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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-10-18 22:27:19 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-10-18 22:27:19 +0000
commit0e6eebd6780015766ed9833bbeaf3dcf97712198 (patch)
tree31112c14c04cac87d7bdd7947e68fe66020e4321
parentfa889ab43e321795242014237078a176a81aff8f (diff)
Importing files for 1.8.0-rc5 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0-rc5@292275 f38db490-d61c-443f-a65b-d21fe96a405b
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+
diff --git a/.version b/.version
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+1.8.0-rc5
diff --git a/ChangeLog b/ChangeLog
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@@ -0,0 +1,25500 @@
+2010-10-18 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc5 Released.
+
+2010-10-18 22:02 +0000 [r292230] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 292229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292229 | lmadsen | 2010-10-18 17:01:16 -0500 (Mon, 18 Oct 2010)
+ | 3 lines Fix typo in the sounds/Makefile. (Issue #17426)
+ ........
+
+2010-10-18 21:55 +0000 [r292227] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 292226 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500
+ (Mon, 18 Oct 2010) | 18 lines 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 21:51 +0000 [r292225] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/sounds.xml, sounds/Makefile, /: Merged revisions 292224
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500
+ (Mon, 18 Oct 2010) | 17 lines 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 ........ ................
+
+2010-10-18 19:50 +0000 [r292188] Russell Bryant <russell@digium.com>
+
+ * main/netsock2.c: Resolve some compiler errors in
+ ast_sockaddr_is_any(). These errors came up once this function
+ was used from within netsock2.c. The errors were like the
+ following: netsock2.c:393: error: dereferencing pointer
+ ‘({anonymous})’ does break strict-aliasing rules The usage of a
+ union here avoids this problem.
+
+2010-10-18 19:16 +0000 [r292155] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c: Fixes build error for systems not supporting
+ IPV6_TCLASS.
+
+2010-10-18 17:15 +0000 [r292122] Matthew Nicholson <mnicholson@digium.com>
+
+ * addons/chan_mobile.c: Fix the cmgr parser. (closes issue 0018152)
+ Reported by: menschentier
+
+2010-10-18 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc4 Released
+
+2010-10-18 16:02 +0000 [r292085] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c: Fixes qos settings for sockets bound to any IPv6
+ or IPv4 address. (closes issue #18099) Reported by: jamesnet
+ Patches: issues_18099_v3.diff uploaded by dvossel (license 671
+
+2010-10-18 15:32 +0000 [r292083] Jeff Peeler <jpeeler@digium.com>
+
+ * pbx/pbx_spool.c: Disable use of inotify for call file handling as
+ it is not working properly. (related to #18089)
+
+2010-10-16 10:47 +0000 [r292050] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * res/res_musiconhold.c, /, configs/musiconhold.conf.sample: Merged
+ revisions 292049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) |
+ 15 lines 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 21:40 +0000 [r292016] Terry Wilson <twilson@digium.com>
+
+ * res/res_srtp.c: Ref/unref res_srtp when we create/destroy a
+ session This avoids unhappy crashing when we try to 'core stop
+ gracefully' and res_srtp tries to unload before chan_sip does.
+ Thanks, Russell! (closes issue #18085) Reported by: st
+
+2010-10-15 20:12 +0000 [r291942] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes peer's host port information being
+ lost on sip reload. (closes issue #18135) Reported by: lmadsen
+ Patches: crazy_ports_v2.diff uploaded by dvossel (license 671)
+ Tested by: lmadsen
+
+2010-10-15 19:50 +0000 [r291940] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * configs/gtalk.conf.sample, /: Merged revisions 291939 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400
+ (Fri, 15 Oct 2010) | 9 lines 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 16:39 +0000 [r291905] Terry Wilson <twilson@digium.com>
+
+ * res/res_jabber.c, /: Merged revisions 291904 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010)
+ | 7 lines 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.
+ ........
+
+2010-10-14 22:09 +0000 [r291827-291829] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c: Set TCLASS field of IPv6 header when sip qos
+ options are set. (closes issue #18099) Reported by: jamesnet
+ Patches: issues_18099_v2.diff uploaded by dvossel (license 671)
+ Tested by: dvossel, jamesnet
+
+ * channels/chan_gtalk.c: Safer xml parsing, treat all clients the
+ same, and better local candidate selection. The gtalk channel
+ driver was doing several unsafe operations in regards to how it
+ parsed incoming XML messages. I have cleaned that code up so it
+ should be much safer now. We now treat all clients types the
+ same. We have no reason to distinguish between GMAIL and GOOGLE
+ VOICE clients anymore because they all work the same way. I also
+ modified how the local ip is found. If no bindaddress is provided
+ in the config file, we attempt to determine the local ip we would
+ use to connect to google.com. If that fails, then we fall back to
+ the ast_find_ourip() function as a last resort. Using the new
+ method makes it much less likely that we would ever advertise a
+ local RTP candidate as a loopback address.
+
+2010-10-14 18:45 +0000 [r291791] Jeff Peeler <jpeeler@digium.com>
+
+ * main/stdtime/localtime.c: Add missing ifdefs for test framework
+ and new locale code. (closes issue #18137) Reported by: ovi
+ Patches: 18137_test_framework_ifdef.patch uploaded by wdoekes
+ (license 717) 18137_localelist_warning.patch uploaded by wdoekes
+ (license 717) Tested by: ovi
+
+2010-10-14 15:15 +0000 [r291758] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_gtalk.c, channels/chan_jingle.c,
+ include/asterisk/acl.h, channels/chan_sip.c,
+ channels/chan_h323.c, main/acl.c: Add the ability for
+ ast_find_ourip to return IPv4, IPv6 or both. While testing
+ chan_gtalk I noticed jabber was using my IPv6 address and not
+ IPv4. When using bindaddr=0.0.0.0 it is possible for
+ ast_find_ourip() to return both IPv6 and IPv4 results. Adding a
+ family parameter gives you the ablility to choose. Since
+ jabber/gtalk/h323 do not support IPv6, we should only return IPv4
+ results. Review: https://reviewboard.asterisk.org/r/973/
+
+2010-10-14 12:08 +0000 [r291725] Russell Bryant <russell@digium.com>
+
+ * doc/tex/secure-calls.tex: Fix a typo - s/seucre/secure/
+
+2010-10-13 23:45 +0000 [r291656] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Merged revisions 291655 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500
+ (Wed, 13 Oct 2010) | 27 lines 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 23:01 +0000 [r291581] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 291580 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291580 | twilson | 2010-10-13 15:58:43 -0700
+ (Wed, 13 Oct 2010) | 28 lines 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 22:46 +0000 [r291578] David Vossel <dvossel@digium.com>
+
+ * channels/chan_gtalk.c: More fixup for chan_gtalk. This patch
+ makes the xml parsing safer.
+
+2010-10-13 22:24 +0000 [r291575] Terry Wilson <twilson@digium.com>
+
+ * Makefile, static-http/mantest.html (added): Add a simple AMI
+ client web page This patch uses the XML docs to parse all of the
+ available AMI commands and allows you to enter the command name
+ and be presented with a form with the available fields. You can
+ then rapidly tab through the fields and submit the command and
+ view the response. It is much faster/easier than having to use
+ telnet for testing purposes.
+
+2010-10-13 20:21 +0000 [r291469-291541] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: The chan_dahdi faxdetect option only works
+ for the first FAX call. The chan_dahdi faxdetect option only
+ works for the first call. After that the option no longer works.
+ The struct dahdi_pvt.callprogress member is the encoded user
+ config setting for the callprogress and faxdetect config options.
+ Changing this value alters the configuration for all following
+ calls until the chan_dahdi.conf file is reloaded. * Fixed the
+ chan_dahdi ast_channel_setoption callback to not change the users
+ faxdetect config setting except for the current call. * Fixed the
+ chan_dahdi ast_channel_queryoption callback to read the active
+ DSP setting of the faxdetect option. * Made actually disable the
+ active faxdetect DSP setting for the current call on the analog
+ port. my_handle_dtmfup() is used for normal analog ports.
+ dahdi_handle_dtmfup() is the legacy code and is no longer used
+ unless in a radio mode. (closes issue #18116) Reported by:
+ seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett
+ (license 664) Review: https://reviewboard.asterisk.org/r/972/
+
+ * channels/chan_misdn.c: Merged revision 291504 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed,
+ 13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the
+ ast_channel. Must get the ast_channel lock before proceeding with
+ release_chan() and release_chan_early() to hold off ast_hangup()
+ from destroying the ast_channel. Missed this change for -r291468.
+ JIRA ABE-2598 JIRA SWP-2317 ..........
+
+ * channels/chan_misdn.c: Merge revision 291468 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed,
+ 13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN
+ call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE
+ --> RELEASE_COMPLETE * Add lock protection around channel list
+ for find/add/delete operations. * Protect misdn_hangup() from
+ release_chan() and vise versa using the release_lock. JIRA
+ ABE-2598 JIRA SWP-2317 ..........
+
+2010-10-13 15:46 +0000 [r291394] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 291393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291393 | russell | 2010-10-13 10:29:21 -0500
+ (Wed, 13 Oct 2010) | 13 lines 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 [r291284] Leif Madsen <lmadsen@digium.com>
+
+ * configs/phoneprov.conf.sample, /: Merged revisions 291280 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010)
+ | 7 lines 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 17:06 +0000 [r291265] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/acl.c: Merged revisions 291264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500
+ (Tue, 12 Oct 2010) | 9 lines 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 16:08 +0000 [r291230] Leif Madsen <lmadsen@digium.com>
+
+ * configs/manager.conf.sample, /: Merged revisions 291229 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010)
+ | 2 lines Add documention that mentions options are defined but
+ not used. (Issue #18101) ........
+
+2010-10-12 15:58 +0000 [r291192-291227] David Vossel <dvossel@digium.com>
+
+ * main/manager.c: Fixes manager.c crash. This issue was caused by
+ improper use of the mansession lock and manession_session lock.
+ These two structures are confusing to begin with so I'm not
+ surprised this occurred. I fixed this by consistently making sure
+ we use each of these locks only to protect the data in the
+ corresponding structure. We had mismatched usage of these locks
+ which resulted in no mutual exclusivity occurring at all. (closes
+ issue #17994) Reported by: vrban Patches:
+ mansession_locking_fix.diff uploaded by dvossel (license 671)
+ Tested by: vrban
+
+ * CHANGES: Update CHANGES to reflect new gtalk.conf options.
+
+ * channels/chan_gtalk.c, include/asterisk/stun.h,
+ configs/gtalk.conf.sample, res/res_stun_monitor.c: Gtalk
+ enhancements and general code cleanup. This patch includes
+ several chan_gtalk enhancements. Two new gtalk.conf options have
+ been added, externip and stunadd. Setting externip allows us to
+ manually specify what the external IP address is outside of a NAT
+ environment. Setting the stunaddr option to a valid stun server
+ allows for that external ip to be retrieved via a STUN server
+ automatically. This external IP is then advertised during call
+ setup as a possible candidate. I have also attempted to clean up
+ chan_gtalk's code so it meets our coding guidelines. During this
+ cleanup I noticed several things that need to be done in the code
+ and made a TODO section at the top of the file.
+
+2010-10-11 18:51 +0000 [r291075-291113] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Move declaration closer to where now used.
+
+ * /, channels/chan_sip.c: Merged revisions 291110-291111 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500
+ (Mon, 11 Oct 2010) | 9 lines 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(). ........ ................ r291111 | rmudgett
+ | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit
+ from handle_request_do() consistent. ................
+
+ * main/cli.c, /: Merged revisions 291073 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010)
+ | 15 lines 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. ........
+
+2010-10-09 23:25 +0000 [r291038] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Add missing
+ option to set calls to be logged in GMT/UTC.
+
+2010-10-09 15:00 +0000 [r291005-291037] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c: small correction for verbose
+ print h.323 packets
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling
+ options per user and peer. Added options for faststart/h.245
+ tunneling per user/peer, properly handle these and global
+ options, correction of handling fs/tunneling fields in signalling
+ responses (issue #17972) Reported by: salecha Patches:
+ fs-tunnel-per-point-3.patch uploaded by may213 (license 454)
+ Tested by: may213, salecha
+
+2010-10-08 20:44 +0000 [r290973] David Vossel <dvossel@digium.com>
+
+ * channels/chan_gtalk.c: Make outbound Google Voice calls. This
+ patch allows for outbound Google Voice calls to be dialed from
+ Asterisk using chan_gtalk. Below is an example dialstring. exten
+ -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In
+ this example, 'asterisk' is the jabber.conf profile configured to
+ connect to your gmail account. In order to receive Google Voice
+ calls make sure to enable 'allowguest=yes' in gtalk.conf.
+
+2010-10-08 15:49 +0000 [r290937-290938] Erin Spiceland <erin@thespicelands.com>
+
+ * addons/res_config_mysql.c: Parentheses around assignment used as
+ truth value, introduced in r290937.
+
+ * addons/res_config_mysql.c, addons/app_mysql.c,
+ configs/res_config_mysql.conf.sample: Add option to
+ res_config_mysql and app_mysql to specify a character set that
+ MySQL should use. (closes issue 17948) Reported by qmax.
+
+2010-10-08 02:56 +0000 [r290864] Jeff Peeler <jpeeler@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 290863 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500
+ (Thu, 07 Oct 2010) | 16 lines 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 22:38 +0000 [r290828-290829] David Vossel <dvossel@digium.com>
+
+ * channels/chan_gtalk.c: Add Philippe Sultan to chan_gtalk author
+ list. Philippe has made some notable contributions to the gtalk
+ channel driver. His name deserves to be listed amoung the authors
+ of that file. Thanks Philippe!
+
+ * channels/chan_gtalk.c: Outbound gtalk calls now work correctly.
+ There was a problem with how the candidates were being built on
+ an outbound call. This patch fixes that.
+
+2010-10-07 20:58 +0000 [r290752] Jason Parker <jparker@digium.com>
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 290751 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290751 | qwell | 2010-10-07 15:57:14 -0500
+ (Thu, 07 Oct 2010) | 16 lines 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 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc3 Released.
+
+2010-10-07 11:00 +0000 [r290713] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 290712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010)
+ | 4 lines Don't crash when Set() is called without a value.
+ Review: https://reviewboard.asterisk.org/r/949/ ........
+
+2010-10-06 21:22 +0000 [r290648-290674] David Vossel <dvossel@digium.com>
+
+ * channels/chan_gtalk.c: Fixes commented out code to use #if 0
+ instead. Thanks to rmudgett for catching this!
+
+ * channels/chan_gtalk.c: Fixes gtalk outbound DTMF to work
+ properly. Outbound DTMF with gtalk needs to be done within the
+ RTP stream. I discovered this after investigating a packet
+ capture from the gmail client. Instead of performing jingle
+ signaling DTMF, the gtalk servers expect all DTMF to arrive on
+ the RTP stream using RFC2833 way of doing things. Chan_gtalk also
+ had an issue with negotiating RTP payload type 106 for the
+ telephony-event and then sending DTMF as payload 101. This has
+ been resolved by always negotiating 101 as the payload type like
+ we do everywhere else. With this patch, incoming google voice
+ calls forwarded to Asterisk via gtalk work.
+
+2010-10-06 18:50 +0000 [r290614] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c: Merged revision 290613 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed,
+ 06 Oct 2010) | 5 lines Eliminate a redundant test for
+ AST_CONTROL_REDIRECTING. Eliminate redundant test for
+ AST_CONTROL_REDIRECTING that prevents running the redirecting
+ interception macro if it is defined. ..........
+
+2010-10-06 13:49 +0000 [r290576] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/file.c: Merged revisions 290575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010)
+ | 8 lines 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 ........
+
+2010-10-06 04:35 +0000 [r290542] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c: Don't try to send RTP when remote_address
+ is null It is possible for ast_rtp_stop() to be called which will
+ clear the remote address and cause the sendto to fail and spam
+ warnings. Don't send in this case.
+
+2010-10-05 22:23 +0000 [r290479-290506] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Fixes uninitialized memory problem in 'iax2
+ set debug peer' option.
+
+ * include/asterisk/jingle.h, channels/chan_gtalk.c,
+ res/res_jabber.c, include/asterisk/jabber.h: Fixes chan_gtalk to
+ work with gmail client This patch was written by Philippe Sultan
+ (phsultan). Thanks for keeping this up to date!
+
+2010-10-05 20:23 +0000 [r290408] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_jabber.c, /: Merged revisions 290396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500
+ (Tue, 05 Oct 2010) | 15 lines 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 20:09 +0000 [r290376-290378] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Resolves dnsmgr memory corruption in
+ chan_iax2. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ afried, russell, dvossel Review:
+ https://reviewboard.asterisk.org/r/965/
+
+ * /, apps/app_directed_pickup.c: Merged revisions 290375 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010)
+ | 10 lines 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 14:15 +0000 [r290066-290289] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Restore run directory for OS X, as well
+ as standardizing some other paths to Mac OS X.
+
+ * pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
+ 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:
+ Merged revisions 290254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010)
+ | 11 lines 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 290201 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500
+ (Mon, 04 Oct 2010) | 9 lines 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 290101 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500
+ (Sun, 03 Oct 2010) | 9 lines 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. ........
+ ................
+
+ * pbx/pbx_spool.c: Get notification only when file is closed, not
+ when created. (closes issue #17924) Reported by: mkeuter Patches:
+ asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946)
+ Tested by: abelbeck
+
+2010-10-02 17:57 +0000 [r290026] Kevin P. Fleming <kpfleming@digium.com>
+
+ * contrib/scripts/get_mp3_source.sh: Allow users to pass additional
+ arguments to the Subversion command that obtains the MP-3 source
+ code. (reported on IRC by jmls)
+
+2010-10-02 08:56 +0000 [r289951] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 289950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör,
+ 02 Okt 2010) | 9 lines 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 04:46 +0000 [r289875] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 289874 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500
+ (Fri, 01 Oct 2010) | 15 lines 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 ........
+ ................
+
+2010-10-02 02:43 +0000 [r289840] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, /, channels/chan_sip.c: Merged revisions
+ 289798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500
+ (Fri, 01 Oct 2010) | 22 lines 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 17:19 +0000 [r289718] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
+ 289704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400
+ (Fri, 01 Oct 2010) | 13 lines 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 16:22 +0000 [r289701] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500
+ (Fri, 01 Oct 2010) | 21 lines 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] Stefan Schmidt <sst@sil.at>
+
+ * 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 20:23 +0000 [r289581] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c: Solaris fixes.
+
+2010-09-30 19:53 +0000 [r289554] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289553 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep
+ 2010) | 4 lines Properly handle channel allocation failures duing
+ invites with replaces. ABE-2588 ........
+
+2010-09-30 19:28 +0000 [r289549] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Merged revision 289547 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu,
+ 30 Sep 2010) | 10 lines In chan_misdn, the
+ DivertingLegInformation2 DivertingNr is garbage when the number
+ is restricted. The same thing happens with
+ DivertingLegInformation1 DivertedTo number. The
+ misdn_PresentedNumberUnscreened_extract() extracted the
+ Unscreened PartyNumber field unconditionally. It now checks the
+ presented number unscreened type to see if the PartyNumber was
+ even present. JIRA ABE-2595 ..........
+
+2010-09-30 17:50 +0000 [r289543] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/localtime.h, main/stdtime/localtime.c,
+ tests/test_time.c, tests/test_utils.c, res/res_agi.c: More
+ Solaris compatibility fixes
+
+2010-09-30 15:39 +0000 [r289426] Russell Bryant <russell@digium.com>
+
+ * apps/app_sms.c, /: Merged revisions 289425 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289425 | russell | 2010-09-30 10:37:29 -0500
+ (Thu, 30 Sep 2010) | 15 lines 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 21:12 +0000 [r289340] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 289339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289339 | qwell | 2010-09-29 16:03:47 -0500
+ (Wed, 29 Sep 2010) | 15 lines 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 20:27 +0000 [r289336] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample, /: Merged revisions 289334 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 Sep 2010)
+ | 1 line Update sample documentation to note md5secret
+ requirements. ........
+
+2010-09-29 20:20 +0000 [r289333] Russell Bryant <russell@digium.com>
+
+ * res/res_config_ldap.c, /: Merged revisions 289332 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29
+ Sep 2010) | 4 lines 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 17:53 +0000 [r289268-289300] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/res_fax.conf.sample: Add 'ecm' to the sample fax config
+ file
+
+ * main/channel.c: Update the CDR record when
+ ast_channel_set_caller_event() is called (related to issue
+ #17569) Reported by: tbelder
+
+2010-09-29 16:16 +0000 [r289253] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Make development error message indicate which
+ channel.
