not really the recommended behavior in RFC
+ 2833. From Section 3.5 of RFC 2833: Limiting the time period of
+ extending the tone is necessary to avoid that a tone "gets
+ stuck". Regardless of the algorithm used, the tone SHOULD NOT be
+ extended by more than three packet interarrival times. A slight
+ extension of tone durations and shortening of pauses is generally
+ harmless. Three seconds will pretty much _always_ be far more
+ than three packet interarrival times. However, that behavior is
+ not required, so I'm going to leave it with our legacy behavior
+ for now. Code from svn/asterisk/team/russell/issue_14460 (closes
+ issue #14460) Reported by: moliveras ........
+
+2009-02-12 16:28 +0000 [r175121] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: Make lock information
+ for ao2_trylock be more useful and gnarly Core show locks
+ information involving an ao2_trylock did not show the function
+ that called ao2_trylock, but would instead show ao2_trylock as
+ the source of the lock. This is not useful when trying to debug
+ locking issues. One bizarre note is that this logic is already in
+ 1.4 but somehow did not get merged to trunk or the 1.6.X
+ branches.
+
+2009-02-12 14:25 +0000 [r175058-175089] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Issue a warning message if our candidate's
+ IP is the loopback address. (closes issue #13985) Reported by:
+ jcovert Tested by: phsultan
+
+ * /, channels/chan_gtalk.c: Merged revisions 175029 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12
+ Feb 2009) | 12 lines Set the initiator attribute to lowercase in
+ our replies when receiving calls. This attribute contains a JID
+ that identifies the initiator of the GoogleTalk voice session.
+ The GoogleTalk client discards Asterisk's replies if the
+ initiator attribute contains uppercase characters. (closes issue
+ #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded
+ by jcovert (license 551) Tested by: jcovert ........
+
+2009-02-11 23:12 +0000 [r174945-174951] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix a bit of odd logic for announcing position.
+ Sync with 1.6.0's logic
+
+ * apps/app_queue.c: Fix odd "thank you" sound playing behavior in
+ app_queue.c If someone has configured the queue to play an
+ position or holdtime announcement, then it is odd and potentially
+ unexpected to hear a "Thank you for your patience" sound when no
+ position or holdtime was actually announced. This fixes the
+ announcement so that the "thanks" sound is only played in the
+ case that a position or holdtime was actually announced. There is
+ a way that the "thank you" sound can be played without a position
+ or holdtime, and that is to set announce-frequency to a value but
+ keep announce-position and announce-holdtime both turned off.
+ (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
+ uploaded by putnopvut (license 60) Tested by: caspy
+
+ * apps/app_dial.c, main/channel.c, main/pbx.c, apps/app_dictate.c,
+ apps/app_waitforsilence.c, include/asterisk/channel.h: Fix 'd'
+ option for app_dial and add new option to Answer application The
+ 'd' option would not work for channel types which use RTP to
+ transport DTMF digits. The only way to allow for this to work was
+ to answer the channel if we saw that this option was enabled. I
+ realized that this may cause issues with CDRs, specifically with
+ giving false dispositions and answer times. I therefore modified
+ ast_answer to take another parameter which would tell if the CDR
+ should be marked answered. I also extended this to the Answer
+ application so that the channel may be answered but not CDRified
+ if desired. I also modified app_dictate and app_waitforsilence to
+ only answer the channel if it is not already up, to help not
+ allow for faulty CDR answer times. All of these changes are going
+ into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the
+ changes except for the change to the Answer application will go
+ in since we do not introduce new features into stable branches
+ (closes issue #14164) Reported by: DennisD Patches: 14164.patch
+ uploaded by putnopvut (license 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/145
+
+2009-02-11 14:44 +0000 [r174844] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Tell the device state core a change happened when
+ a channel is freed but not a specific state. We need to do this
+ because while we know that the freeing of the channel may cause
+ something to become not in use we do not know this for sure.
+ There may be another channel that is still up which would cause
+ it to be in use. (closes issue #13238) Reported by: kowalma
+ Patches: 20090121__bug13238.diff.txt uploaded by Corydon76
+ (license 14) Tested by: alecdavis
+
+2009-02-10 23:17 +0000 [r174764-174805] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Fix potential for stack overflows in
+ app_chanspy.c When using the 'g' or 'e' options, the stack
+ allocations that were used could cause a stack overflow if a
+ spyer stayed on the line long enough without actually
+ successfully spying on anyone. The problem has been corrected by
+ using static buffers and copying the contents of the appropriate
+ strings into them instead of using functions like alloca or
+ ast_strdupa
+
+ * main/manager.c: Fix an fd leak that would occur in HTTP AMI
+ sessions The explanation behind this fix is a bit complicated,
+ and I've already typed it up in the code as a huge comment inside
+ of manager.c, so I'll give the abridged version here. We needed a
+ way to separate action-specific data from session-specific data.
+ Unfortunately, the only way to maintain API compatibility and to
+ not have to change every single manager action was to rename the
+ current mansession structure and wrap it inside a new mansession
+ structure which actually contains action- specific data. (closes
+ issue #14364) Reported by: awk Patches: 14364_better.patch
+ uploaded by putnopvut (license 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/148/
+
+2009-02-10 20:15 +0000 [r174710] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only decrease inringing count if above zero.
+ (issue #13238) Reported by: kowalma
+
+2009-02-10 19:38 +0000 [r174705] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/slinfactory.c, include/asterisk/slinfactory.h: improve
+ slinfactory API to remove implicit sample rate and require
+ explicit sample rate selection by creator of the slinfactory
+
+2009-02-10 18:16 +0000 [r174584] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/jitterbuf.c: Merged revisions 174583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb
+ 2009) | 18 lines Improve behavior of jitterbuffer when
+ maxjitterbuffer is set. This change improves the way the
+ jitterbuffer handles maxjitterbuffer and dramatically reduces the
+ number of frames dropped when maxjitterbuffer is exceeded. In the
+ previous jitterbuffer, when maxjitterbuffer was exceeded, all new
+ frames were dropped until the jitterbuffer is empty. This change
+ modifies the code to only drop frames until maxjitterbuffer is no
+ longer exceeded. Also, previously when maxjitterbuffer was
+ exceeded, dropped frames were not tracked causing stats for
+ dropped frames to be incorrect, this change also addresses that
+ problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
+ by mnicholson (license 96) Tested by: mnicholson Review:
+ http://reviewboard.digium.com/r/144/ ........
+
+2009-02-10 17:48 +0000 [r174543-174580] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Set the type for the peer structure to be a
+ peer as the default. (closes issue #14447) Reported by: triccyx
+
+ * channels/chan_sip.c: Make the logic for inuse and inringing
+ manipluation match that of 1.4. The old broken logic would reset
+ the values back to 0 during certain scenarios causing the wrong
+ state to be reported. (closes issue #14399) Reported by: caspy
+ (issue #13238) Reported by: kowalma
+
+2009-02-10 07:06 +0000 [r174470-174503] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, apps/app_voicemail.c: Fix0ring build
+
+ * apps/app_stack.c: Remove the usage of the KeepAlive app, as it no
+ longer exists.
+
+2009-02-10 04:49 +0000 [r174370-174435] Steve Murphy <murf@digium.com>
+
+ * apps/app_rpt.c: This patch removes the use of AST_PBX_KEEPALIVE
+ from app_rpt.c. (closes issue #14435) Reported by: D_McNaul
+
+ * apps/app_rpt.c: More intptr_t work.
+
+ * /, apps/app_rpt.c: Merged revisions 174369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5
+ lines This patch solves some compiler complaints in both 32 and
+ 64-bit environments. ........
+
+2009-02-09 17:27 +0000 [r174327] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix something I messed up in the merge I
+ just did
+
+2009-02-09 17:26 +0000 [r174325] David Vossel <dvossel@digium.com>
+
+ * apps/app_externalivr.c: Fixes issue with hangups not being sent
+ and external process never terminating. The ignore_hangup,
+ run_dead, and noanswer flags were never initilized to zero
+ causing hangups to never be issued. If the external script
+ expects to be notified of a hangup and never receives one, it
+ runs indefinitely. (closes issue #14251) Reported by: chris-mac
+ Tested by: dvossel
+
+2009-02-09 17:20 +0000 [r174301] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
+ 2009) | 12 lines Don't do an SRV lookup if a port is specified
+ RFC 3263 says to do A record lookups on a hostname if a port has
+ been specified, so that's what we're going to do. See section
+ 4.2. (closes issue #14419) Reported by: klaus3000 Patches:
+ patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000
+ (license 65) ........
+
+2009-02-09 14:49 +0000 [r174219] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 174218 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb
+ 2009) | 4 lines Don't overwrite our pointer to the music class
+ when music on hold stops. We will use this if it starts again to
+ see if we can resume the music where it left off. (closes issue
+ #14407) Reported by: mostyn ........
+
+2009-02-07 16:16 +0000 [r174149] Russell Bryant <russell@digium.com>
+
+ * /, res/snmp/agent.c: Merged revisions 174148 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
+ | 2 lines Fix a race condition that could cause a crash. ........
+
+2009-02-06 23:51 +0000 [r174084] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
+ | 5 lines check ast_strlen_zero() before calling ast_strdupa() in
+ sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter
+ didn't actually upload a properly-formed patch, instead a
+ modified chan_sip.c file was uploaded. I created a patch to
+ determine the changes, then modified the suggested changes to
+ create a proper fix. The summary above is a complete description
+ of the changes. (closes issue #13547) Reported by: tecnoxarxa
+ Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258)
+ Tested by: tecnoxarxa ........
+
+2009-02-06 20:12 +0000 [r174046] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
+ immediate yes/no option to iax.conf This is very similar to the
+ DAHDI immediate=yes option. When the phone is picked up, instead
+ of giving a dialtone it connects directly to the "s" extension.
+ Changes where implemented in chan_iax2.c to directly connect to
+ the "s" extension in the appropriate context when this option is
+ enabled. Examples explaining its use are added to
+ iax2.conf.sample. CHANGES has been updated as well. (closes issue
+ #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk
+ uploaded by jcovert (license 551) iax.conf.sample.patch uploaded
+ by jcovert (license 551) Tested by: jcovert, dvossel Review:
+ http://reviewboard.digium.com/r/143/
+
+2009-02-06 19:28 +0000 [r173974-174041] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_dahdi.c: Don't subscribe to a mailbox on pseudo
+ channels. It is futile. This solves an issue where duplicated
+ pseudo channels would cause a crash because the first one would
+ unsubscribe and the next one would also try to unsubscribe the
+ same subscription. (closes issue #14322) Reported by: amessina
+
+ * /, channels/chan_sip.c: Merged revisions 173967-173968 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
+ lines Some clients do not put the call-id for replaces at the
+ beginning, so support it being anywhere in the string. (closes
+ issue #14350) Reported by: fhackenberger ........ r173968 | file
+ | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
+ debug message I put in by accident. ........
+
+2009-02-06 16:28 +0000 [r173952] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 173917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
+ 2009) | 7 lines Limit the addition of the Contact header in SIP
+ responses according to various SIP RFCs. (closes issue #13602)
+ Reported by: hjourdain Tested by: mnicholson ........
+
+2009-02-06 15:59 +0000 [r173902] Joshua Colp <jcolp@digium.com>
+
+ * main/audiohook.c, apps/app_chanspy.c: Always detach and destroy
+ the whisper and barge audiohooks. Additionally also allow an
+ audiohook to be detached if it has not been attached. (closes
+ issue #14414) Reported by: bluecrow76
+
+2009-02-06 10:55 +0000 [r173848-173858] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c: Add
+ a common implementation of a scheduler context with a dedicated
+ thread. This commit expands the Asterisk scheduler API to include
+ a common implementation of a scheduler context being processed by
+ a dedicated thread. chan_iax2 has been updated to use this new
+ code. Also, as a result, this resolves some race conditions
+ related to the previous chan_iax2 scheduler handling. Related to
+ rev 171452 which resolved the same issues in 1.4. Code from
+ team/russell/sched_thread2 Review:
+ http://reviewboard.digium.com/r/129/
+
+ * main/manager.c: Resolve a memory leak that would occur on an
+ invalid channel given to Action: Status
+
+2009-02-05 23:48 +0000 [r173773-173776] Mark Michelson <mmichelson@digium.com>
+
+ * configs/extensions.conf.sample: Update extensions.conf.sample to
+ be correct. In trunk, the only necessary change pointed out was
+ that the call to ChanIsAvail uses an option that has been
+ removed. For the 1.6.1 branch, however, it appears that the
+ sample file is badly in need of updating since there are |'s used
+ all over the place there. My tentative plan is just to copy
+ trunk's sample config file to those branches since the info there
+ is most up-to-date and should be correct for use in 1.6.1 Thanks
+ to macli in #asterisk-dev for bringing this up
+
+ * apps/app_voicemail.c: Properly set "seen" and "unseen" flags when
+ moving messages from the new to the old folder when using IMAP
+ for voicemail storage (closes issue #13905) Reported by: jaroth
+ Patches: foldermove_v2.patch uploaded by jaroth (license 50)
+
+2009-02-05 21:00 +0000 [r173697] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 173696 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05
+ Feb 2009) | 12 lines Add new configuration option to make shared
+ IMAP mailboxes function as expected. The new option is
+ "imapvmshareid" which is an ID to tag multiple mailboxes using
+ the same IMAP storage location to function as one mailbox. This
+ allows all messages to be retrieved for any user in the group.
+ The patch alters the 'X-Asterisk-VM-Extension' header that is
+ responsible for matching voicemails for a given user. (closes
+ issue #13673) Reported by: howardwilkinson ........
+
+2009-02-05 20:30 +0000 [r173693] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 173692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb
+ 2009) | 12 lines Fix situations where queue members could be
+ autopaused unexpectedly Specifically, this patch prevents us from
+ autopausing members when we receive a busy or congestion frame
+ from them. (closes issue #14376) Reported by: fiddur Patches:
+ 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
+ ........
+
+2009-02-05 19:36 +0000 [r173657] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_sqlite.c: Change the first field, or we don't get
+ the necessary field separation.
+
+2009-02-05 18:48 +0000 [r173507-173593] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_mixmonitor.c: Merged revisions 173592 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu,
+ 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor
+ ........
+
+ * /, apps/app_mixmonitor.c: Merged revisions 173559 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu,
+ 05 Feb 2009) | 25 lines Fix a problem where a channel pointer
+ becomes invalid due to masquerading or hanging up. app_mixmonitor
+ runs its own thread to monitor the channel's activity and write
+ the mixed audio to a file. Since this thread runs independently
+ of the channel, it is possible that the mixmonitor thread's
+ channel pointer will point to freed memory when the channel
+ either is masqueraded or hangs up (technically, both cases are
+ hangups, but we need to handle the cases slightly differently).
+ The solution for this is to employ a datastore, which has the
+ nice benefit of allowing us to hook into channel masquerades and
+ hangups and update our pointer as necessary. If this looks
+ familiar, this same technique is employed in app_chanspy.
+ app_chanspy is a bit more involved since it does a lot more
+ operations on the channel that is being spied upon.
+ app_mixmonitor does have an extra touch that app_chanspy doesn't
+ have, though. Since there is a thread race between the channel's
+ thread and the mixmonitor thread on a hangup, we em- ploy a
+ condition-and-boolean combination to ensure that the channel
+ thread finishes with our structure before the mixmonitor thread
+ attempts to free it. No crashes! (closes issue #14374) Reported
+ by: aragon Patches: 14374.patch uploaded by putnopvut (license
+ 60) Tested by: aragon, putnopvut ........
+
+ * apps/app_queue.c: Fix some areas where the incorrect interface
+ was passed to ast_device_state I swear it feels like I already
+ did this once... (closes issue #14359) Reported by: francesco_r
+
+2009-02-04 21:26 +0000 [r173503] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_jabber.c: Add XML documentation for the applications and
+ functions in res_jabber (closes issue #14405) Reported by: snuffy
+ Patches: xml_jabber.diff uploaded by snuffy (license 35)
+
+2009-02-04 21:25 +0000 [r173502] David Vossel <dvossel@digium.com>
+
+ * channels/iax2-parser.h, channels/chan_iax2.c: Fixes issue with
+ IAX2 transfer not handing off calls. Reverts changes in 116884
+ Fixes issue with IAX2 transfers not taking place. As it was, a
+ call that was being transfered would never be handed off
+ correctly to the call ends because of how call numbers were
+ stored in a hash table. The hash table, "iax_peercallno_pvt",
+ storing all the current call numbers did not take into account
+ the complications associated with transferring a call, so a
+ separate hash table was required. This second hash table
+ "iax_transfercallno_pvt" handles calls being transfered, once the
+ call transfer is complete the call is removed from the transfer
+ hash table and added to the peer hash table resuming normal
+ operations. Addition functions were created to handle storing,
+ removing, and comparing items in the iax_transfercallno_pvt
+ table. The changes reverted in 116884 caused backwards
+ compatibility issues involving iax2 transfer with 1.6.0, 1.4, and
+ 1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel
+
+2009-02-04 21:17 +0000 [r173500] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c, include/asterisk/features.h: Merged revisions
+ 173211 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009)
+ | 17 lines Parking attempts made to one end of a bridge no longer
+ will hang up due to a parking failure. Parking attempts made
+ using either one-touch, or doing either a blind or assisted
+ transfer to the parking extension now keep up the bridge instead
+ of hanging up the attempted parked party. Normal causes for the
+ parking attempt to fail includes the specific specified extension
+ (via PARKINGEXTEN) not being available or if all the parking
+ spaces are currently in use. To avoid having to reverse a
+ masquerade park_space_reserve was made to provide foresight if a
+ parking attempt will succeed and if so reserve the parking space.
+ (closes issue #13494) Reported by: mdu113 Reviewed by Russell:
+ http://reviewboard.digium.com/r/133/ ........
+
+2009-02-04 18:48 +0000 [r173458] Tilghman Lesher <tlesher@digium.com>
+
+ * main/tcptls.c: When using a socket as a FILE *, the stdio
+ functions will sometimes try to do an fseek() on the stream,
+ which is an invalid operation for a socket. Turning off buffering
+ explicitly lets the stdio functions know they cannot do this,
+ thus avoiding a potential error. (closes issue #14400) Reported
+ by: fnordian Patches: tcptls.patch uploaded by fnordian (license
+ 110)
+
+2009-02-04 17:45 +0000 [r173354-173397] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 173396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb
+ 2009) | 3 lines Revert my previous change because it was stupid
+ ........
+
+ * /, apps/app_chanspy.c: Merged revisions 173392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb
+ 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever
+ matter, but it's needed. ........
+
+ * main/file.c: Fix a problem where file playback would cause fds to
+ remain open forever The problem came from the fact that a frame
+ read from a format interpreter was not freed. Adding a call to
+ ast_frfree fixed this. The explanation for why this caused the
+ problem is a bit complex, but here goes: There was a problem in
+ all versions of Asterisk where the embedded frame of a filestream
+ structure was referenced after the filestream was freed. This was
+ fixed by adding reference counting to the filestream structure.
+ The refcount would increase every time that a filestream's frame
+ pointer was pointing to an actual frame of data. When the frame
+ was freed, the refcount would decrease. Once the refcount reached
+ 0, the filestream was freed, and as part of the operation, the
+ open files were closed as well. Thus it becomes more clear why a
+ missing ast_frfree would cause a reference leak and cause the
+ files to not be closed. You may ask then if there was a frame
+ leak before this patch. The answer to that is actually no! The
+ filestream code was "smart" enough to know that since the frame
+ we received came from a format interpreter, the frame had no
+ malloced data and thus didn't need to be freed. Now, however,
+ there is cleanup that needs to be done when we finish with the
+ frame, so we do need to call ast_frfree on the frame to be sure
+ that the refcount for the filestream is decremented
+ appropriately. (closes issue #14384) Reported by: fiddur Patches:
+ 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur,
+ putnopvut
+
+2009-02-04 00:43 +0000 [r173311] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, pbx/pbx_config.c: Ensure that commas placed in the
+ middle of extension character classes do not interfere with
+ correct parsing of the extension. Also, if an unterminated
+ character class DOES make its way into the pbx core (through some
+ other method), ensure that it does not crash Asterisk. (closes
+ issue #14362) Reported by: Nick_Lewis Patches:
+ 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76
+
+2009-02-03 17:35 +0000 [r173169] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Broke up the large conditional blocks so
+ it is easy to see if a function is compiled.
+
+2009-02-03 00:29 +0000 [r173104-173130] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/xml.c, include/asterisk/compiler.h, apps/app_stack.c,
+ include/asterisk/optional_api.h: 1. Make OS X compile cleanly
+ with app_stack. 2. Use curl to download sound files, as curl is
+ installed natively on OS X, whereas wget and fetch are not.
+ (closes issue #14332) Reported by: oej Tested by: Corydon76
+
+ * /, configs/extensions.conf.sample: Merged revisions 173070 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009)
+ | 5 lines Add warning to standard config, that globals may be
+ overridden by other dialplan configuration files. (closes issue
+ #14388) Reported by: macli ........
+
+2009-02-02 23:57 +0000 [r173067] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 173066 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009)
+ | 2 lines Fix a feature inheritance bug I added after code review
+ ........
+
+2009-02-02 23:21 +0000 [r173028-173047] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, CHANGES: Reverting commit number 173028 as there
+ are some potential issues
+
+ * main/manager.c, CHANGES: Add a CLI command to log out a manager
+ user (closes issue #13877) Reported by: eliel Patches:
+ cli_manager_logout.patch.txt uploaded by eliel (license 64)
+ Tested by: eliel, putnopvut
+
+2009-02-02 20:40 +0000 [r172963] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Recorded merge of revisions 172962 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009)
+ | 11 lines channels/chan_dahdi.c * Added doxygen comments to the
+ major dahdi structures. * Fixed PRI using an incorrect string
+ value if the extension delimiter is not present in the Dial()
+ function. * Fixed some uninitialized string variables on FXS
+ ports. configs/chan_dahdi.conf.sample * Updated some
+ documentation. These changes are already in trunk -r172400
+ ........
+
+2009-02-02 19:02 +0000 [r172929] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/features.c, CHANGES,
+ include/asterisk/features.h: This reverts the changes I made for
+ 11583; will reviewboard this before committing again... reopened
+ 11583 until all Russell's issues are resolved.
+
+2009-02-02 18:13 +0000 [r172894] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample: Update the res_ldap.conf file with
+ a better working example. (closes issue #13861) Reported by:
+ scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by
+ blitzrage (license 10) Tested by: jcovert
+
+2009-02-02 17:37 +0000 [r172890] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/features.c, CHANGES,
+ include/asterisk/features.h: This change allows the disconnect
+ feature (as in "one-touch" in features.c) to be used within the
+ dial app, before a call is bridged. Many thanks to sobomax for
+ submitting this patch. Quoting from bug 11582: "So the goal of
+ the patch was to use the user configured feature code during the
+ call setup phase. The original ast_feature_interpret() function
+ is not well suited for this purpose as it uses much call bridge
+ specific data and doesn't separate a detection of feature from a
+ feature handler call. So a new function ast_feature_detect() has
+ been extracted off the ast_feature_interpret() function but
+ keeping the original logic intact except some insignificant
+ changes to locking. "Having created the ast_feature_detect()
+ function the possibility to use feature detection in almost any
+ place of the asterisk code. So a call to this function has been
+ added to wait_for_answer() function of app_dial.so module. This
+ code doesn't call the feature handler however and uses old call
+ leg disconnect logic to make the changes as small and simple as
+ possible to prevent unexpected problems. A disconnect feature
+ currently is the only one supported during call setup as other
+ features as call parking and call transfer don't make much sense
+ during call setup. However if need in some of the features would
+ arise it is much easier to implement as the infrastructure
+ changes are already in place with this patch." I have cleaned up
+ the patch somewhat, and verified that the existing functionality
+ is not harmed, and that the new functionality works. Terry has
+ committed his stuff, and there were no conflicts (see 14274).
+ (closes issue #11583) Reported by: sobomax Patches:
+ patch-apps__app_dial.c uploaded by sobomax (license 359)
+ patch-include__asterisk__features.h uploaded by sobomax (license
+ 359) patch-res__res_features.c uploaded by sobomax (license 359)
+ enable-features-during-call-setup.diff uploaded by sobomax
+ (license 359) 11583.newdiff uploaded by murf (license 17)
+ enable-features-during-call-setup-1.diff uploaded by sobomax
+ (license 359) 11583.latest-patch uploaded by murf (license 17)
+ Tested by: sobomax, murf
+
+2009-02-02 16:42 +0000 [r172855] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix a spelling mistake.
+
+2009-02-02 10:46 +0000 [r172816-172818] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Add a todo. I do need to really check what's
+ going on with this kill-the-user business ;-) Why do we suddenly
+ have two flags to set peer type?
+
+ * channels/chan_sip.c: Small formatting change
+
+ * channels/chan_sip.c: Add some well-needed improvements to the
+ wishlist in the code, so that we can close some bug reports.
+
+2009-02-02 01:41 +0000 [r172778] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_sip.c: The CID lookup feature wasn't actually
+ working properly with dialog-info+xml supporting devices. The
+ devices (snoms, specifically) need to receive a SIP URI instead
+ of just an extension. This adds that functionality.
+
+2009-02-01 02:44 +0000 [r172706-172741] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Blank argument crashes Asterisk (closes
+ issue #14377) Reported by: amorsen
+
+ * funcs/func_strings.c: Don't increment the loop, now that
+ incrementing is taken care of by the decoder function. (closes
+ issue #14363) Reported by: andrew53 Patches:
+ func_strings_filter.patch uploaded by andrew53 (license 519)
+
+2009-01-30 22:22 +0000 [r172598] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/channel.h: Fix redefinition of flag in channel.h
+
+2009-01-30 21:50 +0000 [r172580-172581] Terry Wilson <twilson@digium.com>
+
+ * configs/features.conf.sample: Remove incorrect line from sample
+ config
+
+ * apps/app_dial.c, main/global_datastores.c, main/features.c,
+ include/asterisk/global_datastores.h, CHANGES,
+ configs/features.conf.sample: Merged revisions 172517 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009)
+ | 37 lines Fix feature inheritance with builtin features When
+ using builtin features like parking and transfers, the
+ AST_FEATURE_* flags would not be set correctly for all instances
+ when either performing a builtin attended transfer, or parking a
+ call and getting the timeout callback. Also, there was no way on
+ a per-call basis to specify what features someone should have on
+ picking up a parked call (since that doesn't involve the Dial()
+ command). There was a global option for setting whether or not
+ all users who pickup a parked call should have
+ AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
+ PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
+ variable which can be set either in the dialplan or with setvar
+ in channels that support it. This variable can be set to any
+ combination of 't', 'k', 'w', and 'h' (case insensitive matching
+ of the equivalent dial options), to set what features should be
+ activated on this channel. The patch moves the setting of the
+ features datastores into the bridging code instead of app_dial to
+ help facilitate this. 2) adds global options parkedcallparking,
+ parkedcallhangup, and parkedcallrecording to be similar to the
+ parkedcalltransfers option for globally setting features. 3) has
+ builtin_atxfer call builtin_parkcall if being transfered to the
+ parking extension since tracking everything through multiple
+ masquerades, etc. is difficult and error-prone 4) attempts to fix
+ all cases of return calls from parking and completed builtin
+ transfers not having the correct permissions (closes issue
+ #14274) Reported by: aragon Patches:
+ fix_feature_inheritence.diff.txt uploaded by otherwiseguy
+ (license 396) Tested by: aragon, otherwiseguy Review
+ http://reviewboard.digium.com/r/138/ ........
+
+2009-01-30 18:36 +0000 [r172441-172548] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_aes.c: Parameter position reversed in documentation
+
+ * /, autoconf/ast_func_fork.m4, configure, main/app.c,
+ apps/app_rpt.c, main/asterisk.c: Merged revisions 172438 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009)
+ | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at
+ startup. Otherwise, if Asterisk runs as a non-root user and the
+ administrator does a 'restart now', Asterisk loses the ability to
+ set QOS on packets. (closes issue #14004) Reported by: nemo
+ Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
+ (license 14) Tested by: Corydon76 ........
+
+2009-01-29 23:15 +0000 [r172370-172440] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cli.c: Remove tabs from comment
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
+ channels/chan_dahdi.c * Added doxygen comments to the major dahdi
+ structures. * Fixed PRI and SS7 using an incorrect string value
+ if the extension delimiter is not present in the Dial() function.
+ * Fixed SS7 not checking if the dialed extension is at least as
+ long as the stripmsd option. * Fixed PRI not handling unknown
+ TON/NPI prefix letters correctly. * Fixed some uninitialized
+ string variables on FXS ports. configs/chan_dahdi.conf.sample *
+ Updated some documentation.
+
+ * include/asterisk/say.h: Fixed some doxygen comments
+
+2009-01-29 17:10 +0000 [r172318-172319] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_local.c: Revert two lines that was extra, but only
+ on fridays.
+
+ * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
+ include/asterisk/causes.h, apps/app_queue.c: Fix "cancel answered
+ elsewhere" through app_queue with members in chan_local. Also,
+ implement a private cause code (as suggested by Tilghman). This
+ works with chan_sip, but doesn't propagate through chan_local.
+
+2009-01-29 16:48 +0000 [r172315] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/func_odbc.conf.sample: Better document mode=multirow,
+ based upon a conversation with Jared.
+
+2009-01-29 13:47 +0000 [r172271] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/realtime_pgsql.sql: The realtime_pgsql.sql script
+ is missing a couple of fields. closes issue #14339) Reported by:
+ fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur
+ (license 678)
+
+2009-01-29 13:24 +0000 [r172173-172270] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample, CHANGES: Update documentation
+
+ * include/asterisk/app.h, channels/chan_sip.c, main/app.c: - Make
+ sure we set setvar= variables on outbound calls too, not only
+ inbound calls. - Also, change a function in app.c to return a
+ userful value instead of always returning 0. Patch by fnordian,
+ changed by Corydon76 and myself. This does not close the bug
+ report, as fnordian had an additional change we're still
+ discussing. (related to issue #14059) Reported by: fnordian
+ Patches: chan_sip_hfield.patch uploaded by fnordian (license 110)
+ 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: fnordian, Corydon76, oej
+
+ * channels/chan_sip.c: Make sure register= line supports both port
+ and expiry at the same time. (closes issue #14185) Reported by:
+ Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by
+ Nick (license 657) Tested by: Nick_Lewis
+
+ * /, channels/chan_sip.c: Merged revisions 172169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16
+ lines Make sure that we always add the hangupcause headers. In
+ some cases, the owner was disconnected before we checked for the
+ cause. This patch implements a temporary storage in the pvt and
+ use that instead. The code is based on ideas from code from
+ Adomjan in issue #13385 (Add support for Reason: header) Thanks
+ to Klaus Darillion for testing! (closes issue #14294) related to
+ issue #13385 Reported by: klaus3000 and adomjan Patches:
+ bug14294b.diff uploaded by oej (license 306) Based on
+ 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan
+ (license 487) Tested by: oej, klaus3000 ........
+
+2009-01-28 22:52 +0000 [r172132] Steve Murphy <murf@digium.com>
+
+ * channels/chan_misdn.c: A further correction: cast the sizeof to
+ an int.
+
+2009-01-28 22:48 +0000 [r172131] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c: Fix how we skip fields (to avoid fields
+ which don't exist) when doing an UPDATE. (closes issue #14205)
+ Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: blitzrage
+
+2009-01-28 21:48 +0000 [r172063-172099] Steve Murphy <murf@digium.com>
+
+ * channels/chan_misdn.c: my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2
+ didn't like the \%ld and issued a warning, breaking my dev-mode
+ build. This fixes it.
+
+ * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /,
+ main/features.c, include/asterisk/channel.h: Merged revisions
+ 172030 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) |
+ 46 lines This patch fixes h-exten running misbehavior in
+ manager-redirected situations. What it does: 1. A new Flag value
+ is defined in include/asterisk/channel.h,
+ AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
+ bridge hangup exten code not to run the h-exten there (nor
+ publish the bridge cdr there). It will done at the pbx-loop level
+ instead. 2. In the manager Redirect code, I set this flag on the
+ channel if the channel has a non-null pbx pointer. I did the same
+ for the second (chan2) channel, which gets run if name2 is set...
+ and the first succeeds. 3. I restored the ending of the cdr for
+ the pbx loop h-exten running code. Don't know why it was removed
+ in the first place. 4. The first attempt at the fix for this bug
+ was to place code directly in the async_goto routine, which was
+ called from a large number of places, and could affect a large
+ number of cases, so I tested that fix against a fair number of
+ transfer scenarios, both with and without the patch. In the
+ process, I saw that putting the fix in async_goto seemed not to
+ affect any of the blind or attended scenarios, but still, I was
+ was highly concerned that some other scenarios I had not tested
+ might be negatively impacted, so I refined the patch to its
+ current scope, and jmls tested both. In the process, tho, I saw
+ that blind xfers in one situation, when the one-touch blind-xfer
+ feature is used by the peer, we got strange h-exten behavior. So,
+ I inserted code to swap CDRs and to set the HANGUP_DONT field, to
+ get uniform behavior. 5. I added code to the bridge to obey the
+ HANGUP_DONT flag, skipping both publishing the bridge CDR, and
+ running the h-exten; they will be done at the pbx-loop (higher)
+ level instead. 6. I removed all the debug logs from the patch
+ before committing. 7. I moved the AUTOLOOP set/reset in the
+ h-exten code in res_features so it's only done if the h-exten is
+ going to be run. A very minor performance improvement, but
+ technically correct. (closes issue #14241) Reported by: jmls
+ Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer
+ uploaded by murf (license 17) Tested by: murf, jmls ........
+
+2009-01-28 17:27 +0000 [r171964] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 171963 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28
+ Jan 2009) | 2 lines Clarify log message (suggested by manxpower
+ on #asterisk-dev) ........
+
+2009-01-28 14:39 +0000 [r171838-171925] Olle Johansson <oej@edvina.net>
+
+ * CHANGES: Yep. Documentation is important.
+
+ * apps/app_queue.c: Add final part of previously committed work for
+ answered elsewhere in queue - the missing piece that started with
+ app_dial() earlier on. This is to avoid having the list and
+ counter of missed calls being touched by queue calls. Add the C
+ option to queue() and nothing will be logged on phones that
+ support the Reason: header on SIP cancel, like the SNOM phones.
+
+ * configs/sip.conf.sample: Add some more notes about device
+ matching.
+
+ * /, configs/sip.conf.sample: Merged revisions 171837 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan
+ 2009) | 2 lines Add a better explanation of the difference
+ between the device namespace and the dialplan for newbies.
+ ........
+
+2009-01-28 00:17 +0000 [r171797] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_aes.c: Fix some signedness problems in func_aes.c
+
+2009-01-27 23:28 +0000 [r171793] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: Don't complain about lack of D-channels on
+ PTMP connections
+
+2009-01-27 22:43 +0000 [r171757] David Vossel <dvossel@digium.com>
+
+ * funcs/func_aes.c (added), CHANGES: Adding AES_ENCRYPT and
+ AES_DECRYPT dialplan functions. (closes issue #14301) Reported
+ by: amorsen review: http://reviewboard.digium.com/r/128/
+
+2009-01-27 21:58 +0000 [r171618-171691] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Merged revisions 171689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan
+ 2009) | 39 lines Fix devicestate problems for "always-on" agent
+ channels A revision to chan_agent attempted to "inherit" the
+ device state of the underlying channel in order to report the
+ device state of an agent channel more accurately. The problem
+ with the logic here is that it makes no sense to use this for
+ always-on agents. If the agent is logged in, then to the
+ underlying channel, the agent will always appear to be "in use,"
+ no matter if the agent is on a call or not. The reason is that to
+ the underlying channel, the channel is currently in use on a call
+ to the AgentLogin application. The most common cause that I found
+ for this issue to occur was for a SIP channel to be the
+ underlying channel type for an Agent channel. If the SIP phone
+ re-registers, then the registration will cause the device state
+ core to query the device state of the SIP channel. Since the SIP
+ channel is in use, the Agent channel would also inherit this
+ status. Once the agent channel was set to "in use" there was no
+ way that the device state could change on that channel unless the
+ agent logged out. The solution for this problem is a bit
+ different in 1.4 than it is in the other branches. In 1.4, there
+ will be a one-line fix to make sure that only callback agents
+ will inherit device state from their underlying channel type. For
+ the other branches of Asterisk, since callback support has been
+ removed, there is also no need for device state inheritance in
+ chan_agent, so I will simply be removing it from the code. In
+ addition, the 1.4 source is getting a new comment to help the
+ next person who edits chan_agent.c. I'm adding a comment that a
+ agent_pvt's loginchan field may be used to determine if the agent
+ is a callback agent or not. (closes issue #14173) Reported by:
+ nathan Patches: 14173.patch uploaded by putnopvut (license 60)
+ Tested by: nathan, aramirez ........
+
+ * /, main/slinfactory.c: Merged revisions 171621 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan
+ 2009) | 18 lines Prevent a crash from occurring when a jitter
+ buffer interpolated frame is removed from a slinfactory
+ slinfactory used the "samples" field of an ast_frame in order to
+ determine the amount of data contained within the frame. In
+ certain cases, such as jitter buffer interpolated frames, the
+ frame would have a non-zero value for "samples" but have NULL
+ "data" This caused a problem when a memcpy call in
+ ast_slinfactory_read would attempt to access invalid memory. The
+ solution in use here is to never feed frames into the slinfactory
+ if they have NULL "data" (closes issue #13116) Reported by:
+ aragon Patches: 13116.diff uploaded by putnopvut (license 60)
+ ........
+
+ * apps/app_queue.c: Fix queue crashes that would occur after the
+ calling channel was masqueraded. The data passed to the
+ end_bridge_callback was assumed to be data which was still
+ stack'd. The problem was that with some call features, attended
+ transfers in particular, a new bridge thread is started once the
+ feature completes, meaning that when the end_bridge_callback is
+ called, the end_bridge_callback_data was invalid. To fix this
+ problem, there are two measures taken 1. Instead of pointing to
+ stacked data, we now used heap-allocated data for passing to the
+ end_bridge_callback in app_queue 2. Since bridges can end
+ multiple times on a single logical call, we wait until the final
+ bridge is broken to actually set any queue variables. This is
+ accomplished through reference-counting and the use of an
+ end_bridge_callback_data_fixup function in app_queue.c (closes
+ issue #14260) Reported by: ccesario Patches: 14260.patch uploaded
+ by putnopvut (license 60) Tested by: ccesario
+
+2009-01-27 15:23 +0000 [r171558] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: Handle new VMWI ioctl structure (Now there
+ are two VMWI ioctl calls.) (issue #14104) Reported by: alecdavis
+ Tested by: dbailey
+
+2009-01-27 15:00 +0000 [r171263-171528] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Solving the same issue, but a bit
+ different in trunk... Merged revisions 171527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13
+ lines Use the same branch tag in CANCEL as in INVITE Originally
+ putnopvut implemented some changes in revision 142079 that
+ according to the bug report seemed to have worked then, but
+ somehow fails now. I guess code, as humans, get old and forget
+ stuff. Anyway, this bug caused CANCEL not to work with picky
+ systems. Thanks Fredrik for pointing out where the bug in the SIP
+ messaging was. (closes issue #14346) Reported by: oej Patches:
+ bug14346.diff uploaded by oej (license 306) Tested by: oej
+ ........
+
+ * channels/chan_sip.c: Moving generic setting to friends
+
+ * channels/chan_sip.c: Continue to move variables into the sip_cfg
+ structure to make them easier to handle in the future as a group
+ of settings for a group of devices. At some point, I want one
+ sip_cfg per domain handled, so we can have "group" settings.
+
+ * channels/chan_sip.c: Just moving around variable declarations so
+ that we have all globals in the same place. Default setting is
+ set before we activate the channel or at reloads, not where we
+ declare the variable.
+
+ * /, channels/chan_sip.c: Merged revisions 171264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9
+ lines Don't retransmit 401 on REGISTER requests when
+ alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000
+ Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000 ........
+
+ * main/channel.c: Add extensions and context on manager event when
+ new channel is created.
+
+2009-01-25 23:58 +0000 [r171188] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_oss.c: Merged revisions 171187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009)
+ | 6 lines Correctly track the hookstate (closes issue #13686)
+ Reported by: itiliti Patches: 20081013__bug13686.diff.txt
+ uploaded by Corydon76 (license 14) ........
+
+2009-01-25 16:50 +0000 [r171043-171081] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: dont segfault when a MWI event occurs on
+ a line without a registered device
+
+ * configs/skinny.conf.sample: Make the sample skinny.conf work
+ (closes issue #14325) Reported by: DEA Patches:
+ skinny.conf.sample-trunk.txt uploaded by DEA (license 3)
+
+2009-01-25 13:35 +0000 [r170980] Sean Bright <sean.bright@gmail.com>
+
+ * /, apps/app_page.c: Merged revisions 170979 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan
+ 2009) | 9 lines Resolve a logic error that was causing Page() to
+ crash when more than one channel was specified. (closes issue
+ #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
+ uploaded by seanbright (license 71) Tested by: kc0bvu ........
+
+2009-01-25 02:49 +0000 [r170902-170943] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/utils.h: Change ARRAY_LEN() to be more C++ safe.
+ When the second part of this macro is written as 0[a] instead of
+ a[0], it will force a failure if the macro is used on a C++
+ object that overloads the [] operator.
+
+ * res/res_agi.c: Add a todo to finish the XML docs in this module
+
+2009-01-24 13:55 +0000 [r170837] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/res_odbc.conf.sample: Merged revisions 170836 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009)
+ | 2 lines Remove superfluous implementation note (closes issue
+ #14319) ........
+
+2009-01-23 23:10 +0000 [r170794] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/tex/Makefile: Fix asterisk.pdf generation if branch name has
+ an underscore in it.
+
+2009-01-23 22:58 +0000 [r170790] Russell Bryant <russell@digium.com>
+
+ * doc/tex/Makefile: Don't blow up if a branch name has an
+ underscore in it
+
+2009-01-23 20:56 +0000 [r170677-170720] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/res_odbc.conf.sample: Merged revisions 170719 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan
+ 2009) | 8 lines Add notes to the idlecheck explanation in
+ res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000
+ Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by
+ klaus3000 (license 65) ........
+
+ * /, contrib/i18n.testsuite.conf: Merged revisions 170671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan
+ 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use
+ deprecated syntax * Convert Wait,1 to Wait(1) * Convert
+ SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
+ priorities beyond the first Also added test for Chinese numbers,
+ too. (closes issue #14320) Reported by: dant Patches:
+ i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
+ 670) ........
+
+2009-01-23 20:18 +0000 [r170652] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 170648 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4
+ lines When a channel is answered make sure any indications
+ currently playing stop. Usually the phone would do this but if
+ the channel was already answered then they are being generated by
+ Asterisk and we darn well need to stop them. (closes issue
+ #14249) Reported by: RadicAlish ........
+
+2009-01-23 19:25 +0000 [r170608] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 170588 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23
+ Jan 2009) | 2 lines Additions to AST-2009-001 ........
+
+2009-01-23 19:09 +0000 [r170505-170569] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 170568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4
+ lines When a call is forwarded stop any active indications. The
+ new channel will provide an indication, if need be, itself.
+ (closes issue #14310) Reported by: RadicAlish ........
+
+ * /, channels/chan_sip.c: Merged revisions 170504 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4
+ lines Use the on hold flag to see if the call is on hold or not.
+ It is possible that our address for them will still be valid even
+ though they are on hold. (closes issue #14295) Reported by:
+ klaus3000 ........
+
+2009-01-23 17:46 +0000 [r170501] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_h323.c: let's use SENTINEL where needed
+
+2009-01-23 17:32 +0000 [r170498] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Reset the ast_str used for escape
+ substitution. We need to do this since it is a thread local
+ variable that may contain the value of a previous substitution.
+ (closes issue #14312) Reported by: pj
+
+2009-01-23 17:03 +0000 [r170463] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: We should not do restart messages if we're
+ in PTMP mode
+
+2009-01-23 16:57 +0000 [r170460] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Dont clear the display of skinny phones
+ when not needed. (closes issue #13182) Reported by: pj Patches:
+ 2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak
+ (license 7) Tested by: mvanbaak, pj
+
+2009-01-23 16:35 +0000 [r170457] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: MWI messages included in CID spill was not
+ being properly handled and prevented the call from being
+ processed (issue #14313) Reported by: seandarcy Tested by:
+ dbailey
+
+2009-01-23 15:44 +0000 [r170393] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 170392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan
+ 2009) | 28 lines Fix broken call pickup There was a subtle change
+ in ast_do_masquerade which resulted in failed attempts to pickup
+ calls. The problem was that the value of the AST_FLAG_OUTGOING
+ flag was copied from the clone to the original channel. In the
+ case of call pickup, this meant that the AST_FLAG_OUTGOING flag
+ ended up being cleared on the channel that was attempting to
+ execute the pickup. Because this flag was not set, when ast_read
+ came across an answer frame, it ignored it. The result of this
+ was that the calling channel was never properly answered. This
+ fix changes the behavior in ast_do_masquerade to set the flags on
+ the original channel to the union of the flags on the clone
+ channel. This way, if the AST_FLAG_OUTGOING flag is set on either
+ of the two channels involved in the masquerade, the resulting
+ channel will have the flag set as well. (closes issue #14206)
+ Reported by: francesco_r Patches: 14206.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut
+ ........
