[asterisk-commits] lmadsen: tag 1.6.2.18-rc1 r308630 - /tags/1.6.2.18-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Feb 23 18:01:34 CST 2011
Author: lmadsen
Date: Wed Feb 23 18:01:29 2011
New Revision: 308630
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=308630
Log:
Importing files for 1.6.2.18-rc1 release.
Added:
tags/1.6.2.18-rc1/.lastclean (with props)
tags/1.6.2.18-rc1/.version (with props)
tags/1.6.2.18-rc1/ChangeLog (with props)
Added: tags/1.6.2.18-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.18-rc1/.lastclean?view=auto&rev=308630
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--- tags/1.6.2.18-rc1/ChangeLog (added)
+++ tags/1.6.2.18-rc1/ChangeLog Wed Feb 23 18:01:29 2011
@@ -1,0 +1,29984 @@
+2011-02-23 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.18-rc1 Released.
+
+2011-02-22 15:37 +0000 [r308528] Andrew Latham <lathama at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * apps/app_privacy.c: Resolve a compiler warning.
+
+2011-01-20 15:42 +0000 [r302917] Leif Madsen <lmadsen at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * res/res_timing_timerfd.c: Handle 'Resource temporarily
+ unavailable' error more gracefully.
+
+2011-01-19 15:52 +0000 [r302416] Sean Bright <sean at 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 at 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 at 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 at 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.
[... 29274 lines stripped ...]
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