+
+2010-09-29 15:04 +0000 [r289179] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 289178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500
+ (Wed, 29 Sep 2010) | 15 lines 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 18:18 +0000 [r289104] Tilghman Lesher <tlesher@digium.com>
+
+ * makeopts.in, apps/app_voicemail.c, Makefile, tests/test_time.c,
+ configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, tests/test_utils.c,
+ configure.ac: Solaris compatibility fixes Review:
+ https://reviewboard.asterisk.org/r/942/
+
+2010-09-28 18:18 +0000 [r289099] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 289095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289095 | bbryant | 2010-09-28 14:14:19 -0400
+ (Tue, 28 Sep 2010) | 21 lines 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/ ........ ................
+
+2010-09-28 01:04 +0000 [r289054-289057] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Avoid deadlock processing incoming AOC-E
+ messages. Deadlock avoidance for the owner channel was not done
+ when processing incoming AOC-E messages.
+
+ * channels/sig_pri.c: Revert stuff not ready for commit in
+ -r289054.
+
+ * channels/sig_pri.c, channels/chan_sip.c: Break up long
+ ast_manager_event_multichan() event lines.
+
+2010-09-27 18:37 +0000 [r288961] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Still build SIP, even if res_crypto cannot
+ be built (use, not depend). (closes issue #18062) Reported by: a
+ user on the mailing list
+
+2010-09-27 13:03 +0000 [r288925-288927] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: Fix some documentation typos and spelling errors.
+
+ * res/res_agi.c: Fix a documentation spelling error.
+
+2010-09-24 17:58 +0000 [r288821-288852] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Append Retry-After header on 500 error
+ response to Re-INVITE according to RFC3261 section 14.2. ABE-2301
+
+ * channels/chan_sip.c: Inspect Require header on BYE transaction
+ according to RFC3261 section 8.2.2.3. ABE-2293
+
+2010-09-24 16:02 +0000 [r288748] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 288747 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288747 | twilson | 2010-09-24 08:37:39 -0700
+ (Fri, 24 Sep 2010) | 12 lines 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 13:54 +0000 [r288606-288713] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 288712 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24
+ Sep 2010) | 5 lines Solaris won't printf a NULL. (closes issue
+ #18041) Reported by: asgaroth ........
+
+ * main/asterisk.exports.in: Export timersub for platforms which do
+ not have it
+
+ * include/asterisk/channel.h, cdr/cdr_pgsql.c, /, configure,
+ include/asterisk/autoconfig.h.in, include/asterisk/compat.h,
+ main/strcompat.c, configure.ac: Merged revisions 288637 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500
+ (Thu, 23 Sep 2010) | 9 lines 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 ........
+ ................
+
+ * CHANGES: Add note about the checkhangup option of ${CHANNEL()}
+
+2010-09-23 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc2 Released.
+
+2010-09-23 18:05 +0000 [r288507-288572] Terry Wilson <twilson@digium.com>
+
+ * main/manager.c: Make AMI honor enabled=no (closes issue #18040)
+ Reported by: twilson Review:
+ https://reviewboard.asterisk.org/r/938/
+
+ * channels/chan_local.c, /: Merged revisions 288500 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288500 | twilson | 2010-09-22 16:10:09 -0700
+ (Wed, 22 Sep 2010) | 15 lines 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 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc1 Released.
+
+2010-09-22 17:49 +0000 [r288345-288418] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288417 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500
+ (Wed, 22 Sep 2010) | 11 lines 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 288344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500
+ (Wed, 22 Sep 2010) | 9 lines 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 16:45 +0000 [r288341] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 288340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288340 | russell | 2010-09-22 11:44:13 -0500
+ (Wed, 22 Sep 2010) | 18 lines 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 15:14 +0000 [r288268] Tilghman Lesher <tlesher@digium.com>
+
+ * UPGRADE.txt, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /:
+ Merged revisions 288267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500
+ (Wed, 22 Sep 2010) | 23 lines 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 00:06 +0000 [r288194] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288193 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500
+ (Tue, 21 Sep 2010) | 33 lines 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 22:57 +0000 [r288159] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500
+ (Tue, 21 Sep 2010) | 22 lines 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 22:26 +0000 [r288157] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288147 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue,
+ 21 Sep 2010) | 9 lines 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 21:03 +0000 [r288079-288082] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/tex/partymanip.tex: Add note in party manipulation chapter on
+ interception macros.
+
+ * apps/app_queue.c, apps/app_dial.c: Simplify locking code for
+ REDIRECTING interception macro when forwarding a call. Simplified
+ the locking code by using a local copy of the redirecting party
+ information in app_dial.c:do_forward() and
+ app_queue.c:wait_for_answer() for launching the REDIRECTING
+ interception macro when a call is forwarded. Reduced the lock
+ time of the 'o->chan' and 'in' channels.
+
+ * main/channel.c: Protect channel access in CONNECTED_LINE and
+ REDIRECTING interception macro launch code.
+
+2010-09-21 19:48 +0000 [r288007] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 288006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400
+ (Tue, 21 Sep 2010) | 14 lines 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 19:08 +0000 [r287935] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 287934 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500
+ (Tue, 21 Sep 2010) | 9 lines 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. ........ ................
+
+2010-09-21 19:02 +0000 [r287931] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Revert change in favor of a more targeted fix
+
+2010-09-21 18:32 +0000 [r287929] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Send a "415 Unsupported Media Type" after
+ failure to process sdp due to unknown Content-Encoding header.
+ ABE-2258
+
+2010-09-21 15:53 +0000 [r287897] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Cut-n-paste error in builtin_blindtransfer().
+
+2010-09-21 15:43 +0000 [r287895] Russell Bryant <russell@digium.com>
+
+ * res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
+ main/acl.c: Don't use ast_strdupa() from within the arguments to
+ a function. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ russell Review: https://reviewboard.asterisk.org/r/927/
+
+2010-09-21 15:24 +0000 [r287893] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Anonymous callerid needs a "sip:" uri
+ prefix. (closes issue #17981) Reported by: avalentin Patches:
+ sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
+ (plus an additional fix by me) Tested by: avalentin
+
+2010-09-21 13:41 +0000 [r287863] Russell Bryant <russell@digium.com>
+
+ * main/logger.c: Fix a regression in verbose logger processing.
+
+2010-09-21 04:37 +0000 [r287833] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Don't generate connected line buffer twice for
+ comparison
+
+2010-09-21 00:00 +0000 [r287760] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 287759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400
+ (Mon, 20 Sep 2010) | 23 lines 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 23:51 +0000 [r287757] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Avoid infinite loop with certain local channel
+ connected line updates Compare connected line data before sending
+ a connected line indication to avoid possible loops. Review:
+ https://reviewboard.asterisk.org/r/932/
+
+2010-09-20 23:20 +0000 [r287701] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/channel.c, /: Merged revisions 287685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep
+ 2010) | 18 lines 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 23:14 +0000 [r287683] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: The inalarm flag was not set in sig_analog
+ struct if the port is initially in alarm. Fixed initial inalarm
+ value for sig_analog ports. Along with -r261007, this gets the
+ inalarm flag in sync with chan_dahdi for sig_analog ports.
+ (closes issue #16983)
+
+2010-09-20 22:21 +0000 [r287661] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/channel.c: ast_do_masquerade. Keep channels ao2_container
+ locked while unlink and linking channels. Previously, Masquerade
+ would unlock 'original' and 'clonechan' and allow another masq
+ thread to run. End result would be corrupted memory, and the
+ frequent report 'Bad Magic Number'. (closes issue #17801,#17710)
+ Reported by: notthematrix Patches: Based on bug17801.diff1.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/928
+
+2010-09-20 22:09 +0000 [r287645-287647] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/channel.h, CHANGES, include/asterisk/framehook.h
+ (added), main/channel.c, main/framehook.c (added),
+ funcs/func_frame_trace.c (added): Addition of the FrameHook API
+ (AKA AwesomeHooks) So far all our tools for viewing and
+ manipulating media streams within Asterisk have been entirely
+ focused on audio. That made sense then, but is not scalable now.
+ The FrameHook API lets us tap into and manipulate _ANY_ type of
+ media or signaling passed on a channel present today or in the
+ future. This tool is a step in the direction of expanding
+ Asterisk's boundaries and will help generate some rather
+ interesting applications in the future. In addition to the
+ FrameHook API, a simple dialplan function exercising the api has
+ been included as well. This function is called FRAME_TRACE().
+ FRAME_TRACE() allows for the internal ast_frames read and written
+ to a channel to be output. Filters can be placed on this function
+ to debug only certain types of frames. This function could be
+ thought of as an internal way of doing ast_frame packet captures.
+ Review: https://reviewboard.asterisk.org/r/925/
+
+ * channels/chan_sip.c: Fixes issue with registrations not working
+ properly with pedantic=yes. (closes issue #18017) Reported by:
+ schmidts Patches: issues_18017_v1.diff uploaded by dvossel
+ (license 671) Tested by: schmidts
+
+2010-09-20 21:29 +0000 [r287643] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_skinny.c: Merged revisions 287642 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep
+ 2010) | 8 lines 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 ........
+
+2010-09-20 21:19 +0000 [r287639] Brett Bryant <bbryant@digium.com>
+
+ * main/logger.c: Fixes an error with the logger that caused verbose
+ messages to be spammed to the screen if syslog was configured in
+ logger.conf (closes issue #17974) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/915/
+
+2010-09-20 15:57 +0000 [r287559] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500
+ (Mon, 20 Sep 2010) | 14 lines 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 16:09 +0000 [r287471] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 287470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön,
+ 19 Sep 2010) | 14 lines 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 [r287388] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 287387 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500
+ (Fri, 17 Sep 2010) | 14 lines 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 13:37 +0000 [r287309] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287308 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500
+ (Fri, 17 Sep 2010) | 12 lines 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 ........ ................
+
+2010-09-17 08:44 +0000 [r287269-287271] Jan Kalab <pitlicek@gmail.com>
+
+ * res/res_calendar_ews.c: Events are visible after they were
+ removed from EWS calendar Because we must merge calendar even
+ when it's empty. (closes issue #17786)
+
+ * res/res_calendar_ews.c: Asterisk crashing because of double free
+ when EWS request fails The free is done later in code. I think
+ ast_free() should have built in checks for double free. (closes
+ issue #17782)
+
+ * res/res_calendar_caldav.c, res/res_calendar_ews.c,
+ res/res_calendar_exchange.c, res/res_calendar_icalendar.c:
+ Support for HTTP redirects in calendar's URL libneon does not
+ support HTTP redirects (3xx responses) by default. You must tell
+ it to follow them. Also, another little unsigned int fix. (closes
+ issue #17776) Review: https://reviewboard.asterisk.org/r/921/
+
+2010-09-16 22:04 +0000 [r287195] Jason Parker <jparker@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Don't fail when running the
+ Debian init script directly (as one would normally do). readlink
+ apparently returns 1 when the arg isn't a symlink, which caused
+ the script to exit. (closes issue #17910) Reported by: wurstsalat
+
+2010-09-16 21:57 +0000 [r287193] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Set
+ the default for "autofill" and "shared_lastcall" to "yes" in
+ queues.conf. Review: https://reviewboard.asterisk.org/r/922/
+
+2010-09-16 20:07 +0000 [r287116-287120] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287119 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500
+ (Thu, 16 Sep 2010) | 15 lines 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 287115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500
+ (Thu, 16 Sep 2010) | 15 lines 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 22:17 +0000 [r287056] Terry Wilson <twilson@digium.com>
+
+ * res/res_srtp.c: Don't hang up a call on an SRTP unprotect failure
+ Also make it more obvious when there is an issue en/decrypting.
+ (closes issue #17563) Reported by: Alexcr Patches:
+ res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by:
+ twilson
+
+2010-09-15 20:58 +0000 [r287020] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: fix uninintialized variable
+
+2010-09-15 20:53 +0000 [r287017] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
+ revision 287014 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed,
+ 15 Sep 2010) | 58 lines The handling of call transfer signaling
+ for mISDN PTMP is not fully implemented. The handling of call
+ transfer signaling for mISDN PTMP is not fully implemented. The
+ signaling of number updates with ISDN/DSS1 ECT supplementary
+ services (ETS 300 369-1) comes along with a notification
+ indicator IE and redirection number IE for PTMP. The
+ implementation in the current Asterisk mISDN channel
+ unfortunately can handle these information elements only in a
+ NOTIFY message. These information elements are also signaled in a
+ FACILTY message with a RequestSubaddress facility, when the
+ subscriber is already in the active state (see 9.2.4 and 9.2.5 of
+ ETS 300 369-1). ********** abe_2526_ast.patch * Added support to
+ handle the notification indicator IE and redirection number IE
+ with the RequestSubaddress facility. * Made
+ misdn_update_connected_line() send a NOTIFY message if Asterisk
+ originated the call and it is not connected yet. * Made
+ misdn_update_connected_line() send a FACILITY message if the call
+ is already connected. This patch requires the presence of the
+ associated mISDN patches to compile. I had to enhance mISDN to
+ allow the notification indicator IE and the redirection number IE
+ to be used with a FACILITY message. Earlier versions of the
+ Digium enhanced mISDN are no longer going to work. **********
+ abe_2526_misdn.patch * Made an incoming FACILITY message allow
+ the presence of the notification indicator IE and the redirection
+ number IE. ********** abe_2526_misdnuser_v3.patch * Added support
+ to send and receive a FACILITY message with the notification
+ indicator IE and the redirection number IE. * Added the ability
+ to send a NOTIFY message in PTMP/NT mode to all responding
+ subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches:
+ abe_2526_ast.patch uploaded by rmudgett (license 664)
+ abe_2526_misdn.patch uploaded by rmudgett (license 664)
+ abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526
+ ..........
+
+2010-09-15 20:32 +0000 [r286931-287015] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 286998 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500
+ (Wed, 15 Sep 2010) | 14 lines 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 ........
+ ................
+
+ * CHANGES, channels/chan_iax2.c, channels/sip/include/sip.h,
+ configs/features.conf.sample, channels/chan_mgcp.c,
+ include/asterisk/features.h, channels/chan_dahdi.c,
+ channels/sig_analog.c, channels/chan_sip.c, main/features.c: Add
+ parking extension for non-default parking lots. This is a new
+ feature that allows for parking to custom parking lots to be
+ accessed directly, rather than with channel variables or by
+ changing the default parking lot. The extension is set with the
+ parkext option just as the default parking lot is done. Also, the
+ manager action has been updated to optionally allow a specified
+ parking lot. (closes issue #14882) Reported by: vmikhnevych
+ Patches: patch_14882.txt uploaded by mnick (license 874) modified
+ by me Review: https://reviewboard.asterisk.org/r/884/
+
+2010-09-15 18:29 +0000 [r286904-286905] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_analog.c: Simplify some code in sig_analog.
+
+ * channels/sig_analog.c: Unable to originate calls using E&M over
+ T1. When originating a call from Unit Under Test to Reference
+ Unit using E&M RBS signaling mode, I get the following warning
+ message: "Ring/Off-hook in strange state 3 on channel 1". Fixed
+ the sig_analog outgoing flag. It was never set when sig_analog
+ was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408
+
+2010-09-15 13:05 +0000 [r286868] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Set tohost to the domain specified in the
+ configuration file instead of the IP address of the host we are
+ calling. This fixes a regression introduced in r274783. (closes
+ issue #17960) Reported by: adriavidal Patches:
+ sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested
+ by: mich, mnicholson, adriavidal (closes issue #17676) Reported
+ by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson
+
+2010-09-14 21:57 +0000 [r286834] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Sets subscribed type for outgoing MWI
+ subscriptions so correct Event header is used.
+
+2010-09-14 19:28 +0000 [r286682-286758] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286757 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500
+ (Tue, 14 Sep 2010) | 20 lines 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 286681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500
+ (Tue, 14 Sep 2010) | 14 lines 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 15:30 +0000 [r286647] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Corrected
+ documented CONNECTED_LINE and REDIRECTING party manipulation
+ macro names.
+
+2010-09-14 06:55 +0000 [r286617] Jan Kalab <pitlicek@gmail.com>
+
+ * res/res_calendar_ews.c: Merging events for Exchange web service
+ doesn't work as expected, resulting in only one event in calendar
+ The solution is to use "global" counter of events, since we do
+ new requests for every event and calendar sync after every
+ request. So now we do sync only after last request. (closes issue
+ #17877) Review: https://reviewboard.asterisk.org/r/916/
+
+2010-09-14 05:07 +0000 [r286528-286588] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/realtime/mysql/voicemail_data.sql (added), /,
+ contrib/realtime/mysql/voicemail_messages.sql (added): Merged
+ revisions 286587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010)
+ | 2 lines Add documentation on missing backend tables for
+ Voicemail ........
+
+ * /, main/features.c: Merged revisions 286557 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010)
+ | 2 lines C precedence got me ........
+
+ * /, main/features.c: Merged revisions 286527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010)
+ | 2 lines Refactor conversion to ast_poll() to fix callparking
+ regression. ........
+
+2010-09-13 19:40 +0000 [r286457] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286456 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) |
+ 5 lines Remove "Internal IP" from sip show settings, as it's not
+ at all useful to display. (closes issue #17840) Reported by: oej
+ ........
+
+2010-09-13 15:52 +0000 [r286426] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/chan_dahdi.conf.sample: Update chan_dahdi.conf.sample to
+ reflect new libpri T309 default value.
+
+2010-09-11 17:09 +0000 [r286270] Olle Johansson <oej@edvina.net>
+
+ * /, main/file.c: Merged revisions 286268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör,
+ 11 Sep 2010) | 11 lines 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 22:04 +0000 [r286189] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/channel.h, include/asterisk/pbx.h,
+ include/asterisk/frame.h, channels/chan_local.c,
+ funcs/func_channel.c: Merged revisions 286115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286115 | twilson | 2010-09-10 15:35:25 -0500
+ (Fri, 10 Sep 2010) | 23 lines 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 21:11 +0000 [r286120] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 286117 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400
+ (Fri, 10 Sep 2010) | 11 lines 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 20:55 +0000 [r286118] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, /: Merged revisions 286116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500
+ (Fri, 10 Sep 2010) | 18 lines 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 20:31 +0000 [r286112] Russell Bryant <russell@digium.com>
+
+ * main/db.c: Rate limit calls to fsync() to 1 per second after
+ astdb updates. Astdb was determined to be one of the most
+ significant bottlenecks in SIP registration processing. This
+ patch improved the speed of an astdb load test by 50000% (yes,
+ Fifty-Thousand Percent). On this particular load test setup, this
+ doubled the number of SIP registrations the server could handle.
+ Review: https://reviewboard.asterisk.org/r/825/
+
+2010-09-10 18:31 +0000 [r286025] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Merged revisions 286024 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286024 | tilghman | 2010-09-10 13:30:21 -0500
+ (Fri, 10 Sep 2010) | 9 lines 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 ........ ................
+
+2010-09-10 13:13 +0000 [r285992] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt, CHANGES: Added missing documentation for
+ ExternalIVR feature added in January 2010
+
+2010-09-10 05:32 +0000 [r285931-285962] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/select.h, /: Merged revisions 285961 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010)
+ | 6 lines 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
+ 285930 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500
+ (Thu, 09 Sep 2010) | 14 lines 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 22:52 +0000 [r285819] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, codecs/gsm/Makefile: Merged revisions 285818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285818 | pabelanger | 2010-09-09 18:49:19 -0400
+ (Thu, 09 Sep 2010) | 15 lines 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 20:11 +0000 [r285745] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /: Merged revisions 285744 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285744 | qwell | 2010-09-09 15:09:23 -0500
+ (Thu, 09 Sep 2010) | 16 lines 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 18:51 +0000 [r285640-285711] Brett Bryant <bbryant@digium.com>
+
+ * main/pbx.c, /: Merged revisions 285710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010)
+ | 8 lines 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 285639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285639 | bbryant | 2010-09-09 13:22:25 -0400
+ (Thu, 09 Sep 2010) | 14 lines 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 22:14 +0000 [r285564-285568] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 285567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285567 | dvossel | 2010-09-08 17:11:28 -0500
+ (Wed, 08 Sep 2010) | 9 lines 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: Merged revisions 285563 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010)
+ | 54 lines 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 20:58 +0000 [r285533] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 285532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010)
+ | 8 lines 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 20:43 +0000 [r285527-285530] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c, /, include/asterisk/astobj2.h: Merged
+ revisions 285529 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) |
+ 1 line Follow coding guidelines in moh rescan fix. Also fix the
+ documentation that got me in trouble. ........