+
+2009-01-22 23:23 +0000 [r170351] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: Make sure we don't set the channel to be
+ inalarm for a D-channel drop on PTMP connections
+
+2009-01-22 21:25 +0000 [r170307] Tilghman Lesher <tlesher@digium.com>
+
+ * main/abstract_jb.c: Create logfile safely. (closes issue #14160)
+ Reported by: tzafrir Patches: 20090104__bug14160.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2009-01-22 20:04 +0000 [r170240] Joshua Colp <jcolp@digium.com>
+
+ * /, main/rtp.c: Merged revisions 170239 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7
+ lines Don't crash if RTCP is not enabled on an RTP structure but
+ statistics are output. (closes issue #14234) Reported by: jcovert
+ Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
+ rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........
+
+2009-01-22 17:19 +0000 [r170165] Tilghman Lesher <tlesher@digium.com>
+
+ * /, pbx/pbx_config.c: Merged revisions 170158 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009)
+ | 6 lines Allow global variables after substitution to be as long
+ as other variables. (closes issue #14263) Reported by: markd
+ Patches: 20090120__bug14263.diff.txt uploaded by Corydon76
+ (license 14) ........
+
+2009-01-22 16:52 +0000 [r170148] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 170147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4
+ lines If we are unable to request a DAHDI pseudo channel and we
+ are using the user introduction without review option make sure
+ it gets unset so other code does not blindly assume a DAHDI
+ pseudo channel exists. (closes issue #14282) Reported by:
+ cheesegrits ........
+
+2009-01-22 15:49 +0000 [r170112] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: change VMWI to
+ use new DAHDI_VMWI ioctl call. Change configure script to detect
+ the new ioctl call data structure. (issue #14104) Reported by:
+ alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded
+ by dbailey (license ) Tested by: alecdavis, dbailey
+
+2009-01-22 15:14 +0000 [r170047-170051] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, /: Merged revisions 170050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6
+ lines Do a string comparison instead of pointer comparison since
+ some people specify the context they are actually in as an
+ argument to get around some funkiness. (closes issue #14011)
+ Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga
+ (license 665) ........
+
+ * apps/app_parkandannounce.c: Clear the autoloop flag when parsing
+ and setting the context/extension/priority to go back to. When
+ the channel executes a PBX again we want it to start out at the
+ point we explicitly say and at that point it will not yet be
+ doing autoloop. (closes issue #14304) Reported by: jcovert
+
+2009-01-22 02:10 +0000 [r170007] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: * Adjust some conditionals to balance
+ curly braces. * Other minor changes.
+
+2009-01-22 00:44 +0000 [r169944] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 169943 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009)
+ | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really
+ wanted to ask is whether autoconf detected a static initializer
+ value. This fixes rwlocks on all such platforms (mainly, Mac OS
+ X). (closes issue #13767) Reported by: jcovert Patches:
+ 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: jcovert, Corydon76 ........
+
+2009-01-22 00:23 +0000 [r169910] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Whitespace changes only
+
+2009-01-21 23:25 +0000 [r169869] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, /: Merged revisions 169867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4
+ lines Read lock the contexts to maintain the locking order when
+ we are notified that the state of a device has changed. (closes
+ issue #13839) Reported by: mcallist ........
+
+2009-01-21 23:20 +0000 [r169794-169866] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c: Test commit for test issue #14303
+
+ * main/say.c: Fix a crash when saying certain numbers in Chinese
+ This commit fixes a crash that was occurring when attempting to
+ say a number between 10000 and 100000 due to dividing by 0. This
+ also removes some places where a "zero" is spoken when it should
+ not be. (closes issue #14291) Reported by: dant Patches:
+ say.c-14291.diff uploaded by dant (license 670) Tested by: dant
+
+2009-01-21 22:04 +0000 [r169793] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/tex/extensions.tex: remove duplicated sentence.
+
+2009-01-21 21:53 +0000 [r169791] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Further fix some oddities in sip show users
+ and sip show peers logic ccesario on IRC pointed out that his sip
+ peers were not displayed properly when he would issue the command
+ "sip show peers." The problem was that the onlymatchonip field
+ was used to determine if the endpoint was a "peer" or "user." The
+ tricky part is that a "friend" is supposed to be treated as both
+ a "user" and a "peer" but the logic would not allow "friends" to
+ show up as "peers" since onlymatchonip was set to FALSE for
+ friends. I have modified the sip_peer structure to more
+ explicitly keep track of what type endpoint it is so that the
+ various manager and CLI commands will display the expected
+ information Reported by ccesario via IRC Tested by ccesario
+
+2009-01-21 21:03 +0000 [r169723] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 169722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009)
+ | 8 lines Extra NULLs in the output cause some terminal types to
+ abort in the middle of a color code, causing terminal weirdness.
+ (closes issue #14130) Reported by: coolmig Patches:
+ 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, coolmig ........
+
+2009-01-21 17:21 +0000 [r169673] Steve Murphy <murf@digium.com>
+
+ * utils/refcounter.c: This patch corrects a segfault reported in
+ 14289, due to a null ptr being refd. Yes, seanbright is right in
+ the bug comments, that is the fix. Sorry for this oversight; I
+ guess my personal usage didn't have this happen! murf (closes
+ issue #14289) Reported by: jamesgolovich
+
+2009-01-21 10:49 +0000 [r169620-169625] Russell Bryant <russell@digium.com>
+
+ * /: Remove properties that erroneously got merged into trunk
+
+ * main/tcptls.c: Fix a regression in TCP support. This patch fixes
+ a problem that caused chan_sip to think that every open TCP
+ session was to a remote address of 0.0.0.0:0. (closes issue
+ #14287) Reported by: jamesgolovich Patches: bug-14287.diff.txt
+ uploaded by jamesgolovich (license 176)
+
+2009-01-21 00:33 +0000 [r169557-169611] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix device state parsing issues for channel
+ names with multiple slashes The fix being applied is a bit
+ different for trunk and the 1.6.X branches. For trunk, we only
+ wish to strip off the characters beyond the second slash if the
+ channel is a Local channel (i.e. we are removing the /n from the
+ device name). Other channel technologies with multiple slashes
+ (e.g. DAHDI) need the information after the second slash in order
+ to get the proper device state information. In addition to this
+ fix, the 1.6.X branches are receiving a much more important fix
+ as well. The problem in 1.6.X is that the member's device name
+ was being directly changed instead of having a copy changed. This
+ meant that we would strip off the second slash and trailing
+ characters and then leave the member's device name like that
+ permanently thereafter. (closes issue #14014) Reported by:
+ kebl0155 Patches: 14014_number2.patch uploaded by putnopvut
+ (license 60) Tested by: kebl0155
+
+ * apps/app_queue.c: Use the default timeout for a queue instead of
+ -1 (closes issue #14272) Reported by: timking
+
+ * /, channels/chan_sip.c: Convert the character pointers in a
+ sip_request to be pointer offsets When an ast_str expands to hold
+ more data, any pointers that were pointing to the data prior to
+ the expansion will be pointing at invalid memory. This change
+ makes such pointers used in chan_sip.c instead be offsets from
+ the beginning of the string so that the same math may be applied
+ no matter where in memory the string resides. To help ease this
+ transition, a macro called REQ_OFFSET_TO_STR has been added to
+ chan_sip.c so that given a sip_request and an offset, the string
+ at that offset is returned. (closes issue #14220) Reported by:
+ riksta Tested by: putnopvut Review
+ http://reviewboard.digium.com/r/126/
+
+2009-01-20 19:22 +0000 [r169486-169510] Terry Wilson <twilson@digium.com>
+
+ * main/features.c: Make a proper builtin attended transfer to
+ parking work This is an ugly hack from 1.4 that allows the
+ timeout callback from a parked call to use the right channel name
+ for the callback when the park is done with a builtin attended
+ transfer (that isn't completed early). This hasn't ever worked in
+ trunk and no one has complained yet, so eh.
+
+ * /, main/features.c: Merged revisions 169485 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009)
+ | 6 lines Don't play audio to the channel if we've masqueraded
+ (closes issue #14066) Reported by: bluefox Tested by:
+ otherwiseguy, bluefox ........
+
+2009-01-19 21:42 +0000 [r169438] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/res_odbc.h, funcs/func_odbc.c,
+ include/asterisk/strings.h, res/res_odbc.c: ast_str_SQLGetData is
+ *not* part of the ast_str API, it's part of the ast_odbc API and
+ just happens to use an ast_str as the buffer; move all of it to
+ res_odbc.c and res_odbc.h, renaming appropriately along the way
+ fix some minor coding style issues in strings.h and add some
+ attribute_pure annotations to functions in the ast_str API
+
+2009-01-19 20:14 +0000 [r169367-169369] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: fix assignment in swapmode plug. Spotted and fix
+ provided by ys (closes issue #14129) Reported by: ys Tested by:
+ ys
+
+ * channels/chan_skinny.c: Redo the event-based MWI in chan_skinny.
+ Dan saw regular segfaults with the old implementation and rewrote
+ it to make it really eventbased. I altered it to be trunk
+ compatible and wedhorn gave some feedback and ideas how to make
+ it even better. (closes issue #13821) Reported by: DEA Patches:
+ chan_skinny-mwi-events.txt uploaded by DEA (license 3) Tested by:
+ mvanbaak, DEA "no probs by me" from wedhorn
+
+2009-01-19 20:05 +0000 [r169365] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /, apps/app_userevent.c: Merged revisions 169364
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009)
+ | 4 lines Truncate userevents at the end of a line, when the
+ command exceeds the buffer. (closes issue #14278) Reported by:
+ fnordian ........
+
+2009-01-19 18:36 +0000 [r169327] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: Make asterisk compile on non-amd64 versions of
+ OpenBSD. The HW_PHYSMEM64 is only available in latest OpenBSD
+ and/or amd64 versions of OpenBSD. Use HW_PHYSMEM when
+ HW_PHYSMEM64 is not available. (closes issue #14129) Reported by:
+ ys Patches: 2009011600_physmem64.diff.txt uploaded by mvanbaak
+ (license 7) Tested by: mvanbaak, jtodd
+
+2009-01-19 18:22 +0000 [r169277-169325] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: Get rid of magic number and replace with
+ DAHDI_VMWI_NUMBER_MASK when determining the number of messages
+ pending for MWI call
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
+ enhanced MWI generation to take advantage of new dahdi line
+ reversal MWI ability. (closes issue #14104) Reported by:
+ alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey
+ (license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis, dbailey
+
+2009-01-19 15:54 +0000 [r169211] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 169210 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon,
+ 19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a
+ potential NULL pointer dereference Move the check for if both
+ channels on a local_pvt have generators to below where p->chan is
+ checked for NULLity (NULLness?). This prevents a crash from
+ occurring if p->chan is NULL. (closes issue #14189) Reported by:
+ sascha Patches: 14189.patch uploaded by putnopvut (license 60)
+ Tested by: sascha ........
+
+2009-01-17 18:26 +0000 [r169153] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
+ discriminator for when ring pulse alert signal is used to preface
+ MWI spills This prevents the situation when MWI messages are
+ added to caller ID spills causing the channel to be hung up
+
+2009-01-17 02:52 +0000 [r169116] Sean Bright <sean.bright@gmail.com>
+
+ * pbx/pbx_dundi.c: Change intializer types. Found while working on
+ asterisk-cpp. I have a new favorite error message from g++:
+ pbx_dundi.c:4580: sorry, unimplemented: non-trivial designated
+ initializers not supported I like it when compilers are
+ apologetic.
+
+2009-01-17 01:56 +0000 [r169044-169080] Terry Wilson <twilson@digium.com>
+
+ * main/tcptls.c, main/http.c, include/asterisk/tcptls.h: Fix
+ qualify for TCP peer (closes issue #14192) Reported by:
+ pabelanger Patches: asterisk-bug14192.diff.txt uploaded by
+ jamesgolovich (license 176) Tested by: jamesgolovich
+
+ * channels/chan_sip.c: Fix port :0 added to SIP INVITE URI when
+ outboundproxy used (closes issue #14233) Reported by: chris-mac
+ Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich
+ (license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy
+
+2009-01-16 22:43 +0000 [r168976] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168975 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan
+ 2009) | 18 lines Account for possible NULL pointer when we
+ receive a 408 in response to a REGISTER It may be that by the
+ time we receive a reply to a REGISTER request, the attempt has
+ timed out and thus the registry structure pointed to by the
+ corresponding sip_pvt has gone away. This situation was handled
+ properly for a 200 OK response, but the 408 case assumed that the
+ sip_registry struct was non-NULL, thus potentially causing a
+ crash This commit fixes this assumption and prints out a message
+ to the console if we should receive a late 408 response to a
+ REGISTER (closes issue #14211) Reported by: aborghi Patches:
+ 14211.diff uploaded by putnopvut (license 60) Tested by: aborghi
+ ........
+
+2009-01-16 22:16 +0000 [r168941] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 168716 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009)
+ | 12 lines Convert call to park_call_full to
+ masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
+ return value, we need to use masqueraded parking, otherwise we
+ will try to call ast_hangup() in __pbx_run() and in
+ do_parking_thread() and then promptly crash. (closes issue
+ #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
+ issue #14228) Reported by: kobaz Tested by: otherwiseguy ........
+
+2009-01-16 19:54 +0000 [r168898] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_timing_timerfd.c: Fix a logic error that occur when using
+ the timerfd interface This sequence of events posed a problem
+ timerfd_timer_open timerfd_timer_enable_continuous
+ timerfd_timer_set_rate timerfd_timer_disable_continuous The
+ reason was that the timing module was written under the
+ assumption that timerfd_timer_set_rate would not be called
+ between enabling and disabling continuous mode. What happened in
+ this situation was that timerfd_timer_enable_continuous saved off
+ our previously set timer (in this situation a 0 timer, meaning it
+ never runs out). Then timerfd_timer_disable_continuous would
+ restore this 0 timer, even though it logically should set the
+ timer to be whatever was set in timerfd_timer_set_rate. Now the
+ behavior in timerfd_timer_set_rate is to overwrite the saved
+ timer that may or may not have been set in
+ timerfd_timer_enable_continuous. Even if
+ timerfd_timer_enable_continuous has not been previously called,
+ this will not harm the operation. Thanks to Terry Wilson for
+ discovering the problem and giving me a really great debug
+ capture that pointed out the problem clearly
+
+2009-01-16 18:49 +0000 [r168832] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c, include/asterisk/say.h, apps/app_voicemail.c:
+ Merged revisions 168828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009)
+ | 6 lines Fix the conjugation of Russian and Ukrainian languages.
+ (related to issue #12475) Reported by: chappell Patches:
+ vm_multilang.patch uploaded by chappell (license 8) ........
+
+2009-01-16 17:09 +0000 [r168759-168760] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Fix a spelling mistake.
+
+ * channels/chan_misdn.c: build in dev mode
+
+2009-01-16 00:34 +0000 [r168737-168746] Steve Murphy <murf@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 168745 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) |
+ 14 lines This patch fixes a problem where a goto (or jump, in
+ this case) fails a consistency check because it can't find a
+ matching extension. The problem was a missing instruction to end
+ the range notation in the code where it converts the pattern into
+ a regex and uses the regex code to determine the match. I tested
+ using the AEL code the user supplied, and now, the consistency
+ check passes. (closes issue #14141) Reported by: dimas ........
+
+ * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: This patch
+ allows null args in ast_expr2 func calls, and fixes commas being
+ converted to pipes, which was 1.4 type stuff. If the user says
+ count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); then it
+ won't complain about the empty arg (c,,...) and fabled's patch
+ won't let it swap the commas for pipes. Ran it thru my dialplan
+ and no complaints. (closes issue #14169) Reported by: fabled
+ Patches: function-argument-separator-fix.diff uploaded by fabled
+ (license 448)
+
+2009-01-15 20:18 +0000 [r168734] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_config_odbc.c, build_tools/menuselect-deps.in, configure,
+ funcs/func_odbc.c, configure.ac, cdr/cdr_adaptive_odbc.c,
+ cdr/cdr_odbc.c, makeopts.in, res/res_odbc.c,
+ apps/app_voicemail.c: remove the PBX_ODBC logic from the
+ configure script, and add GENERIC_ODCB logic that includes
+ copying the relevant LIB and INCLUDE data from either UnixODBC or
+ iODBC, based on which was found; if both were found, prefer
+ UnixODBC this stops modules from being linked against both sets
+ of libraries on systems that have both installed
+
+2009-01-15 20:00 +0000 [r168725-168732] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add missing brace
+
+ * channels/chan_sip.c: Fix the compactheaders option in sip.conf
+
+ * channels/chan_sip.c: Remove an unneeded condition for line
+ addition to a SIP request/response In Asterisk 1.4 and 1.6.0, the
+ sip_request structure had a statically allocated buffer to hold
+ the text of the request. There was a check in the add_line
+ function to not attempt to write the line into the buffer if we
+ did not have room for it. In trunk and Asterisk versions starting
+ with 1.6.1, an expandable ast_str structure is used to hold the
+ text. Since it may grow to fit an arbitrarily sized string, this
+ check in add_line is no longer valid. I found this oddity while
+ attempting to fix issue #14220; however, I do not believe that
+ this is the fix for that issue since the output supplied by the
+ reporter did not contain the warning message that would be
+ printed had this condition been satisfied.
+
+2009-01-15 18:47 +0000 [r168722] Olle Johansson <oej@edvina.net>
+
+ * /, configs/extconfig.conf.sample: Merged revisions 168721 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2
+ lines Meetme actually has realtime but wasn't documented ........
+
+2009-01-15 18:39 +0000 [r168719] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/strings.h: Resolve issue with negative vs
+ non-negative length parameters. (closes issue #14245) Reported
+ by: dveiga
+
+2009-01-15 18:08 +0000 [r168711-168712] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Make sure that we have the same terminology
+ in sip.conf.sample and the source code warning. Thanks Nick Lewis
+ for pointing this out in the bug tracker.
+
+ * configs/sip.conf.sample: Clarify some misunderstandings and make
+ it even more clear that you can refer to a peer in the register=
+ line.
+
+2009-01-15 15:33 +0000 [r168705] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_meetme.c: Add a missing unlock and properly handle the
+ 'maxusers' setting on MeetMe conferences. We were using the 'user
+ number' field to compare against the maximum allowed users, which
+ works assuming users with lower user numbers didn't leave the
+ conference. (closes issue #14117) Reported by: sergedevorop
+ Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright
+ (license 71) Tested by: sergedevorop
+
+2009-01-15 13:37 +0000 [r168636-168639] Olle Johansson <oej@edvina.net>
+
+ * CREDITS, CHANGES: Related to issue #14246 Update changes for
+ SIPRemoveHeader()
+
+ * channels/chan_sip.c: Add capability to remove added SIP headers
+ *before* INVITE is generated. (closes issue #14246) Reported by:
+ klaus3000 Patches: 2patch_chan_sip_SIPRemoveHeader_trunk.txt
+ uploaded by klaus3000 (license 65)
+
+ * apps/app_queue.c: Add support for setting the Reason header when
+ cancelling a call in the queue because someone else answered.
+ Previously, only dial() was supported. EDV-102
+
+2009-01-15 00:14 +0000 [r168629] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 168628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan
+ 2009) | 16 lines Fix some crashes from bad datastore handling in
+ app_queue.c * The queue_transfer_fixup function was searching for
+ and removing the datastore from the incorrect channel, so this
+ was fixed. * Most datastore operations regarding the
+ queue_transfer datastore were being done without the channel
+ locked, so proper channel locking was added, too. (closes issue
+ #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
+ putnopvut (license 60) Tested by: ZX81, festr ........
+
+2009-01-14 23:10 +0000 [r168626] Sean Bright <sean.bright@gmail.com>
+
+ * main/cli.c: Don't crash when typing 'core set verbose' or 'core
+ set debug' by themselves. (closes issue #14219) Reported by:
+ jamesgolovich Patches: asterisk-setverbosecrash.diff.txt uploaded
+ by jamesgolovich (license 176)
+
+2009-01-14 21:51 +0000 [r168623] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/misdn/isdn_lib.c: Merged revisions 168622 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009)
+ | 4 lines * Fixed create_process() allocation of process ID
+ values. The allocated process IDs could overflow their respective
+ NT and TE fields. Affects outgoing calls. ........
+
+2009-01-14 21:19 +0000 [r168619] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: This fixes a problem where MWI FSK spills
+ were being injected onto off hook fxs lines. (closes issue
+ #14143) Reported by: alecdavis Patches:
+ chan_dahdi-14143.patch.txt uploaded by dbailey (license ) Tested
+ by: alecdavis
+
+2009-01-14 20:58 +0000 [r168615] Sean Bright <sean.bright@gmail.com>
+
+ * /, contrib/scripts/autosupport: Merged revisions 168614 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan
+ 2009) | 9 lines Update autosupport script to supply info for both
+ Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x
+ and trunk instead of zttest. (closes issue #14132) Reported by:
+ dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by
+ dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded
+ by dsedivec (license 638) ........
+
+2009-01-14 20:51 +0000 [r168613] Steve Murphy <murf@digium.com>
+
+ * /, apps/app_page.c: Merged revisions 168608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1
+ line app_page was failing to compile in dev-mode on my gcc-4.2.4
+ system. This change gets rid of the warning. ........
+
+2009-01-14 20:13 +0000 [r168610] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Restore the "sip show users" and "sip show
+ user" CLI commands (closes issue #14180) Reported by: amorsen
+ Patches: sip_show_users_161v3.diff uploaded by putnopvut (license
+ 60) Tested by: blitzrage, amorsen
+
+2009-01-14 19:36 +0000 [r168609] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: Fix compilation on FreeBSD and OSX This started
+ as work to fix the 'core show sysinfo' CLI command but while
+ working on it oej pointed out that read_credentials did not
+ compile neither. So while being there, fix that as well. Thanks
+ for all the testing oej! (closes issue #14129) Reported by: ys
+ Tested by: oej, mvanbaak
+
+2009-01-14 19:11 +0000 [r168601-168604] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c, /: Merged revisions 168603 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009)
+ | 7 lines Don't read into a buffer without first checking if a
+ value is beyond the end. (closes issue #13600) Reported by: atis
+ Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
+ (license 14) Tested by: atis ........
+
+ * channels/chan_misdn.c: Mostly spacing changes; no functionality
+ change at all.
+
+2009-01-14 02:00 +0000 [r168594] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_page.c: Merged revisions 168593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009)
+ | 20 lines Don't overflow when paging more than 128 extensions
+ The number of available slots for calls in app_page was hardcoded
+ to 128. Proper bounds checking was not in place to enforce this
+ limit, so if more than 128 extensions were passed to the Page()
+ app, Asterisk would crash. This patch instead dynamically
+ allocates memory for the ast_dial structures and removes the
+ (non-functional) arbitrary limit. This issue would have special
+ importance to anyone who is dynamically creating the argument
+ passed to the Page application and allowing more than 128
+ extensions to be added by an outside user via some external
+ interface. The patch posted by a_villacis was slightly modified
+ for some coding guidelines and other cleanups. Thanks,
+ a_villacis! (closes issue #14217) Reported by: a_villacis
+ Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
+ uploaded by a (license 660) Tested by: otherwiseguy ........
+
+2009-01-13 23:57 +0000 [r168591] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_misdn.c: Janitor patch for chan_misdn (make channel
+ variable access safe) (closes issue #12887) Reported by: pputman
+ Patches: chan_misdn_threadsafe.patch uploaded by pputman (license
+ 81)
+
+2009-01-13 23:05 +0000 [r168585-168588] Terry Wilson <twilson@digium.com>
+
+ * res/res_http_post.c: Fully overwrite a same-named file when
+ uploading (closes issue #14190) Reported by: timking
+
+ * Makefile, include/asterisk/options.h, main/asterisk.c: Add option
+ to hide console connect messages (closes issue #14222) Reported
+ by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded
+ by jamesgolovich (license 176) Tested by: otherwiseguy
+
+2009-01-13 22:30 +0000 [r168579] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Clarify a message that app_queue prints and
+ change to a debug-level message The "No one is answering..."
+ verbose message contained 3 numbers that were not explained in
+ any way to whoever was viewing the message. It is more helpful
+ now since the message explains what the numbers mean. Also, the
+ message has been downgraded to "DEBUG" level. (closes issue
+ #14172) Reported by: caio1982 Patches: queue_answering_debug.diff
+ uploaded by caio1982 (license 22)
+
+2009-01-13 22:22 +0000 [r168578] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168551 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009)
+ | 7 lines Don't pass a value with a side effect to a macro
+ (closes issue #14176) Reported by: paraeco Patches:
+ chan_sip.c.diff uploaded by paraeco (license 658) ........
+
+2009-01-13 21:18 +0000 [r168575] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Allow
+ specifying a port number in the user portion of a register =>
+ line in sip.conf With this commit, a register => line in sip.conf
+ may contain a port number in the "user" section of the line.
+ Please see CHANGES and sip.conf.sample for more details regarding
+ this. (closes issue #14198) Reported by: Nick_Lewis Patches:
+ chan_sip.c-domainport2.patch uploaded by Nick (license 657)
+ Tested by: Nick_Lewis
+
+2009-01-13 19:22 +0000 [r168562] Russell Bryant <russell@digium.com>
+
+ * channels/chan_unistim.c, main/pbx.c, apps/app_read.c, /,
+ include/asterisk/indications.h, apps/app_readexten.c,
+ apps/app_disa.c, include/asterisk/channel.h, main/indications.c,
+ main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
+ funcs/func_channel.c, main/app.c, res/snmp/agent.c,
+ res/res_indications.c: Merged revisions 168561 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009)
+ | 2 lines Revert unnecessary indications API change from rev
+ 122314 ........
+
+2009-01-13 17:51 +0000 [r168547] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_logic.c: Merged revisions 168546 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009)
+ | 6 lines If either conditional is NULL, don't try copying it.
+ (closes issue #14226) Reported by: caspy Patches:
+ 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
+ ........
+
+2009-01-13 16:02 +0000 [r168539] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * main/taskprocessor.c: correct a CLI description
+
+2009-01-12 23:45 +0000 [r168526] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_alsa.c: Merged revisions 167095 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31
+ Dec 2008) | 5 lines Repeat attempts to write when we receive
+ -EAGAIN from the driver, as detailed in the ALSA sample code (see
+ http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
+ Reported by: Jerry Geis (via the -users list) Fixed by: me
+ (license 14) ........
+
+2009-01-12 23:12 +0000 [r168523] Mark Michelson <mmichelson@digium.com>
+
+ * main/srv.c: bump the verbosity of a message in srv.c up by one.
+ It used to be at this level prior to a large patch merge which
+ converted ast_verbose calls to ast_verb (closes issue #14221)
+ Reported by: jcovert Patches: srv.c.patch uploaded by jcovert
+ (license 551)
+
+2009-01-12 23:06 +0000 [r168522] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/app.c: Some platforms (notably, the BSDs) have a more
+ efficient implementation called closefrom(3).
+
+2009-01-12 21:51 +0000 [r168508-168517] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 168516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009)
+ | 5 lines (closes issue #13881) Reported by: hoowa Update the app
+ CDR field for AGI commands that are not executing an application
+ via "exec". ........
+
+ * /, channels/chan_agent.c: Merged revisions 168507 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12
+ Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG
+ Tested by: denisgalvao This gits rid of the notion of an
+ owning_app allowing the request and hangup to be initiated by
+ different threads. Originating from an active agent channel
+ requires this. The implementation primarily changes __login_exec
+ to wait on a condition variable rather than a lock. Review:
+ http://reviewboard.digium.com/r/35/ ........
+
+2009-01-12 16:31 +0000 [r168497] Olle Johansson <oej@edvina.net>
+
+ * apps/app_minivm.c: Better to use the proper app name
+
+2009-01-12 15:00 +0000 [r168485] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Merged revisions 168482 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan
+ 2009) | 5 lines I am reverting the fix made in revision 168128
+ (and its upward merges) after being contacted by Olle Johansson
+ and being shown how this fix is incorrect. Thanks to Olle for
+ clearing this up for me. ........
+
+2009-01-12 14:57 +0000 [r168481] Russell Bryant <russell@digium.com>
+
+ * /, configs/indications.conf.sample: Merged revisions 168480 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009)
+ | 2 lines s/ringdance/ringcadence/ for Bulgaria ........
+
+2009-01-12 14:35 +0000 [r168479] Olle Johansson <oej@edvina.net>
+
+ * main/asterisk.c: Don't include swap.h unless we have swapctl
+
+2009-01-10 01:42 +0000 [r168334] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: sizeof for a stringfield is 4. Kinda low for
+ reconstructing a field value.
+
+2009-01-09 23:16 +0000 [r168270] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 168267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan
+ 2009) | 1 line update to use new sound file packages that include
+ license files ........
+
+2009-01-09 23:15 +0000 [r168269] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Spacing change
+
+2009-01-09 23:04 +0000 [r168265] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/sip_nat_settings (added), CHANGES: Add a script
+ to find out the correct settings for Asterisk behind NAT (closes
+ issue #13065) Reported by: tzafrir Patches: sip_nat_settings
+ uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by
+ mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and
+ moi
+
+2009-01-09 22:21 +0000 [r168200] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 168198 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09
+ Jan 2009) | 2 lines Make this compile for mvanbaak ........
+
+2009-01-09 21:53 +0000 [r168193] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168128 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan
+ 2009) | 13 lines Add check_via calls to more request handlers
+ INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
+ checking the topmost Via to determine where to send the response.
+ Adding check_via calls to those request handlers solves this.
+ (closes issue #13071) Reported by: baron Patches: check_via.patch
+ uploaded by baron (license 531) Tested by: baron ........
+
+2009-01-09 21:43 +0000 [r168192] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 168191 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09
+ Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 *
+ Miscellaneous doxygen comments added. ........
+
+2009-01-09 20:25 +0000 [r168142] Terry Wilson <twilson@digium.com>
+
+ * res/res_phoneprov.c: Don't leak memory if phoneprov.conf does not
+ exist (closes issue #14203) Reported by: jamesgolovich Patches:
+ asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich
+ (license 176)
+
+2009-01-09 18:30 +0000 [r168090] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c, include/asterisk/strings.h: When using ast_str
+ with a non-ast_str-enabled API, we need to update the buffer or
+ otherwise, we cannot use ast_str_strlen().
+
+2009-01-09 18:01 +0000 [r168014-168054] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/logger.c: Added a comment to logger.c about where to put
+ includes
+
+ * main/logger.c: Use ast_safe_system() in logger.c instead of
+ system() (closes issue #14194) Reported by: pabelanger
+
+2009-01-09 01:15 +0000 [r167935-167973] Terry Wilson <twilson@digium.com>
+
+ * apps/app_originate.c: Set ORIGINATE_STATUS instead of
+ OUTGOING_STATUS to match the documentation
+
+ * apps/app_dial.c: Set peer context and exten values so MACRO_EXTEN
+ and MACRO_CONTEXT will be set
+
+2009-01-08 22:37 +0000 [r167894] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 167840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009)
+ | 6 lines Don't truncate database results at 255 chars. (closes
+ issue #14069) Reported by: evandro Patches:
+ 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
+ ........
+
+2009-01-08 22:34 +0000 [r167888] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Revert chan_sip changes which were
+ accidentally committed in revision 167792
+
+2009-01-08 21:40 +0000 [r167835-167837] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_minivm.c: Fix variables to comply with documentation
+ changes
+
+ * apps/app_minivm.c: Textual changes, consistency in status
+ variable naming, and other minor bugs. (closes issue #13943)
+ Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded
+ by Marquis (license 32)
+
+2009-01-08 19:48 +0000 [r167792] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, CHANGES, apps/app_queue.c: Add the average
+ talk time for a queue This patch adds the functionality to
+ app_queue of calculating the average amount of time that channels
+ are bridged for a queue. The algorithm used to calculate the
+ average is the same exponential average currently used to
+ calculate the average holdtime. See the CHANGES file to see the
+ methods you may use to view this information. (closes issue
+ #13960) Reported by: coolmig Patches:
+ app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)
+
+2009-01-08 19:44 +0000 [r167791] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, CHANGES: Convert dialplan application
+ DAHDISendCallreroutingFacility to use commas. (closes issue
+ #13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded
+ by eliel (license 64)
+
+2009-01-08 17:26 +0000 [r167700-167720] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 167714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan
+ 2009) | 1 line remove an unnecessary argument to queue_request()
+ ........
+
+ * channels/chan_sip.c: Merged revisions 167620 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan
+ 2009) | 5 lines When a SIP request or response arrives for a
+ dialog with an associated Asterisk channel, and the lock on that
+ channel cannot be obtained because it is held by another thread,
+ instead of dropping the request/response, queue it for later
+ processing when the channel lock becomes available.
+ http://reviewboard.digium.com/r/123/ ........
+
+2009-01-08 14:27 +0000 [r167662] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/sip-friends.sql: Oops... fix the fieldname I
+ changed yesterday to be right.
+
+2009-01-07 22:36 +0000 [r167542-167569] Russell Bryant <russell@digium.com>
+
+ * /, main/file.c: Merged revisions 167566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009)
+ | 2 lines Fix the last couple of places where free() was
+ improperly used directly. ........
+
+ * /, main/file.c: Merged revisions 167554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009)
+ | 2 lines Don't fclose() the file early, the filestream
+ destructor will handle it. ........
+
+ * /, main/file.c: Merged revisions 167545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009)
+ | 2 lines Only try to close the file if one was actually opened
+ ........
+
+ * /, main/file.c: Merged revisions 167541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009)
+ | 4 lines Don't use free() directly. This caused a crash since
+ ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
+ #asterisk-dev ........
+
+2009-01-07 18:20 +0000 [r167478] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_followme.c: Answer the channel if it has not already
+ been answered and we've already found a valid profile for
+ followme. (closes issue #14140) Reported by: dimas Patches:
+ 14140.patch uploaded by dimas
+
+2009-01-07 18:18 +0000 [r167477] Leif Madsen <lmadsen@digium.com>
+
+ * configs/queues.conf.sample: Update queues.conf.sample
+ documentation. Update the queues.conf.sample documentation to
+ mention that you need to preload chan_local.so as well if you
+ plan on using Local channels for queue members, and you're
+ preloading pbx_config.so. (closes issue #14179) Reported by:
+ CrashHD Tested by: CrashHD
+
+2009-01-07 17:35 +0000 [r167442] Russell Bryant <russell@digium.com>
+
+ * /, main/indications.c: Merged revisions 167432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009)
+ | 4 lines Treat an empty string the same way as a NULL country
+ argument. In passing, simplify the handling of returning a
+ default tone zone. ........
+
+2009-01-07 17:05 +0000 [r167416] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: Cleanup fsk spill if off hook is detected
+ during mwi spill. Correct logic error in handling events when
+ sending mwi spill (closes issue #14143) Reported by: alecdavis
+ Patches: chan_dahdi.handle_init_event2.diff.txt uploaded by
+ dbailey
+
+2009-01-07 14:26 +0000 [r167373] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/sip-friends.sql: Update the sip-friends.sql file
+ to use the non-deprecated 'defaultname' instead of 'username' and
+ remove an extra comma that would cause the script to fail as-is
+
+2009-01-06 21:36 +0000 [r167301] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/db.c: Merged revisions 167299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan
+ 2009) | 8 lines Use the correct variable when creating the format
+ string (closes issue #14177) Reported by: nic_bellamy Patches:
+ asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
+ (license 299) ........
+
+2009-01-06 21:02 +0000 [r167265] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 167260 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r167260 | tilghman | 2009-01-06 14:48:05 -0600
+ (Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
+ Jan 2009) | 2 lines Security fix AST-2009-001. ........
+ ................
+
+2009-01-05 16:59 +0000 [r167180] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 167179 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan
+ 2009) | 41 lines A couple of changes to T.38 SDP attribute
+ handling There are some boolean attributes for T.38 such as
+ T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
+ T38FaxTranscodingJBIG. By simply being present, we should treat
+ these as a "true" value. The current code, however, was requiring
+ a 1 or 0 as the value of the attribute in order to parse it. This
+ is due to the fact that there are some T.38 endpoints and
+ gateways that also transmit this information incorrectly. This
+ patch follows the "be liberal in what you accept and strict in
+ what you send" philosophy by accepting both the correctly- and
+ incorrectly-formatted attributes, but only sending information as
+ it is supposed to be sent. It was also discovered that a
+ particular type of T.38 gateway sends some non-standard T.38 SDP
+ attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
+ it used T38MaxDatagram and T38FaxMaxRate respectively. We now
+ will properly accept these attributes as well. Note that there
+ are a lot of patches cited in the below commit message template.
+ This is because the person who submitted these patches is an
+ awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
+ issue #13976) Reported by: linulin Patches:
+ chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
+ by arcivanov (license 648)
+ chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) Tested by: arcivanov ........
+
+2009-01-05 16:44 +0000 [r167176] Tilghman Lesher <tlesher@digium.com>
+
+ * UPGRADE-1.6.txt: More clearly explain that quote marks are no
+ longer necessary. (closes issue #13718) Reported by: davidw
+ Patches: 20081020__bug13718.diff.txt uploaded by Corydon76
+ (license 14) Tested by: blitzrage
+
+2009-01-03 20:29 +0000 [r167125] Jeff Peeler <jpeeler@digium.com>
+
+ * main/asterisk.c: When parsing environment variable
+ ASTERISK_PROMPT, make sure to proceed to the next character when
+ a non format specifier is used (no %). Otherwise, the while loop
+ looking for the null byte will never exit.
+
+2008-12-31 23:07 +0000 [r167061] Sean Bright <sean.bright@gmail.com>
+
+ * doc/CODING-GUIDELINES, include/asterisk.h, channels/h323/README:
+ Mostly just whitespace, but also convert 'CVS' to 'SVN' in a
+ couple places and fix a few typos I found in the
+ CODING_GUIDELINES.
+
+2008-12-31 22:53 +0000 [r167057] Terry Wilson <twilson@digium.com>
+
+ * main/xmldoc.c: Don't forget to free typename
+
+2008-12-31 21:52 +0000 [r167021] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c: Change some incorrect syntax for pri set
+ debug and correct an off-by-one error in ss7 set debug command
+
+2008-12-31 19:39 +0000 [r166954-166958] Tilghman Lesher <tlesher@digium.com>
+
+ * main/ast_expr2.h, main/ast_expr2.c: That was weird...
+
+ * channels/chan_local.c, /, main/ast_expr2.h, main/ast_expr2.c:
+ Merged revisions 166953 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008)
+ | 5 lines Also inherit the musiconhold class. (Closes #14153)
+ Reported by: Jerry Geis, via the users list. Patch by: me
+ (license 14) ........
+
+2008-12-30 20:50 +0000 [r166908] Terry Wilson <twilson@digium.com>
+
+ * res/res_phoneprov.c, doc/sip-retransmit.txt,
+ doc/tex/phoneprov.tex, res/res_http_post.c,
+ phoneprov/polycom_line.xml, doc/realtimetext.txt: Fix some
+ svn:keywords
+
+2008-12-29 18:04 +0000 [r166861] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, apps/app_queue.c: Update app_queue to deal with
+ the removal of AST_PBX_KEEPALIVE When placing a call to a queue
+ which ran a gosub on the member's channel, Asterisk would crash
+ every time, stemming from the fact that the member's channel was
+ being hung up unexpectedly when the Gosub completed. The
+ necessary change was pretty much copied and pasted from
+ app_dial's similar changes made last week. I also took the
+ opportunity to change a LOG_DEBUG message in app_dial to use
+ ast_debug. I am guessing this was due to a direct merge from 1.4
+ that was not corrected to use trunk's preferred syntax.
+
+2008-12-28 15:36 +0000 [r166823] Eliel C. Sardanons <eliels@gmail.com>
+
+ * funcs/func_audiohookinherit.c: Fix a typo in the XML
+ documentation of the AUDIOHOOK_INHERIT dialplan function.
+
+2008-12-28 15:15 +0000 [r166773] Russell Bryant <russell@digium.com>
+
+ * /, channels/misdn_config.c: Merged revisions 166772 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28
+ Dec 2008) | 4 lines Use strncat() instead of an sprintf() in
+ which source and target buffers overlap
+ http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
+ ........
+
+2008-12-24 15:10 +0000 [r166731] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: There is no section 22.2.2 in rfc 3261. I
+ believe 26.2.2 is what was meant: Note that in the SIPS URI
+ scheme, transport is independent of TLS, and thus
+ "sips:alice@atlanta.com;transport=tcp" and
+ "sips:alice@atlanta.com;transport=sctp" are both valid (although
+ note that UDP is not a valid transport for SIPS). The use of
+ "transport=tls" has consequently been deprecated, partly because
+ it was specific to a single hop of the request. This is a change
+ since RFC 2543.
+
+2008-12-23 20:47 +0000 [r166696] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Allow semicolons and extended characters in
+ user-specified SIP headers. (closes issue #14110) Reported by:
+ gork Patches: 20081222__bug14110__2.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: gork, putnopvut
+
+2008-12-23 18:13 +0000 [r166665] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, /, main/features.c,
+ apps/app_macro.c, include/asterisk/pbx.h, apps/app_queue.c,
+ include/asterisk/features.h: Merged revisions 166093 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4 In
+ order to merge this 1.4 patch into trunk, I had to resolve some
+ conflicts and wait for Russell to make some changes to res_agi. I
+ re-ran all the tests; 39 calls in all, and made fairly careful
+ notes and comparisons: I don't want this to blow up some aspect
+ of asterisk; I completely removed the KEEPALIVE from the pbx.h
+ decls. The first 3 scenarios involving feature park; feature xfer
+ to 700; hookflash park to Park() app call all behave the same,
+ don't appear to leave hung channels, and no crashes. ........
+ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) |
+ 131 lines This merges the masqpark branch into 1.4 These changes
+ eliminate the need for (and use of) the KEEPALIVE return code in
+ res_features.c; There are other places that use this result code
+ for similar purposes at a higher level, these appear to be left
+ alone in 1.4, but attacked in trunk. The reason these changes are
+ being made in 1.4, is that parking ends a channel's life, in some
+ situations, and the code in the bridge (and some other places),
+ was not checking the result code properly, and dereferencing the
+ channel pointer, which could lead to memory corruption and
+ crashes. Calling the masq_park function eliminates this danger in
+ higher levels. A series of previous commits have replaced some
+ parking calls with masq_park, but this patch puts them ALL to
+ rest, (except one, purposely left alone because a masquerade is
+ done anyway), and gets rid of the code that tests the KEEPALIVE
+ result, and the NOHANGUP_PEER result codes. While bug 13820
+ inspired this work, this patch does not solve all the problems
+ mentioned there. I have tested this patch (again) to make sure I
+ have not introduced regressions. Crashes that occurred when a
+ parked party hung up while the parking party was listening to the
+ numbers of the parking stall being assigned, is eliminated. These
+ are the cases where parking code may be activated: 1. Feature one
+ touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3.
+ Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi
+ hookflash xfer to 700) 4. Run Park via manager. The interesting
+ testing cases for parking are: I. A calls B, A parks B a. B hangs
+ up while A is getting the numbers announced. b. B hangs up after
+ A gets the announcement, but before the parking time expires c. B
+ waits, time expires, A is redialed, A answers, B and A are
+ connected, after which, B hangs up. d. C picks up B while still
+ in parking lot. II. A calls B, B parks A a. A hangs up while B is
+ getting the numbers announced. b. A hangs up after B gets the
+ announcement, but before the parking time expires c. A waits,
+ time expires, B is redialed, B answers, A and B are connected,
+ after which, A hangs up. d. C picks up A while still in parking
+ lot. Testing this throroughly involves acting all the
+ permutations of I and II, in situations 1,2,3, and 4. Since I
+ added a few more changes (ALL references to KEEPALIVE in the
+ bridge code eliimated (I missed one earlier), I retested most of
+ the above cases, and no crashes. H-extension weirdness. Current
+ h-extension execution is not completely correct for several of
+ the cases. For the case where A calls B, and A parks B, the 'h'
+ exten is run on A's channel as soon as the park is accomplished.
+ This is expected behavior. But when A calls B, and B parks A,
+ this will be current behavior: After B parks A, B is hung up by
+ the system, and the 'h' (hangup) exten gets run, but the channel
+ mentioned will be a derivative of A's... Thus, if A is DAHDI/1,
+ and B is DAHDI/2, the h-extension will be run on channel
+ Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be
+ those relating to Channel A. And, in the case where A is
+ reconnected to B after the park time expires, when both parties
+ hang up after the joyful reunion, no h-exten will be run at all.
+ In the case where C picks up A from the parking lot, when either
+ A or C hang up, the h-exten will be run for the C channel. CDR's
+ are a separate issue, and not addressed here. As to WHY this
+ strange behavior occurs, the answer lies in the procedure
+ followed to accomplish handing over the channel to the parking
+ manager thread. This procedure is called masquerading. In the
+ process, a duplicate copy of the channel is created, and most of
+ the active data is given to the new copy. The original channel
+ gets its name changed to XXX<ZOMBIE> and keeps the PBX
+ information for the sake of the original thread (preserving its
+ role as a call originator, if it had this role to begin with),
+ while the new channel is without this info and becomes a call
+ target (a "peer"). In this case, the parking lot manager thread
+ is handed the new (masqueraded) channel. It will not run an
+ h-exten on the channel if it hangs up while in the parking lot.
+ The h exten will be run on the original channel instead, in the
+ original thread, after the bridge completes. See bug 13820 for
+ our intentions as to how to clean up the h exten behavior.
+ Review: http://reviewboard.digium.com/r/29/ ........
+
+2008-12-23 16:04 +0000 [r166625] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Fix spelling error.
+
+2008-12-23 15:17 +0000 [r166569] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 166568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec
+ 2008) | 12 lines Fix a crash resulting from a datastore with
+ inheritance but no duplicate callback The fix for this is to
+ simply set the newly created datastore's data pointer to NULL if
+ it is inherited but has no duplicate callback. (closes issue
+ #14113) Reported by: francesco_r Patches: 14113.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r ........
+
+2008-12-23 04:32 +0000 [r166533] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 166509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008)
+ | 4 lines Use the integer form of condition for integer
+ comparisons. (closes issue #14127) Reported by: andrew ........
+
+2008-12-22 23:25 +0000 [r166470] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_agi.c: Always use the value of the AGISIGHUP when running
+ an AGI. Prior to this patch, the value of AGISIGUP was not always
+ honored when set on a channel. (closes issue #13711) Reported by:
+ fmueller Patches: 13711.patch uploaded by putnopvut (license 60)
+
+2008-12-22 21:45 +0000 [r166436] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Cosmetic change - don't mix struct
+ initializer styles.
+
+2008-12-22 21:08 +0000 [r166382] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 166380 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon,
+ 22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks
+ and autoservice It has been discovered that if a channel is
+ locked prior to a call to ast_autoservice_stop, then it is likely
+ that a deadlock will occur. The reason is that the call to
+ ast_autoservice_stop has a check built into it to be sure that
+ the thread running autoservice is not currently trying to
+ manipulate the channel we are about to pull out of autoservice.
+ The autoservice thread, however, cannot advance beyond where it
+ currently is, though, because it is trying to acquire the lock of
+ the channel for which autoservice is attempting to be stopped.
+ The gist of all this is that a channel MUST NOT be locked when
+ attempting to stop autoservice on the channel. In this particular
+ case, the channel was locked by a call to ast_read. A call to
+ ast_exists_extension led to autoservice being started and stopped
+ due to the existence of dialplan switches. It may be that there
+ are future commits which handle the same symptoms but in a
+ different location, but based on my looks through the code, it is
+ very rare to see a construct such as this one. (closes issue
+ #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
+ by putnopvut (license 60) Tested by: rtrauntvein Review:
+ http://reviewboard.digium.com/r/107/ ........
+
+2008-12-22 20:26 +0000 [r166273-166377] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Fix a bad typo.
+
+ * main/astobj2.c: Remove some error messages. This is the default
+ handler that is valid to use.
+
+ * /, main/utils.c: Merged revisions 166297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008)
+ | 2 lines Fix up timeout handling in ast_carefulwrite(). ........
+
+ * include/asterisk/utils.h, main/manager.c, main/utils.c: Introduce
+ ast_careful_fwrite() and use in AMI to prevent partial writes.
+ This patch introduces a function to do careful writes on a file
+ stream which will handle timeouts and partial writes. It is
+ currently used in AMI to address the issue that has been
+ reported. However, there are probably a few other places where
+ this could be used. (closes issue #13546) Reported by: srt Tested
+ by: russell http://reviewboard.digium.com/r/104/
+
+ * res/res_musiconhold.c: Re-work ref count handling of MoH classes
+ using astobj2 to resolve crashes. (closes issue #13566) Reported
+ by: igorcarneiro Tested by: russell Review:
+ http://reviewboard.digium.com/r/106/
+
+2008-12-22 16:08 +0000 [r166268] Joshua Colp <jcolp@digium.com>
+
+ * main/dnsmgr.c: Record the previous port in the temporary address
+ structure so that the comparison does not treat the host as
+ having changed even if it did not. This would have been
+ uninitialized before and would have led to a baddddd port.
+ (closes issue #13628) Reported by: pananix Patches:
+ bug13628.patch uploaded by jpeeler (license 325) Tested by: file,
+ blitzrage
+
+2008-12-22 16:07 +0000 [r166267] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_timeout.c, main/file.c: Fix a file playback crash and
+ explicitly initialize values in func_timeout.c A crash was
+ brought up on the bugtracker. The first run through valgrind was
+ full of legitimate complaints of uninitialized values in
+ func_timeout when setting a response timeout. These were fixed
+ but the crash persisted. A second run through showed the real
+ problem. The reference counting used for filestreams was
+ incorrect because there were some missing increments when a frame
+ was read from a format module. (closes issue #14118) Reported by:
+ blitzrage Patches: 14118v2.patch uploaded by putnopvut (license
+ 60) Tested by: blitzrage
+
+2008-12-22 14:16 +0000 [r166258] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: Remove AST_PBX_KEEPALIVE usage from res_agi. This
+ patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The
+ only usage was for the AGI command, "asyncagi break". This patch
+ removes this feature. Normally, a feature would not be removed
+ like this. However, this code is broken and usage of it will
+ result in a memory leak. Usage of this feature will make the AGI
+ code return a result of AST_PBX_KEEPALIVE. The PBX handler
+ assumes that another thread has assumed ownership of the channel.
+ The channel thread will exit without destroying the channel.
+ Unfortunately, _no_ thread has ownership of the channel at this
+ point. There are a couple of serious problems here: 1) The only
+ way to recover the caller is to issue a channel redirect. This
+ will work, but this will be done with a masquerade, and the old
+ ast_channel structure will be lost. 2) Until the channel redirect
+ happens, there is no code servicing the channel. That means
+ nothing is reading audio or handling events coming from the
+ channel. This is very bad. The recommended way to get this same
+ "break" functionality is to issue the redirect while the channel
+ is still being handled by the AGI code. That way, there will be
+ no memory leak, and there will be no period of time that the
+ channel is not being serviced.
+
+2008-12-20 01:37 +0000 [r166219] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxyref.h: Make a note about formatting the
+ review URL in commit messages
+
+2008-12-19 23:45 +0000 [r166092-166162] Mark Michelson <mmichelson@digium.com>
+
+ * main/audiohook.c: Get rid of an extra space. I don't know how
+ this crept back in when I had already fixed it earlier
+
+ * funcs/func_audiohookinherit.c: Remove the verbatim tag from the
+ author line I could have sworn I already did that before,
+ though...
+
+ * main/channel.c, funcs/func_audiohookinherit.c (added),
+ include/asterisk/audiohook.h, main/audiohook.c, CHANGES: Adding a
+ new dialplan function AUDIOHOOK_INHERIT This function is being
+ added as a method to allow for an audiohook to move to a new
+ channel during a channel masquerade. The most obvious use for
+ such a facility is for MixMonitor when a transfer is performed.
+ Prior to the addition of this functionality, if a channel running
+ MixMonitor was transferred by another party, then the recording
+ would stop once the transfer had completed. By using
+ AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the
+ call even after the transfer has completed. It has also been
+ determined that since this is seen by most as a bug fix and is
+ not an invasive change, this functionality will also be
+ backported to 1.4 and merged into the 1.6.0 branches, even though
+ they are feature-frozen. (closes issue #13538) Reported by: mbit
+ Patches: 13538.patch uploaded by putnopvut (license 60) Tested
+ by: putnopvut Review: http://reviewboard.digium.com/r/102/
+
+2008-12-19 21:44 +0000 [r166058] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Add configuration
+ support for half_full DAHDI buffer policy
+
+2008-12-19 18:20 +0000 [r165954] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_record.c: Fix the XML documentation for Record().
+ <value> tags inside <variable> elements must have CDATA and no
+ another XML node.
+
+2008-12-19 15:05 +0000 [r165801-165890] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 165889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008)
+ | 9 lines Ensure that the chanspy datastore is fully initialized.
+ This patch resolved some random crash issues observed by a user
+ on a BSD system (closes issue #14111) Reported by: ys Patches:
+ app_chanspy.c.diff uploaded by ys (license 281) ........
+
+ * include/asterisk/doxyref.h: Disable some automatic links
+ generated by doxygen.
+
+ * include/asterisk/doxyref.h: Introduce commit message formatting
+ guidelines. This documents the recommended outline to use for
+ commit message. It also covers information on special tags that
+ can be used in commit messages, as well as the template
+ functionality that is available on bugs.digium.com. Review:
+ http://reviewboard.digium.com/r/96/
+
+ * /, main/utils.c: Merged revisions 165796 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008)
+ | 11 lines Make ast_carefulwrite() be more careful. This patch
+ handles some additional cases that could result in partial writes
+ to the file description. This was done to address complaints
+ about partial writes on AMI. (issue #13546) (more changes needed
+ to address potential problems in 1.6) Reported by: srt Tested by:
+ russell Review: http://reviewboard.digium.com/r/99/ ........
+
+2008-12-18 21:43 +0000 [r165798] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: (closes issue #13993) Reported by: mika Add
+ ActionID response to ping if sent with request.
+
+2008-12-18 21:41 +0000 [r165797] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 165767 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18
+ Dec 2008) | 8 lines Add mutexes around accesses to the IMAP
+ library interface. This prevents certain crashes, especially when
+ shared mailboxes are used. (closes issue #13653) Reported by:
+ howardwilkinson Patches:
+ asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
+ howardwilkinson (license 590) Tested by: jpeeler ........
+
+2008-12-18 21:21 +0000 [r165792] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_misdn.c,
+ channels/chan_sip.c, pbx/pbx_ael.c, apps/app_queue.c,
+ channels/chan_oss.c: Numerous documentation updates. (closes
+ issue #13970) Reported by: pkempgen Patches:
+ __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage
+ (license 10)
+
+2008-12-18 19:34 +0000 [r165724] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_odbc.c: Fix crashes in res_odbc. The variable "class" was
+ being set NULL just prior to being dereferenced in an ao2_link
+ call. I have moved the setting of the variable to NULL until
+ after the ao2_link call.
+
+2008-12-18 19:33 +0000 [r165662-165723] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: Remove the
+ need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This
+ is part of an effort to completely remove AST_PBX_KEEPALIVE and
+ other similar return codes from the source. While this usage was
+ perfectly safe, there are others that are problematic. Since we
+ know ahead of time that we do not want to PBX to destroy the
+ channel, the PBX API has been changed so that information can be
+ provided as an argument, instead, thus removing the need for the
+ KEEPALIVE return value. Further changes to get rid of KEEPALIVE
+ and related code is being done by murf. There is a patch up for
+ that on review 29. Review: http://reviewboard.digium.com/r/98/
+
+ * /, res/res_musiconhold.c: Merged revisions 165661 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18
+ Dec 2008) | 7 lines Set the process group ID on the MOH process
+ so that all children will get killed (closes issue #14099)
+ Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2
+ uploaded by caspy (license 645) ........
+
+2008-12-18 18:36 +0000 [r165658] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Fix 2 resource leaks and fix another
+ pipe-to-comma conversion
+
+2008-12-18 17:13 +0000 [r165599] Joshua Colp <jcolp@digium.com>
+
+ * /, main/rtp.c: Merged revisions 165591 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4
+ lines Only care about a compatible codec for early bridging if we
+ are actually bridging to another channel. If we are not we
+ actually want to bring the audio back to us. (closes issue
+ #13545) Reported by: davidw ........
+
+2008-12-18 16:36 +0000 [r165541] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Fix reference counts of the class and add an
+ assertion to the end.
+
+2008-12-18 15:25 +0000 [r165502] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/strings.c, include/asterisk/strings.h: Remove duplicate code
+ from the ast_str API. We now use __AST_STR_* to access 'struct
+ ast_str' members, but this must only be used inside the API
+ implementation. (closes issue #14098) Reported by: eliel Patches:
+ ast_str.patch uploaded by eliel (license 64)
+
+2008-12-18 14:23 +0000 [r165433-165469] Russell Bryant <russell@digium.com>
+
+ * apps/app_originate.c: Add a \todo note for app_originate. Jared
+ Smith suggested that we add a way to be able to set variables and
+ functions on the outbound channel. I think that it's a great
+ idea, so I have added it as a todo so that it gets done at some
+ point.
+
+ * apps/app_originate.c (added), CHANGES: Add a new application,
+ Originate. (closes issue #14075) Reported by: rcasas Patches:
+ app_originate.c uploaded by rcasas (license 641), heavily
+ modified by me Tested by: russell Review:
+ http://reviewboard.digium.com/r/95/
+
+2008-12-17 23:39 +0000 [r165397] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_record.c: Add RECORD_STATUS variable, as requested on
+ the -users list. Patch by me (license 14)
+
+2008-12-17 21:46 +0000 [r165326-165330] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_odbc.c: Fix a refcount leak in res_odbc
+
+ * apps/app_meetme.c, res/res_realtime.c: Fix the build
+
+2008-12-17 21:28 +0000 [r165319-165325] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_macro.c: Oops, broke trunk
+
+ * /, apps/app_macro.c: Merged revisions 165317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008)
+ | 4 lines Reverse the fix from issue #6176 and add proper
+ handling for that issue. (Closes issue #13962, closes issue
+ #13363) Fixed by myself (license 14) ........
+
+2008-12-17 21:17 +0000 [r165318] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_meetme.c, res/res_realtime.c, apps/app_directory.c,
+ apps/app_queue.c, apps/app_voicemail.c: Merged revisions 165255
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec
+ 2008) | 7 lines Fix some memory leaks found while looking at how
+ realtime configs are handled. Also cleaned up some coding
+ guidelines violations in app_realtime.c, mostly related to
+ spacing ........
+
+2008-12-17 20:50 +0000 [r165254] Steve Murphy <murf@digium.com>
+
+ * utils/extconf.c: This patch is here committed to satisfy the
+ buildbot, who has a problem with the const.
+
+2008-12-17 19:55 +0000 [r165219] Terry Wilson <twilson@digium.com>
+
+ * res/res_phoneprov.c: Polycom phones close the connection after
+ reading a little bit of the firmware files, we should stop
+ sending in that case. Also, make that case print out a debug
+ statement instead of a scary WARNING.
+
+2008-12-17 19:52 +0000 [r165216] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Call proxy_update so that the IP address
+ gets populated. Sending stuff to 0.0.0.0 is silly! (closes issue
+ #14055) Reported by: chris-mac
+
+2008-12-17 18:49 +0000 [r165180] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: This patch
+ adds a new 'ignoresdpversion' option to sip.conf. When this is
+ enabled (either globally or for a specific peer), chan_sip will
+ treat any SDP data it receives as new data and update the media
+ stream accordingly. By default, Asterisk will only modify the
+ media stream if the SDP session version received is different
+ from the current SDP session version. This option is required to
+ interoperate with devices that have non-standard SDP session
+ version implementations (observed by toc on the bug tracker with
+ Microsoft OCS which always uses 0 as the session version).
+ http://reviewboard.digium.com/r/94/ (closes issue #13958)
+ Reported by: toc Tested by: toc
+
+2008-12-17 17:56 +0000 [r165145] Russell Bryant <russell@digium.com>
+
+ * doc/appdocsxml.dtd: argsep is used as an attribute for an
+ argument, as well
+
+2008-12-17 17:53 +0000 [r165142-165143] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: And actually assign the function to a
+ pointer...
+
+ * apps/app_voicemail.c: Use the create_vm_state_from_user function
+ in a place where it was not being used before. Also, I've moved
+ the urgent folder check in messagecount() up a bit so that the
+ flow is a bit better. This was something I noticed while taking a
+ look at issue #13973, although I don't think this is the
+ underlying cause of the issue.
+
+2008-12-17 16:41 +0000 [r165108] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils: ignore this copied file
+
+2008-12-17 05:04 +0000 [r165039-165071] Steve Murphy <murf@digium.com>
+
+ * utils/Makefile, pbx/pbx_ael.c, utils/ael_main.c, utils/extconf.c,
+ utils/conf2ael.c, utils/check_expr.c: A possibly "horrible fix"
+ for a "horribly broken" situation. As stuff shifts around in the
+ asterisk code, the miscellaneous inclusions from the standalone
+ stuff gets broken. There's no easy fix for this situation. I made
+ sure that everything in utils builds without problem ***AND***
+ that aelparse runs the regressions correctly with the following
+ make menuselect options both on and off: DONT_OPTIMIZE
+ DEBUG_THREADS DEBUG_CHANNEL_LOCKS MALLOC_DEBUG MTX_PROFILE
+ DEBUG_SCHEDULER DEBUG_THREADLOCALS DETECT_DEADLOCKS CHANNEL_TRACE
+ I think from now on, I'm going to #undef all these features in
+ the various utils native files; I guess I could do the same for
+ the copied-in files, surrounded by STANDALONE ifdef. A standalone
+ isn't going to care about threads, mutexes, etc.
+
+ * pbx/ael/ael-test/ref.ael-vtest17,
+ pbx/ael/ael-test/ref.ael-vtest13: fixed the regressions
+
+2008-12-16 23:06 +0000 [r164978] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 164977 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec
+ 2008) | 7 lines After looking through SIP registration code most
+ of the day, this is one of the few things I could find that was
+ just plain wrong. Even though it probably isn't possible for it
+ to happen, it seems weird to have code that checks if a pointer
+ is NULL and then immediately dereferences that pointer if it was
+ NULL. ........
+
+2008-12-16 22:57 +0000 [r164976] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, doc/api-1.6.2-changes.txt (added),
+ funcs/func_logic.c, include/asterisk/pbx.h, utils/extconf.c,
+ CHANGES, configs/extensions.conf.sample: Add timezone to the
+ possible fields in a timespec. (closes issue #14028) Reported by:
+ mostyn Patches: timezone-v2.patch uploaded by mostyn (license
+ 398) (with additional code guideline fixes and a memory leak fix
+ by me - license 14)
+
+2008-12-16 22:45 +0000 [r164942] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_record.c: (closes issue #13669) Reported by: pj Delete
+ file recording if recording terminated from a hangup.
+
+2008-12-16 22:31 +0000 [r164941] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Make a note of the feature request in bug
+ #11157 as per the reporter and oej, and suspend the bug since no
+ one seems to be keen on implementing it any time soon.
+
+2008-12-16 21:39 +0000 [r164821-164882] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c: Merged revisions 164881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008)
+ | 9 lines Fix an issue where DEBUG_THREADS may erroneously report
+ that a thread is exiting while holding a lock. If the last lock
+ attempt was a trylock, and it failed, it will still be in the
+ list of locks so that it can be reported. (closes issue #13219)
+ Reported by: pj ........
+
+ * /, apps/app_macro.c: Merged revisions 164876 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008)
+ | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has
+ been returned. This is a bug I noticed while looking at the code
+ for app_macro. This return code means that another thread has
+ assumed ownership of the channel and it can no longer be touched.
+ (I hate this return code with a passion, by the way.) ........
+
+ * main/asterisk.c: Fix build issues on Linux after sysinfo related
+ changes
+
+2008-12-16 20:47 +0000 [r164809-164814] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Qualify
+ trumps poke per lmadsen.
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
+ configuration options for finer control over how Asterisk handles
+ having to poke all peers at seemingly the same time. (closes
+ issue #13217) Reported by: cervajs
+
+2008-12-16 20:41 +0000 [r164807] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 164806 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008)
+ | 9 lines Add "restart gracefully" to the AMI blacklist of CLI
+ commands. "module unload" was already identified as a command
+ that can not be used from the AMI. "restart gracefully"
+ effectively unloads all modules, and will run in to the same
+ problems. (closes issue #13894) Reported by: kernelsensei
+ ........
+
+2008-12-16 20:08 +0000 [r164802] Michiel van Baak <michiel@vanbaak.info>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/asterisk.c: introduce 'core show sysinfo' for systems that
+ dont have the Linux-ish sysinfo stuff but do have sysctl. (closes
+ issue #13433) Reported by: mvanbaak Patches:
+ 2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license
+ 7) with two free calls replaced with ast_free based on feedback
+ on reviewboard Review: http://reviewboard.digium.com/r/91/
+
+2008-12-16 20:04 +0000 [r164801] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: (closes issue #14076) Reported by: toc Tested by:
+ murf OK, Well this issue has had its share of flip-flopping. I
+ found the following: 1. the code in question, in ext_cmp1 in
+ pbx.c, would not allow two extensions that vary only by any
+ dashes contained within them, to be defined in the same context.
+ 2. for input dialstrings, dashes are NOT ignored. So, skipping
+ them when sorting patterns seemed a bit silly. Thus, you might
+ declare ext 891 in a context, but if you try dialing 8-9-1, it
+ will NOT match 891. So, I proposed to remove the code from
+ ext_cmp1 to skip the spaces and dashes. Just kept us from
+ declaring 891 and 8-9-1 in the same context, forcing users to
+ generate otherwise uselessly obfuscated dialplan code to get the
+ same effect. Then, I tried out 1.4, and found that: 1. you can
+ declare 891 and 8-9-1 in the same context! 2. You can't define
+ 891, and have 8-9-1 match it! Nor can you define 8-9-1, and have
+ 891 match it! So, it appears that my proposal simply restores the
+ pbx to behaving as it did in 1.4.
+
+2008-12-16 19:54 +0000 [r164798] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/safe_asterisk: Set up umask as a possible
+ configuration option. (closes issue #13753) Reported by: irroot
+
+2008-12-16 17:14 +0000 [r164737] Russell Bryant <russell@digium.com>
+
+ * /, main/threadstorage.c, include/asterisk/threadstorage.h: Merged
+ revisions 164736 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008)
+ | 14 lines Fix memory leak and invalid reporting issues with
+ DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was
+ being used within the context of the thread local data
+ destructors. We would go off and allocate more thread local data
+ while the pthread lib was in the middle of destroying it all.
+ This led to a memory leak. Another issue was an invalid argument
+ being provided to the the object_add API call. (closes issue
+ #13678) Reported by: ys Tested by: Russell ........
+
+2008-12-16 16:50 +0000 [r164733] Joshua Colp <jcolp@digium.com>
+
+ * pbx/pbx_config.c: Be more detailed about why the include did not
+ get included. (closes issue #14071) Reported by: kshumard
+ Patches: pbx_config.patch.improvederroroutput.txt uploaded by
+ kshumard (license 92)
+
+2008-12-16 16:00 +0000 [r164675] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 164672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008)
+ | 11 lines Fix a memory leak related to the use of the "setvar"
+ configuration option. The problem was that these variables were
+ being appended to the list of vars on the sip_pvt every time a
+ re-registration or re-subscription came in. Since it's just a
+ waste of memory to put them there unless the request was an
+ INVITE, then the fix is to check the request type before copying
+ the vars. (closes issue #14037) Reported by: marvinek Tested by:
+ russell ........
+
+2008-12-16 15:44 +0000 [r164659] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: When using externhost make sure the port
+ gets set to the bindaddr port if one was not specified in the
+ externhost value itself. (closes issue #13634) Reported by:
+ performer
+
+2008-12-16 15:31 +0000 [r164648] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /: Merged revisions 164634 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5
+ lines I added a sentence to clarify why - and ' ' are ignored in
+ patterns as per bug 14076. Leif says he'll put some stuff about
+ it in the extensions.conf sample, etc. ........
+
+2008-12-16 15:00 +0000 [r164602-164623] Russell Bryant <russell@digium.com>
+
+ * apps/app_minivm.c: Set MINIVM_ACCMESS_STATUS in all cases. Also,
+ remove a variable that was not needed. (closes issue #14081)
+ Reported by: pkempgen
+
+ * /, res/res_musiconhold.c: Merged revisions 164605 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16
+ Dec 2008) | 5 lines Don't try to change working directory if a
+ directory was not configured. (closes issue #14089) Reported by:
+ caspy ........
+
+ * channels/chan_dahdi.c: Fix usage of the DAHDI_VMWI ioctl. (closes
+ issue #14090) Reported by: alecdavis Patches:
+ chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license
+ 585)
+
+2008-12-16 01:52 +0000 [r164565] Sean Bright <sean.bright@gmail.com>
+
+ * doc/tex/odbcstorage.tex: Use tables instead of ASCII art. Also
+ change a bit of minor formatting.
+
+2008-12-15 22:25 +0000 [r164519-164525] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Open a timer before loading configuration
+ so that the trunking configuration option will take effect.
+ (closes issue #14082) Reported by: seandarcy
+
+ * channels/chan_iax2.c: Fix log message to refer to the generic
+ timing interface, not DAHDI specifically (inspired by issue
+ #14082)
+
+ * main/frame.c: Make sure we handle a uint32_t payload in
+ ast_frdup() (closes issue #14080) Reported by: fnordian Patches:
+ frame.patch uploaded by fnordian (license 110)
+
+2008-12-15 21:17 +0000 [r164485] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extconfig.conf.sample, pbx/pbx_realtime.c, CHANGES: Allow
+ disabling pattern match searches within the Realtime dialplan
+ switch. (closes issue #13698) Reported by: fhackenberger Patches:
+ 20081211__bug13698.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: fhackenberger
+
+2008-12-15 20:07 +0000 [r164419-164428] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_page.c: Add an 'i' option to app_page. This option works
+ the same as the 'i' options for app_dial and app_queue, in that
+ they will ignore any attempts by phones to forward the call.
+ (closes issue #13977) Reported by: putnopvut Patches:
+ page_ignore_forwards.patch uploaded by putnopvut (license 60)
+ Tested by: putnopvut, acunningham
+
+ * /, include/asterisk/pbx.h: Merged revisions 164422 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon,
+ 15 Dec 2008) | 3 lines Add the deadlock note to
+ ast_spawn_extension as well ........
+
+ * /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged
+ revisions 164416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec
+ 2008) | 4 lines Add notes to autoservice and pbx doxygen
+ regarding a potential deadlock scenario so that it is avoided in
+ the future ........
+
+2008-12-15 19:48 +0000 [r164417] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, include/asterisk/strings.h: Revert ast_str
+ opacity in chan_sip for now, since something wasn't quite right
+ in the merge.
+
+2008-12-15 19:42 +0000 [r164415] Steve Murphy <murf@digium.com>
+
+ * include/asterisk/strings.h: I was getting this warning during a
+ compile on a 64-bit machine running ubuntu server 8.10, and
+ gcc-4.3.2: [CXXi] chan_vpb.ii -> chan_vpb.oo cc1plus: warnings
+ being treated as errors In file included from
+ /home/murf/asterisk/trunk/include/asterisk/utils.h:671, from
+ chan_vpb.cc:46:
+ /home/murf/asterisk/trunk/include/asterisk/strings.h: In function
+ ‘char* ast_str_truncate(ast_str*, ssize_t)’:
+ /home/murf/asterisk/trunk/include/asterisk/strings.h:479: error:
+ comparison between signed and unsigned integer expressions
+ make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2
+ which this fix silences
+
+2008-12-15 18:12 +0000 [r164351] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 164350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6
+ lines Do not try to unlock a non-existant channel if the transfer
+ fails. (closes issue #13800) Reported by: dwagner Patches:
+ asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license
+ 608) ........
+
+2008-12-15 18:09 +0000 [r164349] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c: When querying for the structure of the CDR
+ table, remove the schema, if it exists. (Closes issue #14058)
+
+2008-12-15 17:24 +0000 [r164312] Joshua Colp <jcolp@digium.com>
+
+ * main/file.c: Use ast_seekstream to return the file stream back to
+ the beginning instead of directly seeking to zero. This is
+ because some audio formats have headers at the front that need to
+ be skipped, which will be done by the format module. (closes
+ issue #14079) Reported by: elguero
+
+2008-12-15 17:21 +0000 [r164272-164309] Russell Bryant <russell@digium.com>
+
+ * channels/h323/ast_h323.cxx, include/asterisk/strings.h: Fix a
+ couple more build issues related to ast_str_opaque
+
+ * pbx/pbx_dundi.c: When a reload is issued, always process the
+ configuration for dundi.conf. The reason is that a reload can be
+ used to refresh DNS lookups for defined peers. Even if the config
+ file hasn't changed, we want to process it for that purpose.
+ (closes issue #13776) Reported by: kombjuder
+
+2008-12-15 16:16 +0000 [r164268-164270] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix a compile warning and a logic error that
+ could have been bad for non-realtime queues
+
+ * apps/app_queue.c: Fix up a few issues with regards to queues *
+ Fix reference counting used in the __queues_show function * Add
+ code to be sure that the "queue show" command does not print
+ information for a realtime queue which has been deleted from the
+ backend * Add a missing unref to the realtime queue loading
+ function for the case where a queue is in the module's container
+ but has been deleted from the realtime backend (closes issue
+ #14033) Reported by: cristiandimache Patches: 14033.patch
+ uploaded by putnopvut (license 60) Tested by: cristiandimache
+
+2008-12-15 15:41 +0000 [r164208-164257] Joshua Colp <jcolp@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, apps/app_fax.c,
+ configure.ac: Make app_fax compatible with newer versions of
+ spandsp. This remains backwards compatible with earlier versions
+ though so do not fret. (closes issue #14073) Reported by:
+ seandarcy
+
+ * main/utils.c: Update to work with new ast_str changes.
+
+2008-12-15 14:40 +0000 [r164202-164203] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 164201 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008)
+ | 31 lines Handle a case where a call can be bridged to a channel
+ that is still ringing. The issue that was reported was about a
+ case where a RINGING channel got redirected to an extension to
+ pick up a call from parking. Once the parked call got taken out
+ of parking, it heard silence until the other side answered.
+ Ideally, the caller that was parked would get a ringing
+ indication. This patch fixes this case so that the caller
+ receives ringback once it comes out of parking until the other
+ side answers. The fixes are: - Make sure we remember that a
+ channel was an outgoing channel when doing a masquerade. This
+ prevents an erroneous ast_answer() call on the channel, which
+ causes a bogus 200 OK to be sent in the case of SIP. - Add some
+ additional comments to explain related parts of code. - Update
+ the handling of the ast_channel visible_indication field. Storing
+ values that are not stateful is pointless. Control frames that
+ are events or commands should be ignored. - When a bridge first
+ starts, check to see if the peer channel needs to be given
+ ringing indication because the calling side is still ringing. -
+ Rework ast_indicate_data() a bit for the sake of readability.
+ (closes issue #13747) Reported by: davidw Tested by: russell
+ Review: http://reviewboard.digium.com/r/90/ ........
+
+ * apps/app_jack.c: Fix build WRT ast_str_opaque
+
+2008-12-14 18:16 +0000 [r164168] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/strings.h: Don't pass a negative to an unsigned
+ type and expect things to work correctly.
+
+2008-12-14 15:26 +0000 [r164054-164137] Sean Bright <sean.bright@gmail.com>
+
+ * doc/tex/cdrdriver.tex: Use a \picture instead of ASCII art.
+
+ * res/snmp/agent.c: Use ast_str_strlen() instead of recalculating
+ the string length.
+
+2008-12-13 13:26 +0000 [r164028] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/snmp/agent.c: nuke another use of the ast_str internals.
+
+2008-12-13 08:36 +0000 [r163991] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, apps/app_meetme.c,
+ funcs/func_strings.c, utils/hashtest.c, cdr/cdr_adaptive_odbc.c,
+ main/utils.c, apps/app_chanisavail.c, include/asterisk/tcptls.h,
+ cdr/cdr_pgsql.c, res/res_http_post.c, apps/app_followme.c,
+ res/res_config_sqlite.c, main/config.c, main/cli.c, main/cdr.c,
+ channels/chan_dahdi.c, res/res_config_odbc.c, main/manager.c,
+ configure, funcs/func_odbc.c, res/res_agi.c, apps/app_dumpchan.c,
+ main/logger.c, main/http.c, main/app.c, apps/app_externalivr.c,
+ res/res_config_ldap.c, include/asterisk/threadstorage.h,
+ cdr/cdr_manager.c, res/res_clialiases.c, utils/refcounter.c,
+ res/res_config_pgsql.c, main/strings.c (added), main/pbx.c,
+ channels/chan_sip.c, main/Makefile, main/translate.c,
+ include/asterisk/cdr.h, apps/app_queue.c, channels/iax2-parser.c,
+ funcs/func_realtime.c, utils/Makefile, res/res_config_curl.c,
+ main/tcptls.c, include/asterisk/app.h, funcs/func_curl.c,
+ utils/hashtest2.c, include/asterisk/strings.h,
+ include/asterisk/pbx.h, main/asterisk.c, main/xmldoc.c,
+ apps/app_voicemail.c, utils/check_expr.c: Merge ast_str_opaque
+ branch (discontinue usage of ast_str internals)
+
+2008-12-13 03:03 +0000 [r163951-163952] Sean Bright <sean.bright@gmail.com>
+
+ * doc/tex/asterisk.tex: This shouldn't have gotten commited. We
+ might want to generate this into a separate file instead of the
+ version controlled one.
+
+ * doc/tex/qos.tex, doc/tex/asterisk.tex: Use actual tables instead
+ of ASCII art ones.
+
+2008-12-13 00:59 +0000 [r163912] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_chanspy.c: Only detach and destroy the whisper
+ audiohooks if they are actually in use.
+
+2008-12-12 23:48 +0000 [r163873] Terry Wilson <twilson@digium.com>
+
+ * apps/app_queue.c: When using realtime queues, app_queue wasn't
+ updating the strategy if it was changed in the realtime backend.
+ This patch resolves the issue for almost all situations. It is
+ currently not supported to switch to the linear strategy via
+ realtime since the ao2_container for members will have been set
+ to have multiple buckets and therefore the members would be
+ unordered. (closes issue #14034) Reported by: cristiandimache
+ Tested by: otherwiseguy, cristiandimache
+
+2008-12-12 23:06 +0000 [r163828] Russell Bryant <russell@digium.com>
+
+ * res/res_clioriginate.c: Add a note to indicate why this only
+ supports one channel for now.
+
+2008-12-12 22:04 +0000 [r163762] Tilghman Lesher <tlesher@digium.com>
+
+ * main/editline/read.c, /, main/asterisk.c: Merged revisions 163761
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008)
+ | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk,
+ but also add a pointer inside editline to look back to
+ asterisk.c, so others don't spend as much time as I did looking
+ (in the wrong place) for the appropriate function. Reported by:
+ ZX81, via the #asterisk-users channel Fixed by: me (license 14)
+ ........
+
+2008-12-12 20:12 +0000 [r163716] Russell Bryant <russell@digium.com>
+
+ * CHANGES, res/res_clioriginate.c: Add a new CLI command, "channel
+ redirect", which is similar in operation to AMI Redirect. Review:
+ http://reviewboard.digium.com/r/89/
+
+2008-12-12 19:16 +0000 [r163675] Steve Murphy <murf@digium.com>
+
+ * channels/chan_dahdi.c: demote always-appearing debug message (for
+ certain boards) to ast_debug lev 3 msg instead
+
+2008-12-12 18:45 +0000 [r163642-163670] Russell Bryant <russell@digium.com>
+
+ * main/tcptls.c, channels/chan_sip.c: Rename a number of
+ tcptls_session variables. There are no functional changes here.
+ The name "ser" was used in a lot of places. However, it is a
+ relic from when the struct was a server_instance, not a
+ session_instance. It was renamed since it represents both a
+ server or client connection.
+
+ * channels/chan_sip.c: Fix a small race condition in
+ sip_tcp_locate(). We must increase the reference count on the
+ tcptls_session _before_ unlocking the thread list.
+
+ * channels/chan_sip.c: Resolve crashes when using SIP TCP/TLS with
+ qualify. The problem was a reference count error on the
+ tcptls_session structure. (closes issue #13989) Reported by:
+ Nugget
+
+2008-12-12 18:17 +0000 [r163629] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: When a device registers we need to unlink
+ them (if linked) from the peers_by_ip container and link them
+ back in since their IP address has changed. This would have
+ manifested itself if you configured a new device (as type=peer),
+ registered, and then tried to place a call from the device. Since
+ the peer was not linked into the peers_by_ip container it would
+ have never been found. (closes issue #13811) Reported by: pj
+
+2008-12-12 17:22 +0000 [r163582-163612] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/res_monitor.c: Document default Monitor file location.
+ (closes issue #14065) Reported by: kshumard Patches:
+ res_monitor.documentation.patch.txt uploaded by kshumard (license
+ 92)
+
+ * channels/chan_skinny.c: Fix codec capability setup in chan_skinny
+ Behaviour now is that general codec config flows to default_line
+ and default_device. [devices] stuff amends default_device and
+ similar for [lines]. These are copied to individual device and
+ line as they are created. Added confcapability and confprefs for
+ the configured stuff which doesn't change as device and so on are
+ connected. prefs are based on line prefs if they exist, else the
+ device prefs are used (prefs identifies codec order). (closes
+ issue #13806) Reported by: pj Patches: codecs.diff uploaded by
+ wedhorn (license 30) Tested by: pj and me
+
+2008-12-12 16:55 +0000 [r163579] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, channels/chan_sip.c: Since chan_sip is callback
+ devicestate driven do not pass in actual states, pass in unknown
+ so we get asked. Additionally do not pass in an actual device
+ state value in ast_setstate since the channel may be callback
+ driven. (closes issue #13525) Reported by: pj
+
+2008-12-12 15:10 +0000 [r163516] Doug Bailey <dbailey@digium.com>
+
+ * configs/phoneprov.conf.sample: Add internationalization to sample
+ configuration file
+
+2008-12-12 14:44 +0000 [r163449-163512] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 163511 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008)
+ | 5 lines Specify uint32_t for variables storing a CRC32 so that
+ it is actually 32 bits on 64-bit machines, as well. (inspired by
+ issue #13879) ........
+
+ * main/channel.c, main/autoservice.c, /,
+ include/asterisk/channel.h: Merged revisions 163448 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12
+ Dec 2008) | 26 lines Resolve issues that could cause DTMF to be
+ processed out of order. These changes come from
+ team/russell/issue_12658 1) Change autoservice to put digits on
+ the head of the channel's frame readq instead of the tail. If
+ there were frames on the readq that autoservice had not yet read,
+ the previous code would have resulted in out of order processing.
+ This required a new API call to queue a frame to the head of the
+ queue instead of the tail. 2) Change up the processing of DTMF in
+ ast_read(). Some of the problems were the result of having two
+ sources of pending DTMF frames. There was the dtmfq and the more
+ generic readq. Both were used for pending DTMF in various
+ scenarios. Simplifying things to only use the frame readq avoids
+ some of the problems. 3) Fix a bug where a DTMF END frame could
+ get passed through when it shouldn't have. If code set
+ END_DTMF_ONLY in the middle of digit emulation, and a digit
+ arrived before emulation was complete, digits would get processed
+ out of order. (closes issue #12658) Reported by: dimas Tested by:
+ russell, file Review: http://reviewboard.digium.com/r/85/
+ ........
+
+2008-12-11 23:38 +0000 [r163384] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 163383 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008)
+ | 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on
+ certain shells, the terminal is messed up. By intercepting those
+ events with a signal handler in the remote console, we can avoid
+ those issues. (closes issue #13464) Reported by: tzafrir Patches:
+ 20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: blitzrage ........
+
+2008-12-11 22:49 +0000 [r163317] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 163316 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec
+ 2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes
+ issue #13819) Reported by: adomjan Patches:
+ pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
+ dundi_clearecache3.diff uploaded by mnicholson (license 96)
+ Tested by: adomjan ........
+
+2008-12-11 21:48 +0000 [r163241-163254] Russell Bryant <russell@digium.com>
+
+ * /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions
+ 163253 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008)
+ | 8 lines Fix some observed slowdowns in dialplan processing. The
+ change is to remove autoservice usage from dialplan functions
+ that do not need it because they do not perform operations that
+ potentially block. (closes issue #13940) Reported by: tbelder
+ ........
+
+ * res/res_timing_pthread.c: Fix a problem where continuous mode
+ will get inadvertently get turned off if set_rate() is used while
+ continuous mode was already turned on. (closes issue #13738)
+ Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix
+ (license 547)
+
+2008-12-11 20:57 +0000 [r163198-163213] Mark Michelson <mmichelson@digium.com>
+
+ * configs/voicemail.conf.sample, apps/app_voicemail.c: Add an
+ option to voicemail.conf to allow urgent messages to be forwarded
+ as not urgent. (closes issue #14063) Reported by: jaroth Patches:
+ urgfwd_v2.patch uploaded by jaroth (license 50)
+
+ * main/features.c: Add an appropriate goto if ast_call fails
+
+2008-12-11 20:07 +0000 [r163171] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Fix the "failed" extension for outgoing calls.
+ The conversion to use ast_check_hangup() everywhere instead of
+ checking the softhangup flag directly introduced this problem.
+ The issue is that ast_check_hangup() checked for tech_pvt to be
+ NULL. Unfortunately, this will be NULL is some valid
+ circumstances, such as with a dummy channel. The fix is simple.
+ Don't check tech_pvt. It's pointless, because the code path that
+ sets this to NULL is when the channel hangup callback gets
+ called. This happens inside of ast_hangup(), which is the same
+ function responsible for freeing the channel. Any code calling
+ ast_check_hangup() better not be calling it after that point, and
+ if so, we have a bigger problem at hand. (closes issue #14035)
+ Reported by: erogoza
+
+2008-12-11 20:02 +0000 [r163168] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Sometimes even Linux needs -lm to link
+ libtonezone, such as when libtonezone is compiled statically.
+ (closes issue #13887) Reported by: tzafrir
+
+2008-12-11 19:40 +0000 [r163166] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Reduce indentation level of
+ ast_feature_request_and_dial
+
+2008-12-11 17:06 +0000 [r163094] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 163092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008)
+ | 11 lines Fix an issue that made it so you could only have a
+ single caller executing a custom feature at a time. This was
+ especially problematic when custom features ran for any
+ appreciable amount of time. The fix turned out to be quite
+ simple. The dynamic features are now stored in a read/write list
+ instead of a list using a mutex. (closes issue #13478) Reported
+ by: neutrino88 Fix suggested by file ........
+
+2008-12-11 16:52 +0000 [r163089] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 163088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008)
+ | 6 lines Don't wait forever, if there's a specified recording
+ timeout. (closes issue #13885) Reported by: bamby Patches:
+ res_agi.c.patch uploaded by bamby (license 430) ........
+
+2008-12-11 16:47 +0000 [r163081-163085] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 163084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec
+ 2008) | 4 lines Revert this cast to long. Using time_t here
+ causes build failures on a FreeBSD 32-bit build. ........
+
+ * /, apps/app_queue.c: Merged revisions 163080 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec
+ 2008) | 14 lines Fix a potential crash due to unsafe datastore
+ handling. This patch also contains a conversion from using long
+ to time_t for representing times for a queue, as well as some
+ whitespace fixes. (closes issue #14060) Reported by: nivek
+ Patches: datastore_fixup.patch.corrected uploaded by nivek
+ (license 636) with slight modification from me Tested by: nivek
+ ........
+
+2008-12-11 15:40 +0000 [r163037] Sean Bright <sean.bright@gmail.com>
+
+ * doc/tex/qos.tex: Fix some of the grammar issues in
+ doc/tex/qos.tex. (closes issue #14049) Reported by: kshumard
+ Patches: doc.tex.qos.tex.patch uploaded by kshumard (license 92)
+ (Slight modifications by seanbright)
+
+2008-12-11 15:05 +0000 [r162997] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: When a device registers to use it is
+ entirely possible that they may be in use, so tell the core that
+ we don't know the devstate and have it ask us for it. (closes
+ issue #13525) Reported by: pj
+
+2008-12-10 23:01 +0000 [r162930] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Previously missing line, now the substitution works
+ correctly
+
+2008-12-10 22:53 +0000 [r162927] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 162926 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10
+ Dec 2008) | 3 lines Oops, inverted logic for a strcasecmp check.
+ Pointed out by mmichelson, thanks! ........
+
+2008-12-10 22:48 +0000 [r162923] Joshua Colp <jcolp@digium.com>
+
+ * res/res_clialiases.c: Fix reloads of aliased CLI commands. Due to
+ changes done to turn it into a single memory allocation we can't
+ just use the existing CLI alias structure. We have to destroy all
+ existing ones and then create new ones. (closes issue #14054)
+ Reported by: pj
+
+2008-12-10 22:48 +0000 [r162922] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Checking global variables here actually overwrote the
+ previous substitution by channel variables, and in any case, was
+ redundant; pbx_substitute_variables_helper ALREADY does
+ substitution for global variables. (closes issue #13327) Reported
+ by: pj
+
+2008-12-10 22:11 +0000 [r162891] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 162874 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10
+ Dec 2008) | 5 lines (closes issue #13229) Reported by:
+ clegall_proformatique Ensure that moh_generate does not return
+ prematurely before local_ast_moh_stop is called. Also, the sleep
+ in mp3_spawn now only occurs for http locations since it seems to
+ have been added originally only for failing media streams.
+ ........
+
+2008-12-10 19:02 +0000 [r162739-162805] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 162804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6
+ lines Fix subscription based MWI up a bit. We only want to put
+ sip: at the beginning of the URI if it is not already there and
+ revert code to ignore destination check if subscribing for MWI.
+ (closes issue #12560) Reported by: vsauer Patches: patch001.diff
+ uploaded by ramonpeek (license 266) ........
+
+ * /, channels/chan_sip.c: Merged revisions 162738 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6
+ lines When a SIP peer unregisters set the expiry time back to 0
+ so that the 200 OK contains an expires of 0. (closes issue
+ #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded
+ by hjourdain (license 583) ........
+
+2008-12-10 17:09 +0000 [r162687] Michiel van Baak <michiel@vanbaak.info>
+
+ * include/asterisk.h, main/asterisk.c, main/cli.c: add tab
+ completion for 'core set debug X filename.c' (closes issue
+ #13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel
+
+2008-12-10 16:39 +0000 [r162664-162667] Mark Michelson <mmichelson@digium.com>
+
+ * doc/tex/misdn.tex, /: Merged revisions 162659 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec
+ 2008) | 8 lines Add missing documentation to misdn.txt (closes
+ issue #14052) Reported by: festr Patches: misdn.txt.patch
+ uploaded by festr (license 443) ........
+
+ * /, channels/chan_sip.c: Merged revisions 162663 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec
+ 2008) | 11 lines Revert fix for issue 13570. It has caused more
+ problems than it helped to fix. (closes issue #13783) Reported
+ by: navkumar (closes issue #14025) Reported by: ffs ........