+
+ * res/res_musiconhold.c, /: Merged revisions 285526 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep
+ 2010) | 8 lines 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 ........
+
+2010-09-08 07:14 +0000 [r285484] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_channel.c: Documentation only
+
+2010-09-07 22:22 +0000 [r285455] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Don't automatically add domains for wildcard
+ bindaddrs. (closes issue #17832) Reported by: oej Patches:
+ 17832-wildcard.diff uploaded by qwell (license 4) Tested by:
+ qwell
+
+2010-09-07 21:20 +0000 [r285373-285386] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c: Don't notify on attribute changes, and change
+ how the queuing mechanism works. Fixes call spools in 1.8.
+ (closes issue #17337) Reported by: loloski Patches:
+ 20100827__issue17337.diff.txt uploaded by tilghman (license 14)
+ (closes issue #17924) Reported by: mkeuter Tested by: mkeuter
+
+ * funcs/func_channel.c: Add CHANNEL(checkhangup) to check whether a
+ channel is in the process of being hanged up. (closes issue
+ #17652) Reported by: kobaz Patches: func_channel.patch uploaded
+ by kobaz (license 834)
+
+2010-09-07 21:08 +0000 [r285371] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix cut-n-paste error.
+
+2010-09-07 20:58 +0000 [r285369] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Add note to 'sip show settings' regarding
+ dual-stack support, and a :: bindaddress. (closes issue #17831)
+ Reported by: oej Patches: 17831-v6wildcardbind.diff uploaded by
+ qwell (license 4)
+
+2010-09-07 20:56 +0000 [r285268-285367] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 285366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285366 | tilghman | 2010-09-07 15:31:41 -0500
+ (Tue, 07 Sep 2010) | 16 lines 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 ........
+ ................
+
+ * include/asterisk/compiler.h, addons/ooh323c/src/ooSocket.h: Fix
+ build on FreeBSD 8.0, take 2.
+
+ * main/poll.c, /: Merged revisions 285267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285267 | tilghman | 2010-09-07 14:07:17 -0500
+ (Tue, 07 Sep 2010) | 11 lines 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 17:54 +0000 [r285197] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 285196 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285196 | bbryant | 2010-09-07 13:49:07 -0400
+ (Tue, 07 Sep 2010) | 17 lines 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] ........
+ ................
+
+2010-09-07 17:47 +0000 [r285195] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Merged revisions 285193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 285192 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........
+ r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010)
+ | 8 lines COLP/CONP and chan_misdn missing update chan_misdn does
+ not update the caller id of the channel if a new connected number
+ or ECT-INFORM (w/ new peer number on call transfer) is received.
+ JIRA ABE-2502 JIRA SWP-2058 ........ ........
+
+2010-09-06 20:10 +0000 [r285161-285162] Russell Bryant <russell@digium.com>
+
+ * configure: regenerate configure script.
+
+ * include/asterisk/autoconfig.h.in, configure.ac: Fix libsrtp -fPIC
+ check for when non-standard prefix is used. Thanks to loompek in
+ #asterisk for reporting the issue and testing this patch.
+
+2010-09-06 06:56 +0000 [r285090] Tilghman Lesher <tlesher@digium.com>
+
+ * BSDmakefile (added), makeopts.in, /: Merged revisions 285089 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285089 | tilghman | 2010-09-06 01:55:17 -0500
+ (Mon, 06 Sep 2010) | 9 lines 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.
+ ........ ................
+
+2010-09-04 18:08 +0000 [r285057] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/cli.h: Add a C++ compatible version of
+ AST_CLI_DEFINE().
+
+2010-09-03 23:19 +0000 [r285017] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Call correct lock function as transferer is
+ a sip_pvt not a channel Both functions are #defined to ao2_lock,
+ but still...
+
+2010-09-03 22:21 +0000 [r285006] David Vossel <dvossel@digium.com>
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/chan_sip.c: Disables auth_options_request option by
+ default. The auth_options_request option was created to do
+ authentication on OPTIONS request just like INVITES are done.
+ Since it has been noted that some endpoints use OPTIONS requests
+ as a way of qualifying a peer and that a 401 authentication
+ response could result in interoperability issues, this option has
+ been disabled by default.
+
+2010-09-03 18:19 +0000 [r284967] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 284958 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03
+ Sep 2010) | 8 lines 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 18:03 +0000 [r284950-284952] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: During OPTIONS authentication, the authpeer
+ does not need to be returned for any reason.
+
+ * configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h,
+ channels/chan_sip.c: authenticate OPTIONS requests just like we
+ would an INVITE OPTIONS requests should be treated the same as an
+ INVITE This includes authentication. This patch adds the ability
+ for incoming out of dialog OPTION requests to be authenticated
+ before providing a response indicating whether an extension is
+ available or not. The authentication routine works the exact same
+ way as it does for incoming INVITEs. This means that if a peer
+ has 'insecure=invite' in their peer definition, the same will be
+ true for the processing of the OPTIONS request. Review:
+ https://reviewboard.asterisk.org/r/881/
+
+2010-09-03 16:28 +0000 [r284921] Terry Wilson <twilson@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 284897 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284897 | twilson | 2010-09-03 11:20:45 -0500
+ (Fri, 03 Sep 2010) | 12 lines 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. ........ ................
+
+2010-09-03 13:07 +0000 [r284849-284852] Jan Kalab <pitlicek@gmail.com>
+
+ * res/res_calendar_ews.c: Calendar categories and priorities:
+ strdupa() fix
+
+ * res/res_calendar_ews.c: Fix for calendar categories and
+ priorities according to ISO C90
+
+ * res/res_calendar_caldav.c, include/asterisk/calendar.h,
+ res/res_calendar_ews.c, res/res_calendar.c,
+ res/res_calendar_icalendar.c: Support for calendar events
+ priorities and categories Review 880
+
+2010-09-02 21:04 +0000 [r284781] Brett Bryant <bbryant@digium.com>
+
+ * main/manager.c, /: Merged revisions 284778 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284778 | bbryant | 2010-09-02 16:54:33 -0400
+ (Thu, 02 Sep 2010) | 14 lines 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 21:02 +0000 [r284779-284780] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Simplified pri_dchannel() poll timeout
+ duration code.
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
+ Made output libpri event names if pri debugging is enabled when
+ sig_pri processes them. * Simplified CLI "pri debug xx span xx"
+ command code and removed redundant debugging enabled messages. *
+ Made CLI "pri debug xx span xx" command only close the debugging
+ log file if it was opened.
+
+2010-09-02 16:56 +0000 [r284705] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284704 | dvossel | 2010-09-02 11:48:51 -0500
+ (Thu, 02 Sep 2010) | 13 lines 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 16:43 +0000 [r284701] Jason Parker <jparker@digium.com>
+
+ * formats/format_wav.c: Add slin16 support for format_wav (new
+ wav16 file extension) (closes issue #15029) Reported by: andrew
+ Patches: wav16.patch uploaded by andrew (license 240) Tested by:
+ qwell, andrew
+
+2010-09-02 16:34 +0000 [r284698] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/tex/channelvariables.tex, doc/tex/partymanip.tex (added),
+ doc/tex/asterisk.tex: Added documentation for CONNECTEDLINE and
+ REDIRECTING functions. (closes issue #17808) Reported by: jtodd
+ Review: https://reviewboard.asterisk.org/r/875/
+
+2010-09-02 16:27 +0000 [r284597-284696] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/ooh323c/src/oochannels.c: Fixing build
+
+ * channels/chan_usbradio.c, /: Merged revisions 284665 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02
+ Sep 2010) | 2 lines Fixing build. ........
+
+ * apps/app_queue.c, /: Merged revisions 284631 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010)
+ | 7 lines 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)
+ ........
+
+ * channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c,
+ apps/app_followme.c, main/loader.c, apps/app_speech_utils.c,
+ pbx/pbx_loopback.c, channels/chan_dahdi.c, funcs/func_aes.c,
+ include/asterisk/module.h, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
+ apps/app_stack.c, channels/chan_mgcp.c, apps/app_voicemail.c,
+ apps/app_adsiprog.c, channels/chan_sip.c, channels/chan_agent.c:
+ When optional_api is non-optional, force dependent modules to be
+ loaded. (closes issue #17707) Reported by: ira Patches:
+ 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/876/
+
+ * include/asterisk/channel.h, res/res_jabber.c, res/res_pktccops.c,
+ main/poll.c, channels/chan_usbradio.c, include/asterisk/select.h
+ (added), channels/chan_phone.c, channels/chan_misdn.c, configure,
+ main/features.c, include/asterisk/poll-compat.h,
+ tests/test_poll.c (added), addons/ooh323c/src/oochannels.c,
+ main/asterisk.c, addons/ooh323c/src/ooSocket.h, main/stun.c,
+ res/res_ais.c, /, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/console_video.c: Merged revisions 284593,284595 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500
+ (Wed, 01 Sep 2010) | 18 lines 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/ ........
+ ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500
+ (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after
+ last commit ................
+
+2010-09-01 21:47 +0000 [r284561] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: During request to dialog matching, verify
+ init_ruri is present before comparing. During request to dialog
+ matching, we attempt a best effort routine for fork detection
+ which requires several elements to be in place. The dialog's
+ initial request uri is one of those elements. Since it is best
+ effort, if the init_ruri is not present for some reason we can
+ not proceed with that routine.
+
+2010-09-01 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta5 released.
+
+2010-09-01 18:44 +0000 [r284477] Terry Wilson <twilson@digium.com>
+
+ * res/res_srtp.c, res/res_rtp_asterisk.c,
+ include/asterisk/res_srtp.h, main/rtp_engine.c,
+ channels/chan_sip.c: Fix SRTP for changing SSRC and multiple
+ a=crypto SDP lines Adding code to Asterisk that changed the SSRC
+ during bridges and masquerades broke SRTP functionality. Also
+ broken was handling the situation where an incoming INVITE had
+ more than one crypto offer. This patch caches the SRTP policies
+ the we use so that we can change the ssrc and inform libsrtp of
+ the new streams. It also uses the first acceptable a=crypto line
+ from the incoming INVITE. (closes issue #17563) Reported by:
+ Alexcr Patches: srtp.diff uploaded by twilson (license 396)
+ Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/878/
+
+2010-09-01 18:16 +0000 [r284415-284473] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 284472 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01
+ Sep 2010) | 5 lines Don't warn on floats and timestamps (closes
+ issue #17082) Reported by: coolmig ........
+
+ * /, channels/chan_sip.c: Merged revisions 284399 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500
+ (Tue, 31 Aug 2010) | 14 lines 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 19:00 +0000 [r284318] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 284317 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284317 | lmadsen | 2010-08-31 13:59:31 -0500
+ (Tue, 31 Aug 2010) | 15 lines 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 22:28 +0000 [r284281] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 284280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010)
+ | 11 lines 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 ........
+
+2010-08-29 07:05 +0000 [r284096-284158] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/res_curl.conf.sample (added): Missed adding this file
+
+ * sounds: Also ignore the checksums
+
+ * configs/cel_odbc.conf.sample (added), cel/cel_adaptive_odbc.c
+ (removed), cel/cel_odbc.c (added),
+ configs/cel_adaptive_odbc.conf.sample (removed): Rename CEL
+ adaptive driver to plain driver, since there isn't another ODBC
+ driver (and the other CEL drivers have adaptive capabilities,
+ anyway).
+
+2010-08-28 21:29 +0000 [r284065] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: Be more flexible with whitespace on AMI action
+ headers. Previously, this code required exactly one space to be
+ after the ':' in headers for an AMI action. This now makes
+ whitespace optional, and allows whitespace that is there to vary
+ in amount. (closes issue #17862) Reported by: cmoye Patches:
+ manager.c.patch_trunk uploaded by cmoye (license 858)
+ manager.c.patch_1.8 uploaded by cmoye (license 858) Tested by:
+ cmoye
+
+2010-08-27 22:37 +0000 [r284032] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284002 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500
+ (Fri, 27 Aug 2010) | 14 lines 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 21:33 +0000 [r283951] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_realtime.c: Print exten@context:priority in verbose
+ messages from pbx_realtime.
+
+2010-08-27 20:31 +0000 [r283882] Jason Parker <jparker@digium.com>
+
+ * main/config.c, addons/res_config_mysql.c, res/res_config_odbc.c,
+ /: Merged revisions 283881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283881 | qwell | 2010-08-27 15:30:27 -0500
+ (Fri, 27 Aug 2010) | 15 lines 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 23:47 +0000 [r283770] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Convert MOH to use generic timers. (closes
+ issue #17726) Reported by: lmadsen Patches:
+ 20100825__issue17726__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: tilghman
+
+2010-08-26 15:26 +0000 [r283692] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283691 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500
+ (Thu, 26 Aug 2010) | 25 lines 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.
+ ........ ................
+
+2010-08-26 13:26 +0000 [r283627-283659] Russell Bryant <russell@digium.com>
+
+ * res/res_odbc.c: Slight improvement to a debug message.
+
+ * keys/iaxtel.pub (removed), keys/freeworlddialup.pub (removed),
+ Makefile: Remove public keys that are no longer useful.
+
+ * configs/manager.conf.sample: Move httptimeout out from in between
+ port and bindaddr.
+
+2010-08-25 22:57 +0000 [r283595] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010)
+ | 7 lines 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. ........
+
+2010-08-25 16:12 +0000 [r283561] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Initialize connect timeout on each time through
+ the loop. (closes issue #17911) Reported by: wurstsalat
+
+2010-08-25 15:54 +0000 [r283559] David Vossel <dvossel@digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 283558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010)
+ | 10 lines 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) ........
+
+2010-08-25 14:55 +0000 [r283527] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Convert ast_log(LOG_DEBUG, ...) to
+ ast_debug(...)
+
+2010-08-24 20:34 +0000 [r283493] David Vossel <dvossel@digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
+ Changes the default behavior for sip.conf's pedantic option from
+ "no" to "yes".
+
+2010-08-24 18:56 +0000 [r283457] Leif Madsen <lmadsen@digium.com>
+
+ * res/res_rtp_asterisk.c, channels/chan_sip.c: Fix issue where TOS
+ is no longer set on RTP packets. Fix issue where the tos is no
+ longer being set on RTP packets through res_rtp_asterisk. (closes
+ issue #17890) Reported by: elguero Patches: qos_18.diff uploaded
+ by elguero (license 37) Review:
+ https://reviewboard.asterisk.org/r/868
+
+2010-08-24 16:11 +0000 [r283382] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500
+ (Tue, 24 Aug 2010) | 18 lines 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. ........
+ ................
+
+2010-08-24 12:49 +0000 [r283350] Russell Bryant <russell@digium.com>
+
+ * funcs/func_odbc.c: Don't attempt to release a NULL ODBC handle.
+
+2010-08-23 21:33 +0000 [r283319] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, cel/cel_adaptive_odbc.c,
+ /: Merged revisions 283318 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010)
+ | 2 lines CDR drivers depend upon res_odbc, not directly on the
+ ODBC libraries ........
+
+2010-08-23 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta4 Released.
+
+2010-08-23 13:35 +0000 [r283177-283241] Russell Bryant <russell@digium.com>
+
+ * configs/cel.conf.sample: Add sample configuration for cel_radius.
+
+ * main/cel.c, include/asterisk/cel.h: Make the AST_CEL_AMA enum
+ match up with the AST_CDR_ ama flag values. Really, having 2
+ enums for this is silly and error prone, demonstrated by the
+ crash that I hit because there was an assumption in the code that
+ the values in each matched up. However, this is a quick fix to
+ get them to match up so it will work.
+
+ * main/cel.c: Don't blow up on an invalid AMA flag.
+
+ * configs/cel_custom.conf.sample: Tack on ${eventextra} to the
+ sample cel_custom.conf.
+
+ * configs/cel_custom.conf.sample: Cut down on excessive quotation.
+
+2010-08-23 12:06 +0000 [r283175] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_stun_monitor.c: Don't fail to start if the config file is
+ missing.
+
+2010-08-23 11:58 +0000 [r283173] Russell Bryant <russell@digium.com>
+
+ * configs/cel_custom.conf.sample: Expand cel_custom.conf.sample.
+ Include the usage of CSV_QUOTE() to ensure data has valid CSV
+ formatting. Also list the special CEL variables that are
+ available for use in the mapping.
+
+2010-08-20 16:51 +0000 [r283050-283125] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Recorded merge of revisions 283124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283124 | rmudgett | 2010-08-20 11:48:10 -0500
+ (Fri, 20 Aug 2010) | 16 lines 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/sig_pri.c, /: Merged revisions 283049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500
+ (Fri, 20 Aug 2010) | 29 lines 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) ........ ................
+
+2010-08-20 12:45 +0000 [r282979-283013] Russell Bryant <russell@digium.com>
+
+ * configs/cel_adaptive_odbc.conf.sample: Fix a typo in a column
+ name.
+
+ * apps/app_celgenuserevent.c: Add an argument missing from the
+ CELGenUserEvent documentation.
+
+2010-08-19 21:07 +0000 [r282891-282895] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500
+ (Thu, 19 Aug 2010) | 18 lines 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: Merged revisions 282890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010)
+ | 5 lines fixes sip peer memory leaks in the peer_by_ip table
+ (issue #17798) ........
+
+2010-08-19 20:01 +0000 [r282860] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500
+ (Thu, 19 Aug 2010) | 23 lines 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 14:44 +0000 [r282826] Tilghman Lesher <tlesher@digium.com>
+
+ * main/netsock2.c: Only output debugging if the debug level is on.
+
+2010-08-19 02:18 +0000 [r282740] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 282730 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282730 | twilson | 2010-08-18 21:14:28 -0500
+ (Wed, 18 Aug 2010) | 9 lines 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 15:28 +0000 [r282671-282672] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h: Use the correct type for aoce_delayhangup bit
+ field.
+
+ * channels/chan_dahdi.c: Use the correct operator when calculating
+ the PRI span devstate.
+
+2010-08-18 13:10 +0000 [r282639] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Properly handle 200 and unknown responses
+ conatined in NOTIFY requests received in response to REFER
+ requests. This patch fixes the way asterisk handles NOTIFY
+ requests received in response to REFER requests. These changes to
+ NOTIFY handler were first introduced in r217482. This new change
+ properly handles the 200 response by queueing an
+ AST_TRANSFER_SUCCESS control frame and also prevents that control
+ frame from being queued when provisional and unknown responses
+ are received. (issue #17486) Reported by: davidw Tested by:
+ mnicholson (issue #12713) Reported by: davidw Review:
+ https://reviewboard.asterisk.org/r/860/
+
+2010-08-18 12:30 +0000 [r282638] Russell Bryant <russell@digium.com>
+
+ * channels/chan_multicast_rtp.c: Split _all_ arguments before
+ parsing them. This fixes multicast RTP paging using linksys mode.
+
+2010-08-18 07:49 +0000 [r282608] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/sig_pri.c, /: Merged revisions 282607 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010)
+ | 9 lines 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 21:36 +0000 [r282543-282577] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010)
+ | 9 lines 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 ........
+
+ * channels/chan_iax2.c: ACCEPT message should respond with the new
+ FORMAT2 ie (closes issue #17804) Reported by: tpanton
+
+ * include/asterisk/unaligned.h: fixes truncated uint64_t value in
+ put_unaligned_uint64_t() function (issue #17804)
+
+2010-08-16 18:01 +0000 [r282470] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/asterisk.tex, doc/tex/sounds.tex (added), /: Merged
+ revisions 282469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010)
+ | 7 lines 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 17:53 +0000 [r282468] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 282467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282467 | twilson | 2010-08-16 12:32:01 -0500
+ (Mon, 16 Aug 2010) | 23 lines 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/
+ ........ ................
+
+2010-08-14 04:53 +0000 [r282366] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/sched.h: Fix our FRACKing
+ issue with chan_iax2 a different way. Review:
+ https://reviewboard.asterisk.org/r/861/
+
+2010-08-13 23:53 +0000 [r282334] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: PRI CCSS may use a stale dial string for
+ the recall dial string. If an outgoing call negotiates a
+ different B channel than initially requested, the saved original
+ dial string was not transferred to the new B channel. CCSS uses
+ that dial string to generate the recall dial string.