+
+2008-12-10 16:11 +0000 [r162619-162660] Joshua Colp <jcolp@digium.com>
+
+ * res/res_http_post.c: FreeBSD also needs libgen.h (closes issue
+ #14051) Reported by: ys Patches: res_http_post.c.diff uploaded by
+ ys (license 281)
+
+ * /, main/rtp.c: Merged revisions 162653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6
+ lines Increment the sequence number on the end packets for
+ RFC2833. After reading the RFC some more and doing some testing I
+ agree with this change. (closes issue #12983) Reported by: vt
+ Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
+ 520) ........
+
+ * channels/chan_sip.c: When transmitting a register set the socket
+ port to the local one for the transport being used, not the port
+ for the remote server. (closes issue #13633) Reported by:
+ performer
+
+2008-12-10 11:34 +0000 [r162583] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/snmp/agent.c: Make res_snmp.so compile on OpenBSD. OpenBSD
+ uses an old version of gcc which throws an error if you use a
+ macro that's not #defined
+
+2008-12-10 01:09 +0000 [r162542] Joshua Colp <jcolp@digium.com>
+
+ * doc/janitor-projects.txt, channels/iax2-parser.c,
+ apps/app_voicemail.c: Finish conversion to using ARRAY_LEN and
+ remove it as a janitor project. (closes issue #14032) Reported
+ by: bkruse Patches: 14032.patch uploaded by bkruse (license 132)
+
+2008-12-09 23:41 +0000 [r162488] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/stringfields.h: it does help if the compiler
+ attribute syntax is correct
+
+2008-12-09 23:10 +0000 [r162466] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 162463 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09
+ Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........
+
+2008-12-09 22:38 +0000 [r162414-162418] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
+ main/asterisk.c: Add some additional Asterisk project developer
+ documentation. After the nightly update of the documentation on
+ asterisk.org, I'll post an update to asterisk-dev with a pointer
+ to the changes. This covers some release branch and commit policy
+ information. None of this should be a surprise, since it's just
+ documenting what we have already been doing.
+
+ * include/asterisk/utils.h, /, main/utils.c, main/asterisk.c:
+ Merged revisions 162413 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008)
+ | 8 lines Remove the test_for_thread_safety() function
+ completely. The test is not valid. Besides, if we actually
+ suspected that recursive mutexes were not working, we would get a
+ ton of LOG_ERROR messages when DEBUG_THREADS is turned on.
+ (inspired by a discussion on the asterisk-dev list) ........
+
+2008-12-09 21:57 +0000 [r162355] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 162348 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09
+ Dec 2008) | 4 lines We appear to have documented tz= in the
+ [general] section of voicemail.conf, without actually having
+ implemented it. Oops. (Reported by Olivier on the -users list)
+ ........
+
+2008-12-09 21:16 +0000 [r162342] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 162341 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4
+ lines Add 'down' as a valid state for directed call pickup. This
+ creeps up when we receive session progress when dialing a device
+ and not ringing. (closes issue #14005) Reported by: ddl ........
+
+2008-12-09 20:59 +0000 [r162291] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 162286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008)
+ | 9 lines Fix an issue where callers on an incoming call on an
+ SLA trunk would not hear ringback. We need to make sure that we
+ don't start writing audio to the trunk channel until we're
+ actually ready to answer it. Otherwise, the channel driver will
+ treat it as inband progress, even though all they are getting is
+ silence. (closes issue #12471) Reported by: mthomasslo ........
+
+2008-12-09 20:46 +0000 [r162275] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 162273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4
+ lines Fix double declaration of 'x' on the PPC platform. (closes
+ issue #14038) Reported by: ffloimair ........
+
+2008-12-09 20:40 +0000 [r162271] Steve Murphy <murf@digium.com>
+
+ * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162264
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1
+ line In discussion with seanbright on #asterisk-dev, I have added
+ a default rule, and an option to suppress the default rule from
+ being generated in the flex output, for the sake of those OS's
+ where they didn't tweak flex's ECHO macro, and the compiler
+ doesn't like it. The regressions are OK with this. ........
+
+2008-12-09 20:30 +0000 [r162266] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 162265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec
+ 2008) | 6 lines If we fail to start a thread for the pbx to run
+ in, we need to be sure to decrease the number of active calls on
+ the system. This fix may relate to ABE-1713, but it is not
+ certain yet. ........
+
+2008-12-09 19:48 +0000 [r162197-162205] Joshua Colp <jcolp@digium.com>
+
+ * /, main/rtp.c: Merged revisions 162204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7
+ lines Make sure that the timestamp for DTMF is not the same as
+ the previous voice frame and do not send audio when transmitting
+ DTMF as this confuses some equipment. (closes issue #13209)
+ Reported by: ip-rob Patches: 13209.diff uploaded by file (license
+ 11) Tested by: ip-rob, bujones ........
+
+ * /, main/rtp.c: Merged revisions 162188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4
+ lines Take video into account when early bridging RTP. (closes
+ issue #13535) Reported by: davidw ........
+
+2008-12-09 18:35 +0000 [r162079-162140] Steve Murphy <murf@digium.com>
+
+ * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162136
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1
+ line Previous fix used ast_malloc and ast_copy_string and messed
+ up the standalone stuff. Fixed. ........
+
+ * res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael_lex.c,
+ res/ael/ael.flex: Merged revisions 162013 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) |
+ 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches:
+ 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme,
+ murf This crash was the result of a few small errors that would
+ combine in 64-bit land to result in a crash. 32-bit land might
+ have seen these combine to mysteriously drop the args to an
+ application call, in certain circumstances. Also, in trying to
+ find this bug, I spotted a situation in the flex input, where, in
+ passing back a 'word' to the parser, it would allocate a buffer
+ larger than necessary. I changed the usage in such situations, so
+ that strdup was not used, but rather, an ast_malloc, followed by
+ ast_copy_string. I removed a field from the pval struct, in u2,
+ that was never getting used, and set in one spot in the code. I
+ believe it was an artifact of a previous fix to make switch cases
+ work invisibly with extens. And, for goto's I removed a '!' from
+ before a strcmp, that has been there since the initial merging of
+ AEL2, that might prevent the proper target of a goto from being
+ found. This was pretty harmless on its own, as it would just
+ louse up a consistency check for users. Many thanks to
+ ckjohnsonme for providing a simplified and complete set of
+ information about the bug, that helped considerably in finding
+ and fixing the problem. Now, to get aelparse up and running again
+ in trunk, and out of its "horribly broken" state, so I can run
+ the regression suite! ........
+
+2008-12-09 16:47 +0000 [r161951-162016] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_disa.c: Merged revisions 162014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008)
+ | 5 lines Allow DISA to handle extensions that start with #.
+ (closes issue #13330) Reported by: jcovert ........
+
+ * /, main/app.c: Merged revisions 161948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008)
+ | 15 lines Fix a problem with GROUP() settings on a masquerade.
+ The previous code carried over group settings from the old
+ channel to the new one. However, it did nothing with the group
+ settings that were already on the new channel. This patch removes
+ all group settings that already existed on the new channel. I
+ have a more complicated version of this patch which addresses
+ only the most blatant problem with this, which is that a channel
+ can end up with multiple group settings in the same category.
+ However, I could not think of a use case for keeping any of the
+ group settings from the old channel, so I went this route for
+ now. (closes AST-152) ........
+
+2008-12-09 14:49 +0000 [r161947] Eliel C. Sardanons <eliels@gmail.com>
+
+ * funcs/func_odbc.c: Avoid allocating memory for a thread that
+ don't need it. Also, this memory was not being freed until the
+ main thread ends. (That is never). (closes issue #14040) Reported
+ by: eliel Patches: func_odbc.c.patch uploaded by eliel (license
+ 64)
+
+2008-12-08 23:04 +0000 [r161911] Brandon Kruse <bkruse@digium.com>
+
+ * main/pbx.c: Note that the recently changed waittime parameter is
+ in milliseconds.
+
+2008-12-08 21:41 +0000 [r161830-161869] Joshua Colp <jcolp@digium.com>
+
+ * formats/format_pcm.c: Add alw as a valid file extension for alaw
+ and ulw as a valid file extension for ulaw. (closes issue #14001)
+ Reported by: henrikw Patches: alw.diff uploaded by henrikw
+ (license 627)
+
+ * contrib/scripts/autosupport.8, contrib/scripts/autosupport:
+ Update autosupport script with a few changes.
+
+2008-12-08 18:49 +0000 [r161790] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: Allocate enough space initially for the message.
+ (closes issue #14027) Reported by: junky Patches: M14027.diff
+ uploaded by junky (license 177)
+
+2008-12-08 18:47 +0000 [r161726-161787] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c: Fix a regression introduced when the PBX timeouts
+ were converted to milliseconds. collect_digits now gets
+ milliseconds fed to it, not seconds. (closes issue #14012)
+ Reported by: dveiga Patches: 14012.patch uploaded by bkruse
+ (license 132)
+
+ * /, channels/chan_sip.c: Merged revisions 161725 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6
+ lines Make the usereqphone option work again. (closes issue
+ #13474) Reported by: mmaguire Patches: 20080912_bug13474.diff
+ uploaded by mmaguire (license 571) ........
+
+2008-12-08 17:23 +0000 [r161721] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Fix a crash that can occur on a transfer in
+ chan_sip when attempting to collect rtp stats. (closes issue
+ #13956) Reported by: chris-mac Tested by: chris-mac
+
+2008-12-08 16:02 +0000 [r161679] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c, CHANGES: Add the ability to play a courtesy
+ tone to the transfer target in a native SIP attended transfer by
+ setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND.
+
+2008-12-08 04:23 +0000 [r161571-161637] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/xmldoc.c: - Fix a leak while printing an argument
+ description. - Avoid printing the name of an argument in the
+ [Arguments] tag if there is no description for that argument.
+
+ * apps/app_voicemail.c: Add voicemail related applications and
+ functions XML documentation: applications: - VoiceMail() -
+ VoiceMailMain() - MailboxExists() - VMAuthenticate() functions: -
+ MAILBOX_EXISTS()
+
+ * apps/app_sms.c: Introduce SMS() application XML documentation.
+
+2008-12-06 21:18 +0000 [r161536] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_speech_utils.c: Move Speech* applications and functions
+ documentation to XML.
+
+2008-12-05 23:24 +0000 [r161493] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_stack.c: If the autoloop flag is set on a channel, then
+ we need to add 1 to the priority when checking if the extension
+ exists. Otherwise, gosubs will fail. This was discovered when
+ investigating an asterisk-users mailing list post made by Gary
+ Hawkins.
+
+2008-12-05 21:08 +0000 [r161349-161427] Sean Bright <sean.bright@gmail.com>
+
+ * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions
+ 161426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r161426 | seanbright | 2008-12-05 16:02:20 -0500
+ (Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
+ 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
+ int). (closes issue #14006) Reported by: alphaque Patches:
+ astobj2.h-patch uploaded by alphaque (license 259) (Slightly
+ modified by seanbright) ........ ................
+
+ * apps/app_voicemail.c: Use ast_free() instead of free(), pointed
+ out by eliel on IRC.
+
+ * apps/app_voicemail.c: When using IMAP_STORAGE, it's important to
+ convert bare newlines (\n) in emailbody and pagerbody to CR-LF so
+ that the IMAP server doesn't spit out an error. This was
+ informally reported on #asterisk-dev a few weeks ago. Reviewed by
+ Mark M. on IRC.
+
+2008-12-05 14:16 +0000 [r161252-161288] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 161287 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008)
+ | 2 lines Fix a NULL format string warning found by buildbot.
+ ........
+
+ * apps/app_minivm.c: Resolve a compiler warning from buildbot about
+ a NULL format string.
+
+2008-12-05 10:31 +0000 [r161218] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/udptl.c, main/frame.c, res/res_musiconhold.c,
+ channels/chan_iax2.c, res/res_jabber.c, res/res_config_sqlite.c,
+ main/config.c, main/cli.c, channels/chan_dahdi.c, main/manager.c,
+ channels/chan_skinny.c, res/res_agi.c, main/features.c,
+ apps/app_minivm.c, pbx/pbx_ael.c, main/logger.c, main/http.c,
+ res/res_realtime.c, channels/chan_alsa.c, res/res_config_ldap.c,
+ apps/app_rpt.c, main/db.c, res/res_config_pgsql.c, main/pbx.c,
+ channels/chan_sip.c, main/translate.c, channels/chan_agent.c,
+ res/res_convert.c, res/res_crypto.c, apps/app_queue.c,
+ channels/chan_oss.c, apps/app_playback.c,
+ channels/chan_usbradio.c, main/file.c, main/astmm.c,
+ pbx/pbx_dundi.c, res/res_indications.c, pbx/pbx_config.c,
+ apps/app_mixmonitor.c, res/res_odbc.c, main/asterisk.c,
+ apps/app_voicemail.c: Janitor, use ARRAY_LEN() when possible.
+ (closes issue #13990) Reported by: eliel Patches: array_len.diff
+ uploaded by eliel (license 64)
+
+2008-12-05 05:41 +0000 [r161181] Tilghman Lesher <tlesher@digium.com>
+
+ * main/config.c: The first file should have a blank config filename
+ in the structure, so that when a save occurs to a different
+ filename, everything goes to the alternate filename, instead of
+ appending to the original. This is important for the AMI command
+ UpdateConfig. (closes issue #13301) Reported by: trevo Patches:
+ 20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
+ 20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license
+ 14) Tested by: Corydon76, blitzrage
+
+2008-12-05 02:47 +0000 [r161147] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_voicemail.c: Check the return value of fread/fwrite so
+ the compiler doesn't complain. Only a problem when IMAP_STORAGE
+ is enabled.
+
+2008-12-04 23:00 +0000 [r161115] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: If
+ 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it
+ exists) after T38 is negotiated. Terry Wilson created the
+ original patch for this functionality, which I slightly modified
+ and added the faxdetect=yes|no configuration option. This patch
+ is only for T38 fax detection and does not do anything for G711
+ over SIP fax detection. By default, this option is disabled.
+ Reviewboard: http://reviewboard.digium.com/r/69/ This
+ functionality is for issue AST-140.
+
+2008-12-04 19:31 +0000 [r161077] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/cli.c: Fix minor coding guidelines introduced with CLI
+ permissions.
+
+2008-12-04 18:32 +0000 [r161014] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/rtp.c: Merged revisions 161013 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008)
+ | 9 lines (closes issue #13835) Reported by: matt_b Tested by:
+ jpeeler This mirrors a check that was present in ast_rtp_read to
+ also be in ast_rtp_raw_write to not schedule sending the receiver
+ report if the remote RTCP endpoint address isn't present in the
+ RTCP structure. Closes AST-142. ........
+
+2008-12-04 16:45 +0000 [r160945] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/callerid.c: Merged revisions 160943 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec
+ 2008) | 15 lines Fix a callerid parsing issue. If someone
+ formatted callerid like the following: "name <number>" (including
+ the quotation marks), then the parts would be parsed as name:
+ "name number: number This is because the closing quotation mark
+ was not discovered since the number and everything after was
+ parsed out of the string earlier. Now, there is a check to see if
+ the closing quote occurs after the number, so that we can know if
+ we should strip off the opening quote on the name. Closes AST-158
+ ........
+
+2008-12-04 16:37 +0000 [r160938] Michiel van Baak <michiel@vanbaak.info>
+
+ * build_tools/cflags-devmode.xml, channels/chan_skinny.c: Add debug
+ flag so skinny debug will show information about packets. We dont
+ want to scare users with this, so we added a devmode compile flag
+ (closes issue #13952) Reported by: wedhorn Patches:
+ packetdebug3.diff uploaded by wedhorn (license 30) Tested by:
+ mvanbaak, wedhorn
+
+2008-12-04 13:45 +0000 [r160896] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_agi.c: Added XML documentation for the following AGI
+ commands: - get option - get variable - hangup - noop
+
+2008-12-04 01:36 +0000 [r160854-160856] Richard Mudgett <rmudgett@digium.com>
+
+ * funcs/func_callerid.c: Jcolp pointed out that num will also match
+ number
+
+ * funcs/func_callerid.c: * Found a couple more places where
+ num/number needed to be done so 1.4 upgraders will not have
+ problems. * Added curly braces and minor tweaks.
+
+2008-12-03 21:58 +0000 [r160791] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 160770 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03
+ Dec 2008) | 2 lines Some compilers warn on null format strings;
+ some don't (caught by buildbot) ........
+
+2008-12-03 21:09 +0000 [r160760] Steve Murphy <murf@digium.com>
+
+ * /, funcs/func_callerid.c: Merged revisions 160703 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec
+ 2008) | 11 lines (closes issue #13597) Reported by: john8675309
+ Patches: patch.13597 uploaded by murf (license 17) Tested by:
+ murf, john8675309 This patch causes the setcid func to update the
+ CDR clid after setting the channel field. I also notice that in
+ trunk, the num/number of 1.4 is left out; I decided to include
+ the option to use either in trunk, so as not to have 1.4
+ upgraders not to have problems. ........
+
+2008-12-03 20:35 +0000 [r160699-160700] Jason Parker <jparker@digium.com>
+
+ * main/manager.c: Another place this is missing
+
+ * main/manager.c: Fix typo when ListCategories returns none.
+ (closes issue #13994) Reported by: mika Patches:
+ ListCategoriesActionPatch.diff uploaded by mika (license 624)
+
+2008-12-03 19:25 +0000 [r160663] Eliel C. Sardanons <eliels@gmail.com>
+
+ * channels/iax2-provision.c: - iax2-provision was not freeing
+ iax_templates structure when unloading the chan_iax2.so module. -
+ Move the code to start using the LIST macros. Review:
+ http://reviewboard.digium.com/r/72 (closes issue #13232) Reported
+ by: eliel Patches: iax2-provision.patch.txt uploaded by eliel
+ (license 64) (with minor changes pointed by Mark Michelson on
+ review board) Tested by: eliel
+
+2008-12-03 18:37 +0000 [r160626] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, apps/app_queue.c, apps/app_stack.c: Add some
+ safety measures when using gosub, especially when using the
+ options for app_dial and app_queue to run a gosub when the call
+ is answered. * Check for the existence of the gosub target in
+ gosub_exec. If it is nonexistent, then this will cause errors
+ when we attempt to actually run the gosub, including a definite
+ memory leak and potential crashes. Return an error in this
+ situation * Check the return value of pbx_exec in app_dial and
+ app_queue before attempting to actually run the gosub routine. If
+ there was an error, we should not attempt to run the gosub. *
+ Change a '|' to a ',' in app_queue. * Add some extra curly braces
+ where they had been missing previously. (closes issue #13548)
+ Reported by: fiddur
+
+2008-12-03 17:48 +0000 [r160562] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_minivm.c: - Add <variable /> tags when naming a channel
+ variable. - Add <filename /> tags when naming a filename. -
+ Simplify the xml formatting putting some enters.
+
+2008-12-03 17:38 +0000 [r160559] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c, /: Merged revisions 160558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008)
+ | 7 lines If an entry is added to the directory during a scan
+ when another entry expires, then that new entry will not be
+ processed promptly, but must wait for either a future entry to
+ start or a current entry's retry to occur. If no other entries
+ exist in the directory (other than the new entries) when a bunch
+ expire, then the new entries must wait until another new entry is
+ added to be processed. This was a rather weird race condition,
+ really. Fixes AST-147. ........
+
+2008-12-03 17:07 +0000 [r160555] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: When investigating issue #13548, I found that
+ gosub handling in app_queue was just completely wrong, mostly
+ because the channel operations being performed were being done on
+ the incorrect channel. With this set of changes, a gosub will
+ correctly run on the answering queue member's channel. There are
+ still crash issues which occur if there are dialplan syntax
+ errors, so I cannot yet close the referenced issue.
+
+2008-12-03 17:01 +0000 [r160481-160552] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c, /: Merged revisions 160551 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008)
+ | 4 lines Don't start scanning the directory until all modules
+ are loaded, because some required modules (channels, apps,
+ functions) may not yet be in memory yet. Fixes AST-149. ........
+
+ * /, channels/chan_sip.c: Merged revisions 160480 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008)
+ | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I
+ guess that having only ip-phones in mind is not a good approach.
+ Since it is possible to have a sip proxy connected to asterisk we
+ could receive a 407 (unauthorized) or 483 (too many hops) as
+ response and dialog ending would not be a good behavior." So
+ modified. ........
+
+2008-12-03 11:01 +0000 [r160447] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_stack.c: - Avoid setting .synopsis and .syntax if we are
+ using XML documentation (or the xml documentation wont be
+ loaded). - Use <variable></variable> to refer to a dialplan
+ variable.
+
+2008-12-02 18:48 +0000 [r160344-160346] Tilghman Lesher <tlesher@digium.com>
+
+ * CHANGES: Info on LOCAL_PEEK function.
+
+ * apps/app_stack.c: Add LOCAL_PEEK function, as requested by
+ lmadsen.
+
+2008-12-02 18:04 +0000 [r160319-160333] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: remove duplicate comment that I
+ accidentally merged
+
+ * channels/chan_dahdi.c: (closes issue #13786) Reported by: tzafrir
+ Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260
+ which fixes not being able to make outgoing calls on some FXO
+ adapters:
+ http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553
+
+2008-12-02 17:56 +0000 [r160208-160308] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 160297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008)
+ | 10 lines When the text does not match exactly (e.g. RTP/SAVP),
+ then the %n conversion fails, and the resulting integer is
+ garbage. Thus, we must initialize the integer and check it
+ afterwards for success. (closes issue #14000) Reported by: folke
+ Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke
+ (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by
+ folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
+ uploaded by folke (license 626) ........
+
+ * main/pbx.c, main/frame.c, /, channels/chan_features.c,
+ include/asterisk/stringfields.h, apps/app_voicemail.c,
+ main/cli.c: Merged revisions 160207 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008)
+ | 3 lines Ensure that Asterisk builds with --enable-dev-mode,
+ even on the latest gcc and glibc. ........
+
+2008-12-01 23:37 +0000 [r160170-160172] Sean Bright <sean.bright@gmail.com>
+
+ * main/manager.c, /: Merged revisions 159976 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008)
+ | 3 lines Get rid of the useless format string and argument in
+ the Bogus/ manager channelname. Noted by kpfleming and name
+ Bogus/manager suggested by eliel ........
+
+ * channels/chan_phone.c: Silence a build warning.
+ (chan_phone.c:810: warning: value computed is not used)
+
+ * utils/smsq.c: Pay attention to the return value of system(), even
+ if we basically ignore it.
+
+2008-12-01 21:23 +0000 [r160097] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Use AST_EXT_LIB_SETUP before using
+ AST_EXT_LIB_CHECK or bad things happen.
+
+2008-12-01 18:52 +0000 [r160062] Eliel C. Sardanons <eliels@gmail.com>
+
+ * configs/cli_permissions.conf.sample (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/cli.h, include/asterisk/_private.h, CHANGES,
+ main/asterisk.c, main/cli.c: Introduce CLI permissions. Based on
+ cli_permissions.conf configuration file, we are able to permit or
+ deny cli commands based on some patterns and the local user and
+ group running rasterisk. (Sorry if I missed some of the testers).
+ Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue
+ #11123) Reported by: eliel Tested by: eliel, IgorG, Laureano,
+ otherwiseguy, mvanbaak
+
+2008-12-01 17:34 +0000 [r159911-160004] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 160003 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01
+ Dec 2008) | 6 lines Apply some logic used in iax2_indicate() to
+ iax2_setoption(), as well, since they both have the potential to
+ send control frames in the middle of call setup. We have to wait
+ until we have received a message back from the remote end before
+ we try to send any more frames. Otherwise, the remote end will
+ consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
+ ........
+
+ * /, .cleancount: Merged revisions 159900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008)
+ | 2 lines Force a "make clean" to avoid a bizarre build issue ...
+ ........
+
+2008-12-01 14:09 +0000 [r159898] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/manager.c, /: Merged revisions 159897 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008)
+ | 4 lines make manager compile on OpenBSD. The last (10th)
+ argument to ast_channel_alloc here should be a pointer and NULL
+ is not really a pointer. ........
+
+2008-11-29 18:33 +0000 [r159853] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_readexten.c: Allow the '#' sign to exist within an
+ extension (inspired by issue #13330)
+
+2008-11-29 17:57 +0000 [r159774-159818] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_vpb.cc, /, main/utils.c, channels/chan_iax2.c,
+ utils/frame.c, include/asterisk/astmm.h, configure,
+ include/asterisk/compat.h, main/features.c,
+ include/asterisk/module.h, main/logger.c,
+ include/asterisk/dlinkedlists.h, main/dns.c,
+ include/asterisk/utils.h, include/asterisk/devicestate.h,
+ channels/chan_sip.c, include/asterisk/dundi.h,
+ include/asterisk/enum.h, configure.ac, channels/chan_agent.c,
+ include/asterisk/config.h, utils/astman.c,
+ include/asterisk/cli.h, include/asterisk/channel.h,
+ include/jitterbuf.h, include/asterisk/manager.h,
+ utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c,
+ include/asterisk/logger.h, Makefile, include/asterisk/res_odbc.h,
+ main/srv.c, channels/chan_misdn.c,
+ include/asterisk/linkedlists.h, main/event.c,
+ include/asterisk/lock.h, include/asterisk/strings.h,
+ utils/extconf.c, makeopts.in, include/asterisk/stringfields.h,
+ main/xmldoc.c, utils/check_expr.c: incorporates r159808 from
+ branches/1.4:
+ ------------------------------------------------------------------------
+ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov
+ 2008) | 7 lines update dev-mode compiler flags to match the ones
+ used by default on Ubuntu Intrepid, so all developers will see
+ the same warnings and errors since this branch already had some
+ printf format attributes, enable checking for them and tag
+ functions that didn't have them format attributes in a consistent
+ way
+ ------------------------------------------------------------------------
+ in addition: move some format attributes from main/utils.c to the
+ header files they belong in, and fix up references to the
+ relevant functions based on new compiler warnings
+
+ * Makefile, funcs/func_sprintf.c (added), main/Makefile,
+ channels/misdn/ie.c, funcs/func_strings.c, UPGRADE.txt,
+ res/res_config_sqlite.c, channels/misdn_config.c, funcs/Makefile:
+ we can now build with -Wformat=2, which found a couple of real
+ bugs because SPRINTF() use non-literal format strings (which
+ cannot be checked), move it into its own module so the rest of
+ func_strings can benefit from format string checking
+
+2008-11-28 14:20 +0000 [r159734] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/Makefile: Make res_config_ldap compile with the official
+ OpenLDAP 2.3.X versions. They removed the LDAP_DEPRECATED define
+ from their source and since we are using a couple of deprecated
+ function calls we should define it with a CFLAG. Tested by me on
+ OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps
+ compiling. It shouldn't break, we only define the LDAP_DEPRECATED
+ with this which is what all 2.2.X and older versions of OpenLDAP
+ did in their own tree.
+
+2008-11-27 20:29 +0000 [r159701] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Removed duplicate code
+
+2008-11-26 22:11 +0000 [r159664-159666] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Make a formatting change to test a new post-commit
+ hook for reviewboard. http://reviewboard.digium.com/r/65/
+
+ * main/pbx.c: Make a formatting change to test a new post-commit
+ hook for reviewboard. http://reviewboard.digium.com/r/65/
+
+ * main/pbx.c: Make a formatting change to test a new post-commit
+ hook for reviewboard. http://reviewboard.digium.com/r/65/
+
+2008-11-26 21:20 +0000 [r159629-159631] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/agi.h, configure,
+ include/asterisk/autoconfig.h.in, contrib/asterisk-ng-doxygen,
+ autoconf/ast_gcc_attribute.m4, configure.ac, res/res_agi.c,
+ apps/app_stack.c, include/asterisk/optional_api.h (added):
+ improve handling of API calls provided by loaded modules through
+ use of some GCC features; this makes app_stack's usage of AGI
+ APIs even cleaner, and will allow it to work 'as expected' either
+ with or without res_agi being loaded reviewed at
+ http://reviewboard.digium.com/r/62
+
+ * main/manager.c, CHANGES: add support for event suppression for
+ AMI-over-HTTP
+
+2008-11-26 19:57 +0000 [r159554] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Add some necessary hangup commands in the case
+ that forwarding a call fails 1) Hang up the original destination
+ if the local channel cannot be requested. 2) Hang up the local
+ channel (in addition to the original destination) if ast_call
+ fails when calling the newly created local channel. This prevents
+ channels from sticking around forever in the case of a botched
+ call forward (e.g. to an extension which does not exist). (closes
+ issue #13764) Reported by: davidw Patches: 13764_v2.patch
+ uploaded by putnopvut (license 60) Tested by: putnopvut, davidw
+
+2008-11-26 19:08 +0000 [r159534] Kevin P. Fleming <kpfleming@digium.com>
+
+ * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules,
+ Makefile.rules: Merged revisions 159476 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov
+ 2008) | 7 lines simplify (and slightly bug-fix) the recent
+ developer-oriented COMPILE_DOUBLE mode ensure that 'make clean'
+ removes dependency files for .i files that are created in
+ COMPILE_DOUBLE mode ........
+
+2008-11-26 18:33 +0000 [r159475] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c: If the config file does not exist, then the first
+ use crashes Asterisk. (closes issue #13848) Reported by:
+ klaus3000 Patches: udptl.c.patch uploaded by eliel (license 64)
+ Tested by: blitzrage
+
+2008-11-26 14:58 +0000 [r159437] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Don't allow for configuration options to
+ overwrite options set via channel variables on a reload. (closes
+ issue #13921) Reported by: davidw Patches: 13921.patch uploaded
+ by putnopvut (license 60) Tested by: davidw
+
+2008-11-26 03:18 +0000 [r159402] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Always parse arguments in park_call_exec so that
+ app_args is valid. This prevents a crash when executing Park from
+ the dialplan with no arguments.
+
+2008-11-25 23:03 +0000 [r159360] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, /, channels/chan_iax2.c: Merged revisions 159316 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) |
+ 15 lines (closes issue #12694) Reported by: yraber Patches:
+ 12694.2nd.diff uploaded by murf (license 17) Tested by: murf,
+ laurav Thanks to file (Joshua Colp) for his IAX fix. the change
+ to cdr.c allows no-answer to percolate up into CDR's, and feels
+ like the right place to locate this fix; if BUSY is done here,
+ no-answer should be, too. ........
+
+2008-11-25 22:45 +0000 [r159276-159317] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+ include/asterisk/dsp.h, CHANGES, main/dsp.c: Add an option,
+ waitfordialtone, for UK analog lines which do not end a call
+ until the originating line hangs up. (closes issue #12382)
+ Reported by: one47 Patches:
+ zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license
+ 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed
+ (license 463) Tested by: fleed
+
+ * /, channels/chan_iax2.c: Merged revisions 159269 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25
+ Nov 2008) | 7 lines Don't try to send a response on a NULL pvt.
+ (closes issue #13919) Reported by: barthpbx Patches:
+ chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
+ barthpbx ........
+
+2008-11-25 21:49 +0000 [r159250] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_followme.c: Make the options for the general and
+ profiles more consistent for the "pls_hold_prompt" option. This
+ does not affect any released version of Asterisk, so there is no
+ need to update the CHANGES file for this. (closes issue #13893)
+ Reported by: eliel
+
+2008-11-25 21:42 +0000 [r159162-159247] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 159246 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600
+ (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008)
+ | 7 lines Regression fix for last security fix. Set the iseqno
+ correctly. (closes issue #13918) Reported by: ffloimair Patches:
+ 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: ffloimair ........ ................
+
+ * pbx/pbx_realtime.c: Don't actually do anything with a negative
+ priority, because we ignore it in the result, anyway.
+
+ * main/pbx.c: Don't limit the length of the hint at the final step
+ (from ~8100 chars max (or ~500 chars max on LOW_MEMORY) to 80
+ chars max). This will allow more channels to be used in a single
+ hint.
+
+2008-11-25 16:18 +0000 [r159093] Terry Wilson <twilson@digium.com>
+
+ * apps/app_festival.c: Add missing variable declaration for PPC
+ code
+
+2008-11-25 05:19 +0000 [r159050-159054] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_readexten.c: Copyright clarification; also, have
+ variable set to "t" or "i" on timeout or invalid extension,
+ respectively. (closes issue #13944) Reported by: chappell
+
+ * channels/chan_usbradio.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/xpmr/xpmr.c, apps/app_rpt.c: Merged revisions 159025 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008)
+ | 3 lines System call ioperm is non-portable, so check for its
+ existence in autoconf. (Closes issue #13863) ........
+
+2008-11-25 03:49 +0000 [r158992] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_usbradio.c: Make chan_usbradio compile under dev
+ mode
+
+2008-11-25 01:01 +0000 [r158959] Sean Bright <sean.bright@gmail.com>
+
+ * funcs/func_dialgroup.c, channels/chan_sip.c,
+ include/asterisk/astobj2.h, res/res_phoneprov.c,
+ main/taskprocessor.c, channels/chan_console.c,
+ channels/chan_iax2.c, apps/app_queue.c, main/astobj2.c,
+ main/config.c, main/manager.c, res/res_timing_pthread.c,
+ main/features.c, res/res_timing_timerfd.c, utils/hashtest2.c,
+ res/res_clialiases.c: This is basically a complete rollback of
+ r155401, as it was determined that it would be best to maintain
+ API compatibility. Instead, this commit introduces
+ ao2_callback_data() which is functionally identical to
+ ao2_callback() except that it allows you to pass arbitrary data
+ to the callback. Reviewed by Mark Michelson via ReviewBoard:
+ http://reviewboard.digium.com/r/64
+
+2008-11-25 00:19 +0000 [r158876-158925] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/file.c: Fix compiling in dev mode.
+
+ * UPGRADE.txt, apps/app_queue.c: Make the Join event from app_queue
+ use CallerIDNum insead of CallerID for indicating the callerid
+ number just like the rest of asterisk. (closes issue #13883)
+ Reported by: davidw
+
+ * main/manager.c, res/res_agi.c, include/asterisk/manager.h: Added
+ EVENT_FLAG_AGI and used it for manager calls in res_agi.c (closes
+ issue #13873) Reported by: fnordian Patches: ami_agievent.patch
+ uploaded by fnordian (license 110)
+
+2008-11-24 21:52 +0000 [r158857] Tilghman Lesher <tlesher@digium.com>
+
+ * main/dsp.c: Add a bit of documentation (thanks, I-MOD) on what
+ the silence threshold constant actually does and what values are
+ valid for it.
+
+2008-11-24 21:27 +0000 [r158851] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/file.c: Make ast_streamfile() check the result of
+ ast_openstream() before doing anything with it. (closes issue
+ #13955) Reported by: chris-mac
+
+2008-11-24 18:11 +0000 [r158808] Terry Wilson <twilson@digium.com>
+
+ * apps/app_minivm.c: This patch adds a new application for sending
+ MWI to phones via Asterisk's event subsystem. Also, the minivm
+ documentation is all converted to use xmldocs. (closes issue
+ #13946) Reported by: Marquis Patches:
+ minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32)
+ Tested by: otherwiseguy, Marquis
+
+2008-11-23 03:36 +0000 [r158754-158756] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: If you enabled
+ 'notifycid' one of the limitations is that the calling channel is
+ only found if it dialed the extension that was subscribed to. You
+ can now specify 'ignore-context' for the 'notifycid' option in
+ sip.conf which will, as it's value implies, ignore the current
+ context of the caller when doing the lookup.
+
+ * channels/chan_sip.c: No need to use a separate structure for this
+ since we can just pass our sip_pvt pointer in directly.
+
+2008-11-22 17:17 +0000 [r158686-158723] Michiel van Baak <michiel@vanbaak.info>
+
+ * funcs/func_realtime.c: last commit worked on OpenBSD but still
+ generated warning on Ubuntu. Initialise a variable so
+ --enable-dev-mode does not complain
+
+ * channels/chan_skinny.c: dont send reorder tone after a device is
+ hungup if a dialout is abandoned or failed. Without this reorder
+ tone will play after hangup and both wedhorn's and my wife have
+ threatened to use an axe on our asterisk box (closes issue
+ #13948) Reported by: wedhorn Patches: switch.diff uploaded by
+ wedhorn (license 30)
+
+ * channels/chan_skinny.c: Add Media Source Update to skinny's
+ control2str (issue #13948)
+
+ * channels/chan_skinny.c: fix a very occasional core dump in
+ chan_skinny found by wedhorn. (issue #13948)
+
+ * funcs/func_realtime.c: make this compile under devmode
+
+2008-11-21 23:40 +0000 [r158606] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 158603 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) |
+ 11 lines In reference to the fix made for 13871, I was merging
+ the fix into 1.6.0 and realized I missed the code in the h-exten
+ block, and didn't catch it because my test case had the h-exten
+ commented out. So, this corrects the code I missed, as a
+ preventative against another crash report. Tested with the
+ h-exten defined, all is well. ........
+
+2008-11-21 23:33 +0000 [r158602-158605] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Allow space within an extension, when the space is
+ within a character class. (requested by lmadsen on -dev, patch by
+ me)
+
+ * main/pbx.c, /: Merged revisions 158600 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008)
+ | 5 lines The passed extension may not be the same in the list as
+ the current entry, because we strip spaces when copying the
+ extension into the structure. Therefore, use the copied item to
+ place the item into the list. (found by lmadsen on -dev, fixed by
+ me) ........
+
+2008-11-21 22:12 +0000 [r158540] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions
+ 158539 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008)
+ | 2 lines When compiling with DEBUG_THREADS, report the real
+ file/func/line for ao2_lock/ao2_unlock ........
+
+2008-11-21 21:47 +0000 [r158484] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 158483 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) |
+ 11 lines (closes issue #13871) Reported by: mdu113 This one is
+ totally my fault. The code doesn't even create a bridge CDR if
+ the channel CDR has POST_DISABLED. I didn't check for that at the
+ end of the bridge. Fixed with a few small insertions. Tested.
+ Looks good. No cdr generated, no crash, no unnecc. data objects
+ created either. ........
+
+2008-11-21 21:06 +0000 [r158482] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: Fix for #13963. Make physical channel
+ mapping unconfigured default
+
+2008-11-21 20:42 +0000 [r158449] Kevin P. Fleming <kpfleming@digium.com>
+
+ * UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, UPGRADE-1.6.txt,
+ CHANGES: as suggested by jtodd, document the purposes of the
+ CHANGES and UPGRADE files
+
+2008-11-21 19:40 +0000 [r158414] Jason Parker <jparker@digium.com>
+
+ * main/manager.c: Make sure we add the Event header for
+ CoreShowChannels. (closes issue #13334) Reported by: srt Patches:
+ 13334_missing_event_header_in_core_show_channel.diff uploaded by
+ srt (license 378)
+
+2008-11-21 17:08 +0000 [r158374] Terry Wilson <twilson@digium.com>
+
+ * cdr/cdr_csv.c: Reloading the config and having no changes still
+ initialized some settings to 0. Initialize settings after doing
+ all of the cfg checks. (closes issue #13942) Reported by: davidw
+ Patches: cdr_diff.txt uploaded by otherwiseguy (license 396)
+ Tested by: davidw
+
+2008-11-21 15:53 +0000 [r158315] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_sip.c: Add fix to prevent crash during reload if
+ there is an outstanding MWI registration message pending.
+
+2008-11-21 01:22 +0000 [r158230-158266] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Use a more expressive constant for a 64-bit
+ scanned int
+
+ * channels/chan_sip.c: Use some magic constants to get the right
+ size for this sscanf statement. Thanks Richard!
+
+ * channels/chan_sip.c: Fix the build for 32-bit systems. %lu is
+ only 32-bits on 32-bit systems, so we need to use %llu instead.
+ Of course %llu is 128-bits on 64-bit systems, so we have to cast
+ to unsigned long long. No harm, but it's sure annoying.
+
+ * channels/chan_sip.c: Change the remote user agent session version
+ variable from an int to a uint64_t. This prevents potential
+ comparison problems from happening if the version string exceeds
+ INT_MAX. This was an apparent problem for one user who could not
+ properly place a call on hold since the version in the SDP of the
+ re-INVITE to place the call on hold greatly exceeded INT_MAX.
+ This also aligns with RFC 2327 better since it recommends using
+ an NTP timestamp for the version (which is a 64-bit number).
+ (closes issue #13531) Reported by: sgofferj Patches: 13531.patch
+ uploaded by putnopvut (license 60) Tested by: sgofferj
+
+2008-11-20 19:41 +0000 [r158188] Sean Bright <sean.bright@gmail.com>
+
+ * res/ael/pval.c: Fix one case where the application argument was
+ not converted from a pipe to a comma. This was causing problems
+ with switch statements with empty expressions. (closes issue
+ #13901) Reported by: smurfix Patches: 20081118_bug13901.diff
+ uploaded by seanbright (license 71) Tested by: seanbright
+ Reviewed by: murf
+
+2008-11-20 18:20 +0000 [r158082-158133] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/file.h, main/frame.c, /, channels/chan_sip.c,
+ main/file.c, include/asterisk/frame.h: Merged revisions 158072
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20
+ Nov 2008) | 2 lines Begin on a crusade to end trailing
+ whitespace! ........
+
+ * /, channels/chan_sip.c: Merged revisions 158071 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov
+ 2008) | 16 lines We don't handle 4XX responses to BYE well.
+ According to section 15 of RFC 3261, we should terminate a dialog
+ if we receive a 481 or 408 in response to our BYE. Since I am
+ aware of at least one phone manufacturer who may sometimes send a
+ 404 as well, I am being liberal and saying that any 4XX response
+ to a BYE should result in a terminated dialog. (closes issue
+ #12994) Reported by: pabelanger Patches: 12994.patch uploaded by
+ putnopvut (license 60) Closes AST-129 ........
+
+2008-11-20 17:53 +0000 [r158078] Ryan Brindley <rbrindley@digium.com>
+
+ * main/config.c: more formatting corrections :: one line for loops
+ and if statements still need {}
+
+2008-11-20 17:48 +0000 [r158072] Terry Wilson <twilson@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, cdr/cdr_sqlite.c, cdr/Makefile,
+ cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_odbc.c,
+ cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c,
+ cdr/cdr_csv.c: Begin on a crusade to end trailing whitespace!
+
+2008-11-20 17:46 +0000 [r158070] Ryan Brindley <rbrindley@digium.com>
+
+ * main/config.c: formatting changes :: one line for loops and if
+ statements should have {}
+
+2008-11-20 17:39 +0000 [r158066] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 158053
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov
+ 2008) | 12 lines Make sure to set the hangup cause on the calling
+ channel in the case that ast_call() fails. For incoming SIP
+ channels, this was causing us to send a 603 instead of a 486 when
+ the call-limit was reached on the destination channel. (closes
+ issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded
+ by putnopvut (license 60) Tested by: blitzrage ........
+
+2008-11-20 17:37 +0000 [r158062] Jeff Peeler <jpeeler@digium.com>
+
+ * main/file.c: (closes issue #12929) Reported by: snyfer This
+ handles the case for a zero length file to attempt to be
+ streamed. Instead of failing from not playing any data, go ahead
+ and return success as ast_streamfile should consider playing
+ nothing a success when there is nothing to play.
+
+2008-11-20 17:37 +0000 [r158061] Jason Parker <jparker@digium.com>
+
+ * README: Whitespace fix
+
+2008-11-20 00:08 +0000 [r157974] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/stdtime, /, main/db1-ast/hash, codecs/gsm/Makefile,
+ Makefile.moddir_rules, main/db1-ast/db, channels/misdn,
+ main/db1-ast/mpool, res/ais, res/Makefile, pbx/Makefile,
+ Makefile.rules, res/snmp, main/stdtime/Makefile, codecs/gsm/src,
+ main/db1-ast/btree, channels/misdn/Makefile, main/db1-ast/recno,
+ res/ael, pbx/ael, channels, main/db1-ast/Makefile: Merged
+ revisions 157859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov
+ 2008) | 7 lines the gcc optimizer frequently finds broken code
+ (use of uninitalized variables, unreachable code, etc.), which is
+ good. however, developers usually compile with the optimizer
+ turned off, because if they need to debug the resulting code,
+ optimized code makes that process very difficult. this means that
+ we get code changes committed that weren't adequately checked
+ over for these sorts of problems. with this build system change,
+ if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is
+ turned on, when a source file is compiled it will actually be
+ preprocessed (into a .i or .ii file), then compiled once with
+ optimization (with the result sent to /dev/null) and again
+ without optimization (but only if the first compile succeeded, of
+ course). while making these changes, i did some cleanup work in
+ Makefile.rules to move commonly-used combinations of flag
+ variables into their own variables, to make the file easier to
+ read and maintain ........
+
+2008-11-20 00:06 +0000 [r157973] Terry Wilson <twilson@digium.com>
+
+ * res/res_timing_timerfd.c: Fix compiling
+
+2008-11-19 23:30 +0000 [r157906-157940] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Add a space to the output
+
+ * apps/app_queue.c: Add a RES_NOT_DYNAMIC case for the CLI command
+ 'queue remove member'
+
+ * CHANGES: Commit CHANGES change I promised when submitting
+ res_timing_timerfd
+
+2008-11-19 22:01 +0000 [r157893] Tilghman Lesher <tlesher@digium.com>
+
+ * CHANGES: Add info about REALTIME_FIELD and REALTIME_HASH
+
+2008-11-19 21:55 +0000 [r157874] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_timing_timerfd.c: Cast this value since a uint64_t is not
+ the same as an unsigned long long on a 64-bit machine. Reported
+ by kpfleming on IRC
+
+2008-11-19 21:54 +0000 [r157870] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_realtime.c: Two new functions, REALTIME_FIELD, and
+ REALTIME_HASH, which should make querying realtime from the
+ dialplan a little more consistent and easy to use. The original
+ REALTIME function is preserved, for those who are already
+ accustomed to that interface. (closes issue #13651) Reported by:
+ Corydon76 Patches: 20081119__bug13651__2.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: blitzrage, Corydon76
+
+2008-11-19 19:37 +0000 [r157820] Mark Michelson <mmichelson@digium.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, res/res_timing_pthread.c,
+ configure.ac, res/res_timing_dahdi.c, res/res_timing_timerfd.c
+ (added), makeopts.in: Merge the changes from the
+ res_timing_timerfd branch. This provides a new timing interface.
+ In order to use it, you must be running a Linux with a kernel
+ version of 2.6.25 or newer and glibc 2.8 or newer. This timing
+ interface is a good alternative if a timing source is necessary
+ (e.g. for IAX trunking) but DAHDI is otherwise unnecessary for
+ the system. For now, this commit contains the actual work done in
+ the res_timing_timerfd branch. There are no notices in the README
+ or CHANGES files yet, but they will be added in my next commit.
+ The timing API of Asterisk also needs to have a bit of work done
+ with regards to choosing which timing interface to use. This
+ commit makes the choice a build-time decision, by only allowing
+ one of the timer interfaces to be chosen in menuselect. It would
+ be preferable if the choice could be made at run-time, however.
+ The preferred timing interface could be loaded and tested, and if
+ it does not work, choice number two may be used instead. That
+ sort of thing. That is beyond the scope of work in this branch
+ though.
+
+2008-11-19 19:25 +0000 [r157818] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_vpb.cc, cdr/cdr_sqlite3_custom.c,
+ channels/iax2-provision.c, cdr/cdr_adaptive_odbc.c,
+ cdr/cdr_pgsql.c, cdr/cdr_radius.c, cdr/cdr_tds.c,
+ channels/misdn_config.c, cdr/cdr_csv.c, channels/chan_usbradio.c,
+ channels/chan_skinny.c, main/logger.c, res/ais/evt.c,
+ pbx/pbx_dundi.c, cdr/cdr_odbc.c, cdr/cdr_custom.c,
+ cdr/cdr_manager.c, main/xmldoc.c, res/res_clialiases.c: Fix
+ checking for CONFIG_STATUS_FILEINVALID so that modules don't
+ crash upon trying to parse an invalid config
+
+2008-11-19 18:28 +0000 [r157784] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Add check for t38_terminal_init in
+ spandsp (not found in 0.0.6, so it should fail reasonably)
+ (closes issue #13473) Reported by: genie Patches:
+ 20080916__bug13473.diff.txt uploaded by Corydon76 (license 14)
+
+2008-11-19 13:45 +0000 [r157706-157743] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_agi.c: correct small bug introduced during API conversion
+
+ * UPGRADE.txt, UPGRADE-1.6.txt: move relevant entries into
+ UPGRADE.txt and resync UPGRADE-1.6.txt with previous branches
+
+ * include/asterisk/agi.h, res/res_agi.c, UPGRADE.txt,
+ UPGRADE-1.6.txt (added), apps/app_stack.c: make some corrections
+ to the ast_agi_register_multiple(), ast_agi_unregister_multiple()
+ and ast_agi_fdprintf() API calls to be consistent with API
+ guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the
+ new UPGRADE.txt contain information about upgrading between
+ Asterisk 1.6 releases
+
+2008-11-19 05:37 +0000 [r157675] Terry Wilson <twilson@digium.com>
+
+ * configs/cdr_adaptive_odbc.conf.sample: Comment out config line
+ that is in a commented out context
+
+2008-11-19 01:02 +0000 [r157639] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/logger.h, main/logger.c, main/utils.c,
+ include/asterisk/strings.h: Starting with a change to ensure that
+ ast_verbose() preserves ABI compatibility in 1.6.1 (as compared
+ to 1.6.0 and versions of 1.4), this change also deprecates the
+ use of Asterisk with FreeBSD 4, given the central use of va_copy
+ in core functions. va_copy() is C99, anyway, and we already
+ require C99 for other purposes, so this isn't really a big change
+ anyway. This change also simplifies some of the core ast_str_*
+ functions.
+
+2008-11-19 00:59 +0000 [r157632] Mark Michelson <mmichelson@digium.com>
+
+ * main/astmm.c: If malloc returns NULL, we need to return NULL
+ immediately or else Asterisk will crash when attempting to
+ dereference the NULL pointer (closes issue #13858) Reported by:
+ eliel Patches: astmm.c.patch uploaded by eliel (license 64)
+
+2008-11-19 00:27 +0000 [r157600] Sean Bright <sean.bright@gmail.com>
+
+ * Makefile, build_tools/make_version, configure, configure.ac,
+ build_tools/make_buildopts_h, makeopts.in: Fix a few build
+ problems on Solaris (and check for an md5 utility in configure
+ instead of the icky loop I was doing before). (closes issue
+ #13842) Reported by: snuffy Patches: bug13842_20081106.diff
+ uploaded by snuffy (license 35) 13842.diff uploaded by seanbright
+ (license 71) Tested by: snuffy
+
+2008-11-18 23:59 +0000 [r157496-157592] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c: This change prevents a crash from
+ occurring if res_musiconhold.so is unloaded and then Asterisk is
+ stopped. The problem was that we are not unregistering the
+ ast_moh_destroy function at exit. (closes issue #13761) Reported
+ by: eliel Patches: res_musiconhold.c.patch uploaded by eliel
+ (license 64)
+
+ * Makefile: Add some missing $(DESTDIR)s to the bininstall target
+ of the Makefile. (closes issue #13875) Reported by: pabelanger
+ Patches: Makefile.155928 uploaded by pabelanger (license 224)
+
+ * apps/app_voicemail.c: Fix the logic for when delete=yes when IMAP
+ storage is in use so that the message is deleted from both local
+ and IMAP storage. (closes issue #13642) Reported by: jaroth
+ Patches: deleteyes.patch uploaded by jaroth (license 50)
+
+ * channels/chan_sip.c: Merged revisions 157503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov
+ 2008) | 13 lines Add some missing invite state changes necessary
+ in the sip_write function. Not setting the invite state correctly
+ on the call was resulting in the Record application leaving empty
+ files. I also have updated the doxygen comment next to the
+ declaration of the INV_EARLY_MEDIA constant to reflect that we
+ also use this state when we *send* a 18X response to an INVITE.
+ (closes issue #13878) Reported by: nahuelgreco Patches:
+ sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
+ (license 162) Tested by: putnopvut ........
+
+ * channels/chan_sip.c: Based on Russell's advice on the
+ asterisk-dev list, I have changed from using a global lock in
+ update_call_counter to using the locks within the sip_pvt and
+ sip_peer structures instead.
+
+2008-11-18 21:15 +0000 [r157460-157463] Jason Parker <jparker@digium.com>
+
+ * Makefile: Remove echo line that is unnecessary (Thanks
+ seanbright).
+
+ * contrib/init.d/rc.archlinux.asterisk: Make this executable
+
+ * Makefile, contrib/init.d/rc.archlinux.asterisk (added): Add init
+ script for ArchLinux (closes issue #13667) Reported by: sherif
+ Patches: archlinux_rc_makefile.patch uploaded by sherif (license
+ 591) archlinux_rc_makefile-2.patch uploaded by mvanbaak (license
+ 7)
+
+2008-11-18 20:23 +0000 [r157427] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: * Add a lock to be used in the
+ update_call_counter function. * Revert logic to mirror 1.4's in
+ the sense that it will not allow the call counter to dip below 0.
+ These two measures prevent potential races that could cause a SIP
+ peer to appear to be busy forever. (closes issue #13668) Reported
+ by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by
+ wolfelectronic (license 586)
+
+2008-11-18 19:16 +0000 [r157366] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 157365 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008)
+ | 6 lines (closes issue #13899) Reported by: akkornel This fix is
+ the result of a bug fix in ast_app_separate_args r124395. If an
+ argument does not exist it should always be set to a null string
+ rather than a null pointer. ........
+
+2008-11-18 18:31 +0000 [r157306] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, channels/chan_local.c, /, main/features.c,
+ include/asterisk/channel.h, apps/app_followme.c: Merged revisions
+ 157305 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov
+ 2008) | 12 lines Fix a crash in the end_bridge_callback of
+ app_dial and app_followme which would occur at the end of an
+ attended transfer. The error occurred because we initially stored
+ a pointer to an ast_channel which then was hung up due to a
+ masquerade. This commit adds a "fixup" callback to the
+ bridge_config structure to allow for end_bridge_callback_data to
+ be changed in the case that a new channel pointer is needed for
+ the end_bridge_callback. ........
+
+2008-11-18 18:07 +0000 [r157302] Steve Murphy <murf@digium.com>
+
+ * main/config.c: (closes issue #13420) Reported by: alex70 Patches:
+ 13420.13539.patch uploaded by murf (license 17) Tested by: murf,
+ awk This fixes two problems: a spurious linefeed insertion
+ probably left over from pre-precomment times. Only generated when
+ category had no previous comments. The other problem: Insertions
+ could get the line-numbering out of whack and generate negative
+ line numbers, causing chunks of line numbers to be emitted, on
+ the scale of the number of lines up to that point in the file. In
+ such cases, abort the looping, and all is well.
+
+2008-11-17 22:25 +0000 [r157253] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c: Can't use items duplicated off the stack frame
+ in an element returned from a function: in these cases, we have
+ to use the heap, or garbage will result. (closes issue #13898)
+ Reported by: alecdavis Patches: 20081114__bug13898__2.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: alecdavis
+
+2008-11-15 19:51 +0000 [r157105-157167] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile.rules: ensure that if a .i file (preprocessed source) is
+ present, the .o file is made from it, not from the .c file (this
+ only works because GNU makes respects the order the rules are
+ defined)
+
+ * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged
+ revisions 157162-157163 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov
+ 2008) | 1 line dist-clean should remove dependency information
+ files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03
+ +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory
+ dist-clean is run, run clean in that directory first, and when
+ running top-level dist-clean, do not run subdirectory clean
+ operations twice ........
+
+ * /, contrib/asterisk-ng-doxygen: Merged revisions 157104 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157104 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov
+ 2008) | 13 lines major update to doxygen configuration file: 1)
+ update to doxygen 1.5.x style file, as used in trunk 2) tell
+ doxygen where are header files are, so include-file processing
+ can be done 3) make all macros that are used to define
+ variables/functions be expanded, so that doxygen will properly
+ document the resulting variable/function 4) make all macros that
+ are used to provide the contents of a variable (structure) be
+ expanded, so that doxygen will be able to document the resulting
+ fields 5) suppress compiler attributes (__attribute__(xxx)) from
+ being seen by doxygen, so it will properly match up function
+ definition and usage (for an example of th effect of this, look
+ at the doxygen docs for ast_log() from before and afte this
+ commit) ........
+
+2008-11-15 15:37 +0000 [r157073] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/xmldoc.c: Avoid a not needed cast, making code more
+ readable.
+
+2008-11-15 04:25 +0000 [r157039-157041] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c, main/features.c, main/taskprocessor.c: Fix a
+ few more places where the case insensitive hash should be used
+ since the comparison is case insensitive.
+
+ * channels/chan_console.c: Use the new case insensitive hash
+ function for console interfaces. The comparison function is case
+ insensitive.
+
+2008-11-14 22:36 +0000 [r157006] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
+ Allow setting static values in CDRs
+
+2008-11-14 21:19 +0000 [r156962] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Revision 155513 of chan_sip.c in trunk
+ inadvertently removed a very important line to set the "len"
+ field for incoming SIP requests. The result was that all incoming
+ SIP messages appeared to be 0-length, meaning Asterisk could do
+ no meaningful processing of anything SIP-related
+
+2008-11-14 17:35 +0000 [r156916-156918] Terry Wilson <twilson@digium.com>
+
+ * res/res_phoneprov.c: Cleanup whitespace issues
+
+ * res/res_phoneprov.c: Use Mark's new ast_str_case_hash function
+ instead of jumping through hoops to do insensitive case lookups
+
+2008-11-14 17:02 +0000 [r156911] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: Ping is missing the standard double-newline after
+ the event. (closes issue #13903) Reported by: kebl0155
+
+2008-11-14 16:53 +0000 [r156883] Mark Michelson <mmichelson@digium.com>
+
+ * UPGRADE.txt, include/asterisk/strings.h, apps/app_queue.c: Fix
+ some refcounting in app_queue.c and change the hashing used by
+ app_queue.c to be case-insensitive. This is accomplished by
+ adding a new case-insensitive hashing function. This was
+ necessary to prevent bad refcount errors (and potential crashes)
+ which would occur due to the fact that queues were initially read
+ from the config file in a case-sensitive manner. Then, when a
+ user issued a CLI command or manager action, we allowed for
+ case-insensitive input and used that input to directly try to
+ find the queue in the hash table. The result was either that we
+ could not find a queue that was input or worse, we would end up
+ hashing to a completely bogus value based on the input. This
+ commit resolves the problem presented in issue #13703. However,
+ that issue was reported against 1.6.0. Since this fix introduces
+ a behavior change, I am electing to not place this same fix in to
+ the 1.6.0 or 1.6.1 branches, and instead will opt for a change
+ which does not change behavior.
+
+2008-11-14 16:34 +0000 [r156874] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: Remove some useless locking and make sure
+ we hangup channels on a link when we get a GRS.
+
+2008-11-14 15:20 +0000 [r156817] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 156816 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri,
+ 14 Nov 2008) | 10 lines If the prompt to reenter a voicemail
+ password timed out, it resulted in the password not being saved,
+ even if the input matched what you gave when first prompted to
+ enter a new password. This is because the return value of
+ ast_readstring was checked, but not checked properly. This bug
+ was discovered by Jared Smith during an Asterisk training course.
+ Thanks for reporting it! ........
+
+2008-11-14 00:43 +0000 [r156690-156756] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_while.c: Merged revisions 156755 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008)
+ | 6 lines ast_waitfordigit() requires that the channel be up, for
+ no good logical reason. This prevents While/EndWhile from working
+ within the "h" extension. Reported by: jgalarneau (for ABE C.2)
+ Fixed by: me ........
+
+ * main/manager.c, /: Merged revisions 156688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008)
+ | 7 lines Provide more space for all the data which can appear in
+ an originating channel name. (closes issue #13398) Reported by:
+ bamby Patches: manager.c.diff uploaded by bamby (license 430)
+ ........
+
+2008-11-13 19:17 +0000 [r156649] Jeff Peeler <jpeeler@digium.com>
+
+ * main/pbx.c: (closes issue #13891) Reported by: smurfix This
+ reverts a change I made in 116297. At the time it seemed the
+ change was required to solve an issue with attempting a transfer
+ but then letting it timeout without dialing any digits. However,
+ I didn't realize that having an empty extension was possible. I'm
+ removing the immediate return that was added in
+ pbx_find_extension if the extension is null.
+
+2008-11-13 19:10 +0000 [r156647] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Command offsets were not changed correctly
+ when the command syntax for 'pri set debug' was changed from 'pri
+ debug'.
+
+2008-11-13 17:07 +0000 [r156612] Mark Michelson <mmichelson@digium.com>
+
+ * configure, autoconf/ast_c_compile_check.m4: Kevin sent a note
+ indicating that this change is not necessary, so I am reverting
+ it
+
+2008-11-13 15:46 +0000 [r156535-156575] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_meetme.c, doc/appdocsxml.dtd, main/xmldoc.c: Introduce
+ XML documentation for: - MeetMe() - MeetMeCount() -
+ MeetMeChannelAdmin() - MeetMeAdmin() - SLAStation() - SLATrunk()
+ - Add an attribute to optionlist 'hasparams' with the same
+ functionality as the one we have in <parameter> and <argument>
+ (the DTD was updated) - Fix a leak when getting an attribute
+ while parsing an <optionlist>.
+
+ * main/xmldoc.c: Fix a typo introduced when changing
+ xmldoc_has_arguments() to xmldoc_has_inside() we need to pass the
+ name of the node that we are looking for.
+
+ * include/asterisk/xml.h, include/asterisk/xmldoc.h, main/xmldoc.c:
+ Remove trailing whitespaces using ':%s/\s\+$//' pointed by
+ seanbright on #asterisk-dev
+
+2008-11-12 23:13 +0000 [r156443] Sean Bright <sean.bright@gmail.com>
+
+ * /: Use the reviewboard:url SVN property so post-review will work
+ without modification.
+
+2008-11-12 21:34 +0000 [r156388] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 156386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008)
+ | 5 lines When using call limits under 1 second, infinite call
+ lengths are allowed, instead. (closes issue #13851) Reported by:
+ ruddy ........
+
+2008-11-12 20:27 +0000 [r156355] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_clialiases.c: - Make alias->real_cmd point to the
+ allocated space outside alias->alias. - Register the aliased cli
+ command (or we will not alias anything). - Use ARRAY_LEN() when
+ possible.
+
+2008-11-12 19:47 +0000 [r156299] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /: Merged revisions 156297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) |
+ 18 lines It turns out that the 0x0XX00 codes being returned for
+ N, X, and Z are off by one, as per conversation with jsmith on
+ #asterisk-dev; he was teaching a class and disconcerted that this
+ published rule was not being followed, with patterns _NXX,
+ _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
+ have been. This change, tested on these 3 patterns now picks the
+ proper one. However, this change may surprise users who set up
+ dialplans based on previous behavior, which has been there for
+ what, 2 and half years or so now. ........
+
+2008-11-12 19:38 +0000 [r156298] Russell Bryant <russell@digium.com>
+
+ * res/res_clialiases.c: Fix a bug caused by using sizeof(pointer)
+ instead of sizeof(the struct)
+
+2008-11-12 19:28 +0000 [r156295] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 156294 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008)
+ | 6 lines If the SLA thread is not started, then reload causes a
+ memory leak. (closes issue #13889) Reported by: eliel Patches:
+ app_meetme.c.patch uploaded by eliel (license 64) ........
+
+2008-11-12 19:11 +0000 [r156290] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 156289 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008)
+ | 3 lines For whatever reason, gcc only warned me about the
+ possible use of an uninitialized variable when compiling 1.6.1.
+ ........
+
+2008-11-12 18:55 +0000 [r156243] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 156229 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12
+ Nov 2008) | 11 lines Revert revision 132506, since it
+ occasionally caused IAX2 HANGUP packets not to be sent, and
+ instead, schedule a task to destroy the iax2 pvt structure 10
+ seconds later. This allows the IAX2 HANGUP packet to be queued,
+ transmitted, and ACKed before the pvt is destroyed. (closes issue
+ #13645) Reported by: dzajro Patches:
+ 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/
+ ........
+
+2008-11-12 18:32 +0000 [r156228] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 156178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008)
+ | 8 lines (closes issue #13173) Reported by: pep This change adds
+ an announce_thread responsible for playing announcements to an
+ existing conference. This allows all announcing to be immediately
+ stopped if necessary but more importantly allows other threads
+ that need to play something to not block. There are multiple
+ benefits to this, but the actual bug is for solving the scenario
+ for a channel to be unusable after hang up for the entire
+ duration of the parting announcement. The parting announcement
+ can be extremely long depending on what the user recorded upon
+ joining the conference. Reviewed by Russell on Review Board:
+ http://reviewboard.digium.com/r/25/ ........
+
+2008-11-12 17:41 +0000 [r156169] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 156167 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov
+ 2008) | 7 lines When doing some tests, I was having a crash at
+ the end of every call if an attended transfer occurred during the
+ call. I traced the cause to the CDR on one of the channels being
+ NULL. murf suggested a check in the end bridge callback to be
+ sure the CDR is non-NULL before proceeding, so that's what I'm
+ adding. ........
+
+2008-11-12 17:38 +0000 [r156166] Russell Bryant <russell@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 156164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008)
+ | 7 lines Move the sanity check that makes sure "always fork" is
+ not set along with the console option to be after the code that
+ reads options from asterisk.conf. This resolves a situation where
+ Asterisk can start taking up 100% when misconfigured. (Thanks to
+ Bryce Porter (x86 on IRC) for letting me log in to his system to
+ figure out what was causing the 100% CPU problem.) ........
+
+2008-11-12 17:28 +0000 [r156162] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/xmldoc.c: - The paramname is a pointer allocated with
+ strdup() or malloc(), so, we need to free it with ast_free().
+
+2008-11-12 15:33 +0000 [r156127] Mark Michelson <mmichelson@digium.com>
+
+ * configure, autoconf/ast_c_compile_check.m4: Add a couple of
+ AC_SUBST calls to the AST_C_COMPILE_CHECK macro. These missing
+ calls were discovered when working on timerfd support in a
+ separate branch.
+
+2008-11-12 13:43 +0000 [r156125] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_agi.c: Add XML documentation for AGI commands: - database
+ deltree - database get - exec - get data - get full variable
+
+2008-11-12 06:46 +0000 [r156120] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/udptl.c, main/pbx.c, channels/chan_sip.c,
+ configs/cli_aliases.conf.sample (added), include/asterisk/cli.h,
+ CHANGES, res/res_jabber.c, main/rtp.c, main/cli.c, main/cdr.c,
+ channels/chan_skinny.c, res/res_agi.c, pbx/pbx_ael.c,
+ pbx/pbx_dundi.c, funcs/func_devstate.c, main/asterisk.c,
+ channels/chan_mgcp.c, res/res_clialiases.c (added): This commit
+ does two things: - Add CLI aliases module to asterisk. - Remove
+ all deprecated CLI commands from the code Initial work done by
+ file. Junk-Y and lmadsen did a lot of work and testing to get the
+ list of deprecated commands into the configuration file.
+ Deprecated CLI commands are now handled by this new module, see
+ cli_aliases.conf for more info about that. ok russellb@ via
+ reviewboard (closes issue #13735) Reported by: mvanbaak
+
+2008-11-12 02:20 +0000 [r156051-156087] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_agi.c, doc/appdocsxml.dtd: - Add 'database del',
+ 'database put' and 'set music' AGI commands XML documentation. -
+ Add to the DTD the possibility to put a parameter inside an
+ <enum>.
+
+ * include/asterisk/agi.h, res/res_agi.c, doc/appdocsxml.dtd,
+ main/xmldoc.c: Implement AGI XML documentation parsing functions.
+ A new <agi> element is used to describe the XML documentation. We
+ have the usual synopsis,syntax,description and seealso for AGI
+ commands. The CLI 'agi show commands' command was changed to show
+ all the documentation se ctions.
+
+2008-11-11 23:32 +0000 [r156017-156018] Pari Nannapaneni <paripurnachand@digium.com>
+
+ * main/manager.c: changing comment style to conform coding
+ guidelines
+
+ * main/manager.c: Patch by Ryan Brindley -- Make sure that manager
+ refuses any duplicate 'new category' requests in updateconfig
+
+2008-11-11 17:57 +0000 [r155967] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/strings.h: use some fancy compiler magic (thanks
+ to Matthew Woehlke on the gcc-help mailing list) to restore
+ type-safety to S_OR by going back to a macro, but preserve the
+ side-effect-safe usage of the macro arguments
+
+2008-11-11 16:46 +0000 [r155934] Doug Bailey <dbailey@digium.com>
+
+ * res/res_phoneprov.c, phoneprov/polycom_line.xml: Add LINEKEYS
+ variable to allow for a user to set the number of keys assigned
+ to a line on a polycom phone
+
+2008-11-11 16:07 +0000 [r155929] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Remove commentary from the issues list for
+ SIP TCP/TLS
+
+2008-11-10 21:14 +0000 [r155863] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 155861 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon,
+ 10 Nov 2008) | 14 lines Channel drivers assume that when their
+ indicate callback is invoked, that the channel on which the
+ callback was called is locked. This patch corrects an instance in
+ chan_agent where a channel's indicate callback is called directly
+ without first locking the channel. This was leading to some
+ observed locking issues in chan_local, but considering that all
+ channel drivers operate under the same expectations, the generic
+ fix in chan_agent is the right way to go. AST-126 ........
+
+2008-11-10 21:12 +0000 [r155763-155862] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_realtime.c: Make documentation of update method match
+ documentation and update update2 method to match. Reported by:
+ atis, via -dev mailing list. Fixed by: me
+
+ * /, doc/valgrind.txt: Merged revisions 155803 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008)
+ | 1 line I got tired of saying this in every single bugnote
+ referring to this file. ........
+
+ * main/editline/readline.c: Fix memory leak when MALLOC_DEBUG is
+ enabled. (closes issue #13864) Reported by: eliel Patches:
+ readline.c.patch uploaded by eliel (license 64)
+
+2008-11-10 13:53 +0000 [r155711] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/pbx.c, main/Makefile, include/asterisk/xmldoc.h (added),
+ include/asterisk/term.h, include/asterisk/_private.h,
+ main/asterisk.c, main/xmldoc.c (added): Move all the XML
+ documentation API from pbx.c to xmldoc.c. Export the XML
+ documentation API: ast_xmldoc_build_synopsis()
+ ast_xmldoc_build_syntax() ast_xmldoc_build_description()
+ ast_xmldoc_build_seealso() ast_xmldoc_build_arguments()
+ ast_xmldoc_printable() ast_xmldoc_load_documentation()
+
+2008-11-09 16:30 +0000 [r155554-155671] Sean Bright <sean.bright@gmail.com>
+
+ * configs/chan_dahdi.conf.sample: Fix this as well. Pointed out by
+ tzafrir.
+
+ * configs/chan_dahdi.conf.sample: Fix some spelling errors, and
+ convert tabs to spaces.
+
+ * main/channel.c, channels/chan_sip.c, apps/app_directed_pickup.c,
+ main/features.c, include/asterisk/channel.h: In order to move
+ away from nested function use, some changes to the recently
+ introduced ast_channel_search_locked need to be made.
+ Specifically, the caller needs to be able to pass arbitrary data
+ which in turn is passed to the callback. This patch addresses all
+ of the nested functions currently in asterisk trunk.
+
+ * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h,
+ apps/app_followme.c, apps/app_queue.c: Merged revisions 155553
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov
+ 2008) | 6 lines Use static functions here instead of nested ones.
+ This requires a small change to the ast_bridge_config struct as
+ well. To understand the reason for this change, see the following
+ post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
+ ........
+
+2008-11-08 21:46 +0000 [r155513-155516] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c, include/asterisk/strings.h: - Check for
+ failure when putting the packet in the ast_str - fix a spelling
+ error in a header file
+
+ * channels/chan_sip.c: Remove some code that is basically a no-op.
+ Code above this already ensures that the buffer is terminated.
+
+2008-11-07 23:41 +0000 [r155467] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Set the invite state to INV_CANCELLED in a
+ place that makes more sense. Where it was set before, it was
+ impossible to actually delay sending a CANCEL if we had not yet
+ received a provisional response to an INVITE. (closes issue
+ #13626) Reported by: atis Patches: 13626.patch uploaded by
+ putnopvut (license 60) Tested by: atis
+
+2008-11-07 22:39 +0000 [r155401] Sean Bright <sean.bright@gmail.com>
+
+ * main/manager.c, channels/chan_sip.c, funcs/func_dialgroup.c,
+ res/res_timing_pthread.c, include/asterisk/astobj2.h,
+ main/features.c, res/res_phoneprov.c, utils/hashtest2.c,
+ channels/chan_console.c, main/taskprocessor.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/astobj2.c, main/config.c: Add ability
+ to pass arbitrary data to the ao2_callback_fn (called from
+ ao2_callback and ao2_find). Currently, passing OBJ_POINTER to
+ either of these mandates that the passed 'arg' is a hashable
+ object, making searching for an ao2 object based on outside
+ criteria difficult. Reviewed by Russell and Mark M. via
+ ReviewBoard: http://reviewboard.digium.com/r/36/
+
+2008-11-07 22:28 +0000 [r155395-155399] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 155398 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008)
+ | 7 lines Clarify error message. (closes issue #13809) Reported
+ by: denke Patches: 20081104__bug13809.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: denke ........
+
+ * funcs/func_odbc.c: Two bugs relating to colnames found by
+ Marquis42 on #asterisk-dev
+
+2008-11-07 21:14 +0000 [r155360] Mark Michelson <mmichelson@digium.com>
+
+ * configs/voicemail.conf.sample: Remove one more instance of the
+ sample configuration lying about what's possible. The tz cannot
+ be set in a context like this. It can only be set in the general
+ section or per-mailbox. Thanks to sasargen on #asterisk-dev for
+ pointing this out
+
+2008-11-07 20:13 +0000 [r155324] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Send call release with unallocated cause
+ instead of normal call clearing, when invalid extension is
+ called. (closes issue #13408) Reported by: adomjan Patches:
+ chan_dahdi.c-ss7-unallocated-2 uploaded by adomjan (license 487)
+
+2008-11-07 16:18 +0000 [r155284] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk/indications.h, res/res_indications.c,
+ main/indications.c: Convert open-coded linked list in indications
+ to the AST_LIST_* macros. This cleans the code up some and should
+ make it more maintainable as time goes on. Reviewed by Russell,
+ Kevin, Mark M., and Tilghman via ReviewBoard:
+ http://reviewboard.digium.com/r/34/
+
+2008-11-07 15:52 +0000 [r155282] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: stringfields conversion for struct sip_peer,
+ as requested :-)
+
+2008-11-07 15:42 +0000 [r155241-155264] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Remove a bogus ast_free() that Kevin
+ noticed. This was probably just left over from pre-astobj2ified
+ chan_sip.
+
+ * include/asterisk/astobj2.h: Clarify which part of OBJ_MULTIPLE is
+ not implemented, and under what case it is perfectly fine to use.
+ (Inspired by a question I received about my last commit.)
+
+ * main/pbx.c, channels/chan_sip.c: Fix some code in chan_sip that
+ was intended to unlink multiple objects from a container. The
+ OBJ_MULTIPLE flag must be provided here. Otherwise, this would
+ only remove a single object.
+
+2008-11-07 03:09 +0000 [r155206] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx/pbx_config.c: correct logic error noticed by rmudgett
+ (thanks!)
+
+2008-11-07 03:02 +0000 [r155175-155204] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/pbx.c: If 'asterisk.conf' is not found, instead of giving
+ up, load documentation for the 'en_US' language (fix my last
+ commit).
+
+ * main/pbx.c: Fix an asterisk crash if no asterisk.conf
+ configuration file is present.
+
+2008-11-06 22:49 +0000 [r155066-155121] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: don't
+ blindly assume that Darwin and Cygwin need GLOB_ABORTED defined;
+ only define it if it is not already defined
+
+ * pbx/pbx_config.c: coding style/guidelines cleanup, plus use new
+ side-effect safe S_OR
+
+ * include/asterisk/strings.h: make S_OR and S_COR safe to use even
+ if the parameters are function calls or have side effects. it
+ still bothers me that these are called S_OR and not something
+ like ast_string_or, but that's water over the bridge
+
+ * channels/chan_dahdi.c: put ifdef protection around the rest of
+ the libpri function calls that were added at the same time as
+ progress_with_cause move parsing of the qsig channel mapping
+ configuration option outside ifdef HAVE_PRI_INBANDDISCONNECT and
+ into a properly ifdef'd block
+
+2008-11-06 19:46 +0000 [r155012] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 155011 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov
+ 2008) | 8 lines The documentation listed the ability to set
+ 'maxmsg' per context. The truth is that you can only set this in
+ the general section or per mailbox. Thus I am updating the sample
+ config file to be more accurate. Thanks to sasargen on IRC for
+ bringing up this issue. ........
+
+2008-11-06 18:19 +0000 [r154967] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/pbx.c: Simplify the output of [See Also]. Functions are
+ printed without parenthesis like: FUNTION Applications are
+ printed with parenthesis like: AppName() Cli commands are printed
+ like: 'core show application' The other type of references are
+ printed as they are inside the <ref> tag.
+
+2008-11-05 22:22 +0000 [r154923-154926] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_directed_pickup.c: Fix some whitespace.
+
+ * apps/app_directed_pickup.c, main/features.c: Update a couple
+ places to use the new ast_channel_search_locked API call.
+
+2008-11-05 22:19 +0000 [r154922] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Don't read history on -rx commands. (Closes
+ issue #13571) Reported by: tzafrir Patch
+ '0001-no-need-for-history-on-asterisk-rx.patch' uploaded by
+ tzafrir.
+
+2008-11-05 22:01 +0000 [r154919] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk.h: Fix a problem found while building res_snmp.
+
+2008-11-05 21:58 +0000 [r154915] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, funcs/func_strings.c, main/app.c,
+ CHANGES: Add LISTFILTER dialplan function, along with supporting
+ documentation. See documentation for more information on how to
+ use it.
+
+2008-11-05 20:45 +0000 [r154875] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Make compilation
+ of chan_dahdi so that it does not require the new
+ pri_progress_with_cause function to have libpri support work.
+
+2008-11-05 20:33 +0000 [r154839] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/res_http_post.c: make this compile on OpenBSD again.
+
+2008-11-05 20:17 +0000 [r154796-154837] Eliel C. Sardanons <eliels@gmail.com>
+
+ * channels/chan_agent.c: Add AgentLogin(), AgentMonitorOutgoing()
+ applications and AGENT() function XML documentation.
+
+ * apps/app_test.c: Add TestClient() and TestServer() applications
+ XML documentation.
+
+ * apps/app_mixmonitor.c: Add more [see also] references based on
+ TFOT.
+
+ * apps/app_macro.c: Add Macro(), MacroExit(), MacroExclusive() and
+ MacroIf() applications XML documentation. (closes issue #13699)
+ Reported by: snuffy Patches: bug13699_20081016.diff uploaded by
+ snuffy (license 35)
+
+2008-11-05 16:11 +0000 [r154687] Steve Murphy <murf@digium.com>
+
+ * main/channel.c, /: Merged revisions 154685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1
+ line This fix was prompted by communication from user, who was
+ seeing thousands of error logs... looks like EAGAIN. Made such
+ uninteresting. ........
+
+2008-11-05 14:37 +0000 [r154467-154647] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/pbx.c, apps/app_privacy.c, apps/app_sayunixtime.c,
+ main/features.c, apps/app_morsecode.c, apps/app_alarmreceiver.c,
+ apps/app_amd.c: Add more SeeAlso references based on TFOT.
+
+ * doc/appdocsxml.dtd: We now can have a reference to a filename
+ inside a <see-also> tag.
+
+ * apps/app_parkandannounce.c: - Add ParkAndAnnounce() application
+ XML documentation.
+
+ * main/pbx.c, apps/app_page.c, apps/app_authenticate.c,
+ apps/app_dumpchan.c, apps/app_disa.c, apps/app_image.c,
+ apps/app_chanspy.c, apps/app_stack.c, apps/app_adsiprog.c: - Add
+ more <see-also> based on TFOT. - Add the 'filename' type to the
+ see-also ref. To be able to reference a filename.
+
+ * apps/app_readfile.c, funcs/func_db.c, apps/app_sendtext.c,
+ funcs/func_blacklist.c, apps/app_url.c, apps/app_queue.c,
+ apps/app_senddtmf.c, apps/app_db.c: - Add some see-also
+ references based on TFOT.
+
+ * apps/app_read.c: - Add Read() application XML documentation.
+
+ * apps/app_followme.c: - Add FollowMe() application XML
+ documentation.
+
+ * apps/app_forkcdr.c, res/res_indications.c: - Add PlayTones() and
+ StopPlayTones() applications XML documentation. - Fix a dot that
+ was outside of the <para> in the ForkCDR() XML documentation.
+
+2008-11-04 23:23 +0000 [r154429] Sean Bright <sean.bright@gmail.com>
+
+ * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h:
+ Introduce a new API call ast_channel_search_locked, which
+ iterates through the channel list calling a caller-defined
+ callback. The callback returns non-zero if a match is found. This
+ should speed up some of the code that I committed earlier today
+ in chan_sip (which is also updated by this commit). Reviewed by
+ russellb and kpfleming via ReviewBoard:
+ http://reviewboard.digium.com/r/28/
+
+2008-11-04 23:03 +0000 [r154366-154428] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Switch to using a thread condition to
+ signal that a child thread is ready for work, rather than a busy
+ wait. (closes issue #13011) Reported by: jpgrayson Patches:
+ chan_iax2_find_idle.patch uploaded by jpgrayson (license 492)
+
+ * /, channels/chan_iax2.c: Merged revisions 154365 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04
+ Nov 2008) | 9 lines On busy systems, it's possible for the values
+ checked within a single line of code to change, unless the
+ structure is locked to ensure a consistent state. (closes issue
+ #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: kowalma ........
+
+2008-11-04 20:12 +0000 [r154329] Eliel C. Sardanons <eliels@gmail.com>
+
+ * Makefile: We need to pass the DTD to xmlstarlet to validate
+ against it the XML. (I thought it was being read within the
+ DOCTYPE definition inside the XML).
+
+2008-11-04 19:07 +0000 [r154268] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 154266 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04
+ Nov 2008) | 4 lines JIRA ABE-1703 mISDN sets the channel to the
+ wrong state when it receives the indication AST_CONTROL_RINGING.
+ ........
+
+2008-11-04 18:59 +0000 [r154260-154264] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_skinny.c, channels/chan_h323.c: Recorded merge
+ of revisions 154263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008)
+ | 3 lines Make the monitor thread non-detached, so it can be
+ joined (suggested by Russell on -dev list). ........
+
+ * include/asterisk/devicestate.h, main/manager.c, apps/app_page.c,
+ include/asterisk/config.h, main/features.c, main/devicestate.c,
+ apps/app_queue.c, main/config.c, apps/app_voicemail.c: Slightly
+ optimize ast_devstate_str and rename global functions
+ devstate2str and config_text_file_save to have an ast_ prefix
+
+2008-11-04 18:06 +0000 [r154225] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_forkcdr.c: Add XML documentation for the ForkCDR()
+ application.
+
+2008-11-04 17:23 +0000 [r154186-154191] Sean Bright <sean.bright@gmail.com>
+
+ * main/pbx.c: GLOB_BRACE is already added to MY_GLOB_FLAGS if it is
+ supported on the platform. This should resolve some build errors
+ on Solaris. (issue #13704) Reported by: dougm
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Allow devices that
+ accept dialog-info+xml (like snoms) to get the Caller ID of the
+ calling party when subscribed to the state of an extension that
+ is ringing. This has some limitations which are documented in
+ sip.conf.sample. (closes issue #13827) Reported by: seanbright
+ Patches: issue13827.patch uploaded by seanbright (license 71)
+ Reviewed by: russellb
+
+ * main/Makefile: Fix build errors.
+
+2008-11-04 15:07 +0000 [r154151] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_vpb.cc, res/res_crypto.c, configure.ac,
+ cdr/cdr_adaptive_odbc.c, channels/chan_oss.c,
+ channels/chan_usbradio.c, res/res_config_odbc.c,
+ apps/app_osplookup.c, funcs/func_odbc.c, configure,
+ build_tools/menuselect-deps.in, channels/chan_alsa.c,
+ makeopts.in, cdr/cdr_odbc.c, res/res_odbc.c,
+ apps/app_voicemail.c: improve configure script to remember the
+ previous value of each dependency in build_tools/menuselect-deps,
+ so that (once it has been written) menuselect can use this
+ information to warn the user when a previously met dependency is
+ no longer met along the way, change tags used in configure
+ script, menuselect-deps and code for various dependencies to be
+ consistently named
+
+2008-11-04 14:38 +0000 [r154149] Eliel C. Sardanons <eliels@gmail.com>
+
+ * channels/chan_dahdi.c: Add XML documentation for: Applications -
+ DAHDISendKeypadFacility() - DAHDISendCallreroutingFacility()
+
+2008-11-03 22:28 +0000 [r154023-154072] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 154066 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03
+ Nov 2008) | 5 lines Attempting to expunge a mailbox when the
+ mailstream is NULL will crash Asterisk. (Closes issue #13829)
+ Reported by: jaroth Patch by: me (modified jaroth's patch)
+ ........
+
+ * /, main/rtp.c: Merged revisions 154060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008)
+ | 3 lines Remove the potential for a division by zero error.
+ (Closes issue #13810) ........
+
+ * funcs/func_odbc.c: Should have passed the string pointer, not the
+ ast_str structure. (closes issue #13830) Reported by: Marquis
+
+2008-11-03 18:02 +0000 [r153983] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Updating docs
+
+2008-11-03 17:11 +0000 [r153947] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_stack.c: Add LOCAL() function XML documentation.
+
+2008-11-03 15:25 +0000 [r153904-153905] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Spaces to replace tabs...
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding a
+ separation of remote authentication and our authentication.
+ remotesecret => our password for a remote service secret => our
+ authentication when someone calls us Secret => still has both
+ functions if remotesecret is not used.
+
+2008-11-03 13:33 +0000 [r153803-153852] Eliel C. Sardanons <eliels@gmail.com>
+
+ * channels/chan_iax2.c: Add XML documentation for: Functions -
+ IAXPEER() - IAXVAR()
+
+ * channels/chan_sip.c: Add XML documentation for: Applications -
+ SIPDtmfMode() - SIPAddHeader() Functions - SIP_HEADER() -
+ SIPPEER() - SIPCHANINFO() - CHECKSIPDOMAIN()
+
+2008-11-03 12:26 +0000 [r153787] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, autoconf/ast_ext_lib.m4: when --without-<foo> is
+ passed to the configure script, explicitly inform menuselect that
+ the package was disabled by the user
+
+2008-11-03 01:01 +0000 [r153747] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_waitforring.c, apps/app_waitforsilence.c, apps/app_db.c,
+ apps/app_ivrdemo.c: Add XML documentation for: - WaitForSilence()
+ - WaitForNoise() - WaitForRing() - IVRDemo() - DBDel() -
+ DBDeltree() (issue #13699) Reported by: snuffy Patches:
+ bug13699_20081016.diff uploaded by snuffy (license 35) (With
+ minor changes)
+
+2008-11-02 23:34 +0000 [r153709] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/agi.h, configure,
+ include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4,
+ configure.ac, include/asterisk/compiler.h, apps/app_stack.c:
+ instead of trying to forcibly load res_agi when app_stack is
+ loaded (even if the administrator didn't want it loaded), use GCC
+ weak symbols to determine whether it was loaded already or not;
+ if it was loaded, then use it.
+
+2008-11-02 20:06 +0000 [r153652] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/features.h: Merged revisions 153651 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008)
+ | 2 lines features.h depends on linkedlists.h, so include it
+ ........
+
+2008-11-02 19:39 +0000 [r153616-153650] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: fix one more warning missed because i did
+ not have new enough libpri installed
+
+ * res/res_musiconhold.c: fix small bug introduced while cleaning up
+ compiler warnings
+
+ * /: mark this revision as merged manually
+
+ * utils/muted.c, apps/app_authenticate.c, res/res_phoneprov.c,
+ main/utils.c, formats/format_wav_gsm.c, res/res_http_post.c,
+ res/res_musiconhold.c, channels/chan_iax2.c, res/res_jabber.c,
+ res/res_config_sqlite.c, utils/frame.c, utils/stereorize.c,
+ main/channel.c, channels/chan_dahdi.c, main/manager.c,
+ res/ael/ael.tab.c, funcs/func_odbc.c, main/ast_expr2f.c,
+ res/res_agi.c, main/http.c, main/logger.c, formats/format_gsm.c,
+ apps/app_adsiprog.c, apps/app_dial.c, channels/chan_sip.c,
+ apps/app_festival.c, formats/format_wav.c, res/ael/ael.y,
+ main/db1-ast/hash/hash_page.c, agi/eagi-test.c, res/res_crypto.c,
+ utils/astman.c, pbx/pbx_lua.c, formats/format_ogg_vorbis.c,
+ utils/astcanary.c, apps/app_queue.c, channels/chan_oss.c,
+ agi/eagi-sphinx-test.c, res/ael/ael_lex.c, channels/chan_h323.c,
+ main/file.c, apps/app_sms.c, pbx/pbx_dundi.c, res/ael/ael.flex,
+ pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c,
+ utils/streamplayer.c, main/asterisk.c, apps/app_voicemail.c:
+ bring over all the fixes for the warnings found by gcc 4.3.x from
+ the 1.4 branch, and add the ones needed for all the new code here
+ too
+
+2008-11-02 06:24 +0000 [r153582] Eliel C. Sardanons <eliels@gmail.com>
+
+ * channels/chan_iax2.c: Add IAX2Provision() application XML
+ documentation.
+
+2008-11-02 05:56 +0000 [r153577-153580] Russell Bryant <russell@digium.com>
+
+ * Makefile: validate-docs is a PHONY target
+
+ * Makefile, configure, configure.ac, makeopts.in: Add a handy
+ makefile target so that you can validate the documentation
+ against the DTD by running "make validate-docs"
+
+ * Makefile: Modify the Makefile logic for extracting documentation.
+ - Build the documentation when you run "make", as opposed to
+ "make install" - Only rebuild the documentation when source code
+ has been changed
+
+2008-11-02 05:10 +0000 [r153541-153543] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_flash.c: Add Flash() application XML documentation.
+
+ * apps/app_talkdetect.c: Fix a typo in the name of the application.
+
+2008-11-02 04:14 +0000 [r153472-153507] Sean Bright <sean.bright@gmail.com>
+
+ * channels/Makefile: There is a troublesome assert() in the
+ alsa/control.h header that causes GCC 4.3.2 to complain that the
+ passed argument will always evaluate to true. So to get things to
+ compile, disable assert when building chan_usbradio.so.
+
+ * apps/app_record.c: Another little one.
+
+2008-11-02 02:55 +0000 [r153362-153470] Russell Bryant <russell@digium.com>
+
+ * apps/app_page.c: fix a typo (thanks sean)
+
+ * apps/app_dial.c, funcs/func_speex.c, apps/app_page.c,
+ apps/app_record.c, funcs/func_env.c, apps/app_dahdiras.c,
+ funcs/func_math.c, funcs/func_strings.c, apps/app_userevent.c,
+ apps/app_exec.c, apps/app_chanspy.c, apps/app_playback.c: Fix
+ various spelling and grammatical issues in documentation
+
+ * apps/app_voicemail.c: - Use a for loop instead of a while loop -
+ Get rid of an unnecessary variable
+
+ * apps/app_directed_pickup.c: Instead of doing a couple of strlen()
+ calls each iteration of the loop, only do it once at the
+ beginning of the function
+
+ * channels/chan_sip.c: Don't ignore the result of find_peer() when
+ looking for a peer by IP in check_peer_ok().