+
+2010-08-13 22:23 +0000 [r282236-282302] David Vossel <dvossel@digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, CHANGES,
+ channels/chan_sip.c: remove current STUN support from chan_sip.c
+ This patch removes the current broken/useless stun support from
+ chan_sip. (closes issue #17622) Reported by: philipp2 Review:
+ https://reviewboard.asterisk.org/r/855/
+
+ * CHANGES: res_stun_monitor and corresponding options CHANGES
+ documentation
+
+ * configs/res_stun_monitor.conf.sample (added),
+ configs/sip.conf.sample, channels/chan_iax2.c,
+ configs/iax.conf.sample, channels/chan_sip.c,
+ include/asterisk/event_defs.h, res/res_stun_monitor.c (added):
+ res_stun_monitor for monitoring network changes behind a NAT
+ device Review: https://reviewboard.asterisk.org/r/854
+
+ * /, channels/chan_sip.c: Merged revisions 282235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010)
+ | 16 lines 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 ........
+
+2010-08-13 16:02 +0000 [r282200-282201] Terry Wilson <twilson@digium.com>
+
+ * configure.ac: Whitespace fix :-/
+
+ * configure, configure.ac: Detect when libsrtp cannot be linked in
+ a shared library The libsrtp build system currently does not
+ produce a shared library or a static library compiled with -fPIC,
+ so on 64-bit systems it is possible that we will get a compile
+ error if libsrtp is installed and res_srtp is selected in
+ menuselect. This patch attempts to detect this situation and
+ provide the user with instructions to work around the problem.
+
+2010-08-12 22:51 +0000 [r282131] Jason Parker <jparker@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 282130 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282130 | qwell | 2010-08-12 17:50:54 -0500
+ (Thu, 12 Aug 2010) | 9 lines 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 22:06 +0000 [r282098] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/ccss.h, main/ccss.c: Separate call completion
+ config parameter allocation and default initialization. If you
+ ever have a need to reset the call completion config parameters
+ to defaults, now you can. And no Virginia, C++ idioms do not
+ always work in C.
+
+2010-08-12 20:41 +0000 [r282066] Russell Bryant <russell@digium.com>
+
+ * CHANGES, main/cli.c: Add a "core reload" CLI command. Review:
+ https://reviewboard.asterisk.org/r/859/
+
+2010-08-12 20:15 +0000 [r282047] David Vossel <dvossel@digium.com>
+
+ * CHANGES, include/asterisk/translate.h, main/cli.c,
+ main/translate.c: improved translation paths for wideband codecs
+ The problem I'm addressing is that Asterisk's current method of
+ building the least cost translation paths between codecs does not
+ take into account sample rate. For instance, it was possible for
+ siren14 (a 32khz codec), to contain the a translation path to
+ siren7 (a 16khz audio codec) that goes through slin at 8khz. In
+ this case Asterisk takes a 32khz codec, down samples it to 8khz
+ and then up samples it to 16khz which is terrible regardless if
+ it is computationally less expensive. This patch now builds
+ translation paths that give priority to maintaining the best
+ possible sample rate before taking into consideration
+ computational cost. This patch also adds cli commands to expose
+ what translation paths are actually being used. Changes: 1.
+ Translation paths will never contain a step that changes the
+ sample rate unless absolutely necessary. 2. When choosing the
+ best codec to make two channels compatible. Shared codecs with
+ the highest sample rate are given priority. 3. A new cli command
+ to show all translation paths available for a specific codec
+ 'core show translation paths [codec name]' has been added. 4.
+ 'core show translation' which displays the translation matrix now
+ includes the new higher bit audio codecs in the table. 5. 'core
+ show channel [channel name]' now displays the translation paths
+ if translation is used. (closes issue #16841) Reported by:
+ dvossel Review: https://reviewboard.asterisk.org/r/842/
+
+2010-08-12 18:03 +0000 [r281982-282015] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Put back pointer value output for ast_debug(), such
+ that it is only removed for verbose output.
+
+ * main/pbx.c: Remove debugging output from verbose messages.
+ Pointer values to internal objects is not terribly useful to
+ users in the verbose messages about adding extensions and
+ contexts.
+
+2010-08-12 03:03 +0000 [r281913] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 281912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500
+ (Wed, 11 Aug 2010) | 27 lines 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 21:12 +0000 [r281875] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 281873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281873 | lmadsen | 2010-08-11 16:09:47 -0500
+ (Wed, 11 Aug 2010) | 14 lines 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) ........ ................
+
+2010-08-11 21:11 +0000 [r281874] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: handle all possible responses to REFER
+ requests (closes issue #17486) Reported by: davidw Patches:
+ Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
+ Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/
+
+2010-08-11 20:30 +0000 [r281870] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_analog.c, channels/sig_analog.h: Fix a call to
+ analog_set_pulsedial() not setting 0 or 1 only. * Also a couple
+ minor tweaks.
+
+2010-08-11 17:54 +0000 [r281764] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 281763 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281763 | lmadsen | 2010-08-11 12:54:09 -0500
+ (Wed, 11 Aug 2010) | 14 lines 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 17:27 +0000 [r281760] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Avoid a deadlock in
+ add_header_max_forwards(). Related to r276951
+
+2010-08-11 15:18 +0000 [r281723] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_readexten.c: Merged revisions 281722 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11
+ Aug 2010) | 7 lines 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) ........
+
+2010-08-11 13:30 +0000 [r281687] <simon.perreault@viagenie.ca>
+
+ * include/asterisk/netsock2.h, configs/sip.conf.sample,
+ channels/sip/config_parser.c, main/netsock2.c: Fix parsing of
+ IPv6 address literals in outboundproxy (closes issue #17757)
+ Reported by: oej Patches: 17757.diff uploaded by sperreault
+ (license 252) sip.conf.diff uploaded by sperreault (license 252)
+ Tested by: oej
+
+2010-08-10 21:47 +0000 [r281568-281650] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
+ Change the default value for alwaysauthreject in sip.conf to
+ "yes". (closes issue #17756) Reported by: oej
+
+ * main/sched.c, /: Merged revisions 281574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010)
+ | 9 lines 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 281567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281567 | russell | 2010-08-10 12:47:13 -0500
+ (Tue, 10 Aug 2010) | 15 lines 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 ........
+ ................
+
+2010-08-10 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta3 Released.
+
+2010-08-10 17:48 +0000 [r281529-281568] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281567 | russell | 2010-08-10 12:47:13 -0500
+ (Tue, 10 Aug 2010) | 15 lines 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 ........
+ ................
+
+ * channels/chan_sip.c: Ensure that the proper external address is
+ used for the RTP destination. (closes issue #17044) Reported by:
+ ebroad Tested by: ebroad Review:
+ https://reviewboard.asterisk.org/r/566/
+
+ * main/cli.c: Resolve a problem with channel name tab completion.
+ Hitting tab without typing any part of a channel name resulted in
+ no results. This now results in getting a full list of active
+ channels, just as it did in previous versions of Asterisk.
+ Review: https://reviewboard.asterisk.org/r/818/
+
+2010-08-10 07:26 +0000 [r281497] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Fixed the issue caused by EXTEN including
+ user parameters.
+
+2010-08-09 23:04 +0000 [r281466] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c: Add some more stuff to copy from 281429.
+
+2010-08-09 20:47 +0000 [r281432] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 281430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010)
+ | 13 lines 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 20:43 +0000 [r281429] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 281391 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500
+ (Mon, 09 Aug 2010) | 20 lines 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/ ........
+ ................
+
+2010-08-09 14:49 +0000 [r281358] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Validate minrate, maxrate, and modem settings
+ before attempting a fax session. FAX-224
+
+2010-08-09 14:31 +0000 [r281356] <simon.perreault@viagenie.ca>
+
+ * configs/sip.conf.sample: Added comment about IPv4-mapped IPv6
+ addresses and the output of netstat.
+
+2010-08-09 12:51 +0000 [r281294-281325] Russell Bryant <russell@digium.com>
+
+ * configs/cdr.conf.sample: Add a couple of default values to the
+ documentation of cdr.conf.
+
+ * configs/cdr.conf.sample: Reorder some options in cdr.conf.sample.
+ Put all of the options that affect the contents of CDRs together,
+ instead of having the batch mode options in the middle of them.
+
+2010-08-06 18:57 +0000 [r281085] Tilghman Lesher <tlesher@digium.com>
+
+ * main/utils.c: Fix alignment of stringfields on the SPARC
+ architecture (closes issue #17789) Reported by: Ian Mason
+ Patches: 20100806__issue17789__2.diff.txt uploaded by tilghman
+ (license 14) Tested by: Ian_Mason
+
+2010-08-05 13:16 +0000 [r281052] Russell Bryant <russell@digium.com>
+
+ * main/cdr.c, /: Merged revisions 281051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010)
+ | 9 lines 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 07:46 +0000 [r280984] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280983
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500
+ (Thu, 05 Aug 2010) | 15 lines 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 ........ ................
+
+2010-08-04 15:11 +0000 [r280909] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Initialize FAXOPT() status variables in sendfax
+ and receivefax instead of when the details structure is created.
+
+2010-08-04 14:04 +0000 [r280809-280879] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c: Check cur value before attempting a deref.
+ (closes issue #17775) Reported by: svinson Patches:
+ 20100804__issue17775.diff.txt uploaded by tilghman (license 14)
+ Tested by: svinson (closes issue #17743) Reported by: tgruenberg
+ Patches: 20100804__issue17775.diff.txt uploaded by tilghman
+ (license 14) Tested by: tgruenberg
+
+ * CHANGES, funcs/func_strings.c: Sneak FIELDNUM() into 1.8. Returns
+ a 1-based index into a list of a specified item. Matches up with
+ FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth
+ Patches: svn-279754.diff uploaded by gareth (license 208) Tested
+ by: gareth, tilghman Review:
+ https://reviewboard.asterisk.org/r/810/
+
+2010-08-03 19:54 +0000 [r280777-280778] <simon.perreault@viagenie.ca>
+
+ * channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes
+ issue #17663) Reported by: oej Patches: diff uploaded by
+ sperreault (license 252) diff2 uploaded by sperreault (license
+ 252) get_domain.diff uploaded by sperreault (license 252)
+
+ * configs/sip.conf.sample: Better documentation related to IPv6.
+ (closes issue #17737) Reported by: oej Patches: doc.diff uploaded
+ by sperreault (license 252) Tested by: mmichelson
+
+2010-08-03 18:48 +0000 [r280742] Russell Bryant <russell@digium.com>
+
+ * addons/Makefile, addons/mp3 (removed),
+ contrib/scripts/get_mp3_source.sh (added): Remove the MP3 decoder
+ source code and replace it with a small shell script. Review:
+ https://reviewboard.asterisk.org/r/836/
+
+2010-08-03 18:42 +0000 [r280624-280740] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/asterisk.sgml, /, doc/asterisk.8, doc/Makefile (added):
+ Merged revisions 280739 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010)
+ | 2 lines Document -B and -W flags and regenerate manpage from
+ sgml ........
+
+ * apps/app_voicemail.c, /: Merged revisions 280671 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02
+ Aug 2010) | 2 lines Allow the pipe, but also allow the comma
+ ........
+
+ * main/Makefile: Make this a little more deterministic... we want
+ the latest value, not just a 1 somewhere.
+
+ * main/Makefile: Apparently, the values in makeopts are sometimes
+ 1:1 and sometimes 1. Compensate for this.
+
+2010-07-29 21:07 +0000 [r280557] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Fix regression introduced in r1664. Give the fax
+ stack time to shutdown and populate the FAXOPT output variables.
+ FAX-222
+
+2010-07-29 20:43 +0000 [r280552] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 280551 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010)
+ | 11 lines 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
+ ........
+
+2010-07-29 20:35 +0000 [r280549] Russell Bryant <russell@digium.com>
+
+ * configs/ccss.conf.sample: Add header to ccss.conf to appease oej.
+ (closes issue #17755) Reported by: oej
+
+2010-07-29 19:47 +0000 [r280519] Sean Bright <sean@malleable.com>
+
+ * channels/sig_pri.c: Fix compilation error in chan_dahdi (strdupa
+ -> ast_strdupa). (closes issue #17751) Reported by: b11d Patches:
+ strdupa_oops.diff uploaded by malcolmd (license 924)
+
+2010-07-29 19:13 +0000 [r280450] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 280449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500
+ (Thu, 29 Jul 2010) | 18 lines 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 18:37 +0000 [r280414-280446] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * tests/test_utils.c: Remove res_crypto dependency.
+
+ * tests/test_utils.c: crypto_loaded_test depends on res_crypto,
+ else test will fail.
+
+2010-07-29 16:25 +0000 [r280391] Russell Bryant <russell@digium.com>
+
+ * main/rtp_engine.c: Don't blow up if get_codec() was not provided
+ in the RTP glue.
+
+2010-07-29 16:07 +0000 [r280346] Jean Galarneau <jgalarneau@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 280345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280345 | jeang | 2010-07-29 11:01:35 -0500
+ (Thu, 29 Jul 2010) | 10 lines 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 15:57 +0000 [r280307-280343] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_usbradio.c: Use PRIx64 instead of PRId64 in format
+ string. related to r280302
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 280306
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul
+ 2010) | 2 lines 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/ ........
+ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames
+ through generic bridges. This change appears to have been
+ unintentionally left out of rev 203699.
+
+2010-07-29 00:45 +0000 [r280302] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_usbradio.c: Use PRId64 with format_t
+
+2010-07-28 20:49 +0000 [r280269] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sip/reqresp_parser.c: Give test category missing leading
+ slash
+
+2010-07-28 20:12 +0000 [r280235] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 280229 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28
+ Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7
+ called_nai and calling_nai config options. ........
+
+2010-07-28 20:03 +0000 [r280233] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 280231 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) |
+ 6 lines 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 19:34 +0000 [r280225] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c: Do rtp/rtcp debugging when it is turned
+ on w/o filtering
+
+2010-07-28 18:24 +0000 [r280195] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 280193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) |
+ 9 lines 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 16:52 +0000 [r280161] Sean Bright <sean@malleable.com>
+
+ * apps/app_queue.c, /: Merged revisions 280160 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul
+ 2010) | 8 lines 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 13:52 +0000 [r280090] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/live_ast, /: Merged revisions 280089 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500
+ (Wed, 28 Jul 2010) | 9 lines 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.
+ ........ ................
+
+2010-07-28 13:01 +0000 [r280058] Russell Bryant <russell@digium.com>
+
+ * res/res_crypto.c: s/init keys/keys init/
+
+2010-07-28 01:37 +0000 [r280023] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_usbradio.c: Resolve compiler warning about
+ formatting (closes issue #17732) Reported by: pabelanger
+
+2010-07-27 22:30 +0000 [r280019-280020] Sean Bright <sean@malleable.com>
+
+ * main/editline/el.h, main/term.c, main/cli.c,
+ main/editline/parse.c, main/editline/tokenizer.c,
+ main/editline/config.sub, main/editline/parse.h,
+ main/editline/tokenizer.h, configure, main/editline/histedit.h,
+ main/editline/sig.c, main/editline/PLATFORMS,
+ main/editline/sig.h, main/editline/key.c, main/editline/editrc.5,
+ main/editline/np/fgetln.c, main/editline/key.h,
+ main/editline/TEST/test.c, main/Makefile,
+ main/editline/configure, main/editline/Makefile.in, configure.ac,
+ main/editline/configure.in, main/editline/readline/readline.h,
+ main/editline/README, main/editline/editline.3,
+ main/editline/vi.c, main/editline/sys.h, main/editline/emacs.c,
+ main/asterisk.c, main/editline/install-sh, main/editline/term.c,
+ main/editline/config.guess, main/editline/read.c,
+ main/editline/term.h, main/editline/map.c,
+ main/editline/np/strlcpy.c, main/editline (added),
+ main/editline/config.h.in, main/editline/read.h,
+ main/editline/tty.c, main/editline/np/unvis.c,
+ main/editline/prompt.c, main/editline/map.h, main/editline/tty.h,
+ main/editline/chared.c, main/editline/prompt.h,
+ main/editline/np/strlcat.c, main/editline/chared.h,
+ main/editline/np, main/editline/TEST, main/editline/refresh.c,
+ main/editline/history.c, main/editline/readline,
+ include/asterisk/term.h, main/editline/refresh.h,
+ main/editline/search.c, main/editline/hist.c,
+ main/editline/search.h, main/editline/hist.h,
+ main/editline/np/vis.c, build_tools/menuselect-deps.in, main,
+ main/editline/readline.c, main/editline/np/vis.h,
+ main/editline/INSTALL, makeopts.in, main/editline/CHANGES,
+ main/editline/common.c, main/xmldoc.c, main/editline/makelist.in,
+ include/asterisk/autoconfig.h.in, main/editline/el.c: Revert
+ r280019 for now - This was poorly executed.
+
+ * include/asterisk/term.h, makeopts.in, main/asterisk.c,
+ main/xmldoc.c, main/cli.c, main/term.c, main/editline (removed),
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
+ main: Add ability to use system libedit and update bundled
+ libedit. The version of libedit that is bundled with asterisk is
+ old and has some bugs. This patch updates the bundled version of
+ libedit within asterisk, and also updates asterisk to use the
+ system libedit instead if one is available (and pkg-config is
+ available). This review integrates several patches from other
+ users specifically kkm and tzafrir. (closes issue #15929)
+ Reported by: kkm Patches: 015929-astcli-editrc-trunk.240324.diff
+ uploaded by kkm (license 888) (issue #16858) Reported by:
+ jw-asterisk (closes issue #17039) Reported by: tzafrir Patches:
+ 0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir
+ (license 46) Review: https://reviewboard.asterisk.org/r/807/
+
+2010-07-27 21:16 +0000 [r279953] Russell Bryant <russell@digium.com>
+
+ * res/ais, main/db1-ast/mpool, Makefile.rules, res/snmp, cdr,
+ formats, codecs/gsm/src, funcs, bridges, codecs/lpc10,
+ main/db1-ast/btree, configure, main/editline, codecs/g722, main,
+ main/db1-ast/recno, channels/sip, makeopts.in, pbx, res, res/ael,
+ channels, main/stdtime, main/editline/np, codecs, utils,
+ main/db1-ast/hash, cel, apps, configure.ac, main/db1-ast/db: Add
+ --enable-coverage option to configure script. This option enables
+ the proper compiler flags for tracking code coverage, which is
+ useful along side automated testing.
+
+2010-07-27 20:57 +0000 [r279949] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c, main/channel.c, /,
+ include/asterisk/audiohook.h: Merged revisions 279946 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500
+ (Tue, 27 Jul 2010) | 24 lines 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 19:50 +0000 [r279916] Russell Bryant <russell@digium.com>
+
+ * channels/sig_pri.c, channels/chan_dahdi.c: Fix inband DTMF
+ detection on outgoing ISDN calls. This is a regression from the
+ sig_pri split from chan_dahdi. When a call is first initiated,
+ the inband DTMF detector is not enabled if it's an outgoing ISDN
+ call. However, it needs to be turned on once the media path
+ starts up. This handling was put back in the open_media()
+ callback of chan_dahdi. In sig_pri, open_media() calls were added
+ to a few places where it was needed, including handling of
+ PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING.
+ Thanks to rmudgett for helping me with the patch!
+
+2010-07-27 18:54 +0000 [r279887] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix parsing error in sip_sipredirect(). The
+ code was written in a way that did a bad job of parsing the port
+ out of a URI. Specifically, it would do badly when dealing with
+ an IPv6 address. In this particular scenario, there was no value
+ from parsing the port out, so I just removed that logic. And
+ while I was messing around in the function, I changed some
+ variable names to be more descriptive. (closes issue #17661)
+ Reported by: oej Patches: 17661.diff uploaded by mmichelson
+ (license 60)
+
+2010-07-27 16:40 +0000 [r279850] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 279849 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul 2010) |
+ 1 line Simply sounds/Makefile some more. ........
+
+2010-07-27 16:09 +0000 [r279817] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c, channels/chan_sip.c: fix sip transaction match
+ with authentication, fix confusing log message when using
+ getaddrinfo
+
+2010-07-27 16:06 +0000 [r279815] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c: Support "channels" in addition to
+ "channel" in chan_dahdi.conf. Review:
+ https://reviewboard.asterisk.org/r/804
+
+2010-07-27 15:15 +0000 [r279785] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 279784 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul
+ 2010) | 14 lines 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 ........
+
+2010-07-27 02:57 +0000 [r279726-279755] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_dahdi.c: If dringXcontext is null, fallback to
+ default context value. (closes issue #17693) Reported by:
+ iasgoscouk Patches: issue17693.patch uploaded by pabelanger
+ (license 224) Tested by: iasgoscouk Review:
+ https://reviewboard.asterisk.org/r/803/
+
+ * main/http.c: Use ast_sockaddr_setnull() when http is not enabled.