+
+ * funcs/func_speex.c, apps/app_dahdibarge.c, funcs/func_rand.c,
+ apps/app_readfile.c, funcs/func_module.c, funcs/func_dialgroup.c,
+ include/asterisk/autoconfig.h.in, funcs/func_env.c,
+ apps/app_dahdiscan.c, apps/app_record.c, funcs/func_strings.c,
+ apps/app_sayunixtime.c, include/asterisk/extconf.h,
+ apps/app_alarmreceiver.c, apps/app_image.c,
+ apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
+ main/config.c, main/term.c, include/asterisk/compat.h, configure,
+ funcs/func_shell.c, apps/app_skel.c, apps/app_dumpchan.c,
+ include/asterisk/module.h, main/features.c, apps/app_amd.c,
+ apps/app_url.c, apps/app_milliwatt.c, apps/app_dial.c,
+ main/pbx.c, include/asterisk/xml.h (added), apps/app_page.c,
+ funcs/func_timeout.c, main/Makefile, apps/app_privacy.c,
+ apps/app_echo.c, apps/app_softhangup.c, apps/app_fax.c,
+ funcs/func_math.c, apps/app_dahdiras.c, configure.ac,
+ apps/app_disa.c, apps/app_morsecode.c, funcs/func_cut.c,
+ apps/app_talkdetect.c, apps/app_transfer.c, apps/app_playback.c,
+ doc/tex/asterisk-conf.tex, Makefile, apps/app_sendtext.c,
+ funcs/func_channel.c, funcs/func_cdr.c, apps/app_zapateller.c,
+ build_tools/get_documentation (added), funcs/func_iconv.c,
+ apps/app_mixmonitor.c, apps/app_chanspy.c, main/asterisk.c,
+ apps/app_cdr.c, funcs/func_base64.c, funcs/func_md5.c,
+ apps/app_dictate.c, apps/app_authenticate.c,
+ apps/app_readexten.c, apps/app_userevent.c, funcs/func_vmcount.c,
+ main/xml.c (added), funcs/func_sha1.c, funcs/func_logic.c,
+ funcs/func_uri.c, apps/app_controlplayback.c, funcs/func_enum.c,
+ apps/app_setcallerid.c, funcs/func_groupcount.c,
+ funcs/func_config.c, funcs/func_volume.c, funcs/func_odbc.c,
+ apps/app_mp3.c, apps/app_directory.c, apps/app_jack.c,
+ apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c,
+ funcs/func_dialplan.c, funcs/func_db.c, funcs/func_version.c,
+ apps/app_festival.c, funcs/func_lock.c, apps/app_waituntil.c,
+ doc, include/asterisk/term.h, include/asterisk/_private.h,
+ apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c,
+ funcs/func_global.c, funcs/func_extstate.c,
+ funcs/func_realtime.c, apps/app_channelredirect.c,
+ funcs/func_blacklist.c, apps/app_directed_pickup.c,
+ include/asterisk/pbx.h, include/asterisk/strings.h, makeopts.in,
+ apps/app_senddtmf.c, funcs/func_devstate.c,
+ funcs/func_callerid.c, doc/appdocsxml.dtd (added),
+ apps/app_verbose.c, apps/app_stack.c: Merge changes from
+ team/group/appdocsxml This commit introduces the first phase of
+ an effort to manage documentation of the interfaces in Asterisk
+ in an XML format. Currently, a new format is available for
+ applications and dialplan functions. A good number of conversions
+ to the new format are also included. For more information, see
+ the following message to asterisk-dev:
+ http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
+
+ * channels/chan_sip.c: Ensure that the sip_pvt properly has its
+ refcount incremented when the scheduler holds a reference to it
+ for session timer processing.
+
+2008-11-01 01:55 +0000 [r153296] Sean Bright <sean.bright@gmail.com>
+
+ * configs/sip.conf.sample: The default in chan_sip for
+ notifyringing is yes, so update the sample conf to reflect that.
+
+2008-10-31 20:05 +0000 [r153223] Mark Michelson <mmichelson@digium.com>
+
+ * main/dial.c, apps/app_page.c, include/asterisk/dial.h, CHANGES: *
+ Fixed timeout logic in the dialing API as setting timeouts had no
+ effect * Updated dialing API documentation to indicate that
+ timeouts are specified in milliseconds * Added a new timeout
+ argument to the Page application. If time expires, any endpoints
+ which have not answered will be hung up.
+
+2008-10-31 18:55 +0000 [r153181] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, main/features.c, include/asterisk/channel.h,
+ apps/app_followme.c, apps/app_queue.c: Recent CDR fixes moved
+ execution of the 'h' exten into the bridging code, so variables
+ that were set after ast_bridge_call was called would not show up
+ in the 'h' exten. Added a callback function to handle setting
+ variables, etc. from w/in the bridging code. Calls back into a
+ nested function within the function calling ast_bridge_call
+ (closes issue #13793) Reported by: greenfieldtech
+
+2008-10-31 17:18 +0000 [r153122-153124] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES:
+ Failover for func_odbc, allowing an INSERT query to be performed
+ when the UPDATE query initially affects 0 rows. (closes issue
+ #13083) Reported by: Corydon76 Patches:
+ 20081031__bug13083.diff.txt uploaded by Corydon76 (license 14)
+
+ * /, channels/chan_sip.c: Merged revisions 153114 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008)
+ | 3 lines Turn off qualify on uncached realtime peers. (Closes
+ issue #13383) ........
+
+2008-10-31 09:31 +0000 [r153057] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Use the ast_str API call to reset the string
+ instead of manually editing its internals (closes issue #13816)
+ Reported by: eliel Patches: channel.c.patch uploaded by eliel
+ (license 64)
+
+2008-10-30 20:59 +0000 [r152993] Sean Bright <sean.bright@gmail.com>
+
+ * /, bootstrap.sh: Merged revisions 152992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct
+ 2008) | 2 lines The -I argument to aclocal needs a space before
+ the include directory name. ........
+
+2008-10-30 20:46 +0000 [r152990] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/timing.h: Add a todo for a new timing API
+ implementation that would work for Linux systems as of kernel
+ 2.6.25 and glibc 2.8
+
+2008-10-30 20:35 +0000 [r152923-152969] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_h323.c: Merged revisions 152958 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30
+ Oct 2008) | 3 lines Cannot join detached threads. See
+ http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
+ (Closes issue #13400) ........
+
+ * channels/chan_local.c, /: Merged revisions 152922 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r152922 | tilghman | 2008-10-30 14:43:38 -0500 (Thu, 30
+ Oct 2008) | 6 lines Unlock before returning, when extension
+ doesn't exist. (closes issue #13807) Reported by: eliel Patches:
+ chan_local.c.patch uploaded by eliel (license 64) ........
+
+2008-10-30 19:40 +0000 [r152887-152920] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix the sip_peer reference count with
+ respect to scheduler entries for scheduling peer pokes, and
+ scheduling peer poke expirations.
+
+ * channels/chan_sip.c: Fix the sip_peer reference count with
+ respect to scheduler entries for registration expirations.
+
+ * include/asterisk/sched.h: Fix a bug in AST_SCHED_REPLACE_UNREF().
+ The reference count of the object _must_ be increased before
+ creating the scheduler entry. Otherwise, you create a race
+ condition where the reference count may hit zero and the object
+ can disappear out from under you. This could also would have
+ incorrectly decreased the reference count in the case that the
+ scheduler add failed.
+
+2008-10-30 19:23 +0000 [r152879] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: I just noticed this construct and thought it
+ was silly to have a bunch of case statements with duplicated code
+ in each case. Instead, just use the built-in fallthrough
+ capability of case statements and reduce the code to a single
+ instance
+
+2008-10-30 19:21 +0000 [r152875-152877] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Modify the documentation of the sip_registry
+ struct - Remove a comment that says that the monitor thread is
+ the only one that ever touches these objects. This is no longer
+ the case with TCP. Also, I would eventually like to get the
+ scheduler in its own thread, so this is just a poor assumption to
+ make. - Note that reference counting of these objects with
+ respect to scheduler entries is not complete. There are some
+ leaked references when deleting scheduler entries.
+
+ * funcs/func_db.c: - spaces to tabs - add some braces - remove
+ unnecessary cast
+
+2008-10-30 16:54 +0000 [r152809-152812] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/cdr.c, /: Merged revisions 152811 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct
+ 2008) | 3 lines instead of comparing the string pointer to 0,
+ let's compare the value that was actually parsed out of the
+ string (found by sparse) ........
+
+ * include/asterisk/buildinfo.h (added): try to get this committed
+ before the buildbot complains about a broken tree
+
+ * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
+ main/dial.c, main/dnsmgr.c, main/buildinfo.c,
+ codecs/lpc10/chanwr.c, utils/astcanary.c,
+ channels/misdn/isdn_lib.c, main/asterisk.c, apps/app_adsiprog.c:
+ fix a few small things found by using sparse
+
+2008-10-30 16:38 +0000 [r152807] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c, CHANGES, configs/features.conf.sample: After
+ seeing another problem in #asterisk stemming from the low default
+ value of featuredigittimeout, I decided it was high time to
+ change it. I have changed the default to 2000 ms based on a
+ suggestion from Leif Madsen.
+
+2008-10-30 04:26 +0000 [r152689-152765] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Set up an example stdexten that
+ preserves the original context and extension in the CDR. (Related
+ to issue #13799) Reported by: davidw
+
+ * CHANGES, apps/app_directory.c: Pay attention to the
+ searchcontexts entry in voicemail.conf (related to AST-125)
+
+ * main/pbx.c: Track down and fix annoying lock errors
+
+2008-10-29 20:53 +0000 [r152646] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_directory.c: If there was no named defined in a
+ voicemail.conf mailbox entry, then app_directory would crash when
+ attempting to read that entry from the file. We now check for the
+ NULL or empty string properly so that there will be no crash.
+ (closes issue #13804) Reported by: bluecrow76
+
+2008-10-29 05:47 +0000 [r152605] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, /, apps/app_queue.c,
+ configs/features.conf.sample: Merged revisions 152538 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) |
+ 14 lines A little documentation cross-ref between features and
+ dial and queue... I wasted some time (stupidly) trying to get the
+ one-touch parking stuff working, because it didn't occur to me
+ that I had to also have the corresponding options in the dial
+ command! Duh! (In all this time, I never set this up before!) So,
+ to keep some poor fool from suffering the same fate, I made the
+ features.conf.sample file mention the corresponding opts in
+ dial/queue; and the docs for dial/app specifically mention the
+ corresponding decls in the feature.conf file. I hope this doesn't
+ spoil some vast, eternal plan... ........
+
+2008-10-29 05:34 +0000 [r152569] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 152539 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008)
+ | 7 lines Fix an incorrect usage of sizeof() (closes issue
+ #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch
+ uploaded by andrew53 (license 519) ........
+
+2008-10-29 05:01 +0000 [r152536] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, /, main/features.c, include/asterisk/pbx.h,
+ apps/app_queue.c, include/asterisk/features.h: Merged revisions
+ 152535 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) |
+ 46 lines The magic trick to avoid this crash is not to try to
+ find the channel by name in the list, which is slow and resource
+ consuming, but rather to pay attention to the result codes from
+ the ast_bridge_call, to which I added the
+ AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when
+ a channel is parked. Why? because CDR's aren't generated via
+ parking, so nothing is needed, but if a transfer occurred, there
+ are critical things I need. If you get AST_PBX_KEEPALIVE, then
+ don't touch the channel pointer. If you get
+ AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
+ don't touch the peer pointer. Updated the several places where
+ the results from a bridge were not being properly obeyed, and
+ fixed some code I had introduced so that the results of the
+ bridge were not overridden (in trunk). All the places that
+ previously tested for AST_PBX_NO_HANGUP_PEER now have to check
+ for both AST_PBX_NO_HANGUP_PEER and
+ AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
+ parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
+ while A is getting the parking slot announcement, immediately
+ after being put on hold. 2. A calls B; B answers; A parks B; B
+ hangs up after A has been hung up, but before the park times out.
+ 3. A calls B; B answers; B parks A; A hangs up while B is getting
+ the parking slot announcement, immediately after being put on
+ hold. 4. A calls B; B answers; B parks A; A hangs up after B has
+ been hung up, but before the park times out. No crash. I also ran
+ the scenarios above against valgrind, and accesses looked good.
+ ........
+
+2008-10-28 22:33 +0000 [r152467] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 152463 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28
+ Oct 2008) | 3 lines Quoting in the wrong direction (Fixes
+ AST-107) ........
+
+2008-10-28 22:26 +0000 [r152448] Doug Bailey <dbailey@digium.com>
+
+ * configs/phoneprov.conf.sample: Add more polycom firmware files to
+ static mapping
+
+2008-10-28 21:38 +0000 [r152369-152442] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c: Only re-add the io port if it was closed,
+ otherwise reload causes a memory leak. (closes issue #13785)
+ Reported by: eliel Patches: chan_mgcp.c.patch uploaded by eliel
+ (license 64)
+
+ * apps/app_dial.c, /: Merged revisions 152368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008)
+ | 8 lines Reset all DIAL variables back to blank, in case Dial is
+ called multiple times per call (which could otherwise lead to
+ inconsistent status reports). (closes issue #13216) Reported by:
+ ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76
+ (license 14) Tested by: ruddy ........
+
+2008-10-27 23:31 +0000 [r152287] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 152286 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27
+ Oct 2008) | 2 lines Buffer policy setting for half is not needed.
+ ........
+
+2008-10-27 21:34 +0000 [r152134-152216] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 152215 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27
+ Oct 2008) | 6 lines Inherit ALL elements of CallerID across a
+ local channel. (closes issue #13368) Reported by: Peter Schlaile
+ Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
+ (license 14) ........
+
+ * apps/app_stack.c: Set ARGC in subroutines with the number of
+ arguments passed.
+
+ * apps/app_stack.c: Oops, only delete the ARG variables once upon
+ release. The following section would have removed them again
+ (removing variables from 2 stack frames, instead of just one).
+
+2008-10-27 16:03 +0000 [r152132] Jason Parker <jparker@digium.com>
+
+ * apps/app_transfer.c: Remove options argument parsing/syntax (it
+ isn't used any longer) (closes issue #13789) Reported by: IgorG
+ Patches: app_transfer.c.diff uploaded by IgorG (license 20)
+
+2008-10-26 20:25 +0000 [r152060] Sean Bright <sean.bright@gmail.com>
+
+ * /, funcs/func_strings.c: Merged revisions 152059 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun,
+ 26 Oct 2008) | 7 lines Since passing \0 as the second argument to
+ strchr is valid (and will match the trailing \0 of a string) we
+ need to check that first, otherwise we end up with incorrect
+ results. Fix suggested by reporter. (closes issue #13787)
+ Reported by: meitinger ........
+
+2008-10-26 10:23 +0000 [r151980-152020] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Trying to fix the user/peer matching
+ correctly. This will need some testing before getting merged into
+ 1.6.1
+
+ * channels/chan_sip.c: Moving more variables to the sip_cfg
+ structure, as I have some future ideas for the usage of that
+ structure.
+
+ * channels/chan_sip.c: Doxygen changes and some formatting.
+
+2008-10-25 11:02 +0000 [r151906] Russell Bryant <russell@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 151905 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008)
+ | 8 lines Move AMI initialization to occur after loading modules.
+ This prevents a deadlock when someone tries to initiate a module
+ reload from the AMI just as Asterisk is starting. (closes issue
+ #13778) Reported by: hotsblanc Fix suggested by hotsblanc
+ ........
+
+2008-10-23 21:27 +0000 [r151830] Terry Wilson <twilson@digium.com>
+
+ * funcs/func_odbc.c: allow to compile under --enable-dev-mode (gcc
+ didn't actually complain when I was using ccache...)
+
+2008-10-23 15:54 +0000 [r151762] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/vmdb.sql: Clarify documentation, following merge
+ of realtime_update2 branch
+
+2008-10-23 15:38 +0000 [r151739-151761] Olle Johansson <oej@edvina.net>
+
+ * CHANGES: Thanks russellb for reminding an old man....
+
+ * channels/chan_sip.c, doc/tex/channelvariables.tex: Adding a small
+ new feature. Setting _SIPFROMDOMAIN in a channel will set the
+ domain we use for the URI in the outbound call leg.
+
+2008-10-23 15:28 +0000 [r151732] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Simplify some nested functions, as suggested
+ by Russell on -dev
+
+2008-10-23 15:09 +0000 [r151722] Doug Bailey <dbailey@digium.com>
+
+ * res/res_http_post.c: Add patch to handle how IE7 issues POST
+ requests using Window path spec including backslash delimiters
+
+2008-10-22 22:11 +0000 [r151682] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c, CHANGES: Added debugging CLI functions
+
+2008-10-22 20:45 +0000 [r151642] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_sip.c: revert the changes in issue #13705 - it's
+ being re-opened as while the results fixed the complaint in the
+ issue, it introduced other more undesirable issues than what was
+ already reported
+
+2008-10-22 20:05 +0000 [r151601] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/live_ast (added): Add a contributed script for
+ running Asterisk without installing it, first. (closes issue
+ #11680) Reported by: tzafrir Patches: live_ast_6 uploaded by
+ tzafrir (license 46)
+
+2008-10-22 20:05 +0000 [r151600] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c: Change some logical ands to bitwise ands
+ and add messages alerting that a channel is being ignored if the
+ PROC_DAHDI_NOCHAN option is set in process_dahdi. (closes issue
+ #13759) Reported by: smurfix Patches: dahdi.patch uploaded by
+ smurfix (license 547)
+
+2008-10-22 17:45 +0000 [r151554-151555] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Print out the right var in the log message
+
+ * channels/chan_sip.c: Fix this check to use the proper variable
+ (the result from get_in_brackets)
+
+2008-10-22 15:08 +0000 [r151420-151512] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: The logic of a strncasecmp call was
+ reversed. (closes issue #13706) Reported by: andrew53 Patches:
+ sip_notify_from_rfc3265.patch uploaded by andrew53 (license 519)
+
+ * channels/chan_sip.c: Make the sip_standard_port function more
+ granular by allowing separate type and port arguments. This is
+ necessary because when building our From and Contact headers, we
+ need to be absolutely sure that we are placing our source port
+ there and not the peer's source port. (closes issue #12761)
+ Reported by: asbestoshead Patches:
+ patch-chan-sip-contact-port.txt uploaded by asbestoshead (license
+ 455)
+
+ * channels/chan_sip.c: Get this compiling in dev-mode
+
+ * channels/chan_sip.c: If a peer uses any transport other than UDP,
+ then MWI will fail for that peer since sip_alloc will allocate a
+ sip_pvt with a default transport of UDP. This change resets the
+ socket type immediately after allocating the sip_pvt in
+ sip_send_mwi_from_peer, so that the proceeding call to
+ create_addr_from_peer does not fail right away. The socket data
+ from the peer is properly copied to the sip_pvt in
+ create_addr_from_peer. (closes issue #13710) Reported by:
+ andrew53 Patches: sip_notify_use_tcp.patch uploaded by andrew53
+ (license 519)
+
+ * channels/chan_sip.c: When attempting to resolve hostnames, we
+ need to be sure to remove any parameters from the string so that
+ name resolution succeeds. (closes issue #13727) Reported by:
+ fnordian Patches: resolvewithouturiparameter.patch uploaded by
+ fnordian (license 110)
+
+2008-10-21 15:20 +0000 [r151371] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_mixmonitor.c: Default file modes should always be full
+ read and write, to allow the system administrator to make the
+ decision of what permissions will actually be given, through the
+ use of the process umask. (Closes issue# 13751)
+
+2008-10-21 11:02 +0000 [r151327] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_sip.c: Fix configuration parsing so type=friend
+ still identifies "friend" as a peer even though it is now a
+ legacy configuration verb. (closes issue #13705) reported by:
+ blitzrage patched by: bweschke
+
+2008-10-20 05:07 +0000 [r151246] BJ Weschke <bweschke@btwtech.com>
+
+ * pbx/pbx_config.c, main/config.c: Do NOT attempt to do anything
+ with the ast_config struct when it's been returned as INVALID by
+ the config file interpreter. (closes issue #13741)
+
+2008-10-20 05:00 +0000 [r151242-151243] Kevin P. Fleming <kpfleming@digium.com>
+
+ * autoconf/ast_check_pwlib.m4, /, autoconf/ast_check_openh323.m4,
+ configure.ac: Merged revisions 151241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct
+ 2008) | 2 lines rename this macro to properly reflect what it
+ does ........
+
+ * autoconf/ast_prog_egrep.m4, autoconf/ast_c_define_check.m4,
+ autoconf/ast_ext_tool_check.m4 (added),
+ autoconf/ast_check_mandatory.m4 (added), /,
+ autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4,
+ autoconf/ast_prog_sed.m4, acinclude.m4 (removed),
+ autoconf/ast_check_pwlib.m4, autoconf (added),
+ autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure,
+ autoconf/ast_gcc_attribute.m4, bootstrap.sh,
+ autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4,
+ autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4: Merged
+ revisions 151240 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct
+ 2008) | 3 lines break up acinclude.m4 into individual files,
+ which will make it easier to maintain, easier to add new macros
+ (less patching) and will ease maintenance of these macros across
+ Asterisk branches ........
+
+2008-10-19 20:30 +0000 [r151188-151190] BJ Weschke <bweschke@btwtech.com>
+
+ * /: Block 151167 from coming forward into the /trunk this is a 1.4
+ fix only.
+
+ * /: Block 151100 from coming forward into the /trunk this is a 1.4
+ fix only.
+
+2008-10-19 19:11 +0000 [r151101] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
+ apps/app_externalivr.c, include/asterisk/tcptls.h: cleaup of the
+ TCP/TLS socket API: 1) rename 'struct server_args' to 'struct
+ ast_tcptls_session_args', to follow coding guidelines 2) make
+ ast_make_file_from_fd() static and rename it to something that
+ indicates what it really is for (again coding guidelines) 3)
+ rename address variables inside 'struct ast_tcptls_session_args'
+ to be more descriptive (dare i say it... coding guidelines) 4)
+ change ast_tcptls_client_start() to use the new 'remote_address'
+ field of the session args for the destination of the connection,
+ and use the 'local_address' field to bind() the socket to the
+ proper source address, if one is supplied 5) in chan_sip, ensure
+ that we pass in the PP address we are bound to when creating
+ outbound (client) connections, so that our connections will
+ appear from the correct address
+
+2008-10-19 13:10 +0000 [r151060] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: dont segfault when placing a call to a
+ line that has no registered device.
+
+2008-10-19 07:20 +0000 [r151019] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Adding changes from train and flight back
+ home from SIPit23 in Lannion, France. - Additional comments on
+ TCP/TLS implementation - Some additions for new drafts/rfcs (no
+ new functionality really, mostly documentation) - Other random
+ small fixes
+
+2008-10-18 10:27 +0000 [r150930-150971] Michiel van Baak <michiel@vanbaak.info>
+
+ * Makefile: Make sure we support nested functions and generation of
+ trampolines under OpenBSD. (closes issue #13724) Reported by:
+ mvanbaak
+
+ * contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.suse.asterisk: dont use deprecated commands in
+ the init scripts. (closes issue #13720) Reported by:
+ decryptus_proformatique Patches:
+ contrib_initd_module_reload.patch uploaded by decryptus (license
+ 555) With mods by me to fix stop commands as well
+
+2008-10-18 03:35 +0000 [r150773-150887] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_authenticate.c, CHANGES: Give app_authenticate the
+ ability to select a prompt other than the default. (closes issue
+ #13734) reported and patched by: jvandal
+
+ * main/manager.c, /: Using the GetVar handler in AMI is potentially
+ dangerous (insta-crash [tm]) when you use a dialplan function
+ that requires a channel and then you don't provide one or provide
+ an invalid one in the Channel: parameter. We'll handle this
+ situation exactly the same way it was handled in pbx.c back on
+ r61766. We'll create a bogus channel for the function call and
+ destroy it when we're done. If we have trouble allocating the
+ bogus channel then we're not going to try executing the function
+ call at all and run the risk of crashing. (closes issue #13715)
+ reported by: makoto patch by: bweschke
+
+ * doc/manager_1_1.txt, CHANGES, apps/app_queue.c: The QueueEntry
+ event now has the uniqueid of the channel included. (closes issue
+ #13731) reported and patched by: caio1982
+
+2008-10-17 21:48 +0000 [r150731] Matthew Fredrickson <creslin@digium.com>
+
+ * configure, configure.ac: Update configure check to check for new
+ function in libpri (pri_progress_with_cause)
+
+2008-10-17 21:35 +0000 [r150729] Jason Parker <jparker@digium.com>
+
+ * codecs/codec_adpcm.c, codecs/ex_g722.h (added),
+ codecs/codec_gsm.c, codecs/ex_adpcm.h (added), codecs/ex_alaw.h
+ (added), codecs/ex_g726.h (added), codecs/ex_gsm.h (added),
+ codecs/slin_ulaw_ex.h (removed), codecs/slin_lpc10_ex.h
+ (removed), codecs/codec_resample.c, codecs/slin_g722_ex.h
+ (removed), codecs/g722_slin_ex.h (removed), codecs/ex_ulaw.h
+ (added), codecs/adpcm_slin_ex.h (removed), codecs/ex_ilbc.h
+ (added), codecs/slin_adpcm_ex.h (removed), codecs/g726_slin_ex.h
+ (removed), codecs/slin_g726_ex.h (removed), codecs/codec_lpc10.c,
+ codecs/gsm_slin_ex.h (removed), codecs/slin_gsm_ex.h (removed),
+ codecs/codec_a_mu.c, codecs/codec_g722.c, codecs/ex_lpc10.h
+ (added), codecs/codec_alaw.c, codecs/codec_speex.c,
+ codecs/codec_g726.c, include/asterisk/slin.h (added),
+ codecs/ex_speex.h (added), codecs/slin_resample_ex.h (removed),
+ codecs/ulaw_slin_ex.h (removed), codecs/slin_ilbc_ex.h (removed),
+ codecs/ilbc_slin_ex.h (removed), codecs/lpc10_slin_ex.h
+ (removed), codecs/codec_ulaw.c, codecs/codec_ilbc.c,
+ codecs/speex_slin_ex.h (removed), codecs/slin_speex_ex.h
+ (removed): Merge codec_consistency branch. This should make
+ sample usage much happier.
+
+2008-10-17 17:31 +0000 [r150664] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/cli.c: Fix CLI command 'channel request hangup' Prodded on
+ IRC by Russell and fixed by eliel (closes issue #13730) Reported
+ by: eliel Patches: main_cli.patch uploaded by eliel (license 64)
+
+2008-10-17 17:25 +0000 [r150640] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Merge in
+ patch for #13454. Includes CallRereouting dialplan application,
+ option for discard of remote hold messages, and using the
+ alternate logical channel mapping in Q.SIG instead of the default
+ physical channel mapping.
+
+2008-10-17 17:09 +0000 [r150580-150635] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Make helper call a little safer (suggested
+ by Russell on IRC)
+
+ * include/asterisk/sched.h, channels/chan_iax2.c: Fix the FRACK!
+ warnings in chan_iax2 when POKE/LAGRQ packets are not answered.
+
+2008-10-17 08:42 +0000 [r150469-150510] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Adding some additional thoughts on
+ configuration changes to TCP/TLS
+
+ * Makefile: Make sure we support nested functions with GCC 4.01
+ OS/X. This might not be OS/X only, but I'll leave it to kpfleming
+ to add this to the configure script for testing.
+
+2008-10-17 06:00 +0000 [r150426] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c, UPGRADE.txt, configs/skinny.conf.sample,
+ CHANGES: Break up skinny.conf into seperate sections for devices
+ and lines. (closes issue #13412) Reported by: wedhorn Patches:
+ config-restruct-v4.diff uploaded by wedhorn (license 30)
+
+2008-10-17 04:28 +0000 [r150384] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: Fix option handling code. (closes issue
+ #11040) Reported by: DEA Patches: rt-meetme-flag-fixes-v2.txt
+ uploaded by DEA (license 3) with additional fixes by me
+
+2008-10-17 00:18 +0000 [r150311] Mark Michelson <mmichelson@digium.com>
+
+ * doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Add an
+ IAXregistry manager command. See doc/manager_1_1.txt for more
+ details of this command. (closes issue #13326) Reported by: ib2
+ Patches: bug13326_trunk_20080822.diff uploaded by snuffy (license
+ 35)
+
+2008-10-17 00:14 +0000 [r150309] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: Initialize character arrays as they are not
+ guaranteed to be set.
+
+2008-10-17 00:13 +0000 [r150207-150307] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: After a long discussion on #asterisk-bugs,
+ it seems kind of odd that a channel would be named after the
+ originating port. For endpoints that always include ":5060" as
+ part of the From: header, it will mean that you have a ton of
+ channels with names like "SIP/5060-3ea38a8b." I am boldly moving
+ forward with this change in trunk, but I'm not touching other
+ branches with this one since this definitely would qualify as a
+ behavior change. If there is a problem with this commit, and I
+ haven't seen the obvious reason why you'd want to name the
+ channel after the port from which the call originated, then
+ please feel free to revert this
+
+ * main/manager.c, /: Merged revisions 150304 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct
+ 2008) | 6 lines Reverting changes from commits 150298 and 150301
+ since I was mistakenly under the assumption that dialplan
+ functions *always* required that a channel be present. I need to
+ go home earlier, I think :) ........
+
+ * main/manager.c: Merged revisions 150298,150301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct
+ 2008) | 10 lines Don't try to call a dialplan function's read
+ callback from the manager's GetVar handler if an invalid channel
+ has been specified. Several dialplan functions, including CHANNEL
+ and SIP_HEADER, do not check for NULL-ness of the channel being
+ passed in. (closes issue #13715) Reported by: makoto ........
+ r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct
+ 2008) | 3 lines And don't forget to return on the error condition
+ ........
+
+ * apps/app_sms.c: Answer the channel prior to checking for the 'a'
+ option in app_sms. (closes issue #13675) Reported by: alecdavis
+ Patches: app_sms.bug13675.148985.diff.txt uploaded by alecdavis
+ (license 585)
+
+ * apps/app_skel.c: Updating app_skel.c to follow coding guidelines
+ with regards to braces used on if statements. (closes issue
+ #13696) Reported by: alecdavis Patches:
+ app_skel.bug13696B.115850.diff.txt uploaded by alecdavis (license
+ 585)
+
+ * channels/chan_iax2.c: Remove an odd redundant comparison
+
+ * configure, configure.ac: Change configure script to search for
+ openais in both /usr/lib and /usr/lib64 since some distros place
+ 64-bit libraries only in the /usr/lib64 directory. (closes issue
+ #13721) Reported by: jcollie Patches:
+ 0007-Look-in-64bit-dirs-for-openais.patch uploaded by jcollie
+ (license 412)
+
+ * channels/chan_sip.c: INVITES with proxy auth were sent with a
+ different branch than what was in the invite_branch of a sip_pvt,
+ meaning that if a CANCEL were sent later, the branch in the
+ CANCEL would not match the branch in the latest INVITE sent out,
+ leading to some endpoints responding to the CANCEL with a 481.
+ (closes issue #13714) Reported by: fnordian Patches:
+ invite_branch.patch uploaded by fnordian (license 110)
+
+2008-10-16 16:04 +0000 [r150125] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 150124 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16
+ Oct 2008) | 1 line Fix memory leak found by customer ........
+
+2008-10-16 15:48 +0000 [r150118-150121] Terry Wilson <twilson@digium.com>
+
+ * configs/modules.conf.sample: This is nolonger needed
+
+ * res/res_phoneprov.c: func_strings isn't a dependency of this
+ module anymore
+
+2008-10-16 15:02 +0000 [r150052] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: ensure that type=peer entries are only
+ matched on IP/port, not on name (after oej audits all the calls
+ to find_peer() to make sure that forcenamematch is set correctly
+ in each case)
+
+2008-10-16 15:00 +0000 [r150008-150051] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Doxygen addition
+
+ * channels/chan_sip.c: Add some notes on problems with the TCP/TLS
+ implementation
+
+2008-10-16 13:28 +0000 [r149917-149981] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: return this logic to where it used to be,
+ *after* the dialog->needdestroy flag has been determined to be
+ set; otherwise, we generate these debug messages every time we
+ inspect every active dialog
+
+ * channels/chan_sip.c: some additional debugging tools added at
+ SIPit23: - move all setting of 'needdestroy' on dialog structures
+ into the history - report all tags involved when a pedantic check
+ fails on a REFER
+
+ * res/res_phoneprov.c: inter-module dependencies should be included
+ in the source code, not just in sample config files
+
+ * res/res_phoneprov.c: correct file name in message
+
+ * configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES:
+ support relative paths in musiconhold.conf, which makes moh work
+ by default when Asterisk was configured using --prefix and 'make
+ samples' is run
+
+2008-10-15 21:36 +0000 [r149848] BJ Weschke <bweschke@btwtech.com>
+
+ * /: Blocking 149840 from coming forward.
+
+2008-10-15 20:55 +0000 [r149802] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Make the sip_proxy struct reference counted.
+ This is necessary to allow for a sip_pvt to maintain a reference
+ to a sip_peer's outboundproxy even after the peer has been freed.
+ (closes issue #13700) Reported by: fnordian Patches: 13700.patch
+ uploaded by putnopvut (license 60) Tested by: fnordian
+
+2008-10-15 20:14 +0000 [r149756] BJ Weschke <bweschke@btwtech.com>
+
+ * configs/agents.conf.sample, /: Merged revisions 149683 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008)
+ | 4 lines An update to the documentation/example of
+ agents.conf.sample with the correct parameter for this feature as
+ defined in chan_agent.c (closes issue #13709) ........
+
+2008-10-15 19:07 +0000 [r149588-149687] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Permit data fields to contain more than 255
+ characters. (closes issue #13631) Reported by: seanbright
+ Patches: 20081015__bug13631.diff.txt uploaded by Corydon76
+ (license 14) Tested by: blitzrage
+
+ * funcs/func_odbc.c: Only set buf to blank before the goto.
+
+ * codecs/lpc10/lpcini.c: When using MALLOC_DEBUG, codec_lpc10 leaks
+ memory, because it matches a library malloc() with an ast_free
+ (which, of course, doesn't match up with known allocated memory,
+ so the free fails). (closes issue #13702) Reported by: eliel
+ Patches: codec_lpc10_lpcini.c uploaded by eliel (license 64)
+
+ * apps/app_echo.c: Minor spacing change (closes issue #13697)
+ Reported by: alecdavis Patches: app_echo.bug13697.103249.diff.txt
+ uploaded by alecdavis (license 585)
+
+2008-10-15 13:52 +0000 [r149542] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Adding a note about a missing part of
+ "kill-the-user" - I got lost in the Ao2 world... We're going to
+ try to get time to fix this and kpfleming believes that there's
+ code in ao2 so that we can solve it...
+
+2008-10-15 11:26 +0000 [r149384-149487] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 149452 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct
+ 2008) | 3 lines fix some problems when parsing SIP messages that
+ have the maximum number of headers or body lines that we support
+ ........
+
+ * configure, configure.ac: reverting this change... had not read
+ the commit list yet, didn't realize the code had been upgraded
+
+ * configure, configure.ac: do complete version check for SpanDSP,
+ since the app_fax code is not compatible with 0.0.6 yet
+
+ * apps/app_stack.c: building this module depends on res_agi being
+ built as well
+
+2008-10-15 07:45 +0000 [r149342] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Fixing sytax errors ;-)
+
+2008-10-14 23:57 +0000 [r149201-149279] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, CHANGES: When specifying an invalid timeout to
+ Dial, take it to mean that no timeout is desired. (closes issue
+ #13625) Reported by: atis
+
+ * /, channels/chan_sip.c: Merged revisions 149266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r149266 | mmichelson | 2008-10-14 18:43:58 -0500 (Tue, 14 Oct
+ 2008) | 4 lines Change this warning to an error message.
+ Suggestion comes from Sean Bright. Thanks Sean! ........
+
+ * /, channels/chan_sip.c: Merged revisions 149207 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct
+ 2008) | 9 lines Call register_peer_exten even in the case that
+ the peer's IP/port does not change. (closes issue #13309)
+ Reported by: dimas Patches: v2-13309.patch uploaded by dimas
+ (license 88) ........
+
+ * /, include/asterisk/audiohook.h, main/audiohook.c: Merged
+ revisions 149204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct
+ 2008) | 12 lines Add a tolerance period for sync-triggered
+ audiohooks so that if packetization of audio is close (but not
+ equal) we don't end up flushing the audiohooks over small
+ inconsistencies in synchronization. Related to issue #13005, and
+ solves the issue for most people who were experiencing the
+ problem. However, a small number of people are still experiencing
+ the problem on long calls, so I am not closing the issue yet
+ ........
+
+ * /, apps/app_queue.c: Merged revisions 149200 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct
+ 2008) | 12 lines Update the queue with the correct number of
+ calls and whether the call was completed within the service level
+ when a transfer takes place. This way, we do not "break" the
+ leastrecent and fewestcalls strategies by not logging a call
+ until after the transferred call has ended. (closes issue #13395)
+ Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
+ by Marquis (license 32) ........
+
+2008-10-14 22:38 +0000 [r149199] Tilghman Lesher <tlesher@digium.com>
+
+ * main/hashtab.c, pbx/pbx_spool.c, channels/chan_sip.c,
+ include/asterisk/chanvars.h, include/asterisk/config.h,
+ include/asterisk/strings.h, res/res_indications.c,
+ include/asterisk/hashtab.h, main/chanvars.c, main/config.c: Add
+ additional memory debugging to several core APIs, and fix several
+ memory leaks found with these changes. (Closes issue #13505,
+ closes issue #13543) Reported by: mav3rick, triccyx Patches:
+ 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: mav3rick, triccyx
+
+2008-10-14 21:08 +0000 [r149131] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 149130 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct
+ 2008) | 7 lines Don't allow reserved characters to be used in
+ register lines in sip.conf. (closes issue #13570) Reported by:
+ putnopvut ........
+
+2008-10-14 20:16 +0000 [r149062] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_waitforsilence.c: Merged revisions 149061 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008)
+ | 6 lines Check correct values in the return of ast_waitfor();
+ also, get rid of a possible memory leak. (closes issue #13658)
+ Reported by: explidous Patch by: me ........
+
+2008-10-14 19:35 +0000 [r149040] Leif Madsen <lmadsen@digium.com>
+
+ * doc/manager_1_1.txt: Add missing documentation for
+ SipShowRegistry action and RegistryEntry event. (closes issue
+ #13342) Reported and patch by: Laureano
+
+2008-10-14 19:03 +0000 [r148917-148988] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 148987 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14
+ Oct 2008) | 2 lines Some compilers warn, some don't. Fixing.
+ ........
+
+ * apps/app_sms.c: App is ignoring 'p' parameter -- initial pause.
+ (closes issue #13617) Reported by: alecdavis Patches:
+ app_sms.13oct.diff.txt uploaded by alecdavis (license 585)
+
+ * /, apps/app_voicemail.c: Merged revisions 148916 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14
+ Oct 2008) | 4 lines Ensure that mail headers are 7-bit clean,
+ even when UTF-8 characters are used in headers like 'Subject' and
+ 'To'. Closes AST-107. ........
+
+2008-10-14 17:38 +0000 [r148913] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 148912 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue,
+ 14 Oct 2008) | 9 lines Deadlock prevention in chan_local. (closes
+ issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded
+ by putnopvut (license 60) Tested by: tacvbo ........
+
+2008-10-14 15:15 +0000 [r148868] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_fax.c: API differences in spandsp 0.0.6pre1 and higher
+ (closes issue #13688) Reported by: irroot Patches:
+ app_fax-span6.patch uploaded by irroot (license 52) with minor
+ modifications by me
+
+2008-10-14 15:00 +0000 [r148867] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix reference count issue that Russell
+ brought up in SIP MWI NOTIFY support. Bump the reference count up
+ before we add it to the scheduler, duh.
+
+2008-10-14 14:18 +0000 [r148825] Doug Bailey <dbailey@digium.com>
+
+ * phoneprov/polycom.xml: Allow MWI registration for all configured
+ lines.
+
+2008-10-14 11:31 +0000 [r148695-148754] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix some references to the owner of a
+ private structure that may not be present
+
+ * Makefile, /: Merged revisions 148736 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct
+ 2008) | 3 lines on Ubuntu (at least), recent versions of ld in
+ binutils delete all debugging symbols when -x is supplied; since
+ the reasons why -x is being passed are lost in the mists of time,
+ remove it so debugging will work properly ........
+
+ * channels/chan_sip.c: this structure should be static
+
+ * channels/chan_sip.c: ensure that *all* fields in the req
+ structure are cleared out before reusing it; has_to_tag was not
+ cleared, which caused the second incoming call over a TCP socket
+ to fail if pedantic checking was enabled
+
+2008-10-14 09:16 +0000 [r148679] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Adding some clarifications
+
+2008-10-14 08:06 +0000 [r148612] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/translate.c: Merged revisions 148611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct
+ 2008) | 3 lines it would be nice if this message printing code
+ had actually been tested before it was committed... ........
+
+2008-10-14 00:08 +0000 [r148570] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_curl.c, res/res_config_pgsql.c,
+ res/res_config_odbc.c, include/asterisk/config.h,
+ res/res_realtime.c, include/asterisk/strings.h,
+ res/res_config_ldap.c, res/res_config_sqlite.c, main/config.c,
+ apps/app_voicemail.c: Merge realtime_update2 branch, which adds a
+ new realtime API call named 'update2', which permits updates
+ which match across multiple columns, instead of requiring all
+ tables to have a single unique identifier. All of the other API
+ calls with the exception of 'update' already had the ability to
+ match on multiple fields, so it was a missing and very desireable
+ feature that an API call implementing an update should have this,
+ too. This does not change any outward performance of Asterisk,
+ but it should make life easier for application developers who use
+ the RealTime framework.
+
+2008-10-13 17:14 +0000 [r148519] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: Hmmm. Nobody (but me) is interested in seeing the
+ trie info when they do 'dialplan show ...' (even with debug set
+ to non-zero); so I set up a 'dialplan debug [context]' cli
+ command instead, to explicitly show just the trie info. I even
+ added an extension_exists() call to make sure the trie info is
+ built. I moved the explanatory header to above the extension loop
+ to ensure it only prints once. And it will do this now, whether
+ debug is set or not. I removed the trie printing from the
+ 'dialplan show' command entirely.
+
+2008-10-13 15:56 +0000 [r148471-148474] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: - Doxygen formatting. (tss tss) - Fixing
+ language
+
+ * main/tcptls.c, channels/chan_sip.c: Highlightning even more bugs
+ in the current tcp/tls implementation.
+
+ * channels/chan_sip.c: Sending a 403 after a 200 is considered very
+ bad. (found at SIPit)
+
+2008-10-12 09:19 +0000 [r148425] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/res_agi.c: fix the 'agi show commands' CLI function. (closes
+ issue #13666) Reported by: eliel Patches: res_agi.c.patch
+ uploaded by eliel (license 64)
+
+2008-10-10 21:21 +0000 [r148373-148376] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: The logic used when checking a peer got
+ changed subtly in the "kill the user" commit and caused calls
+ relying on the insecure setting to not work properly. I changed
+ for finding a peer back to how it was prior to that commit.
+ (closes issue #13644) Reported by: pj Patches:
+ 13644_trunkv2.patch uploaded by putnopvut (license 60) Tested by:
+ pj
+
+ * channels/chan_sip.c: Make sure that the inUse and inRinging
+ fields for a sip peer cannot go below zero. This is a regression
+ from 1.4 and so it will be applied to 1.6.0 as well. (closes
+ issue #13668) Reported by: mjc
+
+2008-10-10 18:59 +0000 [r148268-148329] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_config.c: Reset continuation items at the beginning of
+ each context (suggested by kpfleming).
+
+ * CHANGES, pbx/pbx_config.c: Add keyword "same", which allows you
+ to create multiple steps in a dialplan, without needing to
+ respecify an extension pattern multiple times. (closes issue
+ #13632) Reported by: blitzrage Patches:
+ 20081006__bug13632.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: blitzrage, Corydon76
+
+ * /, apps/app_voicemail.c: Merged revisions 148257 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10
+ Oct 2008) | 7 lines User not notified of temporary greeting, if
+ ODBC storage is in use. (closes issue #13659) Reported by:
+ moliveras Patches: 20081009__bug13659.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: moliveras ........
+
+2008-10-10 00:42 +0000 [r148200] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk.h, main/tdd.c, main/cryptostub.c,
+ res/res_config_sqlite.c, apps/app_voicemail.c: Don't include
+ logger.h in asterisk.h by default as it is causing problems
+ building app_voicemail. Instead, include it where it is needed.
+ This turned out to be a relatively minor issue because other
+ headers include logger.h as well. Need to test -addons before
+ merging this back to 1.6.0. (closes issue #13605) Reported by:
+ tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright
+ (license 71) Tested by: mmichelson
+
+2008-10-09 23:54 +0000 [r148144-148160] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: The priority was unnecessary for the manager
+ atxfer, so it has been removed. Furthermore, now we actually use
+ the Context argument passed to set the transfer context and don't
+ error out if no context is specified. This addresses the actual
+ problems outlined in issue 12158. Regarding the other points
+ brought up, regarding the inability to not transfer to extensions
+ which cannot be represented by DTMF, it is not enough of a
+ constraint that it is worth attempting to rework the feature.