+ Otherwise, ast_tcptls_server_start() will still start http.
+ (closes issue #17708) Reported by: pabelanger Patches: http.patch
+ uploaded by pabelanger (license 224)
+
+2010-07-26 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta2 Released.
+
+2010-07-26 23:29 +0000 [r279689] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * UPGRADE.txt, CHANGES: Updated documentation for FAX logger level.
+
+2010-07-26 23:03 +0000 [r279658] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile (added), /, sounds/Makefile.380 (removed),
+ configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+ (removed), configure.ac: Merged revisions 279657 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul
+ 2010) | 5 lines 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 21:53 +0000 [r279636] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Ignore a control subclass of -1 in
+ ast_waitfordigit_full().
+
+2010-07-26 21:20 +0000 [r279599-279619] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 279609 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26
+ Jul 2010) | 2 lines Dunno why this worked on my machine, but it
+ works better this way. ........
+
+ * res/res_config_ldap.c, /: Merged revisions 279597 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26
+ Jul 2010) | 13 lines 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 ........
+
+ * /: Reverting property remove
+
+2010-07-26 20:58 +0000 [r279598] Gavin Henry <ghenry@suretecsystems.com>
+
+ * /: Merged revisions 279597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/1.6.2
+ -----------------------------------------------------------------------
+ r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) |
+ 13 lines Apply all patches in:
+ https://issues.asterisk.org/view.php?id=13573 [^] (closes issue
+ 0013573) 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 19:59 +0000 [r279568] David Vossel <dvossel@digium.com>
+
+ * channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c: transaction matching using top
+ most Via header This patch modifies the way chan_sip.c does
+ transaction to dialog matching. Asterisk now stores information
+ in the top most Via header of the initial incoming request and
+ compares that against other Requests that have the same call-id.
+ This results in Asterisk being able to detect a forked call in
+ which it has received multiple legs of the fork. I completely
+ stripped out the previous matching code and made the comparisons
+ a little more explicit and easier to understand. My comments in
+ the code should offer all the details involving this patch. This
+ patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
+ find multiple dialogs with the same call-id. Since the callback
+ function was returning (CMP_MATCH | CMP_STOP) only the first item
+ found was being returned. I fixed this by making a new callback
+ function for finding multiple dialogs that only returns
+ (CMP_MATCH) on a match allowing for multiple items to be
+ returned. Review: https://reviewboard.asterisk.org/r/776/
+
+2010-07-26 19:51 +0000 [r279566] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * UPGRADE.txt, CHANGES, configs/logger.conf.sample: Add
+ documentation for FAX logger level. (closes issue #17715)
+ Reported by: vrban Patches: 17715.patch uploaded by pabelanger
+ (license 224) Tested by: vrban
+
+2010-07-26 19:18 +0000 [r279562] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile (removed), /, sounds/Makefile.380 (added),
+ configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+ (added), configure.ac: Merged revisions 279561 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010)
+ | 2 lines Use a special Makefile for noobs who still have GNU
+ Make 3.80. ........
+
+2010-07-26 16:04 +0000 [r279504] Mark Michelson <mmichelson@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sip/reqresp_parser.c: Allow for systems without locale
+ support to be usable. A recent change to SIP URI comparison code
+ added a locale-specific string comparison to the mix, and certain
+ systems do not support such functions. This fix allows for those
+ systems to still use Asterisk 1.8 (closes issue #17697) Reported
+ by: pprindeville Patches: asterisk-trunk-bugid17697.patch
+ uploaded by pprindeville (license 347) Tested by: mmichelson
+
+2010-07-26 15:43 +0000 [r279502] Sean Bright <sean@malleable.com>
+
+ * autoconf/ast_ext_lib.m4, /: Merged revisions 279501 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon,
+ 26 Jul 2010) | 5 lines Expand the correct value within
+ AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
+ ........
+
+2010-07-26 03:27 +0000 [r279472] Tilghman Lesher <tlesher@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_g726.c,
+ formats/format_jpeg.c, formats/format_siren14.c,
+ formats/format_gsm.c, formats/format_g719.c,
+ formats/format_g729.c, formats/format_sln.c,
+ formats/format_wav.c, formats/format_ogg_vorbis.c: Formats need
+ to load before apps, because some apps call
+ ast_format_str_reduce() at load time.
+
+2010-07-25 21:26 +0000 [r279442] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * tests/test_func_file.c: Add trailing backslash to silence warning
+ message.
+
+2010-07-25 18:21 +0000 [r279390-279410] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_odbc.c: Don't re-register CDR module on reload. (closes
+ issue #17304) Reported by: jnemeth Patches:
+ 20100507__issue17304.diff.txt uploaded by tilghman (license 14)
+ Tested by: jnemeth
+
+ * main/logger.c: Don't assume qlog is open. (closes issue #17704)
+ Reported by: vrban Patches: issue17704.patch uploaded by
+ pabelanger (license 224) Tested by: vrban
+
+2010-07-24 23:58 +0000 [r279348] Bradley Latus <brad.latus@gmail.com>
+
+ * doc/asterisk.8: Minor update to man page
+
+2010-07-24 20:47 +0000 [r279273-279314] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile: Remove duplicate -c flag when using $(INSTALL) (closes
+ issue #17695) Reported by: pabelanger Patches: Makefile.diff
+ uploaded by pabelanger (license 224)
+
+ * include/asterisk/netsock2.h: Check if ast_sockaddr is NULL then
+ return. (closes issue #17677) Reported by: outcast Patches:
+ issue0017677.patch uploaded by pabelanger (license 224) Tested
+ by: elguero
+
+ * main/manager.c: Default sin_family to AF_INET for TCP / TLS
+ Bindaddress. Otherwise, 'manager show settings' will generate
+ errors if manager is not enabled.
+
+2010-07-23 22:20 +0000 [r279227] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279207 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500
+ (Fri, 23 Jul 2010) | 14 lines 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 ........ ................
+
+2010-07-12 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta1 Released.
+
+2010-07-23 18:56 +0000 [r279113] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Silly 64-bit compilers (who uses 64-bit anyway?)
+
+2010-07-23 18:23 +0000 [r279056-279094] Russell Bryant <russell@digium.com>
+
+ * /: fix up properties on 1.8 branch
+
+ * / (added): Create a branch for Asterisk 1.8.
+
+ ___ _ _ _ _ ___
+ / _ \ ___| |_ ___ _ __(_)___| | __ / | ( _ )
+ | |_| / __| __/ _ \ '__| / __| |/ / | | / _ \
+ | _ \__ \ || __/ | | \__ \ < | || (_) |
+ |_| |_|___/\__\___|_| |_|___/_|\_\ |_(_)___/
+
+2010-07-23 17:05 +0000 [r278982-278985] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_check_pwlib.m4, /, configure, configure.ac: 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 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 16:33 +0000 [r278980] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c,
+ channels/sip/include/reqresp_parser.h: SIP URI comparison fixes.
+ This initially was created to work around the issue of using a
+ string comparison instead of a binary comparison for IP
+ addresses. It evolved a bit when test cases were created and it
+ was discovered that comparison of URI parameters was not working
+ exactly as it should. sip_uri_cmp() and its helpers have been
+ moved to reqresp_parser.c and a new test has been added. (closes
+ issue #17662) Reported by: oej Review:
+ https://reviewboard.asterisk.org/r/792
+
+2010-07-23 16:19 +0000 [r278957] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h, res/res_config_odbc.c,
+ configs/extconfig.conf.sample, CHANGES, main/config.c,
+ res/res_odbc.c, configs/res_odbc.conf.sample: Merge the realtime
+ failover branch
+
+2010-07-23 16:07 +0000 [r278947] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * doc/asterisk.8: Some left-over hyphen-minus fixes in the man page
+
+2010-07-23 15:57 +0000 [r278944-278945] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: ... just kidding. Enable SIP by default. :-)
+
+ * channels/chan_sip.c: Disable SIP support by default for Asterisk
+ 1.8.
+
+2010-07-23 15:52 +0000 [r278943] Mark Michelson <mmichelson@digium.com>
+
+ * addons/chan_ooh323.c: Well, who knew chan_ooh323 used udptl? I
+ sure didn't!
+
+2010-07-23 15:41 +0000 [r278942] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Rename sig_pri_pri to sig_pri_span. More descriptive of concept.
+
+2010-07-23 15:16 +0000 [r278908] Mark Michelson <mmichelson@digium.com>
+
+ * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h,
+ channels/sip/include/sip.h: Allow IPv6 addresses for UDPTL
+ streams. Review: https://reviewboard.asterisk.org/r/795
+
+2010-07-23 13:37 +0000 [r278875] Olle Johansson <oej@edvina.net>
+
+ * res/res_config_ldap.c: Minor corrections to the LDAP realtime
+ driver Review: https://reviewboard.asterisk.org/r/798/ Thanks
+ Mark for a quick review!
+
+2010-07-23 13:26 +0000 [r278873] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile, agi/Makefile, sounds/Makefile: Portability updates for
+ Makefiles. When possible, use $(INSTALL). This allows us to use
+ the functionality within install for setting directory / file
+ permissions, a requirement for unprivileged installation. Also
+ move any directory we plan to create within the installdirs
+ macro. Plus various other formatting issues. (issue #17436)
+ Reported by: pabelanger Patches: non-root.patch.v8 uploaded by
+ pabelanger (license 224) Tested by: pabelanger Review:
+ https://reviewboard.asterisk.org/r/654/
+
+2010-07-23 11:01 +0000 [r278809-278841] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: missed FXS kewl
+ start polarityswitch when finally on hook. (issue #17318)
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+ channels/sig_analog.c, channels/sig_analog.h: Support FXS module
+ Polarity Reversal on remote party Answer and Hangup FXS lines
+ normally connect to a telephone. However, when FXS lines are
+ routed to an external PBX or Key System to act as "external" or
+ "CO" lines, it is extremely difficult, if not impossible for the
+ external PBX to know when the call has been disconnected without
+ receiving a polarity reversal on the line. Now using
+ answeronpolarityswitch and hanguponpolarityswitch keywords that
+ previously were used only for FXO ports, now applies like
+ functionality for an FXS port, but from the connected equipment's
+ point of view. (closes issue #17318) Reported by: armeniki
+ Patches: fxs_linepolarity.diff5.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/797/
+
+2010-07-22 21:16 +0000 [r278777] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: DNID not cleared when channel hang up
+ (Affects PRI and SS7) The "dahdi show channels" CLI command still
+ reports the DNID of the previous call even if the call is already
+ hang up. The "dahdi show channels" command of older releases
+ clear the DNID once the channel is hang up. Regression from the
+ sig_analog/sig_pri extraction from chan_dahdi. (closes issue
+ #17623) Reported by: klaus3000 Patches: issue17623.patch uploaded
+ by rmudgett (license 664) Tested by: rmudgett
+
+2010-07-22 19:45 +0000 [r278708] Jeff Peeler <jpeeler@digium.com>
+
+ * main/xmldoc.c: 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 14:58 +0000 [r278620] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: 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 ........
+
+2010-07-22 14:56 +0000 [r278619] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: update sip subscription debug message to a
+ warning message If the Expire header of a SUBSCRIBE is less that
+ our expiremin, a log warning will be displayed.
+
+2010-07-22 05:29 +0000 [r278579] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/doxyref.h: Add the full current set of CDR
+ drivers
+
+2010-07-21 19:16 +0000 [r278539] David Vossel <dvossel@digium.com>
+
+ * tests/test_func_file.c: make func_file unit test's category
+ consistent with other tests
+
+2010-07-21 19:11 +0000 [r278538] Terry Wilson <twilson@digium.com>
+
+ * channels/iax2-parser.h, include/asterisk/crypto.h,
+ main/aescrypt.c (removed), include/asterisk/aes_internal.h
+ (removed), funcs/func_aes.c, res/res_crypto.c, main/aestab.c
+ (removed), main/aesopt.h (removed), include/asterisk/aes.h
+ (removed), main/aeskey.c (removed), pbx/pbx_dundi.c,
+ channels/chan_iax2.c, res/res_crypto.exports.in,
+ pbx/dundi-parser.h: Remove built-in AES code and use optional_api
+ instead Review: https://reviewboard.asterisk.org/r/793/
+
+2010-07-21 18:52 +0000 [r278536] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: send "423 Interval too small" Response to
+ Subscribe with Expires less that min allowed [RFC3265]3.1.6.1....
+ The notifier MAY also check that the duration in the "Expires"
+ header is not too small. If and only if the expiration interval
+ is greater than zero AND smaller than one hour AND less than a
+ notifier- configured minimum, the notifier MAY return a "423
+ Interval too small" error which contains a "Min-Expires" header
+ field. The "Min- Expires" header field is described in SIP [1].
+
+2010-07-21 17:44 +0000 [r278501] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.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_trunk_new.diff uploaded by
+ frawd (license 610) Tested by: frawd Review:
+ https://reviewboard.asterisk.org/r/699/
+
+2010-07-21 16:15 +0000 [r278465] Russell Bryant <russell@digium.com>
+
+ * res/res_timing_pthread.c: 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 15:56 +0000 [r278463] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: 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
+
+2010-07-21 15:54 +0000 [r278426-278462] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax_spandsp.c: Properly show the current page being
+ transfered for 'fax show session'
+
+ * channels/chan_sip.c: Properly set the port number for UDPTL media
+ sessions.
+
+ * res/res_fax.c: Don't print failure status when the remote end
+ hangs up, it may not be an actual failure.
+
+2010-07-21 13:02 +0000 [r278425] Russell Bryant <russell@digium.com>
+
+ * main/features.c, UPGRADE.txt, configs/features.conf.sample:
+ Update documentation for 'comebacktoorigin' in featuers.conf. The
+ documentation for this option did not match the code. Fix that
+ along with some minor cleanups to the code along the way.
+ Document a slight change in behavior (to something that was
+ previously undocumented) in UPGRADE.txt.
+
+2010-07-21 06:45 +0000 [r278393] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Change order so that it more closely
+ matches the related SIP command. (closes issue #17648) Reported
+ by: GMLudo Review: https://reviewboard.asterisk.org/r/789/
+
+2010-07-21 03:53 +0000 [r278361] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: include stat.h for everybody, needed for
+ device2chan
+
+2010-07-20 23:23 +0000 [r278275-278307] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, main/logger.c, CHANGES,
+ contrib/realtime/mysql/queue_log.sql (added),
+ configs/logger.conf.sample: Separate queue_log arguments into
+ separate fields, and allow the text file to be used, even when
+ realtime is used. (closes issue #17082) Reported by: coolmig
+ Patches: 20100720__issue17082.diff.txt uploaded by tilghman
+ (license 14) Tested by: coolmig
+
+ * /, apps/app_voicemail.c: 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) ........
+
+2010-07-20 22:38 +0000 [r278274] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Reference correct struct member for unlikely
+ event PRI_EVENT_CONFIG_ERR.
+
+2010-07-20 22:26 +0000 [r278272] Tilghman Lesher <tlesher@digium.com>
+
+ * main/autoservice.c, /, main/features.c,
+ include/asterisk/channel.h: 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) ........
+
+2010-07-20 21:41 +0000 [r278234] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes sip CANCEL race condition If Asterisk
+ sends a 4xx error and the other side sends a CANCEl before
+ receiving the 4xx and responding with the ACK, Asterisk will
+ process the CANCEL and send a 487 Request Terminated as a new
+ final response to the INVITE. Since we are issuing a new final
+ response to the INVITE, the old one must be pretend_acked else it
+ will keep retransmitting.
+
+2010-07-20 21:01 +0000 [r278168] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: This commit contains several changes to the way
+ output channel variables are handled. FAX output channel
+ variables will now match the values reported by FAXOPT() and
+ should be set in all failure and success cases. This commit also
+ contains a few modifications to the way FAXOPT() variables are
+ populated in a few spots and fixes for some reference count leaks
+ of the session details structure in some failure cases. Also
+ found and fixed more cases where FAXOPT(status) may not have
+ gotten set. FAX-214 FAX-203
+
+2010-07-20 19:35 +0000 [r278132] Tilghman Lesher <tlesher@digium.com>
+
+ * cel/cel_pgsql.c, cdr/cdr_sqlite3_custom.c, channels/chan_local.c,
+ res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c,
+ res/res_calendar_caldav.c, formats/format_sln16.c,
+ formats/format_wav_gsm.c, channels/chan_iax2.c, main/config.c,
+ main/loader.c, res/res_rtp_multicast.c, channels/chan_dahdi.c,
+ res/res_smdi.c, channels/chan_skinny.c,
+ include/asterisk/module.h, formats/format_pcm.c,
+ channels/chan_alsa.c, formats/format_h263.c, res/res_curl.c,
+ cdr/cdr_odbc.c, formats/format_jpeg.c, res/res_speech.c,
+ formats/format_gsm.c, cdr/cdr_manager.c, formats/format_g719.c,
+ res/res_calendar_exchange.c, cel/cel_tds.c, formats/format_wav.c,
+ channels/chan_bridge.c, channels/chan_agent.c,
+ formats/format_ogg_vorbis.c, res/res_monitor.c,
+ res/res_calendar_ews.c, res/res_config_curl.c,
+ channels/chan_misdn.c, funcs/func_curl.c,
+ res/res_timing_kqueue.c, formats/format_g726.c, main/asterisk.c,
+ res/res_odbc.c, cel/cel_adaptive_odbc.c, res/res_calendar.c,
+ cel/cel_radius.c, channels/chan_multicast_rtp.c,
+ apps/app_meetme.c, formats/format_sln.c, res/res_musiconhold.c,
+ channels/chan_gtalk.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
+ res/res_jabber.c, res/res_config_sqlite.c,
+ formats/format_siren7.c, cdr/cdr_csv.c, formats/format_ilbc.c,
+ res/res_config_odbc.c, cel/cel_manager.c, cel/cel_custom.c,
+ cdr/cdr_sqlite.c, res/res_agi.c, res/res_timing_timerfd.c,
+ apps/app_confbridge.c, formats/format_h264.c,
+ res/res_config_ldap.c, addons/chan_mobile.c,
+ formats/format_siren14.c, cdr/cdr_custom.c, channels/chan_mgcp.c,
+ res/res_rtp_asterisk.c, res/res_config_pgsql.c,
+ res/res_calendar_icalendar.c, channels/chan_sip.c,
+ cdr/cdr_syslog.c, res/res_fax.c, res/res_crypto.c,
+ res/res_adsi.c, include/asterisk/config.h, pbx/pbx_lua.c,
+ channels/chan_console.c, apps/app_queue.c, cdr/cdr_tds.c,
+ res/res_srtp.c, channels/chan_jingle.c, formats/format_vox.c,
+ res/res_timing_pthread.c, channels/chan_h323.c,
+ cel/cel_sqlite3_custom.c, formats/format_g723.c,
+ funcs/func_devstate.c, formats/format_g729.c,
+ addons/res_config_mysql.c: Add load priority order, such that
+ preload becomes unnecessary in most cases
+
+2010-07-20 18:11 +0000 [r278051-278096] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/install_prereq: Add a package to install_prereq.
+
+ * channels/chan_local.c: Only call ast_channel_cc_params_init() if
+ allocating a channel succeeds.
+
+2010-07-20 16:50 +0000 [r278024] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: 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 21:07 +0000 [r277945] Jean Galarneau <jgalarneau@digium.com>
+
+ * /, main/features.c: 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) ........
+
+2010-07-19 17:16 +0000 [r277872-277873] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample,
+ channels/sip/include/sip.h: Fix port setting of external address
+ in SIP. There are two changes here: 1. Since the externip setting
+ can now have a port attached to it, calling it "externip" is
+ misleading. The option is now documented and parsed as
+ "externaddr." This also extends to the "matchexterniplocally"
+ setting. It is now documented and parsed as
+ "matchexternaddrlocally." The old names for the options may still
+ be used, but they are no longer used in the sip.conf.sample file.
+ 2. If no port is set for the externaddr, and UDP is the transport
+ to be used, then we will set the port of the externaddr to that
+ of the udpbindaddr. This was how things worked prior to the IPv6
+ merge, so this is a regression fix. (closes issue #17665)
+ Reported by: mmichelson Patches: 17665.diff#2 uploaded by
+ pprindeville (license 347) Tested by: pprindeville
+
+ * tests/test_acl.c: Remove the fe80:1234::1234 test case from
+ test_acl.c The ACL test was failing on Mac OS X because it would
+ convert the above invalid link-local address into fe80::1234
+ while reporting no error from getaddrinfo(). Linux does not do
+ this.
+
+2010-07-19 14:39 +0000 [r277837] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Fix regression with distinctive ring
+ detection. The issue here is that passing an array to a function
+ prohibits the ARRAY_LEN macro from returning the real size. To
+ avoid this the size is now defined and use of ARRAY_LEN is
+ avoided. (closes issue #15718) Reported by: alecdavis Patches:
+ bug15718.patch uploaded by jpeeler (license 325)
+
+2010-07-19 14:17 +0000 [r277814] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/acl.h, main/netsock2.c, main/manager.c,
+ channels/chan_sip.c, channels/chan_skinny.c, tests/test_acl.c,
+ main/acl.c, include/asterisk/netsock2.h, configs/sip.conf.sample,
+ channels/chan_iax2.c: Make ACLs IPv6-capable. ACLs can now be
+ configured to match IPv6 networks. This is only relevant for ACLs
+ in chan_sip for now since other channel drivers do not support
+ IPv6 addressing. However, once those channel drivers are
+ outfitted to support IPv6 addressing, the ACLs will already be
+ ready for IPv6 support. https://reviewboard.asterisk.org/r/791
+
+2010-07-17 17:42 +0000 [r277773-277775] Tilghman Lesher <tlesher@digium.com>
+
+ * /, autoconf/ast_func_fork.m4, configure,
+ include/asterisk/autoconfig.h.in: 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, /,
+ include/asterisk/config.h, main/config.c,
+ addons/res_config_mysql.c: 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 ........
+
+2010-07-17 13:10 +0000 [r277703] Russell Bryant <russell@digium.com>
+
+ * Makefile, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, makeopts.in: Allow xmllint to be used for XML docs
+ validation. xmllint seems to be more commonly available since it
+ comes with libxml2.
+
+2010-07-17 00:03 +0000 [r277667] Bradley Latus <brad.latus@gmail.com>
+
+ * res/res_fax.c: Update res_fax.c to be a good xml citizen. (closes
+ issues #17667) Reported by: snuffy
+
+2010-07-16 23:23 +0000 [r277657] Tim Ringenbach <tim.ringenbach@gmail.com>
+
+ * main/features.c: 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 21:24 +0000 [r277530] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: 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 [r277488] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c: 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 [r277484] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/sched.h, main/sched.c: Finally, a method that
+ really fixes the assertions in chan_iax2.c related to cancelling
+ lagid. No, replacing usleep(1) with sched_yield() did not have an
+ effect.
+
+2010-07-16 20:27 +0000 [r277467] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: 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 20:25 +0000 [r277452] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
+ (added): Add documentation for MOH realtime fields
+
+2010-07-16 19:32 +0000 [r277409] Matthew Nicholson <mnicholson@digium.com>
+
+ * tests/test_devicestate.c: updated devicestate test for device
+ state changes
+
+2010-07-16 19:22 +0000 [r277366] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c: 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 18:31 +0000 [r277331] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: 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 [r277263] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: 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 18:05 +0000 [r277250-277262] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c: Print f->subclass.integer instead of f->subclass.
+ (fix build breakage introduced in r277250)
+
+ * main/channel.c, /: 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 17:13 +0000 [r277183] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_amd.c: 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 16:25 +0000 [r277175] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/reqresp_parser.c: Fix up some weird indentation
+ problems in reqresp_parser.c
+
+2010-07-16 15:20 +0000 [r277143] Sean Bright <sean@malleable.com>
+
+ * main/translate.c: 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
+
+2010-07-16 13:40 +0000 [r277103] Eliel C. Sardanons <eliels@gmail.com>
+
+ * CREDITS: Add Despegar.com (my main sponsor) to the CREDITS file.
+
+2010-07-16 13:32 +0000 [r276950-277102] Olle Johansson <oej@edvina.net>
+
+ * main/dnsmgr.c, main/srv.c: Formatting changes
+
+ * channels/chan_sip.c: Formatting fixes
+
+ * configs/sip.conf.sample: Clarify syntax changes
+
+ * CREDITS: Adding a few more to the list of CREDITS
+
+ * channels/chan_sip.c: Formatting changes (guideline corrections)
+ Found a unused bag of curly brackets under my table. I always
+ wondered where they had gone. They where indeed needed in
+ chan_sip.c
+
+ * CREDITS: Adding a few more credits
+
+ * channels/chan_sip.c, doc/tex/channelvariables.tex,
+ configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Add
+ ability to configure the Max-Forwards header in the dialplan, as
+ well as in sip.conf configuration for the channel and for
+ devices. The Max-Forwards header is used to prevent loops in a
+ SIP network. Each intermediary, like SIP proxys and SBCs,
+ decrement this counter and detects when it reaches zero, at which
+ point the SIP request is nicely killed in a SIP-friendly way.
+ Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel
+ for the review and good advice.
+
+ * CHANGES, apps/app_queue.c: Add a dialplan function to check if a
+ queue exists: QUEUE_EXISTS Review:
+ https://reviewboard.asterisk.org/r/777/
+
+2010-07-16 06:04 +0000 [r276910-276911] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_jabber.c: And yet one more
+
+ * res/res_jabber.c: "Item may be used uninitialized in this
+ function."
+
+2010-07-16 05:42 +0000 [r276909] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix reversed logic of if statement. Found
+ based on message from Philip Prindeville on the Asterisk
+ Developers mailing list.
+
+2010-07-16 05:38 +0000 [r276830-276908] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Detect the --dynamic-list flag a bit
+ better
+
+ * configure, main/Makefile, configure.ac, makeopts.in: Fix build on
+ FreeBSD
+
+ * tests/test_utils.c: Fix trunk build for Mac OS X 10.6
+
+ * contrib/realtime/mysql/iaxfriends.sql,
+ contrib/realtime/mysql/meetme.sql,
+ contrib/realtime/postgresql/realtime.sql,
+ contrib/realtime/mysql/sipfriends.sql: Allow ipaddress to contain
+ the maximum IPv6 address. Also, update meetme to the full list of
+ supported fields.
+
+ * configure, autoconf/ast_gcc_attribute.m4: Quote AC_SUBST within
+ m4_ifval, so it does not get prematurely expanded. (closes issue
+ #17654) Reported by: pprindeville Patches: issue17654.diff
+ uploaded by qwell (license 4) Tested by: qwell, pprindeville
+
+2010-07-15 20:21 +0000 [r276788] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: 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.
+
+2010-07-15 19:46 +0000 [r276731-276769] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, configure.ac: Define LLONG_MAX on
+ systems that do not have it. (closes issue #17644) Reported by:
+ pprindeville
+
+ * configure, main/Makefile, autoconf/ast_gcc_attribute.m4,
+ configure.ac, makeopts.in: Fix linking asterisk on CentOS 5,
+ which is using gcc 4.1.1. Gcc 4.1.2 has the real fix. Review:
+ https://reviewboard.asterisk.org/r/790/
+
+2010-07-15 13:51 +0000 [r276653] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: 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. ........
+
+2010-07-15 12:21 +0000 [r276616] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/install_prereq: Add lua5.1 to the handy dandy
+ list of packages.
+
+2010-07-14 22:58 +0000 [r276571] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: 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 22:32 +0000 [r276570] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
+ main/acl.c: Fix errors where incorrect address information was
+ printed. ast_sockaddr_stringiy_fmt (which is call by all
+ ast_sockaddr_stringify* functions) uses thread-local storage for
+ storing the string that it creates. In cases where
+ ast_sockaddr_stringify_fmt was being called twice within the same
+ statement, the result of one call would be overwritten by the
+ result of the other call. This usually was happening in
+ printf-like statements and was resulting in the same stringified
+ addressed being printed twice instead of two separate addresses.
+ I have fixed this by using ast_strdupa on the result of stringify
+ functions if they are used twice within the same statement. As
+ far as I could tell, there were no instances where a pointer to
+ the result of such a call were saved anywhere, so this is the
+ only situation I could see where this error could occur.
+
+2010-07-14 21:29 +0000 [r276531] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_h323.c: Make compile again.
+
+2010-07-14 21:11 +0000 [r276490-276493] Tilghman Lesher <tlesher@digium.com>
+
+ * main/loader.c: Oops, merge reverted this fix.
+
+ * include/asterisk/adsi.h, include/asterisk/agi.h,
+ include/asterisk/crypto.h, main/asterisk.dynamics, main/Makefile,
+ tests/test_utils.c, main/adsistub.c (removed), main/cryptostub.c
+ (removed), res/res_adsi.c, res/res_crypto.c,
+ res/res_crypto.exports.in (added), res/res_adsi.exports.in,
+ main/loader.c, include/asterisk/optional_api.h: Remove the old
+ stub files, preferring the optional_api method. (closes issue
+ #17475) Reported by: tilghman Review:
+ https://reviewboard.asterisk.org/r/695/
+
+2010-07-14 20:15 +0000 [r276441] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/loader.c: Don't try to call an embedded module's
+ backup_globals() function until after confirming it exists.
+
+2010-07-14 19:51 +0000 [r276439] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: handle special case were "200 Ok" to pending
+ INVITE never receives ACK Unlike most responses, the 200 Ok to a
+ pending INVITE Request is acknowledged by an ACK Request. If the
+ ACK Request for this Response is not received the previous
+ behavior was to immediately destroy the dialog and hangup the
+ channel. Now in an effort to be more RFC compliant, instead of
+ immediately destroying the dialog during this special case,
+ termination is done with a BYE Request as the dialog is
+ technically confirmed when the 200 Ok is sent even if the ACK is
+ never received. The behavior of immediately hanging up the
+ channel remains. This only affects how dialog termination
+ proceeds for this one special case. RFC 3261 section 13.3.1.4 "If
+ the server retransmits the 2xx response for 64*T1 seconds without
+ receiving an ACK, the dialog is confirmed, but the session SHOULD
+ be terminated. This is accomplished with a BYE, as described in
+ Section 15."
+
+2010-07-14 16:58 +0000 [r276393] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_vpb.cc, channels/chan_sip.c,
+ include/asterisk/channel.h, channels/sig_pri.c,
+ channels/chan_iax2.c, main/cel.c, channels/chan_oss.c,
+ main/channel.c, main/cdr.c, channels/chan_jingle.c,
+ channels/chan_usbradio.c, channels/chan_dahdi.c,
+ channels/chan_phone.c, channels/sig_analog.c,
+ channels/chan_misdn.c, channels/chan_skinny.c,
+ channels/chan_h323.c, res/snmp/agent.c, apps/app_amd.c,
+ funcs/func_callerid.c, channels/sig_ss7.c, channels/chan_mgcp.c:
+ Expand the caller ANI field to an ast_party_id Expand the ani
+ field in ast_party_caller and ast_party_connected_line to an
+ ast_party_id. This is an extension to the ast_callerid
+ restructuring patch in review:
+ https://reviewboard.asterisk.org/r/702/ Review:
+ https://reviewboard.asterisk.org/r/744/
+
+2010-07-14 16:40 +0000 [r276392] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: collapse debug code in retrans_pkt into
+ separate lines I've been working in this function a bunch lately,
+ and these huge debug strings are getting annoying.
+
+2010-07-14 16:39 +0000 [r276391] Richard Mudgett <rmudgett@digium.com>
+
+ * res/snmp/agent.c: Make compile again.
+
+2010-07-14 16:36 +0000 [r276389] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Do not skip sending MWI for a peer if an
+ address is defined. Really just a merge mistake from IPv6
+
+2010-07-14 16:09 +0000 [r276349] Tim Ringenbach <tim.ringenbach@gmail.com>
+
+ * cel/cel_pgsql.c, doc/tex/celdriver.tex, doc/tex/cdrdriver.tex:
+ Fix documentation for pgsql cel and cdr, and slightly improve
+ pgsql_cel. Change the documented pgsql schema to use "timestamp"
+ instead of "time", as the latter is only a time without a date.
+ Added some missing columns for cel's pgsql schema, and corrected
+ spelling on some others. Updated cel's uniqueid size to be the
+ same as the cdr. Added id column to cel's pgsql schema and
+ updated code to allow unknown columns to get their default value
+ instead of forcing 0 or empty string. Added microseconds to the
+ timestamp cel logs to pgsql. Review:
+ https://reviewboard.asterisk.org/r/734
+
+2010-07-14 15:48 +0000 [r276347] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c, addons/chan_ooh323.c,
+ apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c,
+ channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c,
+ channels/sig_analog.h, apps/app_amd.c, channels/sig_ss7.c,
+ apps/app_dial.c, main/pbx.c, apps/app_privacy.c, apps/app_fax.c,
+ channels/chan_agent.c, apps/app_disa.c,
+ include/asterisk/channel.h, apps/app_talkdetect.c, main/cel.c,
+ funcs/func_redirecting.c (removed), channels/chan_misdn.c,
+ apps/app_macro.c, apps/app_zapateller.c, apps/app_voicemail.c,
+ channels/chan_unistim.c, tests/test_substitution.c,
+ channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c,
+ apps/app_readexten.c, channels/chan_gtalk.c, apps/app_followme.c,
+ include/asterisk/callerid.h, main/cdr.c, main/channel.c,
+ channels/chan_phone.c, main/dial.c, apps/app_setcallerid.c,
+ apps/app_osplookup.c, main/manager.c, apps/app_minivm.c,
+ res/res_agi.c, main/app.c, apps/app_rpt.c, channels/chan_mgcp.c,
+ apps/app_parkandannounce.c, apps/app_while.c,
+ funcs/func_dialplan.c, channels/chan_sip.c, UPGRADE.txt,
+ channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c,
+ channels/chan_oss.c, channels/chan_usbradio.c,
+ channels/chan_jingle.c, funcs/func_blacklist.c,
+ apps/app_directed_pickup.c, main/file.c,
+ funcs/func_connectedline.c (removed), channels/chan_h323.c,
+ main/callerid.c, res/snmp/agent.c, apps/app_sms.c,
+ apps/app_stack.c, funcs/func_callerid.c: ast_callerid
+ restructuring The purpose of this patch is to eliminate struct
+ ast_callerid since it has turned into a miscellaneous collection
+ of various party information. Eliminate struct ast_callerid and
+ replace it with the following struct organization: struct
+ ast_party_name { char *str; int char_set; int presentation;
+ unsigned char valid; }; struct ast_party_number { char *str; int
+ plan; int presentation; unsigned char valid; }; struct
+ ast_party_subaddress { char *str; int type; unsigned char
+ odd_even_indicator; unsigned char valid; }; struct ast_party_id {
+ struct ast_party_name name; struct ast_party_number number;
+ struct ast_party_subaddress subaddress; char *tag; }; struct
+ ast_party_dialed { struct { char *str; int plan; } number; struct
+ ast_party_subaddress subaddress; int transit_network_select; };
+ struct ast_party_caller { struct ast_party_id id; char *ani; int
+ ani2; }; The new organization adds some new information as well.
+ * The party name and number now have their own presentation value
+ that can be manipulated independently. ISDN supplies the
+ presentation value for the name and number at different times
+ with the possibility that they could be different. * The party
+ name and number now have a valid flag. Before this change the
+ name or number string could be empty if the presentation were
+ restricted. Most channel drivers assume that the name or number
+ is then simply not available instead of indicating that the name
+ or number was restricted. * The party name now has a character
+ set value. SIP and Q.SIG have the ability to indicate what
+ character set a name string is using so it could be presented
+ properly. * The dialed party now has a numbering plan value that
+ could be useful to have available. The various channel drivers
+ will need to be updated to support the new core features as
+ needed. They have simply been converted to supply current
+ functionality at this time. The following items of note were
+ either corrected or enhanced: * The CONNECTEDLINE() and
+ REDIRECTING() dialplan functions were consolidated into
+ func_callerid.c to share party id handling code. * CALLERPRES()
+ is now deprecated because the name and number have their own
+ presentation values. * Fixed app_alarmreceiver.c
+ write_metadata(). The workstring[] could contain garbage. It also
+ can only contain the caller id number so using
+ ast_callerid_parse() on it is silly. There was also a typo in the
+ CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse()
+ on the channel's caller id number string. ast_callerid_parse()
+ alters the given buffer which in this case is the channel's
+ caller id number string. Then using ast_shrink_phone_number()
+ could alter it even more. * Fixed caller ID name and number
+ memory leak in chan_usbradio.c. * Fixed uninitialized char arrays
+ cid_num[] and cid_name[] in sig_analog.c. * Protected access to a
+ caller channel with lock in chan_sip.c. * Clarified intent of
+ code in app_meetme.c sla_ring_station() and dial_trunk(). Also
+ made save all caller ID data instead of just the name and number
+ strings. * Simplified cdr.c set_one_cid(). It hand coded the
+ ast_callerid_merge() function. * Corrected some weirdness with
+ app_privacy.c's use of caller presentation. Review:
+ https://reviewboard.asterisk.org/r/702/
+
+2010-07-14 11:51 +0000 [r276268] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/voicemail.conf.sample: 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 22:18 +0000 [r276219] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: chan_sip: RFC
+ compliant retransmission timeout Retransmission of packets should
+ not be based on how many packets were sent, but instead on a
+ timeout period. Depending on whether or not the packet is for a
+ INVITE or NON-INVITE transaction, the number of packets sent
+ during the retransmission timeout period will be different, so
+ timing out based on the number of packets sent is not accurate.
+ This patch fixes this by removing the retransmit limit and only
+ stopping retransmission after a timeout period is reached. By
+ default this timeout period is 64*(Timer T1) for both INVITE and
+ non-INVITE transactions. For more information on sip timer values
+ refer to RFC3261 Appendix A. Review:
+ https://reviewboard.asterisk.org/r/749/
+
+2010-07-13 21:42 +0000 [r276206] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/include/dialog.h, channels/chan_sip.c: Revert early
+ destruction of RTP sessions Some code improperly assumes that the
+ sessions are still there, so revert the change until I can find
+ all of them and fix them.
+
+2010-07-13 19:15 +0000 [r276124-276127] Russell Bryant <russell@digium.com>
+
+ * /: Recorded merge of revisions 276126 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r276126 | russell | 2010-07-13 14:14:54 -0500 (Tue, 13 Jul 2010)
+ | 2 lines Only reset a CDR that exists. ........
+
+ * /, main/features.c: 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 19:05 +0000 [r276114-276122] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c: Oops, XML documentation fix.
+
+ * funcs/func_env.c: It really cannot fail in the places below, but
+ the stupid compiler doesn't know that.
+
+ * funcs/func_env.c: Weird compiler error on Bamboo.
+
+ * funcs/func_env.c, CHANGES, tests/test_func_file.c (added): FILE()
+ now supports line-mode and writing (altering) files. (closes
+ issue #16461) Reported by: skyman Patches:
+ 20100622__issue16461.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/737/
+
+2010-07-13 17:37 +0000 [r276074] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: 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 17:11 +0000 [r275998] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/include/dialog.h, channels/chan_sip.c: Destroy RTP
+ fds when we schedule final dialog destruction Since we are only
+ keeping the dialog around for retransmissions at this point and
+ there is no possibility that we are still handling RTP, go ahead
+ and destroy the RTP sessions. Keeping them alive for 32 past when
+ they are used is unnecessary and can lead to problems with having
+ too many open file descriptors, etc.
+
+2010-07-13 16:53 +0000 [r275995] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: 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 14:48 +0000 [r275910] Tilghman Lesher <tlesher@digium.com>
+
+ * 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, 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: 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. ........
+
+2010-07-13 11:41 +0000 [r275863] Russell Bryant <russell@digium.com>
+
+ * configs/voicemail.conf.sample,
+ contrib/scripts/voicemailpwcheck.py (added): Add example script
+ for use with the externpasscheck voicemail.conf option. (closes
+ issue #17628) Reported by: lmadsen Tested by: russell, lmadsen
+ Review: https://reviewboard.asterisk.org/r/774/
+
+2010-07-12 23:27 +0000 [r275816] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't try to ref authpeer when it isn't set
+
+2010-07-12 17:54 +0000 [r275725] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Add which ITU spec specifies the numbering plan.