+ (closes issue #12158) Reported by: davidw
+
+ * apps/app_voicemail.c: Read the callerid in the correct order and
+ make sure to read the Urgent flag value from the IMAP headers.
+ (closes issue #13652) Reported by: jaroth Patches:
+ imapheaders.patch uploaded by jaroth (license 50)
+
+2008-10-09 23:25 +0000 [r148120] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/res_ldap.conf.sample: Fix example schema (closes issue
+ #12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded
+ by flyn (license 503)
+
+2008-10-09 23:15 +0000 [r148112] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/features.c: Merged revisions 146026 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) |
+ 18 lines (closes issue #13579) Reported by: dwagner (closes issue
+ #13584) Reported by: dwagner Tested by: murf, putnopvut The
+ thought occurred to me that the res= from the extension spawn was
+ ending up being returned from the bridge. "Thou shalt not poison
+ the return value". Made the change and it appears to allow blind
+ xfers to work as normal. If I'm wrong, reopen the bugs. But it
+ looks good to me! Many thanks to putnopvut for helping me
+ reproduce this! ........
+
+2008-10-09 21:47 +0000 [r148000-148071] Tilghman Lesher <tlesher@digium.com>
+
+ * formats/format_wav.c, apps/app_minivm.c, channels/chan_agent.c,
+ main/file.c, res/res_monitor.c, apps/app_voicemail.c: Reverting
+ format addition for now
+
+ * apps/app_minivm.c, channels/chan_agent.c, main/file.c,
+ res/res_monitor.c, apps/app_voicemail.c: Fudges for wav16, just
+ like wav49
+
+ * formats/format_wav.c: Add native 16kHz format for wav file
+ format. (Closes issue #13657)
+
+ * sounds/sounds.xml, sounds/Makefile: Publish MOH files in sln16
+ format
+
+ * /, apps/app_voicemail.c: Merged revisions 147997 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09
+ Oct 2008) | 4 lines When blank, callerid name and number should
+ display "unknown caller" in voicemail emails. (Closes issue
+ #13643) ........
+
+2008-10-09 19:27 +0000 [r147952] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: (closes issue #13139) Reported by: krisk84
+ Tested by: krisk84 This change prevents a call that is placed in
+ the parkinglot to be picked up before the PBX is finished. If
+ another extension dials the parking extension before the PBX
+ thread has completed at minimum warnings will occur about the PBX
+ not properly being terminated. At worst, a crash could occur.
+
+2008-10-09 17:48 +0000 [r147899] Michiel van Baak <michiel@vanbaak.info>
+
+ * include/asterisk/endian.h: only include this for OpenBSD. At
+ least FreeBSD is borked when including it (closes issue #13649)
+ Reported by: ys
+
+2008-10-09 17:46 +0000 [r147896] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Remove "second form" of
+ extensions, as it no longer applies. Also, cleanup the grammar,
+ formatting, and introduce several clarifications to the text.
+ (Closes issue #13654)
+
+2008-10-09 17:04 +0000 [r147854] Terry Wilson <twilson@digium.com>
+
+ * phoneprov/000000000000.cfg, res/res_phoneprov.c,
+ configs/phoneprov.conf.sample: Make phoneprov case-insensitive to
+ remove the func_strings dependency of the default config
+
+2008-10-09 17:01 +0000 [r147853] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_dahdi.c, channels/chan_misdn.c,
+ channels/chan_h323.c: fix some CLI commands we borked during
+ devcon2008 Thanks rmudget for letting me know and providing hints
+ on how to fix it best.
+
+2008-10-09 14:17 +0000 [r147807] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES,
+ include/asterisk/autoconfig.h.in, channels/vcodecs.c,
+ main/translate.c, configure.ac, channels/console_video.c,
+ channels/chan_iax2.c, main/astobj2.c, channels/chan_oss.c,
+ main/rtp.c, main/config.c, main/cli.c, channels/chan_usbradio.c,
+ configure, channels/console_gui.c, utils/extconf.c: (closes issue
+ #13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by
+ nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by
+ nickpeirson (license 579) Tested by: nickpeirson, murf 1.
+ replaced all refs to bzero and bcopy to memset and memmove
+ instead. 2. added a note to the CODING-GUIDELINES 3. add two
+ macros to asterisk.h to prevent bzero, bcopy from creeping back
+ into the source 4. removed bzero from configure, configure.ac,
+ autoconfig.h.in
+
+2008-10-09 01:43 +0000 [r147760-147761] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample: *whistle*
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
+ support for subscribing to a voice mailbox on a remote SIP server
+ and making the new/old message count available to local devices.
+ (issue #AST-77)
+
+2008-10-08 22:32 +0000 [r147714] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_meetme.c: Some small tweaks regarding realtime
+ conference announcements. (closes issue #13522) Reported by: DEA
+ Patches: meetme-rt-fixes.txt uploaded by DEA (license 3)
+
+2008-10-08 22:26 +0000 [r147689] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 147681 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08
+ Oct 2008) | 3 lines when parsing a text configuration option,
+ ensure that the buffer on the stack is actually large enough to
+ hold the legal values of that option, and also ensure that
+ sscanf() knows to stop parsing if it would overrun the buffer
+ (without these changes, specifying "buffers=...,immediate" would
+ overflow the buffer on the stack, and could not have worked as
+ expected) ........
+
+2008-10-08 20:07 +0000 [r147635] Sean Bright <sean.bright@gmail.com>
+
+ * configs/voicemail.conf.sample: Add some examples of IMAP
+ accounts.
+
+2008-10-08 19:08 +0000 [r147592] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_sms.c: Correct a typo in the help; also, ensure that the
+ date and time are correctly set, if not specified in the message.
+ (Closes issue #13594, closes issue #13595) Reported by: alecdavis
+ Patches: 20081001__bug13595.diff.txt uploaded by Corydon76
+ (license 14) Tested by: alecdavis
+
+2008-10-08 14:53 +0000 [r147518] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_speech_utils.c: Merged revisions 147517 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct
+ 2008) | 2 lines If we receive DTMF make sure that the state of
+ the speech structure goes back to being not ready. (issue
+ #LUMENVOX-8) ........
+
+2008-10-08 12:28 +0000 [r147476] Bradley Latus <brad.latus@gmail.com>
+
+ * configs/iax.conf.sample: Adjust commented default trunkmtu value
+ to match documentation above it
+
+2008-10-08 12:15 +0000 [r147388-147457] Sean Bright <sean.bright@gmail.com>
+
+ * funcs/func_curl.c, apps/app_meetme.c, cdr/cdr_adaptive_odbc.c,
+ res/res_odbc.c: Keep up with shadow warnings. One day I'll
+ actually enable this in the Makefile.
+
+ * utils/Makefile: When echoing our copies, strip off ASTTOPDIR from
+ the front of the source file.
+
+ * apps/app_dial.c, channels/chan_dahdi.c, channels/chan_iax2.c:
+ Move the DAHDI-to-DAHDI operator mode check from app_dial into
+ chan_dahdi so we don't have to hardcode anything. (closes issue
+ #13636) Reported by: seanbright Patches: 13636.diff uploaded by
+ seanbright (license 71) Reviewed by: russellb, putnopvut
+
+2008-10-07 20:15 +0000 [r147266-147347] Michiel van Baak <michiel@vanbaak.info>
+
+ * configure, configure.ac: Make format_vorbis_ogg compile on
+ OpenBSD (closes issue #13639) Reported by: mvanbaak Patches:
+ 2008100700_oggsupportOBSD.diff.txt uploaded by mvanbaak (license
+ 7) 2008100700_oggsupportOBSD-configurescript.diff.txt uploaded by
+ mvanbaak (license 7) Tested by: mvanbaak
+
+ * channels/Makefile: make this work on OpenBSD
+
+ * configure, configure.ac: Make sure the configs on OpenBSD are in
+ /etc/asterisk by default (closes issue #13641) Reported by: jtodd
+
+ * contrib/scripts/safe_asterisk_restart,
+ contrib/scripts/safe_asterisk: use pkill instead of killall to be
+ more portable
+
+2008-10-07 18:00 +0000 [r147265] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_voicemail.c: This was flawed. The issue that I was
+ trying to address was addressed by adding the imapsecret alias
+ for imappassword. Will rethink this one and give it another shot
+ on a rainy day TBD.
+
+2008-10-07 17:49 +0000 [r147264] Michiel van Baak <michiel@vanbaak.info>
+
+ * CHANGES: fix wording as pointed out by Corydon
+
+2008-10-07 17:44 +0000 [r147262] Tilghman Lesher <tlesher@digium.com>
+
+ * UPGRADE.txt, include/asterisk/options.h, main/asterisk.c,
+ main/term.c: Allow people to select the old console behavior of
+ white text on a black background, by using the startup flag '-B'.
+
+2008-10-07 16:52 +0000 [r147191-147194] Sean Bright <sean.bright@gmail.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 147193 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r147193 | seanbright | 2008-10-07 12:48:30 -0400 (Tue,
+ 07 Oct 2008) | 2 lines Make 'imapsecret' an alias to
+ 'imappassword' in voicemail.conf. ........
+
+ * apps/app_voicemail.c: Or not.
+
+ * apps/app_voicemail.c: There was a boo-boo in TFOT that is causing
+ some confusion on the mailing lists so include 'imapsecret' as an
+ alias to 'imappassword' (and print a little notice nudging users
+ toward the right option name).
+
+2008-10-07 16:04 +0000 [r147146] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Explicitly setting these fields to NULL was done
+ because I wasn't sure if they would be NULL otherwise. Since they
+ will be set automatically, removing.
+
+2008-10-07 14:59 +0000 [r147050-147099] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_voicemail.c: If we encounter something in mailbox
+ options that we don't grok, then spit out a warning instead of
+ just silently ignoring it.
+
+ * apps/app_dial.c: Make sure to compare the correct number of
+ characters when special-casing our DAHDI operator mode stuff.
+ Technically, it would work fine, as 'DAH' is currently unique
+ amongst our channel technologies, but as Jared points out:
+ <@jsmith> Sure... as long as the technology starts whith DAH....
+ but it could be DAHDOO!
+
+2008-10-07 02:02 +0000 [r147011] Richard Mudgett <rmudgett@digium.com>
+
+ * funcs/func_callerid.c: Independent change from branch issue8824
+ that is not part of COLP. (-r142574 rmudgett)
+
+2008-10-07 00:02 +0000 [r146970] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: A blind transfer to the parking thread would
+ cause a segfault because copy_request accesses dst->data w/o
+ being able to tell whether it is proerly initialized
+
+2008-10-06 23:21 +0000 [r146928] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/threadstorage.h: Update documentation;
+ AST_THREADSTORAGE() in trunk only takes a single argument.
+
+2008-10-06 23:14 +0000 [r146925] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/res_config_odbc.c, build_tools/menuselect-deps.in, configure,
+ funcs/func_odbc.c, include/asterisk/autoconfig.h.in,
+ configure.ac, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c,
+ makeopts.in, res/res_odbc.c, apps/app_voicemail.c: All ODBC parts
+ can now use either unixodbc or iodbc. This allows for the ODBC
+ parts to work on OpenBSD as well. 99.99% of the work is done by
+ seanbright (bow, bow) and I actually did nothing but test and
+ yell at him that it still didn't work :) Thanks for helping out !
+
+2008-10-06 23:08 +0000 [r146875-146923] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c, res/res_agi.c, include/asterisk/features.h:
+ Similar to r143204, masquerade the channel in the case of Park
+ being called from AGI.
+
+ * include/asterisk/endian.h: Mvanbaak said this was needed to
+ compile on OpenBSD, so put it in the OpenBSD section.
+
+ * main/features.c: This commit squashes together three commits
+ because the wrong approach was originally used. (One of the
+ commits was only one line.) 1) r143204: The main change here was
+ to masquerade the channel if the channel that was to be parked
+ was running a PBX on it. The PBX thread can then maintain full
+ control of the channel (the zombie) as it expects to while
+ allowing the parking thread full control of the real (parked)
+ channel. 2) r143270: Changed park_call_full to hold the
+ parkinglot lock a little longer, which protects the parkeduser
+ struct from being freed out from underneath. Made sure that the
+ parking extension is added to the parking context while holding
+ the lock thereby ensuring that there are no spurious warnings
+ from removal attempts when a hangup occurs while the parking lot
+ is being announced. 3) r143475: (the one liner) compare peer and
+ chan instead of looking at the parked user (pu), which could have
+ possibly already have been freed by the parking thread
+
+ * main/features.c: fix some comment placement
+
+ * main/features.c: Explicitly set args in park_call_exec NULL so in
+ the case of no options being passed in, there is no garbage
+ attempted to be used. Also, do not set args to unknown value
+ again if there are no options passed in.
+
+2008-10-06 21:18 +0000 [r146807] Michiel van Baak <michiel@vanbaak.info>
+
+ * include/asterisk/endian.h: make aescrypt.c compile on OpenBSD
+ again
+
+2008-10-06 21:09 +0000 [r146802] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c, /,
+ channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c,
+ funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c,
+ funcs/func_callerid.c, apps/app_speech_utils.c: Merged revisions
+ 146799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008)
+ | 8 lines Dialplan functions should not actually return 0, unless
+ they have modified the workspace. To signal an error (and no
+ change to the workspace), -1 should be returned instead. (closes
+ issue #13340) Reported by: kryptolus Patches:
+ 20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
+ ........
+
+2008-10-06 17:32 +0000 [r146738] Sean Bright <sean.bright@gmail.com>
+
+ * configure, configure.ac: Pretty-print a couple configure options
+
+2008-10-06 16:52 +0000 [r146713] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 146711 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r146711 | tilghman | 2008-10-06 11:51:21 -0500 (Mon, 06
+ Oct 2008) | 9 lines Check whether an extension exists in the
+ _call method, rather than the _alloc method, because we need to
+ evaluate the callerid (since that data affects whether an
+ extension exists). (closes issue #13343) Reported by: efutch
+ Patches: 20080915__bug13343.diff.txt uploaded by Corydon76
+ (license 14) Tested by: efutch ........
+
+2008-10-06 16:03 +0000 [r146644] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Merged revisions 146643 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r146643 | kpfleming | 2008-10-06 10:57:49 -0500 (Mon, 06 Oct
+ 2008) | 8 lines ensure that the private structure for pseudo
+ channels is created without 'leaking' configuration data from
+ other configured channels (closes issue #13555) Reported by:
+ jeffg Patches: issue_13555.patch uploaded by kpfleming (license
+ 421) Tested by: jeffg ........
+
+2008-10-06 15:29 +0000 [r146640] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c: This
+ commit introduces a change to how the "joinempty" and
+ "leavewhenempty" options are configured in queues.conf. Instead
+ of using vague terms like "yes," "no," "loose," and "strict," we
+ now accept a comma-separated list of values to determine when to
+ consider a member available. Extended details can be found in the
+ queues.conf.sample file. Note also that the above four referenced
+ values are still accepted for backwards-compatibility, but are
+ mapped internally to the new method of representing the option.
+ AST-105
+
+2008-10-06 00:36 +0000 [r146555-146597] Sean Bright <sean.bright@gmail.com>
+
+ * utils/Makefile: Make NOISY_BUILD work for the calls to cp in
+ utils/Makefile
+
+ * utils/Makefile: Quote arguments to cp so we can handle spaces in
+ our paths.
+
+2008-10-05 22:11 +0000 [r146514] Russell Bryant <russell@digium.com>
+
+ * utils/muted.c: Make this build on my mac.
+
+2008-10-05 21:21 +0000 [r146449] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Recorded merge of revisions 146448 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r146448 | qwell | 2008-10-05 16:17:44 -0500 (Sun, 05 Oct 2008) |
+ 1 line Fix silly formatting. ........
+
+2008-10-05 01:59 +0000 [r146312-146407] Sean Bright <sean.bright@gmail.com>
+
+ * build_tools/make_buildopts_h: This is far from optimal, but I
+ just found a FreeBSD system without md5 installed on it. So look
+ around for all of the different binaries that we could possibly
+ use. I'd wager this gets completely replaced by someone else in
+ less than 24 hours... :)
+
+ * main/asterisk.c: Fix a bug with the last item in CLI history
+ getting duplicated when read from the .asterisk_history file (and
+ subsequently being duplicated when written). We weren't checking
+ the result of fgets() which meant that we read the same line
+ twice before feof() actually returned non- zero. Also, stop
+ writing out an extra blank line between each item in the history
+ file, fix a minor off-by-one error, and use symbolic constants
+ rather than a hardcoded integer.
+
+ * configs/sip_notify.conf.sample: Add ability to remotely reboot
+ snom phones. Also cleaned up and reorganized
+ sip_notify.conf.sample a bit as well. Tested snom reboot on snom
+ 360 and verified snom-check-cfg worked as well. (closes issue
+ #13601) Reported by: mjc Tested by: seanbright
+
+2008-10-03 22:40 +0000 [r146242] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: remove superfluous reference counting operations
+ in manage_parkinglot since ao2_interator_next increments the ref
+ count automatically
+
+2008-10-03 22:10 +0000 [r146198] Sean Bright <sean.bright@gmail.com>
+
+ * main/cli.c: Resolve a subtle bug where we would never
+ successfully be able to get the first item in the CLI entry list.
+ This was preventing '!' from showing up in either 'help' or in
+ tab completion. (closes issue #13578) Reported by: mvanbaak
+
+2008-10-03 18:30 +0000 [r146081] Tilghman Lesher <tlesher@digium.com>
+
+ * CHANGES: document meetme schedule changes (related to issue
+ #11040)
+
+2008-10-03 17:36 +0000 [r146053] Michiel van Baak <michiel@vanbaak.info>
+
+ * CHANGES: put a note in CHANGES about the cli_cleanup done during
+ AstriDevCon
+
+2008-10-03 17:35 +0000 [r146052] Terry Wilson <twilson@digium.com>
+
+ * main/dial.c: The dialing API should inherit datastores as well as
+ variables
+
+2008-10-02 19:30 +0000 [r145959-145962] Russell Bryant <russell@digium.com>
+
+ * CHANGES: The 'P' command for ExternalIVR was also added in 1.6.0
+
+ * CHANGES: TCP support for ExternalIVR went in to 1.6.1, not 1.6.0
+
+2008-10-02 18:02 +0000 [r145915] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_meetme.c: fix the 'meetme list', 'meetme list concise',
+ 'meetme list $confno' and 'meetme list $confno concise' CLI
+ commands (closes issue #13586) Reported by: john8675309 Help and
+ feedback from eliel, thanks!
+
+2008-10-02 17:16 +0000 [r145846] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Permit
+ the syntax and synopsis fields to be set (for func_odbc).
+
+2008-10-02 16:42 +0000 [r145842] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_meetme.c: make this compile under devmode again
+
+2008-10-02 15:28 +0000 [r145771] Sean Bright <sean.bright@gmail.com>
+
+ * configure, configure.ac: This is much cleaner, methinks.
+
+2008-10-02 15:17 +0000 [r145752] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_odbc.c: Merged revisions 145751 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r145751 | tilghman | 2008-10-02 10:13:21 -0500 (Thu, 02 Oct 2008)
+ | 3 lines Some sanity checks that may have led to prior crashes,
+ found by codefreeze-lap (murf) on IRC. Also some cleanup of
+ incorrectly-used constants. ........
+
+2008-10-01 23:48 +0000 [r145692] Sean Bright <sean.bright@gmail.com>
+
+ * configure, configure.ac: Try a test compile using the GMime
+ library. Some distros install gmime-config in the base package
+ instead of the -devel package. Now we print a notice and disable
+ GMime support instead of bombing during the main compilation.
+ (closes issue #13583) Reported by: arkadia
+
+2008-10-01 23:02 +0000 [r145649] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, funcs/func_strings.c,
+ include/asterisk/localtime.h, main/stdtime/localtime.c: Add
+ schedule extensions to app_meetme. In addition, the reporter
+ found a problem within strptime(3), which we are correcting here
+ with ast_strptime(). (closes issue #11040) Reported by: DEA
+ Patches: 20080910__bug11040.diff.txt uploaded by Corydon76
+ (license 14) Tested by: DEA
+
+2008-10-01 22:23 +0000 [r145553-145606] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Okay, this should really do it now. While I did
+ manage to fix blind transfers with my last commit here, I also
+ caused an unwanted side-effect. That is, only the first priority
+ of the 'h' extension would be executed when a blind transfer
+ occurred instead of all priorities. Essentially, my last commit
+ corrected the return value of ast_bridge_call. However, the
+ implementation still was not 100% correct. Now it is.
+
+ * main/features.c: if (!(x) == 0) is the same as if (x).
+
+ * main/features.c: The logic surrounding the return value of
+ ast_spawn_extension within ast_bridge_call was reversed. This
+ problem was observed when a blind transfer placed from the callee
+ channel of a test call failed. While the problem I am solving
+ here is exactly the same as what was reported in issue #13584,
+ the difference is that this fix I am applying is trunk-only.
+ Issue #13584 was reported against the 1.4 branch, and my tests of
+ 1.4's blind transfers appear to work fine.
+
+2008-10-01 17:26 +0000 [r145487] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 145479
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r145479 | lmadsen | 2008-10-01 13:18:30 -0400 (Wed, 01 Oct 2008)
+ | 6 lines Update the realtime_pgsql.sql script to create the
+ setinterfacevar column. (closes issue #13549) Reported by: fiddur
+ ........
+
+2008-10-01 15:44 +0000 [r145428] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_sms.c: Initializing buffer prevents a segfault when
+ arguments are incomplete. (closes issue #13471) Reported by:
+ alecdavis Patches: 20080916__bug13471.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: alecdavis
+
+2008-10-01 14:44 +0000 [r145381] Mark Michelson <mmichelson@digium.com>
+
+ * Makefile: Too many times have I mistyped the word 'install' as
+ 'isntall' Now this typo shall no longer be a problem since 'make
+ isntall' just builds the 'install' target.
+
+2008-10-01 12:29 +0000 [r145329] Russell Bryant <russell@digium.com>
+
+ * CHANGES: tabs to spaces
+
+2008-09-30 22:21 +0000 [r145249] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: (closes issue #13337) Reported by: pj Tested
+ by: pj Set transport to SIP_TRANSPORT_UDP mode if not specified
+ which fixes calls to get_transport returning UNKNOWN.
+
+2008-09-30 21:32 +0000 [r145226] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c, CHANGES: Add support for call pickup on Snom
+ phones. Asterisk now includes a magic call-id in the dialog-info
+ event package used with extension state subscriptions on Snom
+ phones. Then, when the phone sends an INVITE with Replaces for
+ the special callid, Asterisk will perform a pickup on the
+ extension that was subscribed to. The original code on this issue
+ was submitted by xylome. However, contributions have been made by
+ (at least) mgernoth and pkempgen. The final patch was written by
+ seanbright, and includes the necessary logic to allow this work
+ in a technology independent way. (closes issue #5014) Reported
+ by: xylome Patches: issue5014-trunk.diff uploaded by seanbright
+ (license 71)
+
+2008-09-30 21:00 +0000 [r145200] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.h, doc/tex/misdn.tex,
+ channels/chan_misdn.c, channels/misdn/isdn_lib.c: * Miscellaneous
+ formatting changes to make v1.4 and trunk more merge compatible
+ in the mISDN area. channels/chan_misdn.c * Eliminated redundant
+ code in cb_events() EVENT_SETUP
+
+2008-09-28 23:32 +0000 [r145121] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_unistim.c, res/res_config_pgsql.c,
+ apps/app_meetme.c, res/ais/clm.c, res/res_limit.c,
+ main/taskprocessor.c, channels/chan_console.c, apps/app_queue.c,
+ channels/chan_oss.c, main/astobj2.c, main/cli.c,
+ channels/chan_dahdi.c, main/manager.c, channels/chan_misdn.c,
+ channels/chan_features.c, res/res_agi.c, channels/chan_h323.c,
+ res/ais/evt.c, res/res_config_ldap.c, apps/app_mixmonitor.c,
+ res/res_clioriginate.c: Merge the cli_cleanup branch. This work
+ is done by lmadsen, junky and mvanbaak during AstriDevCon. This
+ is the second audit the CLI got, and this time lmadsen made sure
+ he had _ALL_ modules loaded that have CLI commands in them.
+
+2008-09-28 21:39 +0000 [r145076] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c: Change several improper "sizeof" to
+ "strlen", as sizeof in that context would incorrectly use the
+ size of a pointer, rather than the length of a string. (Closes
+ issue #13574)
+
+2008-09-28 17:08 +0000 [r145027] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: rename chandup() and clarify its usage
+
+2008-09-27 16:17 +0000 [r144949-144951] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils/Makefile: remove incorrect comment
+
+ * agi/Makefile, utils/Makefile, include/asterisk/astmm.h: fix bugs
+ caused by r144949 when MALLOC_DEBUG is defined
+
+ * include/asterisk.h, /, main/Makefile, main/ast_expr2.y,
+ Makefile.moddir_rules, utils/astman.c, main/ast_expr2.c,
+ Makefile, utils/Makefile, main/ast_expr2f.c, pbx/pbx_ael.c,
+ main/astmm.c, utils/ael_main.c, main/stdtime/localtime.c,
+ utils/extconf.c, main/ast_expr2.fl: Merged revisions
+ 144924-144925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep
+ 2008) | 6 lines improve header inclusion process in a few small
+ ways: - it is no longer necessary to forcibly include
+ asterisk/autoconfig.h; every module already includes asterisk.h
+ as its first header (even before system headers), which serves
+ the same purpose - astmm.h is now included by asterisk.h when
+ needed, instead of being forced by the Makefile; this means
+ external modules will build properly against installed headers
+ with MALLOC_DEBUG enabled - simplify the usage of some of these
+ headers in the AEL-related stuff in the utils directory ........
+ r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep
+ 2008) | 2 lines fix some minor issues with rev 144924 ........
+
+2008-09-27 00:49 +0000 [r144879] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_dahdi.c, apps/app_queue.c: fix a couple of CLI
+ commands that did not have a help description.
+
+2008-09-26 23:12 +0000 [r144829] Joshua Colp <jcolp@digium.com>
+
+ * configs/rtp.conf.sample: Update documentation to include default
+ setting. This is for you jtodd!
+
+2008-09-26 18:02 +0000 [r144482-144681] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_lua.c: (closes issue #13564) Reported by: mnicholson
+ Patches: pbx_lua9.diff uploaded by mnicholson (license 96) Many
+ thanks to Matt for his upgrade to the lua dialplan option! the
+ Description from the bug: This patch adds a stack trace to errors
+ encountered while executing lua extensions. The patch also
+ handles out of memory errors reported by lua.
+
+ * main/pbx.c, /: Merged revisions 144677 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) |
+ 12 lines (closes issue #13563) Reported by: mnicholson Patches:
+ found1.diff uploaded by mnicholson (license 96) This patch was
+ mainly meant to apply to trunk and 1.6.x, but I'm applying it to
+ 1.4 also, which should be a perfectly harmless fix to the vast
+ majority of users who are not using external switches, but the
+ few who might be affected will not have to go to the pain of
+ filing a bug report. ........
+
+ * utils/build-extensions-conf.lua (removed): Matt suggests we
+ remove utils/build-extensions-conf.lua, as per bug 12961, it is
+ no longer necessary.
+
+ * main/pbx.c, funcs/func_cut.c, channels/chan_oss.c,
+ apps/app_playback.c: (closes issue #13557) Reported by:
+ nickpeirson The user attached a patch, but the license is not yet
+ recorded. I took the liberty of finding and replacing ALL index()
+ calls with strchr() calls, and that involves more than just
+ main/pbx.c; chan_oss, app_playback, func_cut also had calls to
+ index(), and I changed them out. 1.4 had no references to index()
+ at all.
+
+ * pbx/pbx_lua.c: (closes issue #13559) Reported by: mnicholson
+ Patches: pbx_lua8.diff uploaded by mnicholson (license 96)
+
+ * pbx/pbx_lua.c, configs/extensions.lua.sample,
+ include/asterisk/hashtab.h: I added a little verbage to hashtab
+ for the hashtab_destroy func. It was pretty sparsely documented.
+ This update fleshes out the pbx_lua module, to add the switch
+ statements to the extensions in the extensions.lua file, as well
+ as removing them when the module is unloaded. Many thanks to Matt
+ Nicholson for his fine contribution!
+
+ * pbx/pbx_lua.c: (closes issue #13558) Reported by: mnicholson
+ Considering that the example extensions.lua used nothing but
+ ["12345"] notation, and that the resulting error message: [Sep 24
+ 17:01:16] ERROR[12393]: pbx_lua.c:1204 exec: Error executing lua
+ extension: attempt to call a nil value is not very informative as
+ to the nature of the problem, I think this bug fix is a big win!
+
+2008-09-25 01:46 +0000 [r144357] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Recorded merge of revisions 144356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r144356 | tilghman | 2008-09-24 20:44:47 -0500 (Wed, 24 Sep 2008)
+ | 6 lines Backport Hebrew language to voicemail. (closes issue
+ #13155) Reported by: greenfieldtech Patches:
+ voicemail-hebrew-patch-1.4-SVN.c.patch uploaded by greenfieldtech
+ (license 369) ........
+
+2008-09-24 22:05 +0000 [r144314] Doug Bailey <dbailey@digium.com>
+
+ * res/res_phoneprov.c: Blanch the 404 error message for those with
+ no sense of humor
+
+2008-09-24 08:42 +0000 [r144257] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 144238 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24
+ Sep 2008) | 1 line improved helptext of misdn_set_opt. ........
+
+2008-09-24 06:43 +0000 [r144199] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_curl.c: Create a 'hashcompat' option that permits the
+ results of a CURL() able to be passed directly into the HASH()
+ function. Requested via the -users list, and committed at
+ Astricon in the Code Zone.
+
+2008-09-23 23:33 +0000 [r144149] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a conflict in flag values
+
+2008-09-23 16:52 +0000 [r144067] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 144066 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) |
+ 29 lines (closes issue #13489) Reported by: DougUDI Tested by:
+ murf (closes issue #13490) Reported by: seanbright Tested by:
+ murf (closes issue #13467) Reported by: edantie Tested by: murf,
+ edantie, DougUDI This crash happens because we are unsafely
+ handling old pointers. The channel whose cdr is being handled,
+ has been hung up and destroyed already. I reorganized the code a
+ bit, and tried not to lose the fork-cdr-chain concepts of the
+ previous code. I now verify that the 'previous' channel (the
+ channel we had when the bridge was started), still exists, by
+ looking it up by name in the channel list. I also do not try to
+ reset the CDR's of channels involved in bridges. Testing shows it
+ solves the crash problem, and should not negatively impact
+ previous fixes involving CDR's generated during/after blind
+ transfers. (The reason we need to reset the CDR's on the
+ "beginning" channels in the first place). ........
+
+2008-09-23 15:37 +0000 [r144025] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: When a promiscuous redirect contained both a
+ user and host portion in the Contact URI and specifies a
+ transport, the parsing done in parse_moved_contact resulted in a
+ malformed URI. This commit fixes the parsing so that a proper
+ Dial string may be formed when the forwarded call is placed.
+ (closes issue #13523) Reported by: mattdarnell Patches:
+ 13523v2.patch uploaded by putnopvut (license 60) Tested by:
+ mattdarnell
+
+2008-09-22 22:50 +0000 [r143904] Sean Bright <sean.bright@gmail.com>
+
+ * /, formats/format_pcm.c: Merged revisions 143903 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r143903 | seanbright | 2008-09-22 18:49:00 -0400 (Mon,
+ 22 Sep 2008) | 8 lines Use the advertised header size in .au
+ files instead of just assuming they are 24 bytes (the minimum).
+ (closes issue #13450) Reported by: jamessan Patches:
+ pcm-header.diff uploaded by jamessan (license 246) ........
+
+2008-09-21 09:53 +0000 [r143799-143843] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/tex/privacy.tex: fix privacymanager example so it shows how
+ to use the PRIVACYMRGSTATUS variable
+
+ * doc/tex/privacy.tex: document the new context argument for
+ privacymanager so people can do pattern matching on the input
+
+ * doc/tex/privacy.tex: fix privacy documentation. We no longer do
+ priority jumping +101
+
+ * channels/chan_skinny.c: make 'module unload chan_skinny.so'
+ actually work. (closes issue #13524) Reported by: wedhorn
+ Patches: unload.diff uploaded by wedhorn (license 30) With small
+ tweak by me to prevent a crash
+
+2008-09-20 00:52 +0000 [r143737] Sean Bright <sean.bright@gmail.com>
+
+ * /, contrib/scripts/vmail.cgi: Merged revisions 143736 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r143736 | seanbright | 2008-09-19 20:50:10 -0400 (Fri, 19 Sep
+ 2008) | 9 lines Make vmail.cgi work with mailboxes defined in
+ users.conf, too. (closes issue #13187) Reported by: netvoice
+ Patches: 20080911__bug13187.diff.txt uploaded by Corydon76
+ (license 14) (Slightly modified to take alchamist's comments on
+ mantis into account) Tested by: msales, alchamist, seanbright
+ ........
+
+2008-09-19 21:41 +0000 [r143697] Steve Murphy <murf@digium.com>
+
+ * /: This blocks 143674 from trunk; it appears to already done in
+ trunk, since ast_odbc_direct_execute creates a new stmt for each
+ attempt.
+
+2008-09-19 15:43 +0000 [r143609] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: We should only unsubscribe to the device
+ state event subscription if we have previously subscribed.
+ Otherwise a segfault will occur. (closes issue #13476) Reported
+ by: jonnt Patches: 13476.patch uploaded by putnopvut (license 60)
+ Tested by: jonnt
+
+2008-09-18 23:41 +0000 [r143559] Steve Murphy <murf@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 143534 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r143534 | murf | 2008-09-18 16:11:51 -0600 (Thu, 18 Sep 2008) | 1
+ line A micro-fix, in sip_park_thread, where d is freed before the
+ func is done using it. ........
+
+2008-09-17 20:57 +0000 [r143405] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 143404 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r143404 | tilghman | 2008-09-17 15:55:47 -0500 (Wed, 17
+ Sep 2008) | 6 lines When callerid is blank, we want to use
+ "unknown caller" in those cases, too. (closes issue #13486)
+ Reported by: tomo1657 Patches: 20080917__bug13486.diff.txt
+ uploaded by Corydon76 (license 14) ........
+
+2008-09-17 20:25 +0000 [r143340-143400] Mark Michelson <mmichelson@digium.com>
+
+ * main/astmm.c: If attempting to free a NULL pointer when
+ MALLOC_DEBUG is set, don't bother searching for a region to free,
+ just immediately exit. This has the dual benefit of suppressing a
+ warning message about freeing memory at (nil) and of optimizing
+ the free() replacement by not having to do any futile searching
+ for the proper region to free. (closes issue #13498) Reported by:
+ pj Patches: 13498.patch uploaded by putnopvut (license 60) Tested
+ by: pj
+
+ * /, main/rtp.c: Merged revisions 143337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep
+ 2008) | 6 lines Allow for "G.729" if offered in an SDP even
+ though it is not RFC 3551 compliant. Some Cisco switches will
+ send this in an SDP, and it doesn't hurt to be able to accept
+ this. ........
+
+2008-09-15 21:31 +0000 [r143141] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 143140 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r143140 | tilghman | 2008-09-15 16:29:32 -0500 (Mon, 15
+ Sep 2008) | 6 lines Set the raw formats at the same time as the
+ other formats. (closes issue #13240) Reported by: jvandal
+ Patches: 20080813__bug13240.diff.txt uploaded by Corydon76
+ (license 14) ........
+
+2008-09-14 22:16 +0000 [r143082] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: plug a couple of memleaks in chan_skinny.
+ (closes issue #13452) Reported by: pj Patches: memleak5.diff
+ uploaded by wedhorn (license 30) Tested by: wedhorn, pj, mvanbaak
+ (closes issue #13294) Reported by: pj
+
+2008-09-13 14:15 +0000 [r143034] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_osplookup.c: Everytime a compile fails, a puppy dies.
+
+2008-09-13 13:54 +0000 [r142992-143031] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, channels/chan_iax2.c, channels/iax2-parser.c:
+ Repair IAXVAR implementation so that it works again (regression?)
+ (closes issue #13354) Reported by: adomjan Patches:
+ 20080828__bug13354.diff.txt uploaded by Corydon76 (license 14)
+ 20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license
+ 14) Tested by: Corydon76, adomjan
+
+ * channels/chan_unistim.c, main/udptl.c, apps/app_meetme.c,
+ res/res_snmp.c, codecs/codec_adpcm.c, res/res_phoneprov.c,
+ codecs/codec_gsm.c, apps/app_alarmreceiver.c,
+ channels/chan_gtalk.c, res/res_http_post.c,
+ res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c,
+ res/res_jabber.c, main/enum.c, res/res_config_sqlite.c,
+ main/config.c, main/loader.c, main/cdr.c, channels/chan_dahdi.c,
+ channels/chan_phone.c, res/res_smdi.c, main/manager.c,
+ funcs/func_config.c, apps/app_osplookup.c,
+ channels/chan_skinny.c, funcs/func_odbc.c, main/features.c,
+ apps/app_minivm.c, main/http.c, channels/chan_alsa.c,
+ apps/app_amd.c, apps/app_directory.c, res/res_config_ldap.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, codecs/codec_lpc10.c,
+ res/res_config_pgsql.c, main/dnsmgr.c, codecs/codec_g722.c,
+ channels/chan_sip.c, apps/app_festival.c, codecs/codec_speex.c,
+ codecs/codec_alaw.c, res/res_adsi.c, include/asterisk/config.h,
+ channels/chan_agent.c, codecs/codec_g726.c,
+ channels/chan_console.c, apps/app_queue.c, channels/chan_oss.c,
+ main/rtp.c, apps/app_playback.c, channels/chan_jingle.c,
+ channels/chan_h323.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c,
+ res/res_indications.c, main/asterisk.c, res/res_odbc.c,
+ main/dsp.c, apps/app_voicemail.c: Create a new config file
+ status, CONFIG_STATUS_FILEINVALID for differentiating when a file
+ is invalid from when a file is missing. This is most important
+ when we have two configuration files. Consider the following
+ example: Old system: sip.conf users.conf Old result New result
+ ======== ========== ========== ========== Missing Missing SIP
+ doesn't load SIP doesn't load Missing OK SIP doesn't load SIP
+ doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK
+ Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid
+ SIP loads incompletely SIP doesn't load Invalid Missing SIP
+ doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP
+ doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So
+ in the case when users.conf doesn't load because there's a typo
+ that disrupts the syntax, we may only partially load users,
+ instead of failing with an error, which may cause some calls not
+ to get processed. Worse yet, the old system would do this with no
+ indication that anything was even wrong. (closes issue #10690)
+ Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded
+ by Corydon76 (license 14)
+
+2008-09-12 22:24 +0000 [r142929] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 142927 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r142927 | jpeeler | 2008-09-12 17:22:28 -0500 (Fri, 12
+ Sep 2008) | 6 lines (closes issue #12965) Reported by: rlsutton2
+ Prevents local channels from playing MOH at each other which was
+ causing ast_generic_bridge to loop much faster. ........
+
+2008-09-12 20:49 +0000 [r142866] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 142865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008)
+ | 11 lines Create rules for disallowing contacts at certain
+ addresses, which may improve the security of various
+ installations. As this does not change any default behavior, it
+ is not classified as a direct security fix for anything within
+ Asterisk, but may help PBX admins better secure their SIP
+ servers. (closes issue #11776) Reported by: ibc Patches:
+ 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, blitzrage ........
+
+2008-09-12 18:22 +0000 [r142808] Michiel van Baak <michiel@vanbaak.info>
+
+ * /: Recorded merge of revisions 142807 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142807 | mvanbaak | 2008-09-12 19:59:25 +0200 (Fri, 12 Sep 2008)
+ | 2 lines fix copyright year range ........
+
+2008-09-12 16:54 +0000 [r142741-142748] Tilghman Lesher <tlesher@digium.com>
+
+ * main/app.c: When checking for an encoded character, make sure the
+ string isn't blank, first. (Closes issue #13470)
+
+ * /, apps/app_voicemail.c: Merged revisions 142744 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r142744 | tilghman | 2008-09-12 11:38:02 -0500 (Fri, 12
+ Sep 2008) | 4 lines Missing merge from 1.2 fixes errant exit on
+ DTMF, only when language is Italian (cf commit 34242) (Closes
+ issue #7353) ........
+
+ * /, main/file.c: Merged revisions 142740 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142740 | tilghman | 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008)
+ | 4 lines Don't return a free'd pointer, when a file cannot be
+ opened. (closes issue #13462) Reported by: wackysalut ........
+
+2008-09-12 04:50 +0000 [r142676] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, /, main/features.c,
+ include/asterisk/channel.h, apps/app_queue.c: Merged revisions
+ 142675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) |
+ 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is
+ a "second attempt" to restore the previous "endbeforeh" behavior
+ in 1.4 and up. In order to capture information concerning all the
+ legs of transfers in all their infinite combinations, I was
+ forced to this particular solution by a chain of logical
+ necessities, the first being that I was not allowed to rewrite
+ the CDR mechanism from the ground up! This change basically
+ leaves the original machinery alone, which allows IVR and local
+ channel type situations to generate CDR's as normal, but a
+ channel flag can be set to suppress the normal running of the h
+ exten. That flag would be set by the code that runs the h exten
+ from the ast_bridge_call routine, to prevent the h exten from
+ being run twice. Also, a flag in the ast_bridge_config struct
+ passed into ast_bridge_call can be used to suppress the running
+ of the h exten in that routine. This would happen, for instance,
+ if you use the 'g' option in the Dial app. Running this routine
+ 'early' allows not only the CDR() func to be used in the h
+ extension for reading CDR variables, but also allows them to be
+ modified before the CDR is posted to the backends. While I dearly
+ hope that this patch overcomes all problems, and introduces no
+ new problems, reality suggests that surely someone will have
+ problems. In this case, please re-open 13251 (or 13289), and
+ we'll see if we can't fix any remaining issues. ** trunk note:
+ some code to suppress the h exten being run from app_queue was
+ added; for the 'continue' option available only in trunk/1.6.x.
+ ........
+
+2008-09-12 00:49 +0000 [r142635] Sean Bright <sean.bright@gmail.com>
+
+ * cdr/cdr_adaptive_odbc.c: Build under dev-mode
+
+2008-09-11 23:12 +0000 [r142576] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 142575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) |
+ 20 lines (closes issue #13364) Reported by: mdu113 Well,
+ fundamentally, the problems revealed in 13364 are because of the
+ ForkCDR call that is done before the dial. When the bridge is in
+ place, it's dealing with the first (and wrong) cdr in the list.
+ So, I wrote a little func to zip down to the first non-locked cdr
+ in the chain, and thru-out the ast_bridge_call, these results are
+ used instead of raw chan->cdr and peer->cdr pointers. This
+ shouldn't affect anyone who isn't forking cdrs before a dial, and
+ should correct the cdr's of those that do. So, this change ends
+ up correcting the dstchannel and userfield; the disposition was
+ fixed by a previous patch, it was OK coming into this problem.
+ ........
+
+2008-09-11 21:45 +0000 [r142536] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
+ Add usegmtime, as per the recent -users list discussion, and also
+ add my explanation to the file, since that additional text helps
+ people understand the concept.
+
+2008-09-10 22:11 +0000 [r142475] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 142474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) |
+ 30 lines (closes issue #12318) Reported by: krtorio I made a
+ small change to the code that handles local channel situations.
+ In that code, I copy the answer time from the peer cdr, to the
+ bridge_cdr, but I wasn't also copying the disposition from the
+ peer cdr. So, Now I copy the disposition, and I've tested against
+ these cases: 1. phone 1 never answers the phone; no cdr is
+ generated at all. this should show up as a manager command
+ failure or something. 2. phone 2 never answers. CDR is generated,
+ says NO ANSWER 3. phone 2 is busy. CDR is generated, says BUSY 4.
+ phone 2 answers: CDR is generated, times are correct; disposition
+ is ANSWERED, which is correct. The start time is the time that
+ the manager dialed the first phone. The answer time is the time
+ the second phone picks up. I purposely left the cid and src
+ fields blank; since this call really originates from the manager,
+ there is no 'easy' data to put in these fields. If you feel
+ strongly that these fields should be filled in, re-open this bug
+ and I'll dig further. ........
+
+2008-09-10 19:09 +0000 [r142417] Sean Bright <sean.bright@gmail.com>
+
+ * /, configure, acinclude.m4: Merged revisions 142416 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r142416 | seanbright | 2008-09-10 15:05:46 -0400 (Wed,
+ 10 Sep 2008) | 9 lines Fix detection of PWLIB and OpenH323
+ version when spacing in the headers isn't consistent. (closes
+ issue #13426) Reported by: bamby Patches: detect_openh323.diff
+ uploaded by bamby (license 430) (Modified by me to use sed
+ instead of tr) ........
+
+2008-09-10 16:55 +0000 [r142359] Tilghman Lesher <tlesher@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 142358 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142358 | tilghman | 2008-09-10 11:54:29 -0500 (Wed, 10 Sep 2008)
+ | 2 lines Publish new extra sounds version. ........
+
+2008-09-10 16:41 +0000 [r142318-142355] Russell Bryant <russell@digium.com>
+
+ * /, main/sched.c: Merged revisions 142354 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008)
+ | 7 lines It is a normal situation that a task gets put in the
+ scheduler that should run as soon as possible. Accept "0" as an
+ acceptable time to run, and also treat negative as "run now", and
+ don't print a debug message about it. (inspired by a message
+ asking about the "request to schedule in the past" debug message
+ on the -dev list) ........