+
+2010-07-12 17:21 +0000 [r275682] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: 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 15:37 +0000 [r275626] Leif Madsen <lmadsen@digium.com>
+
+ * cdr/cdr_pgsql.c: 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
+
+2010-07-12 14:55 +0000 [r275587] Mark Michelson <mmichelson@digium.com>
+
+ * main/netsock2.c: Allow netsock2.c to compile on systems that do
+ not define AI_NUMERICSERV. (closes issue #17617) Reported by:
+ pprindeville Patches: asterisk-trunk-bugid17617.patch uploaded by
+ pprindeville (license 347)
+
+2010-07-12 04:16 +0000 [r275551] TransNexus OSP Development <support@transnexus.com>
+
+ * configs/osp.conf.sample, apps/app_osplookup.c: Added support for
+ indirect work mode.
+
+2010-07-10 20:49 +0000 [r275509] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_meetme.c: When creating a conference for a unit test, it
+ is not mandatory to open a dahdi pseudo channel, so if we fail
+ doing it, continue creating the conference.
+
+2010-07-10 14:48 +0000 [r275424-275467] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Make indentation consistent, move some queue features to
+ the queue section.
+
+ * CREDITS, channels/chan_unistim.c, configs/unistim.conf.sample,
+ CHANGES: Add support for devices with less than 3 lines on the
+ LCD. (closes issue #17600) Reported by: minaguib Patches:
+ ast_unistim_height_v2.patch uploaded by minaguib (license 1078)
+ Tested by: minaguib
+
+ * main/features.c, configs/features.conf.sample: 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
+
+2010-07-09 20:58 +0000 [r275385] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix error in parsing SIP registry strings
+ from ASTdb. It was essentially an off-by-one error. The easiest
+ way to fix this was to use the handy-dandy
+ AST_NONSTANDARD_RAW_ARGS macro to parse the pieces of the
+ registration string out. Tested and it works wonderfully.
+
+2010-07-09 20:01 +0000 [r275312] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, channels/chan_iax2.c: Get more information
+ about the Bamboo test failures
+
+2010-07-09 19:58 +0000 [r275309-275310] Russell Bryant <russell@digium.com>
+
+ * main/features.c: Add missing ao2_iterator_destroy().
+
+ * apps/app_voicemail.c: Fix compile error.
+
+2010-07-09 19:46 +0000 [r275308] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix port parsing in check_via. If a Via
+ header contained an IPv6 address, we would not properly parse the
+ port. We would instead get the information after the first colon
+ in the address. (closes issue #17614) Reported by: oej Patches:
+ diff uploaded by sperreault (license 252)
+
+2010-07-09 19:32 +0000 [r275307] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * CHANGES, apps/app_voicemail.c: Include rdnis in msgXXXX.txt file.
+ (closes issue #17566) Reported by: outcast Patches:
+ voicemail-rdnis.patch uploaded by outcast (license 1071) Tested
+ by: outcast
+
+2010-07-09 19:29 +0000 [r275294] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix an issue where the port for p->ourip was
+ being set to 0. This should fix all the CDR tests that were not
+ passing. When they would originate a call, all fields in the
+ INVITE that contained the source port would have the port set to
+ 0. Most troubling of these was the Contact header. Tests are
+ passing locally now and should also pass on the bamboo build
+ agents.
+
+2010-07-09 19:21 +0000 [r275249] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: 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 18:55 +0000 [r275227] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, channels/chan_iax2.c: Weird, no output and
+ Bamboo still fails...
+
+2010-07-09 18:24 +0000 [r275186] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/loader.c: 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 18:21 +0000 [r275172] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, channels/chan_iax2.c: Add some diagnostic
+ feedback to our data tests
+
+2010-07-09 18:11 +0000 [r275147] Russell Bryant <russell@digium.com>
+
+ * configs/features.conf.sample: Move parking lot sample config out
+ from the middle of dynamic features sample config.
+
+2010-07-09 17:50 +0000 [r275144] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/loader.c: 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 ........
+
+2010-07-09 17:00 +0000 [r275105] Tilghman Lesher <tlesher@digium.com>
+
+ * main/netsock2.c, tests/test_substitution.c, tests/test_heap.c,
+ apps/app_meetme.c, tests/test_gosub.c, funcs/func_strings.c,
+ tests/test_event.c, channels/sip/reqresp_parser.c,
+ channels/chan_iax2.c, tests/test_stringfields.c,
+ tests/test_time.c, tests/test_devicestate.c, tests/test_utils.c,
+ main/features.c, res/res_agi.c, include/asterisk/netsock2.h,
+ tests/test_astobj2.c, channels/chan_sip.c,
+ tests/test_ast_format_str_reduce.c, tests/test_app.c,
+ funcs/func_math.c, include/asterisk/channel.h,
+ tests/test_sched.c, tests/test_pbx.c, tests/test_strings.c,
+ main/data.c, tests/test_skel.c, tests/test_acl.c,
+ channels/sip/dialplan_functions.c, tests/test_aoc.c, main/test.c,
+ channels/sip/config_parser.c, res/res_timing_kqueue.c,
+ apps/app_voicemail.c: Kill some startup warnings and errors and
+ make some messages more helpful in tracking down the source.
+
+2010-07-09 16:39 +0000 [r275104] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Return logic of sip_debug_test_addr() to its
+ original functionality.
+
+2010-07-09 16:05 +0000 [r275028] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_dial.c, /: 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 15:35 +0000 [r275022] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/test.h, /, main/test.c: 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. ........
+
+2010-07-09 14:27 +0000 [r274984] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/reqresp_parser.c: Fix sip_uri_parse test comparison.
+ Part of the change with the IPv6 changes is to treat a host:port
+ as a single 'domain' entity. This test was not updated to have
+ the correct expectation after calling parse_uri().
+
+2010-07-09 13:30 +0000 [r274909-274947] <simon.perreault@viagenie.ca>
+
+ * channels/chan_sip.c: Copy the address into the peer structure
+ after we set the default port
+
+ * main/netsock2.c: Sadly we can't dereference a pointer cast and
+ use it as an lvalue without getting this warning (at least with
+ gcc 4.4.4): netsock2.c:492: warning: dereferencing pointer
+ ‘({anonymous})’ does break strict-aliasing rules So we're back to
+ using memcpy()...
+
+2010-07-09 12:48 +0000 [r274907] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/indications.h: Extend length limit on country
+ name in indications.conf.
+
+2010-07-09 11:06 +0000 [r274866] Olle Johansson <oej@edvina.net>
+
+ * configs/cdr.conf.sample, cdr/cdr_csv.c: Make it possible to
+ disable individual cdr files per accountcode in cdr_csv Review:
+ https://reviewboard.asterisk.org/r/678/
+
+2010-07-08 23:46 +0000 [r274827-274828] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_jingle.c, channels/chan_h323.c,
+ channels/chan_gtalk.c: Fix calls of ast_sockaddr_from_sin() from
+ IPv6 integration.
+
+ * addons/chan_ooh323.c: Fix compile of chan_ooh323.c from IPv6
+ integration.
+
+2010-07-08 22:16 +0000 [r274783-274786] Mark Michelson <mmichelson@digium.com>
+
+ * /: And the automerge property.
+
+ * /: Delete properties I merged during v6-new merge.
+
+ * channels/chan_unistim.c, include/asterisk/acl.h, main/netsock2.c
+ (added), channels/sip/include/dialog.h,
+ channels/chan_multicast_rtp.c, addons/chan_ooh323.c,
+ main/rtp_engine.c, /, channels/sip/reqresp_parser.c,
+ include/asterisk/tcptls.h, channels/chan_gtalk.c,
+ channels/chan_iax2.c, main/config.c, res/res_rtp_multicast.c,
+ main/manager.c, channels/chan_skinny.c,
+ channels/sip/include/globals.h, main/http.c, main/app.c,
+ include/asterisk/netsock2.h (added), apps/app_externalivr.c,
+ configs/sip.conf.sample, include/asterisk/rtp_engine.h,
+ channels/sip/include/sip.h, channels/chan_mgcp.c,
+ channels/sip/include/reqresp_parser.h, res/res_rtp_asterisk.c,
+ main/dnsmgr.c, channels/chan_sip.c, include/asterisk/config.h,
+ main/acl.c, CHANGES, channels/chan_jingle.c, main/tcptls.c,
+ channels/sip/dialplan_functions.c, channels/chan_h323.c,
+ include/asterisk/dnsmgr.h: Add IPv6 to Asterisk. This adds a
+ generic API for accommodating IPv6 and IPv4 addresses within
+ Asterisk. While many files have been updated to make use of the
+ API, chan_sip and the RTP code are the files which actually
+ support IPv6 addresses at the time of this commit. The way has
+ been paved for easier upgrading for other files in the near
+ future, though. Big thanks go to Simon Perrault, Marc Blanchet,
+ and Jean-Philippe Dionne for their hard work on this. (closes
+ issue #17565) Reported by: russell Patches:
+ asteriskv6-test-report.pdf uploaded by russell (license 2)
+ Review: https://reviewboard.asterisk.org/r/743
+
+2010-07-08 22:05 +0000 [r274773-274782] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Generate a correct AstData string for
+ ast_callerid.cid_ton
+
+ * main/channel.c: Fix trunk compile.
+
+2010-07-08 14:48 +0000 [r274727] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/pbx.c, channels/chan_sip.c, apps/app_meetme.c,
+ include/asterisk/indications.h, channels/chan_agent.c,
+ include/asterisk/channel.h, include/asterisk/cdr.h,
+ include/asterisk/data.h, channels/chan_iax2.c, apps/app_queue.c,
+ main/indications.c, main/channel.c, main/cdr.c,
+ channels/chan_dahdi.c, main/data.c, res/res_odbc.c,
+ apps/app_voicemail.c: Implement AstData API data providers as
+ part of the GSOC 2010 project, midterm evaluation. Review:
+ https://reviewboard.asterisk.org/r/757/
+
+2010-07-07 20:09 +0000 [r274686] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes some ref count issues introduced by
+ r274539
+
+2010-07-07 18:32 +0000 [r274595-274639] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Add missing conditional around chan_dahdi
+ mfcr2_skip_category config parameter.
+
+ * channels/chan_dahdi.c, /: 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 16:40 +0000 [r274540] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Set proper FAXOPT(status), FAXOPT(statusstr), and
+ FAXOPT(error) values where possible. Previously some failure
+ cases did not result in proper FAXOPT values. FAX-203
+
+2010-07-07 16:21 +0000 [r274539] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Use the relatedpeer field of a sip_pvt
+ during INVITE processing. Review:
+ https://reviewboard.asterisk.org/r/629
+
+2010-07-07 07:07 +0000 [r274492] TransNexus OSP Development <support@transnexus.com>
+
+ * configs/osp.conf.sample, doc/osp.txt: Changed OSP TCP port from
+ 1080 to 5045.
+
+2010-07-07 06:32 +0000 [r274418-274491] Tilghman Lesher <tlesher@digium.com>
+
+ * CHANGES, apps/app_voicemail.c: Also run the externnotify script
+ when the pollmailboxes thread notices a change.
+
+ * /, configs/say.conf.sample: 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 22:23 +0000 [r274316] Jeff Peeler <jpeeler@digium.com>
+
+ * /, configs/sip.conf.sample: 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 22:15 +0000 [r274284] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, UPGRADE.txt: 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/ ........ (no option for
+ trunk, just changing the behavior)
+
+2010-07-06 22:09 +0000 [r274281] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Status shows all non-CRC4 lines as
+ "yellow", even if "yellow" was not in the bitfield.
+
+2010-07-06 19:53 +0000 [r274243] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Properly detect and report invalid maxrate and
+ maxrate values in the FAXOPT dialplan function. Also make
+ fax_rate_str_to_int() return an unsigned int and return 0 instead
+ of -1 in the event of an error. FAX-202
+
+2010-07-06 14:31 +0000 [r274164] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c, /: 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 ........
+
+2010-07-06 06:01 +0000 [r274053] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Uh, yeah.
+
+2010-07-05 13:53 +0000 [r273886] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, main/config.c: 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 02:36 +0000 [r273714-273830] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /, channels/chan_agent.c,
+ channels/chan_h323.c, include/asterisk/lock.h: 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 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: The switch fallthrough could create some
+ errorneous situations, so best to force directly to the default
+ case.
+
+2010-07-02 15:57 +0000 [r273641] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c, channels/chan_misdn.c,
+ channels/chan_sip.c, main/say.c, main/fixedjitterbuf.c,
+ res/res_agi.c, channels/chan_h323.c, main/utils.c,
+ channels/chan_iax2.c, addons/chan_mobile.c, apps/app_rpt.c,
+ channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c,
+ apps/app_while.c: Fix various typos reported by Lintian (Also fix
+ the typos in the comments)
+
+2010-07-01 22:16 +0000 [r273566] Russell Bryant <russell@digium.com>
+
+ * /, main/datastore.c: 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 20:28 +0000 [r273522] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: 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 ........
+
+2010-07-01 19:34 +0000 [r273464] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Properly handle failures of fax->start_session()
+ FAX-177
+
+2010-07-01 16:40 +0000 [r273427] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: correct handling
+ of get_destination return values A failure when calling the
+ get_destination can mean multiple things. If the extension is not
+ found, a 404 error is appropriate, but if the URI scheme is
+ incorrect, a 404 is not approperiate. This patch adds the
+ get_destination_result enum to differentiate between these and
+ other failure types. The only logical difference in this patch is
+ that we now send a "416 Unsupported URI scheme" response instead
+ of a "404" when the scheme is not recognized. This indicates to
+ the initiator of the INVITE to retry the request with a correct
+ URI.
+
+2010-07-01 15:12 +0000 [r273355] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: 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 14:37 +0000 [r273270-273352] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c: Fixed whitespace problems
+
+ * main/manager.c: Altered my comment about TCP_NODELAY
+
+ * addons/chan_mobile.c: Don't free written frames in chan_mobile's
+ mbl_write() function. (closes issue #16430) Reported by: azbest
+ Tested by: azbest
+
+ * main/manager.c: Set TCP_NODELAY on manager TCP sockets to prevent
+ delays on outgoing packets. This regression was introduced in
+ r48338. AST-359
+
+2010-06-30 17:28 +0000 [r273233] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_rtp_asterisk.c: 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
+
+2010-06-30 17:17 +0000 [r273197-273198] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/config.h: Remove unnecessary if test in
+ CV_DSTR()
+
+ * include/asterisk/config.h: Misc doxygen cleanup in config.h
+
+2010-06-30 01:07 +0000 [r273054-273144] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: 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: 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 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 ........
+
+ * /: 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) ........
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Exclude libical for insufficient versions.
+
+ * main/pbx.c: 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 20:44 +0000 [r272981] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: send a 400 Bad Request on malformed sip
+ request RFC 2361 section 24.4.1 send a 400 Bad Request if the
+ request can not be understood due to malformed syntax. Currently
+ we simply ignore a packet with a missing callid, to, from, or via
+ header. Instead of ignoring we now send the 400 Bad request.
+
+2010-06-28 21:50 +0000 [r272923-272926] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/asterisk.c: 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 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 18:38 +0000 [r272880] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c,
+ channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h: rfc compliant sip option
+ parsing + new unit test RFC 3261 section 8.2.2.3 states that if
+ any unsupported options are found in the Require header field, a
+ "420 (Bad Extension)" response should be sent with an Unsupported
+ header field containing only the unsupported options. This is not
+ currently being done correctly. Right now, if Asterisk detects
+ any unsupported sip options in a Require header the entire list
+ of options are returned in the Unsupported header even if some of
+ those options are in fact supported. This patch fixes that by
+ building an unsupported options character buffer when parsing the
+ options that can be sent with the 420 response. A unit test
+ verifying this functionality has been created. Some code
+ refactoring was required. Review:
+ https://reviewboard.asterisk.org/r/680/
+
+2010-06-28 17:33 +0000 [r272805] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: 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 15:33 +0000 [r272684] Russell Bryant <russell@digium.com>
+
+ * doc/tex/chan-mobile.tex (added), doc/tex/celdriver.tex,
+ doc/tex/chan_mobile.tex (removed), doc/tex/cdrdriver.tex,
+ doc/tex/asterisk.tex, doc/tex/cel-doc.tex: Use the underscore
+ package so that underscores do not need to be escaped.
+
+2010-06-28 14:55 +0000 [r272652] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: code guidelines cleanup for retrans_pkt()
+ function I am doing work in this function. I noticed a large
+ number of coding guidline fixes that needed to be made. Rather
+ than have those changes distract from my functional changes I
+ decided to separate these into a separate patch.
+
+2010-06-25 20:18 +0000 [r272568] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/voicemail_odbc_postgresql.txt: 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
+ ........
+
+2010-06-25 19:42 +0000 [r272558] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, include/asterisk/res_fax.h: Implemement support
+ for handling multiple documents when sending.
+
+2010-06-25 19:39 +0000 [r272557] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: chan_sip: more accurate retransmissions
+ RFC3261 states that Timer A should start at 500ms (T1) by
+ default. In chan_sip this value initially started at 1000ms and I
+ changed it to 500ms recently. After doing that I noticed in my
+ packet captures that it still occasionally retransmitted starting
+ at 1000ms instead of 500ms like I told it to. This occurs because
+ the scheduler runs in the do_monitor thread. If a new
+ retransmission is added while the do_monitor thread is sleeping
+ then it may not detect that retransmission for nearly 1000ms. To
+ fix this I just poke the do_monitor thread to wake up when a new
+ packet is sent reliably requiring retransmits. The thread then
+ detects the new scheduler entry and adjusts its sleep time to
+ account for it. Review: https://reviewboard.asterisk.org/r/747
+
+2010-06-25 19:17 +0000 [r272533] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile: Symlink sounds files, to save disk space, when
+ multiple tarballs/checkouts are on the same system.
+
+2010-06-24 22:11 +0000 [r272447] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: 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 23:09 +0000 [r272370] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: 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 22:36 +0000 [r272368] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_queue.c: 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 21:53 +0000 [r272260-272332] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: 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
+
+ * res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael_lex.c,
+ res/ael/ael.flex: 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 21:06 +0000 [r272257-272259] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_meetme.c: Fix previous merge. ast_test_flag !=
+ ast_test_flag64
+
+ * /, apps/app_meetme.c: 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 ........
+
+2010-06-23 20:59 +0000 [r272254-272256] Terry Wilson <twilson@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in: Update configure
+ when changing autconf m4 files...
+
+ * autoconf/ast_ext_tool_check.m4: 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/
+
+2010-06-23 20:35 +0000 [r272243-272252] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/manager.c: Correct manager variable 'EventList' case.
+ (closes issue #17520) Reported by: kobaz Patches: manager.patch
+ uploaded by kobaz (license 834) Tested by: lmadsen
+
+ * configs/say.conf.sample: Add localization support for Spanish
+ (closes issue #17548) Reported by: cjacobsen Patches:
+ say.conf.sample.diff uploaded by cjacobsen (license 1029)
+
+2010-06-23 19:59 +0000 [r272218] Tim Ringenbach <tim.ringenbach@gmail.com>
+
+ * channels/chan_local.c: Add new AMI command LocalOptimizeAway.
+ This command lets you request a "/n" local channel optimize
+ itself out of the way anyway. Review:
+ https://reviewboard.asterisk.org/r/732/
+
+2010-06-23 18:45 +0000 [r272148-272150] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c: D'oh! Defaultenabled FTL.
+
+ * /: Recorded merge of revisions 272147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010)
+ | 5 lines Backport part of revision 136715 to fix callerid in
+ voicemail text files (IMAP only). (closes issue #16945) Reported
+ by: mneuhauser ........
+
+2010-06-23 18:39 +0000 [r272146] Terry Wilson <twilson@digium.com>
+
+ * apps/app_meetme.c: Don't start the sla thread unless we realy
+ need it
+
+2010-06-23 18:25 +0000 [r272145] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c: Load all lines from realtime, not just the
+ first one. (closes issue #17144) Reported by: nahuelgreco
+ Patches: 20100513__issue17144__trunk.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman
+
+2010-06-23 17:21 +0000 [r272109] Terry Wilson <twilson@digium.com>
+
+ * apps/app_meetme.c: 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
+
+2010-06-23 17:08 +0000 [r272090] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add extra protection for reinvite glare
+ scenario. Testing proved that if Asterisk sent a connected line
+ reinvite, and the endpoint to which the reinvite were being sent
+ sent a reinvite, Asterisk would not properly respond with a 491
+ response. The reason is that on connected line reinvites, we set
+ the dialog's invitestate to INV_CALLING to prevent Asterisk from
+ sending a rapid flurry of connected line reinvites. For other
+ reinvites we do not do this. Because of the current invitestate,
+ when Asterisk received the reinvite, we interpreted this as a
+ spiraled INVITE, and thus did not behave properly. The fix for
+ this is to not enter the loop detection or spiral logic in
+ handle_request_invite if the channel state is currently up. This
+ way, no mid-call reinvites will be misinterpreted, no matter what
+ the nature of the reinvite may have been.