+
+ * CHANGES: Move last change to CHANGES up to the 1.6.2 section
+
+2008-09-09 22:08 +0000 [r142280] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * configs/jabber.conf.sample, CHANGES, res/res_jabber.c: Disable
+ autoprune by default. (closes issue #13411) Reported by: caio1982
+ Patches: res_jabber_autoprune1.diff uploaded by caio1982 (license
+ 22) Tested by: caio1982
+
+2008-09-09 19:16 +0000 [r142219] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 142218 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142218 | mmichelson | 2008-09-09 14:15:28 -0500 (Tue, 09 Sep
+ 2008) | 14 lines Make sure that the branch sent in CANCEL
+ requests matches the branch of the INVITE it is cancelling.
+ (closes issue #13381) Reported by: atca_pres Patches:
+ 13381v2.patch uploaded by putnopvut (license 60) Tested by:
+ atca_pres (closes issue #13198) Reported by: rickead2000 Tested
+ by: rickead2000 ........
+
+2008-09-09 17:30 +0000 [r142181] Richard Mudgett <rmudgett@digium.com>
+
+ * main/callerid.c: Cleaned up comment
+
+2008-09-09 17:15 +0000 [r142080-142146] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: This is the trunk version of the patch to close
+ issue 12979. The difference between this and the 1.6.0 and 1.6.1
+ versions is that this is a much more invasive change. With this,
+ we completely get rid of the interfaces list, along with all its
+ helper functions. Let me take a moment to say that this change
+ personally excites me since it may mean huge steps forward
+ regarding proper lock order in app_queue without having to strew
+ seemingly unnecessary locks all over the place. It also results
+ in a huge reduction in lines of code and complexity. Way to go
+ Brett! (closes issue #12979) Reported by: sigxcpu Patches:
+ 20080710_issue12979_queue_custom_state_interface_trunk_2.diff
+ uploaded by bbryant (license 36) Tested by: sigxcpu, putnopvut
+
+ * /, channels/chan_sip.c: Merged revisions 142079 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep
+ 2008) | 21 lines When determining if codecs used by SIP peers
+ allow the media to be natively bridged, use the jointcapability
+ instead of the peercapability. It seems that the intent of using
+ the peercapability was to expand the choice of codecs for the
+ call to increase the chances of being able to native bridge the
+ channels. The problem is that if a codec were settled on for the
+ native bridge and that wasn't a codec that was configured to be
+ used by Asterisk for that peer, then Asterisk would send a
+ REINVITE with no codecs in the SDP which is a bug no matter how
+ you slice it. (closes issue #13076) Reported by: ramonpeek
+ Patches: 13076.patch uploaded by putnopvut (license 60) Tested
+ by: tbelder ........
+
+2008-09-09 15:44 +0000 [r142064] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 142063 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008)
+ | 5 lines Ensure that the stored CDR reference is still valid
+ after the bridge before poking at it. Also, keep the channel
+ locked while messing with this CDR. (fixes crashes reported in
+ issue #13409) ........
+
+2008-09-09 12:34 +0000 [r142000] Bradley Latus <brad.latus@gmail.com>
+
+ * include/asterisk/astobj2.h: Minor fix to doco
+
+2008-09-09 12:32 +0000 [r141995-141998] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Use ast_debug for debug messages. I was
+ wondering why debug messages weren't showing up when I had set
+ the debug level high for just app_queue.c. It's because we were
+ only checking the global option_debug variable instead of using
+ the awesome macro which checks both the global and file-specific
+ value
+
+ * channels/chan_oss.c: Fix a memory leak in chan_oss (closes issue
+ #13311) Reported by: eliel Patches: chan_oss.c.patch uploaded by
+ eliel (license 64)
+
+2008-09-09 01:47 +0000 [r141949] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Modify ast_answer() to not hold the channel lock
+ while calling ast_safe_sleep() or when calling ast_waitfor().
+ These are inappropriate times to hold the channel lock. This is
+ what has caused "could not get the channel lock" messages from
+ chan_sip and has likely caused a negative impact on performance
+ results of SIP in Asterisk 1.6. Thanks to file for pointing out
+ this section of code. (closes issue #13287) (closes issue #13115)
+
+2008-09-08 23:00 +0000 [r141810-141906] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Optimization: The only reason we should check
+ member status is if the queue has a joinempty or a leavewhenempty
+ setting which could cause the caller to not join the queue or
+ exit the queue. Prior to this patch, we could potentially
+ traverse the entire queue's member list for no reason since even
+ if the members are currently not available in some way we're
+ going to let the caller join the queue anyway.
+
+ * channels/chan_sip.c: Um, apparently I didn't actually finish
+ merging before committing. Bad bad bad
+
+ * /, channels/chan_sip.c: Merged revisions 141809 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep
+ 2008) | 14 lines Fix pedantic mode of chan_sip to only check the
+ remote tag of an endpoint once a dialog has been confirmed. Up
+ until that point, it is possible and legal for the far-end to
+ send provisional responses with a different To: tag each time.
+ With this patch applied, these provisional messages will not
+ cause a matching problem. (closes issue #11536) Reported by: ibc
+ Patches: 11536v2.patch uploaded by putnopvut (license 60)
+ ........
+
+2008-09-08 21:05 +0000 [r141807] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 141806 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008)
+ | 7 lines When doing an async goto, detect if the channel is
+ already in the middle of a masquerade. This can happen when
+ chan_local is trying to optimize itself out. If this happens,
+ fail the async goto instead of bursting into flames. (closes
+ issue #13435) Reported by: geoff2010 ........
+
+2008-09-08 20:18 +0000 [r141745] Jason Parker <jparker@digium.com>
+
+ * Makefile, /, redhat (removed): Merged revisions 141741 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) |
+ 8 lines Remove RPM package targets from Makefile (and all
+ associated parts). This has never worked in 1.4, and we decided
+ that it makes no sense to be done here. There are many distros
+ out there that already have "proper" spec files that can be
+ (re)used. Closes issue #13113 Closes issue #10950 Closes issue
+ #10952 ........
+
+2008-09-08 17:13 +0000 [r141682] Sean Bright <sean.bright@gmail.com>
+
+ * build_tools/make_buildopts_h: Quote the arguments to grep so that
+ sh on various platforms doesn't choke on the special characters
+ (like ^). (closes issue #13417) Reported by: dougm Patches:
+ 13417.make_buildopts_h.patch uploaded by seanbright (license 71)
+ Tested by: dougm
+
+2008-09-07 00:04 +0000 [r141626] Michiel van Baak <michiel@vanbaak.info>
+
+ * funcs/func_curl.c: make func_curl.c compile under devmode.
+
+2008-09-06 20:19 +0000 [r141566] Steve Murphy <murf@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 141565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1
+ line This fix comes from Joshua Colp The Brilliant, who, given
+ the trace, came up with a solution. This will most likely will
+ close 13235 and 13409. I'll wait till Monday to verify, and then
+ close these bugs. ........
+
+2008-09-06 15:40 +0000 [r141504-141507] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_curl.c: Get rid of the casts that cause warnings on
+ OpenBSD. The compiler is errantly detecting warnings when we
+ redefine a structure each time it is used, even though the
+ structure is identical. Reported by: mvanbaak, via #asterisk-dev
+
+ * /, res/res_agi.c: Merged revisions 141503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008)
+ | 4 lines Reverting behavior change (AGI should not exit non-zero
+ on SUCCESS) (closes issue #13434) Reported by: francesco_r
+ ........
+
+2008-09-06 12:03 +0000 [r141464] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_sip.c, channels/chan_iax2.c, main/cli.c: Some fixes
+ to autocompletion in some commands. Changes applied by this
+ patch: - Fix autocompletion in 'sip prune realtime', sip peers
+ where never auto completed. Now we complete this command with:
+ 'sip prune realtime peer' -> all | like | sip peers Also I have
+ modified the syntax in the usage, was wrong... - Pass
+ ast_cli_args->argv and ast_cli_args->argc while running
+ autocompletion on CLI commands (CLI_GENERATE). With this we avoid
+ comparisons on ast_cli_args->line like this: strcasestr(a->line,
+ " description") strcasestr(a->line, "descriptions ")
+ strcasestr(a->line, "realtime peer"), and so on.. Making the code
+ more confusing (check the spaces in description!). The only thing
+ we must be sure is to first check a->pos or a->argc. - Fix 'iax2
+ prune realtime' autocompletion, now we autocomplete this command
+ with 'all' & 'iax2 peers', check a look that iax2 peers where all
+ the peers, now only the ones in the cache.. (closes issue #13133)
+ Reported by: eliel Patches: clichanges.patch uploaded by eliel
+ (license 64)
+
+2008-09-05 22:03 +0000 [r141367-141425] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_curl.c: Fix func_curl compilation
+
+ * /, channels/chan_agent.c: Merged revisions 141366 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri,
+ 05 Sep 2008) | 7 lines Agent's should not try to call a channel's
+ indicate callback if the channel has been hung up. It will likely
+ crash otherwise ABE-1159 ........
+
+2008-09-05 19:12 +0000 [r141328] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_curl.c, CHANGES: Add the CURLOPT dialplan function,
+ which permits setting various options for use with the CURL
+ dialplan function. (closes issue #12920) Reported by: davevg
+ Patches: 20080904__bug12920.diff.txt uploaded by Corydon76
+ (license 14) Tested by: Corydon76, davevg
+
+2008-09-05 14:18 +0000 [r141115-141157] Steve Murphy <murf@digium.com>
+
+ * main/channel.c, /: Merged revisions 141156 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1
+ line A small change to prevent double-posting of CDR's; thanks to
+ Daniel Ferrer for bringing it to our attention ........
+
+ * pbx/ael/ael-test/ref.ael-vtest25 (added), /,
+ pbx/ael/ael-test/ael-vtest25/extensions.ael,
+ pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c,
+ pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged
+ revisions 141094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) |
+ 70 lines (closes issue #13357) Reported by: pj Tested by: murf
+ (closes issue #13416) Reported by: yarns Tested by: murf If you
+ find this message overly verbose, relax, it's probably not meant
+ for you. This message is meant for probably only two people in
+ the whole world: me, or the poor schnook that has to maintain
+ this code because I'm either dead or unavailable at the moment.
+ This fix solves two reports, both having to do with embedding a
+ function call in a ${} construct. It was tricky because the
+ funccall syntax has parenthesis () in it. And up till now, the
+ 'word' token in the flex stuff didn't allow that, because it
+ would tend to steal the LP and RP tokens. To be truthful, the
+ "word" token was the trickiest, most unstable thing in the whole
+ lexer. I was lucky it made this long without complaints. I had to
+ choose every character in the pattern with extreme care, and I
+ knew that someday I'd have to revisit it. Well, the day has come.
+ So, my brilliant idea (and I'm being modest), was to use the
+ surrounding ${} construct to make a state machine and capture
+ everything in it, no matter what it contains. But, I have to now
+ treat the word token like I did with comments, in that I turn the
+ whole thing into a state-machine sort of spec, with new contexts
+ "curlystate", "wordstate", and "brackstate". Wait a minute,
+ "brackstate"? Yes, well, it didn't take very many regression
+ tests to point out if I do this for ${} constructs, I also have
+ to do it with the $[] constructs, too. I had to create a separate
+ pcbstack2 and pcbstack3 because these constructs can occur inside
+ macro argument lists, and when we have two state machines
+ operating on the same structures we'd get problems otherwise. I
+ guess I could have stopped at pcbstack2 and had the brackstate
+ stuff share it, but it doesn't hurt to be safe. So, the pcbpush
+ and pcbpop routines also now have versions for "2" and "3". I had
+ to add the {KEYWORD} construct to the initial pattern for "word",
+ because previously word would match stuff like "default7",
+ because it was a longer match than the keyword "default". But,
+ not any more, because the word pattern only matches only one or
+ two characters now, and it will always lose. So, I made it the
+ winner again by making an optional match on any of the keywords
+ before it's normal pattern. I added another regression test to
+ make sure we don't lose this in future edits, and had to fix just
+ one regression, where it no longer reports a 'cascaded' error,
+ which I guess is a plus. I've given some thought as to whether to
+ apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
+ decided to put it in 1.4 because one of the bug reports was
+ against 1.4; and it is unexpected that AEL cannot handle this
+ situation. It actually reduced the amount of useless "cascade"
+ error messages that appeared in the regressions (by one line,
+ ehhem). There is a possible side-effect in that it does now do
+ more careful checking of what's in those ${} constructs, as far
+ as matching parens, and brackets are concerned. Some users may
+ find a an insidious problem and correct it this way. This should
+ be exceedingly rare, I hope. ........
+
+2008-09-04 17:27 +0000 [r141039] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c, res/res_agi.c: Merged revisions 141028 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008)
+ | 7 lines (closes issue #11979) Fixes multiple parking problems:
+ Crash when executing a park on an extension dialed by AGI due to
+ not returning the proper return code. Crash when using a builtin
+ feature that was a subset of a enabled dynamic feature. Crash due
+ to always hanging up the peer despite the fact that the peer was
+ supposed to be parked. ........
+
+2008-09-03 20:16 +0000 [r140975] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix some locking order issues in app_queue.
+ This was brought up by atis on IRC a while ago.
+
+2008-09-03 18:06 +0000 [r140938] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c, CHANGES: Added 'skinny show lines
+ verbose' This will print the subs and their status for every line
+ (if any). wedhorn did most of the work with his patch which
+ introduced 'skinny show debug' but a discussion on IRC stated
+ that it should be added to 'skinny show lines' Input on the
+ output format by Qwell on IRC. (closes issue #13344) Reported by:
+ wedhorn
+
+2008-09-03 14:41 +0000 [r140860-140887] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix compilation
+
+ * /, apps/app_voicemail.c: Merged revisions 140850 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r140850 | mmichelson | 2008-09-03 09:29:15 -0500 (Wed,
+ 03 Sep 2008) | 9 lines Fix voicemail forwarding when using ODBC
+ storage. (closes issue #13387) Reported by: moliveras Patches:
+ 13387.patch uploaded by putnopvut (license 60) Tested by:
+ putnopvut, moliveras ........
+
+2008-09-03 14:01 +0000 [r140824] Steve Murphy <murf@digium.com>
+
+ * res/ael/pval.c, main/pbx.c, res/ael/ael.tab.c, res/ael/ael.y,
+ res/ael/ael.tab.h: In these changes, I have added some
+ explanation of changes to the Set and MSet apps, so people aren't
+ so shocked and surprised when they upgrade from 1.4 to 1.6. Also,
+ for the sake of those upgrading from 1.4 to 1.6 with AEL, I
+ provide automatic support for the "old" way of using Set(), that
+ still does the exact same old thing with quotes and backslashes
+ and so on as 1.4 did, by having AEL compile in the use of MSet()
+ instead of Set(), everywhere it inserts this code. But, if the
+ app_set var is set to 1.6 or higher, it uses the "new",
+ non-evaluative Set(). This only usually happens if the user
+ manually inserts this into the asterisk.conf file, or runs the
+ "make samples" command.
+
+2008-09-03 13:48 +0000 [r140821] Sean Bright <sean.bright@gmail.com>
+
+ * cdr/cdr_sqlite.c: Move some duplicated code into a separate
+ function. Also try to do some wacky stuff in the commit message,
+ like: a newline \n a bell \a a tab \t a format specification %p
+ That is all.
+
+2008-09-03 13:41 +0000 [r140817-140820] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Formatting change to test something on the svn server
+
+ * /, main/poll.c: Merged revisions 140816 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008)
+ | 4 lines Don't freak out if the poll emulation receives NULL for
+ the pollfds array (closes issue #13307) Reported by: jcovert
+ ........
+
+2008-09-02 23:48 +0000 [r140752] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 140751 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r140751 | mmichelson | 2008-09-02 18:47:49 -0500 (Tue,
+ 02 Sep 2008) | 6 lines After adding the context checking to
+ app_voicemail for IMAP storage, I left out a crucial place to
+ copy the context to the vm_state structure. This is the
+ correction. ........
+
+2008-09-02 23:44 +0000 [r140691-140749] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, /: Merged revisions 140747 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1
+ line I am turning the warnings generated in ast_cdr_free and
+ post_cdr into verbose level 2 messages. Really, they matter
+ little to end users. You either get the CDR's you wanted, or you
+ don't, and it is a bug. For trunk, I am going one step further.
+ These messages were pretty worthless even for debug, so I'm
+ completely removing them. ........
+
+ * main/channel.c, /: Merged revisions 140690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1
+ line After reconsidering, with respect to 13409, ast_cdr_detach
+ should be OK, better in fact, than ast_cdr_free, which generates
+ lots of useless warnings that will undoubtably generate
+ complaints. Hmmm. It doesn't hush the useless warnings, but it
+ does allow control of posting via the detach and post routines,
+ for those possible situations, where you'd want to post
+ single-channel cdrs. ........
+
+ * main/channel.c, main/pbx.c, /: Merged revisions 140670 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) |
+ 14 lines (closes issue #13409) Reported by: tomaso Patches:
+ asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license
+ 564) I basically spent the day, verifying that this patch solves
+ the problem, and doesn't hurt in non-problem cases. Why valgrind
+ did not plainly reveal this leak absolutely mystifies and stuns
+ me. Many, many thanks to tomaso for finding and providing the
+ fix. ........
+
+2008-09-02 18:15 +0000 [r140606] Sean Bright <sean.bright@gmail.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 140605 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue,
+ 02 Sep 2008) | 8 lines Make sure to use the correct length of the
+ mohinterpret and mohsuggest buffers when copying configuration
+ values. (closes issue #13336) Reported by:
+ decryptus_proformatique Patches:
+ chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
+ by decryptus (license 555) ........
+
+2008-09-02 15:11 +0000 [r140563-140566] Russell Bryant <russell@digium.com>
+
+ * codecs/codec_resample.c, apps/app_jack.c: Update instructions for
+ getting libresample
+
+ * res/ais/lck.c (removed), res/ais/ckpt.c (removed), res/ais/amf.c
+ (removed): I'm not sure how these files got to trunk (probably my
+ fault), but they should not be here
+
+2008-09-02 14:41 +0000 [r140559] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_sip.c: When a call is rejected because of
+ call-limit, the channel driver is behaving as expected, so we
+ shouldn't report it as an error. Change to LOG_NOTICE instead.
+
+2008-08-29 17:53 +0000 [r140491] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c, CHANGES: Added the option s to the Park
+ application which will silence the announcement of the parking
+ space number. Also, fixes the bug of just clearing the flags
+ instead of actually parsing the arguments to Park.
+
+2008-08-29 17:47 +0000 [r140418-140489] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, res/ais/lck.c, /, channels/chan_sip.c,
+ funcs/func_dialgroup.c, res/res_timing_pthread.c,
+ main/features.c, res/res_phoneprov.c, utils/hashtest2.c,
+ channels/chan_console.c, main/taskprocessor.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/config.c: Merged revisions 140488 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug
+ 2008) | 22 lines After working on the ao2_containers branch, I
+ noticed something a bit strange. In all cases where we provide a
+ callback function to ao2_container_alloc, the callback function
+ would only return 0 or CMP_MATCH. After inspecting the
+ ao2_callback() code carefully, I found that if you're only
+ looking for one specific item, then you should return CMP_MATCH |
+ CMP_STOP. Otherwise, astobj2 will continue traversing the current
+ bucket until the end searching for more matches. In cases like
+ chan_iax2 where in 1.4, all the peers are shoved into a single
+ bucket, this makes for potentially terrible performance since the
+ entire bucket will be traversed even if the peer is one of the
+ first ones come across in the bucket. All the changes I have made
+ were for cases where the callback function defined was passed to
+ ao2_container_alloc so that calls to ao2_find could find a unique
+ instance of whatever object was being stored in the container.
+ ........
+
+ * main/file.c: Allow for video files to be opened as well as audio
+ files. (closes issue #13372) Reported by: epicac Patches:
+ 13372.patch uploaded by putnopvut (license 60) Tested by: epicac
+
+ * /, apps/app_voicemail.c: Merged revisions 140421 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r140421 | mmichelson | 2008-08-29 11:01:07 -0500 (Fri,
+ 29 Aug 2008) | 12 lines Add context checking when retrieving a
+ vm_state. This was causing a problem for people who had
+ identically named mailboxes in separate voicemail contexts. This
+ commit affects IMAP storage only. (closes issue #13194) Reported
+ by: moliveras Patches: 13194.patch uploaded by putnopvut (license
+ 60) Tested by: putnopvut, moliveras ........
+
+ * channels/chan_sip.c: Merged revisions 140417 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140417 | mmichelson | 2008-08-29 10:26:52 -0500 (Fri, 29 Aug
+ 2008) | 10 lines Fix SIP's parsing so that if a port is specified
+ in a string to Dial(), it is not ignored. (closes issue #13355)
+ Reported by: acunningham Patches: 13355v2.patch uploaded by
+ putnopvut (license 60) Tested by: acunningham ........
+
+2008-08-27 23:23 +0000 [r140355] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c: Oops
+
+2008-08-27 20:11 +0000 [r140301] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Merged revisions 140299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug
+ 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when
+ in pedantic mode. The problem was that the wrong tags would be
+ compared depending on the direction of the call. (closes issue
+ #13353) Reported by: flefoll Patches:
+ chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
+ (license 244) ........
+
+2008-08-26 21:59 +0000 [r140246] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: Move the mwi send thread functionality
+ back into the do_monitor thread so that it is easier to manage
+ CID spill resources when do_monitor needs to be killed. (closes
+ issue #13213) Reported by: bbryant
+
+2008-08-26 18:48 +0000 [r140205] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 140056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26
+ Aug 2008) | 9 lines (closes issue #12071) Reported by: tzafrir
+ Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested
+ by: tzafrir, jpeeler This patch fixes closing open file
+ descriptors in the case of an error. ........
+
+2008-08-26 18:46 +0000 [r140201] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_followme.c: OpenBSD compat fix (reminded by mvanbaak on
+ #asterisk-dev)
+
+2008-08-26 18:11 +0000 [r140169] Russell Bryant <russell@digium.com>
+
+ * Makefile: Fix building menuselect-tree with PRINT_DIR set. We
+ _must_ use the --quiet flag here, or else some arbitrary text
+ will end up in the resulting menuselect-tree file and things will
+ explode.
+
+2008-08-26 18:05 +0000 [r140167] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/followme.conf.sample, apps/app_followme.c: Standardize
+ the option names for consistency (but continue to work with the
+ existing names for backwards compatibility). (closes issue
+ #13370) Reported by: jsturtevant
+
+2008-08-26 16:10 +0000 [r140061] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 140060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140060 | russell | 2008-08-26 11:07:58 -0500 (Tue, 26 Aug 2008)
+ | 6 lines Fix some bogus scheduler usage in chan_sip. This code
+ used the return value of a completely unrelated function to
+ determine whether the scheduler should be run or not. This would
+ have caused the scheduler to not run in cases where it should
+ have. Also, leave a note about another scheduler issue that needs
+ to be addressed at some point. ........
+
+2008-08-26 15:57 +0000 [r140057] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, configs/cdr.conf.sample, CHANGES,
+ include/asterisk/options.h: (closes issue #13366) Reported by:
+ erousseau This was a reasonable enhancement request, which was
+ easy to implement. Since it's an enhancement, it could only be
+ applied to trunk. Basically, for accounting where "initiated"
+ seconds are billed for, if the microseconds field on the end time
+ is greater than the microseconds field for the answer time, add
+ one second to the billsec field. The implementation was requested
+ by erousseau, and I've implemented it as requested. I've updated
+ the CHANGES, the cdr.conf.sample, and the .h files accordingly,
+ to accept and set a flag for the corresponding new option. cdr.c
+ adds in the extra second based on the usec fields if the option
+ is set. Tested, seems to be working fine.
+
+2008-08-26 15:29 +0000 [r140053] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 140051 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r140051 | russell | 2008-08-26 10:27:23 -0500 (Tue, 26
+ Aug 2008) | 15 lines Fix a race condition with the IAX scheduler
+ thread. A lock and condition are used here to allow newly
+ scheduled tasks to wake up the scheduler just in case the new
+ task needs to run sooner than the current wakeup time when the
+ thread is sleeping. However, there was a race condition such that
+ a newly scheduled task would not properly wake up the scheduler
+ or affect the wake up period. The order of execution would have
+ been: 1) Scheduler thread determines wake up time of N ms. 2)
+ Another thread schedules a task and signals the condition, with
+ an execution time of < N ms. 3) Scheduler thread locks and goes
+ to sleep for N ms. By moving the sleep time determination to
+ inside the critical section, this possibility is avoided.
+ ........
+
+2008-08-25 23:13 +0000 [r139981] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, doc/asterisk.8, include/asterisk/options.h,
+ main/asterisk.c, main/term.c: Optional light colored background,
+ for those who use black on white terminals. (closes issue #13306)
+ Reported by: Corydon76 Patches: 20080814__bug13306__3.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: Corydon76, pkempgen
+
+2008-08-25 21:48 +0000 [r139928] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, /: Merged revisions 139927 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139927 | jpeeler | 2008-08-25 16:47:33 -0500 (Mon, 25 Aug 2008)
+ | 3 lines Fix a typo I made. Lesson learned, apply the patch if
+ one exists. ........
+
+2008-08-25 21:32 +0000 [r139915] Sean Bright <sean.bright@gmail.com>
+
+ * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged
+ revisions 139909 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug
+ 2008) | 9 lines Some versions of awk (nawk, for example) don't
+ like empty regular expressions so be slightly more verbose.
+ (closes issue #13374) Reported by: dougm Patches: 13374.diff
+ uploaded by seanbright (license 71) Tested by: dougm ........
+
+2008-08-25 20:59 +0000 [r139870] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 139869 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008)
+ | 2 lines Make SIPADDHEADER() propagate indefinitely ........
+
+2008-08-25 17:24 +0000 [r139832] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Add output of variables to AgentRingNoAnswer
+ manager event if eventwhencalled is set to "vars" in queues.conf.
+ Yay for consistency. (closes issue #13369) Reported by: srt
+ Patches: 13369_agentringnoanswer_variables.diff uploaded by srt
+ (license 378)
+
+2008-08-25 16:02 +0000 [r139775] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/followme.txt (added), apps/app_followme.c: Realtime
+ capabilities for the Find-Me-Follow-Me application. (closes issue
+ #13295) Reported by: Corydon76 Patches:
+ 20080813__followme_realtime_enabled.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: dferrer
+
+2008-08-25 15:54 +0000 [r139770] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /, main/features.c: Merged revisions 139764 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9
+ lines This patch reverts the changes made via 139347, and 139635,
+ as users are seeing adverse difference. I will un-close 13251.
+ Back to the drawing board/ concept/ beginning/ whatever! ........
+
+2008-08-24 16:26 +0000 [r139704-139707] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c: Memory leak
+
+ * cdr/cdr_pgsql.c: Eliminate open coding of ast_str
+
+2008-08-22 22:32 +0000 [r139627-139662] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 139635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6
+ lines I found some problems with the code I committed earlier,
+ when I merged them into trunk, so I'm coming back to clean up.
+ And, in the process, I found an error in the code I added to
+ trunk and 1.6.x, that I'll fix using this patch also. ........
+
+ * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions
+ 139347 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) |
+ 47 lines (closes issue #13251) Reported by: sergee Tested by:
+ murf THis is a bold move for a static release fix, but I wouldn't
+ have made it if I didn't feel confident (at least a *bit*
+ confident) that it wouldn't mess everyone up. The reasoning goes
+ something like this: 1. We simply cannot do anything with CDR's
+ at the current point (in pbx.c, after the __ast_pbx_run loop).
+ It's way too late to have any affect on the CDRs. The CDR is
+ already posted and gone, and the remnants have been cleared. 2. I
+ was very much afraid that moving the running of the 'h' extension
+ down into the bridge code (where it would be now practical to do
+ it), would result in a lot more calls to the 'h' exten, so I
+ implemented it as another exten under another name, but found, to
+ my pleasant surprise, that there was a 1:1 correspondence to the
+ running of the 'h' exten in the pbx_run loop, and the new spot at
+ the end of the bridge. So, I ifdef'd out the current 'h' loop,
+ and moved it into the bridge code. The only difference I can see
+ is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this
+ is still an important decision point, I can replicate it if there
+ are complaints. To be perfectly honest, the KEEPALIVE situation
+ is not totally clear to me, and how it relates to a post-bridge
+ situation is less clear. I suspect the users will point out
+ everything in total clarity if this steps on anyone's toes! 3. I
+ temporarily swap the bridge_cdr into the channel before running
+ the 'h' exten, which makes it possible for users to edit the cdr
+ before it goes out the door. And, of course, with the
+ endbeforehexten config var set, the users can also get at the
+ billsec/duration vals. After the h exten finishes, the cdr is
+ swapped back and processing continues as normal. Please, all who
+ deal with CDR's, please test this version of Asterisk, and file
+ bug reports as appropriate! ........ I also made a little fix to
+ the app_dial's 'e' option, that is related to my updates.
+
+2008-08-22 21:57 +0000 [r139622-139624] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, /: Merged revisions 139621 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008)
+ | 5 lines (closes issue #13359) Reported by: Laureano Patches:
+ originate_channel_check.patch uploaded by Laureano (license 265)
+ ........
+
+ * main/features.c: remove extra comma typo
+
+2008-08-22 20:20 +0000 [r139457-139563] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: The -1 return value from incomplete or
+ improper headers for the SipNotify manager command was causing
+ the current manager session to become disconnected. Change the
+ return value to 0 for these cases. Also change a test for a NULL
+ pointer to be ast_strlen_zero instead. (closes issue #13351)
+ Reported by: Laureano Patches: sipnotify_action_fix.patch
+ uploaded by Laureano (license 265)
+
+ * main/features.c: Add missing unique id to ParkedCallGiveUp and
+ ParkedCallTimeOut manager events (closes issue #13358) Reported
+ by: srt Patches: 13358_parking_events.diff uploaded by srt
+ (license 378)
+
+ * /, include/asterisk/threadstorage.h: Merged revisions 139553 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug
+ 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is
+ selected (closes issue #13298) Reported by: snuffy Patches:
+ bug13298_20080822.diff uploaded by snuffy (license 35) ........
+
+ * /, channels/chan_iax2.c: Merged revisions 139466 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri,
+ 22 Aug 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........
+
+ * /, channels/chan_iax2.c: Merged revisions 139456 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri,
+ 22 Aug 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting
+ from incorrect locking order between iax2_pvt and ast_channel
+ structures. AST-13 ........
+
+2008-08-21 23:41 +0000 [r139391] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 139387 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21
+ Aug 2008) | 3 lines Fixes loop that could possibly never exit in
+ the event of a channel never being able to be opened or specify
+ after a restart. (closes issue #11017) ........
+
+2008-08-21 23:00 +0000 [r139345-139346] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * apps/app_receivefax.c (removed), apps/app_sendfax.c (removed):
+ oops
+
+ * apps/app_receivefax.c (added), apps/app_sendfax.c (added):
+ initiate T38 negotiation in FaxSend; use channel variables; other
+ stuff too
+
+2008-08-21 09:55 +0000 [r139281] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Fix two memory leaks in chan_gtalk, thanks
+ Eliel! (closes issue #13310) Reported by: eliel Patches:
+ chan_gtalk.c.patch uploaded by eliel (license 64)
+
+2008-08-20 22:16 +0000 [r139215] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 139213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008)
+ | 11 lines Fix a crash in the ChanSpy application. The issue here
+ is that if you call ChanSpy and specify a spy group, and sit in
+ the application long enough looping through the channel list, you
+ will eventually run out of stack space and the application with
+ exit with a seg fault. The backtrace was always inside of a
+ harmless snprintf() call, so it was tricky to track down.
+ However, it turned out that the call to snprintf() was just the
+ biggest stack consumer in this code path, so it would always be
+ the first one to hit the boundary. (closes issue #13338) Reported
+ by: ruddy ........
+
+2008-08-20 22:06 +0000 [r139210] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Fix output of sipshowpeer manager response.
+ (closes issue #13346) Reported by: srt Patches:
+ 13346_malformed_sip_show_peer_response.diff uploaded by srt
+ (license 378)
+
+2008-08-20 20:03 +0000 [r139153-139154] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: Remove extraneous debugging messages.
+
+ * codecs/codec_dahdi.c: Fix bug where the samples were not accurate
+ when in G723 mode, which would cause the timestamp field of the
+ RTP header to be invalid.
+
+2008-08-20 17:25 +0000 [r139083] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, /: Merged revisions 139074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) |
+ 12 lines (closes issue #13263) Reported by: brainy Tested by:
+ murf The specialized reset routine is tromping on the flags field
+ of the CDR. I made a change to not reset the DISABLED bit. This
+ should get rid of this problem. ........
+
+2008-08-20 16:16 +0000 [r139020] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: fix unholding phones after hangup on
+ older cisco phones. Patch by wedhorn.
+
+2008-08-20 15:38 +0000 [r138887-139016] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 139015 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug
+ 2008) | 6 lines sip_read should properly handle a NULL return
+ from sip_rtp_read. (closes issue #13257) Reported by: travishein
+ ........
+
+ * /, channels/chan_agent.c: Merged revisions 138942 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r138942 | mmichelson | 2008-08-19 18:17:17 -0500 (Tue,
+ 19 Aug 2008) | 11 lines Reset agent_pvt variables back to the
+ values in agents.conf (from what the corresponding channel
+ variables were set to) when the agent logs out. (closes issue
+ #13098) Reported by: davidw Patches:
+ 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by
+ bbryant (license 36) Tested by: davidw ........
+
+ * /, apps/app_chanspy.c: Merged revisions 138886 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug
+ 2008) | 23 lines Add a lock and unlock prior to the destruction
+ of the chanspy_ds lock to ensure that no other threads still have
+ it locked. While this should not happen under normal
+ circumstances, it appears that if the spyer and spyee hang up at
+ nearly the same time, the following may occur. 1.
+ ast_channel_free is called on the spyee's channel. 2. The chanspy
+ datastore is removed from the spyee's channel in
+ ast_channel_free. 3. In the spyer's thread, the spyer attempts to
+ remove and destroy the datastore from the spyee channel, but the
+ datastore has already been removed in step 2, so the spyer
+ continues in the code. 4. The spyee's thread continues and calls
+ the datastore's destroy callback, chanspy_ds_destroy. This
+ involves locking the chanspy_ds. 5. Now the spyer attempts to
+ destroy the chanspy_ds lock. The problem is that in step 4, the
+ spyee has locked this lock, meaning that the spyer is attempting
+ to destroy a lock which is currently locked by another thread.
+ The backtrace provided in issue #12969 supports the idea that
+ this is possible (and has even occurred). This commit does not
+ close the issue, but should help in preventing one type of crash
+ associated with the use of app_chanspy. ........
+
+2008-08-19 16:56 +0000 [r138851] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: chan_skinny now respects callwaiting=no
+ (closes issue #12691) Reported by: sbisker Patches:
+ callwaitingv1.diff uploaded by wedhorn (license 30) Tested by:
+ wedhorn on old skinny phones, mvanbaak on 7960 and 7905 with
+ latest firmware
+
+2008-08-19 16:31 +0000 [r138815-138845] Steve Murphy <murf@digium.com>
+
+ * res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h,
+ utils/ael_main.c, utils/conf2ael.c: Oops. put a decl in a
+ generated file. My bad, but fixed now.
+
+ * main/pbx.c, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h:
+ These changes are in regards to bug 13249, where users are being
+ surprised by the changes made to the Set app in trunk/1.6.x, as
+ they come from the 1.4 world. They are only bitten if they write
+ their AEL dialplan in the 1.4 world, and then carry it over to a
+ trunk/1.6.x installation where a "make samples" was executed, or
+ where they hand-edited the asterisk.conf file and added the
+ [compat] category with app_set = 1.6 (or higher). (this commit
+ does not totally solve 13249, at least not yet) The change
+ involves issueing a single warning while the AEL file is loading,
+ if: 1. app_set is present in the config file, and set to 1.6 or
+ higher. 2. there are double quotes in an assignment statement (eg
+ x = "hi there";) 3. the warning was not already issued. The
+ standalone app, aelparse, does not (yet) issue this warning. I'd
+ have to have it read in the asterisk.conf file, and that's a bit
+ of hassle. I'll add it if users request it, tho.
+
+2008-08-19 15:58 +0000 [r138814] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Mention JID rather than SreenName in help
+ messages
+
+2008-08-19 00:10 +0000 [r138775-138780] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_sip.c: Let it compile now, too (woops)
+
+ * channels/chan_sip.c: And remove code we don't need anymore.
+
+ * channels/chan_sip.c: While we're at it, make this machine
+ parseable too.
+
+ * channels/chan_sip.c: Change event header to RegistrationTime to
+ be more consistent (and avoid breaking existing frameworks).
+ Pointed out by Laureano on #asterisk-dev.
+
+2008-08-18 21:07 +0000 [r138738] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
+ doc/tex/misdn.tex, channels/chan_misdn.c,
+ configs/misdn.conf.sample, channels/misdn/isdn_lib.c,
+ channels/misdn_config.c: channels/chan_misdn.c * Made
+ bearer2str() use allowed_bearers_array[] * Made use the causes.h
+ defines instead of hardcoded numbers. * Made use Asterisk
+ presentation indicator values if either of the mISDN presentation
+ or screen options are negative. * Updated the misdn_set_opt
+ application option descriptions. * Renamed the awkward Caller ID
+ presentation misdn_set_opt application option value not_screened
+ to restricted. Deprecated the not_screened option value.
+ channels/misdn/isdn_lib.c * Made use the causes.h defines instead
+ of hardcoded numbers. * Fixed some spelling errors and typos. *
+ Added all defined facility code strings to fac2str().
+ channels/misdn/isdn_lib.h * Added doxygen comments to struct
+ misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
+ comments to struct misdn_stack. channels/misdn_config.c
+ configs/misdn.conf.sample * Updated the mISDN presentation and
+ screen parameter descriptions. doc/tex/misdn.tex * Updated the
+ misdn_set_opt application option descriptions. * Fixed some
+ spelling errors and typos.
+
+2008-08-18 20:23 +0000 [r138687-138694] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: Change the queue
+ timeout priority logic into less ugly and confusing code pieces.
+ Clarify the logic within queues.conf.sample. (closes issue
+ #12690) Reported by: atis Patches: queue_timeoutpriority.patch
+ uploaded by atis (license 242)
+
+ * apps/app_queue.c: Merged revisions 138685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug
+ 2008) | 10 lines Change the inequalities used in app_queue with
+ regards to timeouts from being strict to non-strict for more
+ accuracy. (closes issue #13239) Reported by: atis Patches:
+ app_queue_timeouts_v2.patch uploaded by atis (license 242)
+ ........
+
+2008-08-18 15:54 +0000 [r138631] Jason Parker <jparker@digium.com>
+
+ * Makefile: Remove option that isn't valid here.
+
+2008-08-18 02:13 +0000 [r138518] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: add missing define for SS7 in
+ dahdi_restart
+
+2008-08-17 14:12 +0000 [r138442-138482] Sean Bright <sean.bright@gmail.com>
+
+ * main/features.c: Move Uniqueid to the end of the event for those
+ that rely on the position of the name/value pairs, pointed out by
+ snuffy-home on #asterisk-commits. For those of you who rely on
+ the position of name/value pairs in manager events... stop...
+ that is why associative arrays were invented.
+
+ * main/features.c: Add Uniqueid header to ParkedCall manager event.
+ (closes issue #13323) Reported by: srt Patches:
+ 13323_unique_id_for_parkedcalls_event.diff uploaded by srt
+ (license 378)
+
+ * main/rtp.c: Add missing colons to RTCPReceived and RTCPSent
+ manager events. (closes issue #13319) Reported by: srt Patches:
+ 13319_rtcp_manager_event_headers.diff uploaded by srt (license
+ 378)
+
+ * channels/chan_iax2.c: Fix the output of the JitterBufStats
+ manager event. (closes issue #13324) Reported by: srt Patches:
+ 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt
+ (license 378)
+
+ * configs/cdr_tds.conf.sample: Since it's introduction in revision
+ 3497, cdr_tds has *never* read the port configuration option from
+ cdr_tds.conf. So go ahead and remove it from the sample config.
+
+2008-08-16 13:07 +0000 [r138409-138412] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Fix compilation warnings (found with
+ dev-mode)
+
+ * main/pbx.c: Also make sure hinting won't crash on reload. (Closes
+ issue #13312)
+
+2008-08-16 01:13 +0000 [r138311-138361] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 138360 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15
+ Aug 2008) | 1 line fixes use count to properly decrement if an
+ active dahdi channel is destroyed allowing module to be unloaded
+ ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 138119,138151,138238
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008)
+ | 4 lines Fixes the dahdi restart functionality. Dahdi restart
+ allows one to restart all DAHDI channels, even if they are
+ currently in use. This is different from unloading and then
+ loading the module since unloading requires the use count to be
+ zero. Reloading the module is different in that the signalling is
+ not changed from what it was originally configured. Also, this
+ fixes not closing all the file descriptors for D-channels upon
+ module unload (which would prevent loading the module
+ afterwards). (closes issue #11017) ........ r138151 | jpeeler |
+ 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared
+ static mutexes using AST_MUTEX_DEFINE_STATIC macro ........
+ r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008)
+ | 1 line initialize condition variable ss_thread_complete using
+ ast_cond_init ........
+
+2008-08-15 22:54 +0000 [r138206-138260] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 138258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008)
+ | 8 lines More fixes for realtime peers. (closes issue #12921)
+ Reported by: Nuitari Patches: 20080804__bug12921.diff.txt
+ uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: Corydon76 ........
+
+ * main/pbx.c, configs/extensions.conf.sample: Remove deprecated
+ syntax from sample config file (closes issue #13314) Reported by:
+ kue
+
+2008-08-15 20:12 +0000 [r138155] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to
+ dfd to match 1.4 (left over from DAHDI transition)
+
+2008-08-15 19:36 +0000 [r138086-138148] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Change free to ast_free_ptr, too
+
+ * main/pbx.c: e->data can be NULL, so use the safe version of
+ ast_strdup() (closes issue #13312) Reported by: pj
+
+ * channels/chan_sip.c: regseconds is actually stored as the epoch
+ time, not registration length
+
+2008-08-15 15:09 +0000 [r138028] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 138027 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008)
+ | 9 lines Ensure that when a hangup occurs in autoservice, that a
+ hangup frame gets properly deferred to be read from the channel
+ owner when it gets taken out of autoservice. (closes issue
+ #12874) Reported by: dimas Patches: v1-12874.patch uploaded by
+ dimas (license 88) ........
+
+2008-08-15 15:03 +0000 [r138024] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 138023 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15
+ Aug 2008) | 8 lines Additional check for more string specifiers
+ than arguments. (closes issue #13299) Reported by: adomjan
+ Patches: 20080813__bug13299.diff.txt uploaded by Corydon76
+ (license 14) func_strings.c-sprintf.patch uploaded by adomjan
+ (license 487) Tested by: adomjan ........
+
+2008-08-14 22:43 +0000 [r137987] Russell Bryant <russell@digium.com>
+
+ * doc/tex/Makefile: Fix a bashism that causes an error when trying
+ to build the pdf on ubuntu
+
+2008-08-14 18:47 +0000 [r137933] Sean Bright <sean.bright@gmail.com>
+
+ * cdr/cdr_sqlite3_custom.c: Fix memory leak in cdr_sqlite3_custom.
+ (closes issue #13304) Reported by: eliel Patches: sqlite.patch
+ uploaded by eliel (license 64) (Slightly modified by me)
+
+2008-08-14 18:12 +0000 [r137901] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Prepare for adding 1.6.2 changes
+
+2008-08-14 16:52 +0000 [r137848] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 137847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14
+ Aug 2008) | 9 lines When creating the secondary subchannel name,
+ it is necessary to compare to the existing channel name without
+ the "Zap/" or "DAHDI/" prefix, since our test string is also
+ without that prefix. (closes issue #13027) Reported by: dferrer
+ Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer
+ (license 525) (Slightly modified by me, to compensate for both
+ names) ........
+
+2008-08-14 15:32 +0000 [r137812] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Make sure we set the socket port, so we
+ don't try to use <ip address>:0. (closes issue #13255) Reported
+ by: falves11 Patches: 13255-socketport.diff uploaded by qwell
+ (license 4) Tested by: falves11
+
+2008-08-14 15:03 +0000 [r137780] Sean Bright <sean.bright@gmail.com>
+
+ * cdr/cdr_tds.c: If we detect that we are no longer connected, try
+ to reconnect a few times before giving up. This relies on the
+ timeout settings in the freetds.conf file and, unfortunately, on
+ a recent version of FreeTDS (0.82 or newer). I either need to
+ change the current execs to be non-blocking (which I do not want
+ to do) or we have to force people to run with the latest and
+ greatest of FreeTDS. I'm on the fence...
+
+2008-08-14 14:15 +0000 [r137732] Russell Bryant <russell@digium.com>
+
+ * /, configs/sip.conf.sample: Merged revisions 137731 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14
+ Aug 2008) | 4 lines Comments in this config file were aligned
+ only if your tab size was set to 8. So, convert tabs to spaces so
+ that things should be aligned regardless of what tab size you use
+ in your editor. ........
+
+2008-08-14 02:03 +0000 [r137680] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, Zaptel-to-DAHDI.txt: Merged revisions 137679 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug
+ 2008) | 1 line forgot one module name that changed ........
+
projects / thirdparty / asterisk.git / commitdiff