+
+2010-06-22 23:20 +0000 [r272052] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c: Don't try to lock/unlock an uninitialized
+ lock on a dahdi_pri. This small changes prevents
+ destroy_all_channels() from accessing a lock on an unused
+ dahdi_pri struct, resolving a ton of ERRORs that get spewed out
+ when shutting Asterisk down gracefully.
+
+2010-06-22 22:11 +0000 [r271905-272014] David Vossel <dvossel@digium.com>
+
+ * pbx/pbx_config.c: fixes issue with 'dialplan remove extension
+ blah' segfaulting with tab completion (closes issue #17440)
+ Reported by: kobaz
+
+ * channels/chan_sip.c: ignore CANCEL request after having already
+ received final response to INVITE RFC 3261 section 9 states that
+ a CANCEL has no effect on a request to a UAS that has already
+ given a final response. This patch checks to make sure there is a
+ pending invite before allowing a CANCEL request to be processed,
+ otherwise it responds to the CANCEL with a "481 Call/Transaction
+ Does Not Exist". Review: https://reviewboard.asterisk.org/r/697/
+
+ * main/manager.c: minor fixes for white/black event filters This
+ fixes a ref count leak in event filters and checks for a filter
+ container allocation failure during session creation.
+
+2010-06-22 17:35 +0000 [r271903] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: 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 16:29 +0000 [r271868] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, configs/manager.conf.sample, CHANGES: Add regular
+ expression filtering for manager events. This patch as documented
+ in the sample config allows one to optionally apply white, black,
+ or both types of filtering to manager events. The new
+ 'eventfilter' option is set per user. (closes issue #14861)
+ Reported by: fnordian Patches: eventfilter3.patch uploaded by
+ fnordian (license 110), modified by me Review:
+ https://reviewboard.asterisk.org/r/673/
+
+2010-06-22 16:28 +0000 [r271833-271867] Russell Bryant <russell@digium.com>
+
+ * res/ais/clm.c, res/ais/evt.c: 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.
+
+ * res/res_fax.c: Change the method of retrieving the Asterisk
+ version string. Using this method makes it so res_fax doesn't
+ have to be rebuilt on every svn update.
+
+2010-06-22 15:46 +0000 [r271831] David Vossel <dvossel@digium.com>
+
+ * main/features.c: 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 15:08 +0000 [r271690-271764] Matthew Nicholson <mnicholson@digium.com>
+
+ * CHANGES: Updated the CHANGES file documenting the addition of a
+ configurable port in the dundi config file.
+
+ * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: 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 ........
+
+ * /, channels/chan_sip.c, include/asterisk/strings.h,
+ channels/sip/include/sip.h: 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 22:41 +0000 [r271657] Tilghman Lesher <tlesher@digium.com>
+
+ * build_tools/menuselect-deps.in, configure, configure.ac,
+ res/res_timing_kqueue.c: Conflict kqueue on OS X, since it
+ doesn't work there yet, anyway.
+
+2010-06-21 21:58 +0000 [r271625] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_speex.c, codecs/ex_speex.h,
+ contrib/editors/asterisk.vim: add speex 16khz sample frame so
+ codec cost can be calculated (closes issue #17534) Reported by:
+ fabled Patches: speex-wb-sample.diff uploaded by fabled (license
+ 448)
+
+2010-06-21 20:46 +0000 [r271554] Jeff Peeler <jpeeler@digium.com>
+
+ * res/ael/pval.c, /: 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 ........
+
+2010-06-21 20:46 +0000 [r271551-271553] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c: fixes crash
+ when From header URI is missing "sip:" (closes issue #17437)
+ Reported by: klaus3000 Patches: sip_crash uploaded by dvossel
+ (license 671) Tested by: klaus3000
+
+ * res/res_rtp_asterisk.c: fixes logic error introduced by slin16
+ sip support
+
+2010-06-21 05:10 +0000 [r271520] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_saycounted.c (added), CHANGES: Add new application for
+ declining counting words in multiple languages. (closes issue
+ #16869) Reported by: chappell Patches: app_say_counted-20100317.c
+ uploaded by chappell (license 8) Tested by: chappell
+
+2010-06-18 21:32 +0000 [r271483] Jeff Peeler <jpeeler@digium.com>
+
+ * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: 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 ........
+
+2010-06-18 18:59 +0000 [r271341] David Vossel <dvossel@digium.com>
+
+ * main/file.c: file.c was truncating audio file formats to the
+ lower 32bits.
+
+2010-06-18 18:36 +0000 [r271336] Jeff Peeler <jpeeler@digium.com>
+
+ * /: Recorded merge of revisions 271335 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010)
+ | 13 lines Eliminate deadlock potential in dahdi_fixup(). (This
+ is a backport of 269307, committed to trunk by rmudgett.) 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. (closes issue #17261) Reported by: aragon
+ ........
+
+2010-06-17 21:23 +0000 [r271231-271300] David Vossel <dvossel@digium.com>
+
+ * channels/sip/reqresp_parser.c: fixes some coding guideline issue
+
+ * channels/sip/include/dialog.h, channels/chan_sip.c,
+ channels/sip/include/sip.h: retransmit response to BYE requests
+ until timer J expires According to RFC 3261 section 17.2.2, which
+ describes non-INVITE server transaction, when a dialog enters the
+ Completed state it must destroy the dialog after Timer J (T1*64)
+ fires. For a BYE transaction Asterisk terminates the dialog
+ immediately during sip_hangup() when it should be waiting T1*64
+ ms. This results in some odd behavior. For instance if Asterisk
+ receives a BYE and transmits a 200ok in response, if the endpoint
+ never receives the 200ok it will retransmit the BYE to which
+ Asterisk responds with a "481 Call leg/transaction does not
+ exist" because the dialog is already gone. To resolve this I made
+ a function called sip_scheddestroy_final(). This differs slightly
+ from sip_schedestroy() in that it enables a flag that will
+ prevent the destruction from ever being rescheduled or canceled
+ afterwards. It also prevents the pvt's needdestroy flag from
+ being set which triggers the destruction of the dialog within the
+ do_monitor thread(). By using this function we are guaranteed
+ destruction will not occur until the scheduled time. This allows
+ Asterisk to respond to any possible retransmits for a dialog
+ after we process the initial BYE request for T1*64 ms. Other
+ changes: I removed two instances where sip_cancel_destroy is used
+ right before calling sip_scheddestroy. sip_scheddestroy always
+ calls sip_cancel_destroy before scheduling the new destruction so
+ it is completely unnecessary. Review:
+ https://reviewboard.asterisk.org/r/694/
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, CHANGES: adds support
+ for slin16 in sip (closes issue #16153) Reported by: kfister
+ Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license
+ 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested
+ by: kfister, malcolmd
+
+ * main/channel.c, res/res_rtp_asterisk.c, main/frame.c,
+ main/rtp_engine.c, codecs/codec_speex.c, CHANGES,
+ include/asterisk/frame.h: adds speex 16khz audio support (closes
+ issue #17501) Reported by: fabled Patches:
+ asterisk-trunk-speex-wideband-v2.patch uploaded by fabled
+ (license 448) Tested by: malcolmd, fabled, dvossel
+
+2010-06-17 15:34 +0000 [r271192] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_analog.c: Change expected operation from error to
+ debug message
+
+2010-06-17 00:30 +0000 [r271089] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_meetme.c: option w[(secs)] incorrectly capitalized in
+ xmldoc (closes issue #17516) Reported by: karlfife
+
+2010-06-16 22:37 +0000 [r271056] David Vossel <dvossel@digium.com>
+
+ * channels/sip/reqresp_parser.c: addition of more parse_uri test
+ cases
+
+2010-06-16 21:17 +0000 [r270987] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, configs/extensions.conf.sample: Merged revisions 270979 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun
+ 2010) | 4 lines Fixed typo in macro-page Reported to
+ #asterisk-dev by a student of jsmith. ........
+
+2010-06-16 21:12 +0000 [r270981-270983] Jason Parker <jparker@digium.com>
+
+ * channels/chan_agent.c: Fix the actual place that was pointed out,
+ for previous commit.
+
+ * /, channels/chan_agent.c: Merged revisions 270980 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun
+ 2010) | 4 lines Need to lock the agent chan before access its
+ internal bits. Pointed out by russellb on asterisk-dev mailing
+ list. ........
+
+2010-06-16 20:34 +0000 [r270974] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/dnsmgr.c, main/acl.c: Set sin_family to AF_INET when doing
+ lookups, also reset sin_port the first time the ip address
+ changes. (closes issue #17496) Reported by: ManChicken (closes
+ issue #15827) Reported by: DennisD Patches: dnsmgr_15827.patch
+ uploaded by chappell (license 8) Tested by: DennisD, gentlec,
+ damage, wimpy
+
+2010-06-16 19:03 +0000 [r270940] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, res/res_rtp_asterisk.c, main/frame.c,
+ main/rtp_engine.c, channels/chan_sip.c, CHANGES,
+ channels/chan_iax2.c, include/asterisk/frame.h,
+ formats/format_g719.c (added): addition of G.719 pass-through
+ support (closes issue #16293) Reported by: malcolmd Patches:
+ g719.passthrough.patch.7 uploaded by malcolmd (license 924)
+ format_g719.c uploaded by malcolmd (license 924)
+
+2010-06-16 18:43 +0000 [r270936] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_agi.c, CHANGES: MSG_OOB flag on HANGUP packet removed.
+ Per Tilghman's request on IRC (#asterisk-bugs). (closes issue
+ #17506) Reported by: brycebaril Tested by: pabelanger, tilghman
+
+2010-06-16 17:36 +0000 [r270867] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 270866 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16
+ Jun 2010) | 22 lines fixes chan_iax2 race condition There is code
+ in chan_iax2.c that attempts to guarantee that only a single
+ active thread will handle a call number at a time. This code
+ works once the thread is added to an active_list of threads, but
+ we are not currently guaranteed that a newly activated thread
+ will enter the active_list immediately because it is left up to
+ the thread to add itself after frames have been queued to it.
+ This means that if two frames come in for the same call number at
+ the same time, it is possible for them to grab two separate
+ threads because the first thread did not add itself to the
+ active_list fast enough. This causes some pretty complex
+ problems. This patch resolves this race condition by immediately
+ adding an activated thread to the active_list within the network
+ thread and only depending on the thread to remove itself once it
+ is done processing the frames queued to it. By doing this we are
+ guaranteed that if another frame for the same call number comes
+ in at the same time, that this thread will immediately be found
+ in the active_list of threads. Review:
+ https://reviewboard.asterisk.org/r/720/ ........
+
+2010-06-16 16:45 +0000 [r270836] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_analog.c: Fix no call waiting caller ID Clearing the
+ callwaitcas flag in analog_call was causing the incoming D digit
+ to be ignored which triggers sending the caller ID.
+
+2010-06-16 15:05 +0000 [r270801] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * doc/tex/channelvariables.tex: Update formatting for
+ channelvariables.tex (closes issue #17511) Reported by: klaus3000
+ Patches: channelvariables.tex-patch.txt uploaded by klaus3000
+ (license 65) Tested by: pabelanger
+
+2010-06-15 22:48 +0000 [r270726] Russell Bryant <russell@digium.com>
+
+ * channels/sig_analog.c: Don't blow up if an ast_channel doesn't
+ get allocated.
+
+2010-06-15 21:42 +0000 [r270658-270692] Terry Wilson <twilson@digium.com>
+
+ * main/http.c: Don't continue sending the file when there has been
+ an error If there is a problem with a firmware file, Polycom
+ phones will close the connection. We were continuing to send the
+ file anyway. There should be no reason to continue sending a file
+ if there is an error writing it. (closes issue #16682) Reported
+ by: lmadsen
+
+ * res/res_phoneprov.c: Don't send files twice and remove extra \r\n
+ from header After the manager http auth changes, we forgot to
+ remove the manual sending of the file. Also, ast_http_send adds
+ two \r\n to the header that is passed to it, so a trailing \r\n
+ is removed from the Content-type header. It might be better to
+ change ast_http_send, but I don't like changing the behavior of
+ an API function. (closes issue #17239) Reported by: cjacobsen
+ Patches: patch2.diff uploaded by cjacobsen (license 1029) Tested
+ by: lathama, cjacobsen
+
+ * channels/chan_sip.c: Make contactdeny apply to src ip when
+ nat=yes chan_sip's "contactdeny" feature screens the "to be
+ registered contact". In case of nat=yes it should not use the
+ address information from the Contact header (which is not used at
+ all for routing), but the source IP address of the request. Thus,
+ if nat=yes and a client sends a request from a denied IP address
+ (e.g. by spoofing the src-IP address) it can bypass the
+ screening. This commit makes contactdeny apply to the src ip when
+ nat=yes instead. (closes issue #17276) Reported by: klaus3000
+ Patches: patch-asterisk-trunk-contactdeny.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000
+
+2010-06-15 18:26 +0000 [r270519-270584] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 270583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010)
+ | 5 lines Variables have always been case-sensitive, so we should
+ not be removing case-insensitive matches. Bug reported via the
+ -dev list. See
+ http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html
+ ........
+
+ * res/res_jabber.c: Argh, mixed declarations and code.
+
+ * configs/jabber.conf.sample, include/asterisk/jabber.h,
+ doc/distributed_devstate-XMPP.txt (added), CHANGES,
+ res/res_jabber.c: Add distributed devicestate via the XMPP
+ protocol. (closes issue #15757) Reported by: Marquis Patches:
+ distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
+ Tested by: Marquis, lmadsen, marcelloceschia Review:
+ https://reviewboard.asterisk.org/r/351/
+
+2010-06-15 12:51 +0000 [r270443] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 270442 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010)
+ | 1 line Move information about zonemessages into the
+ [zonemessages] section. ........
+
+2010-06-14 21:33 +0000 [r270332] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 270331 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon,
+ 14 Jun 2010) | 14 lines Properly play first file in sort list.
+ When using sort=alpha we would always skip the first file in the
+ list first time through. We now check for that properly. (closes
+ issue #17470) Reported by: pabelanger Patches: sort.aplha.patch
+ uploaded by pabelanger (license 224) Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/703/ ........
+
+2010-06-14 20:51 +0000 [r270298] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
+ Extract sig_ss7_init_linkset() to sig_ss7. Also found a place
+ where sig_pri_init_pri() was inlined and called it instead.
+
+2010-06-14 19:41 +0000 [r270260] Jason Parker <jparker@digium.com>
+
+ * channels/chan_agent.c: Add option to get untruncated channel name
+ from AGENT function. The "channel" option would chop the channel
+ name at the last '-', which made it useless for something like a
+ channel transfer from the dialplan. The "fullchannel" option will
+ return the channel name as-is. ABE-2218
+
+2010-06-14 15:55 +0000 [r270219] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add digit
+ manipulation tag support to chan_dahdi/sig_pri like chan_misdn.
+ Add the append_msn_to_cid_tag option to chan_dahdi like
+ chan_misdn. Review: https://reviewboard.asterisk.org/r/696/
+
+2010-06-13 09:16 +0000 [r270184] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * autoconf/ast_check_pwlib.m4, configure: bashism in configure
+ script Theoretically the ./configure script is a pure
+ bourne-shell script. Practically it may be run by bash if /bin/sh
+ is not good enough. But we should not count on it. See bug report
+ for the gory details. (closes issue #17485) Patches:
+ 0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by
+ tzafrir (license 46)
+
+2010-06-13 01:53 +0000 [r270042-270151] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Reverting patch and reopening issue #16155, as patch breaks
+ FreeBSD / OSX builds.
+
+ * /, doc/HOWTO_collect_debug_information.txt: Merged revisions
+ 270078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun
+ 2010) | 2 lines Fix typo in example ........
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Use
+ pkg-config to find gmime libraries This way the libraries can be
+ found even if they are in non-standard locations. (closes issue
+ #16155) Reported by: jcollie Patches:
+ 0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch
+ uploaded by jcollie (license 412) Tested by: jsmith, tilghman,
+ pabelanger
+
+2010-06-11 18:31 +0000 [r269936-269976] Tilghman Lesher <tlesher@digium.com>
+
+ * main/frame.c, /: Merged revisions 269960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010)
+ | 8 lines For SpeeX, 0 bits remaining is valid and does not need
+ an emitted warning. (closes issue #15762) Reported by: nblasgen
+ Patches: issue15672.patch uploaded by pabelanger (license 224)
+ Tested by: nblasgen ........
+
+ * CHANGES, main/db.c: Add DBGetComplete event after a
+ DBGetResponse. (closes issue #16965) Reported by: rrb3942
+ Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003)
+
+ * main/logger.c: Remove lines from the output related to the
+ backtrace itself.
+
+2010-06-10 20:30 +0000 [r269889] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile, makeopts.in: Remove ASTBINDIR variable (closes issue
+ #17031) Reported by: pabelanger Patches:
+ Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224)
+ Tested by: pabelanger, tilghman
+
+2010-06-10 19:34 +0000 [r269749-269822] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 269821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun
+ 2010) | 19 lines Fix potential crash when writing raw SLIN audio
+ on a PLC-enabled channel. The issue here was that the frame
+ created when adjusting for PLC had no offset to its audio data.
+ If this frame were translated to another format prior to being
+ sent out an RTP socket, all went well because the translation
+ code would put an appropriate offset into the frame. However, if
+ the SLIN audio were not translated before being sent out the RTP
+ socket, bad things would happen. Specifically, the
+ ast_rtp_raw_write makes the assumption that the frame has at
+ least enough of an offset that it can accommodate an RTP header.
+ This was not the case. As such, data was being written prior to
+ the allocation, likely corrupting the data the memory allocator
+ had written. Thus when the time came to free the data, all hell
+ broke loose. ....Well, Asterisk crashed at least. The fix was
+ just what one would expect. Offset the data in the frame by a
+ reasonable amount. The method I used is a bit odd since the data
+ in the frame is 16 bit integers and not bytes. I left a big ol'
+ comment about it. This can be improved on if someone is
+ interested. I was more interested in getting the crash resolved.
+ ........
+
+ * doc/tex/plc.tex (added), doc/tex/asterisk.tex: Add documentation
+ explaining PLC in Asterisk. Review:
+ https://reviewboard.asterisk.org/r/688/
+
+2010-06-10 13:17 +0000 [r269711] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c: Fix an off by one error that caused a unit
+ test to occasionally crash.
+
+2010-06-10 12:28 +0000 [r269707] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/logger.c: Ensure that 'logger show channels' works properly
+ when wildcards are used in logger.conf.
+
+2010-06-10 08:15 +0000 [r269636] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/logger.c, utils/extconf.c, main/asterisk.c: Merged
+ revisions 269635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010)
+ | 9 lines Ensure restartable system calls can restart (BSD signal
+ semantics). This eliminates the annoying <beep> on the console.
+ (closes issue #17477) Reported by: jvandal Patches:
+ 20100610__issue17477.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2010-06-10 00:32 +0000 [r269417-269602] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c: Attempt to fix a FreeBSD build error by
+ including sys/stat.h.
+ http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log
+
+ * main/lock.c: Attempt to fix FreeBSD build problem.
+
+ * /, channels/chan_oss.c: Merged revisions 269495 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010)
+ | 2 lines Don't stop Asterisk if chan_oss fails to register
+ 'Console' (due to another channel driver already claiming it).
+ ........
+
+ * include/asterisk/event.h, main/event.c: Resolve an invalid memory
+ read on an event. Valgrind pointed out that attempting to get an
+ IE value from an event that has no IEs produces an invalid memory
+ read past the end of the event. Thanks to mmichelson for pointing
+ the problem out to me and then testing the fix.
+
+2010-06-09 17:32 +0000 [r269346] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * contrib/init.d/rc.debian.asterisk, /, main/term.c: Merged
+ revisions 269334 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun
+ 2010) | 12 lines 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 #16784) Reported by: pabelanger Patches:
+ 20100530__issue16784__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: pabelanger, tilghman ........
+
+2010-06-09 17:06 +0000 [r269307-269308] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
+ Add missing API function to sig_ss7: sig_ss7_fixup().
+
+ * channels/chan_dahdi.c: 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.
+
+2010-06-09 15:09 +0000 [r269271] David Vossel <dvossel@digium.com>
+
+ * res/res_musiconhold.c: fixes crash in moh when cachertclasses
+ flag is used The result for moh_register was not verified to
+ guarantee the mohclass as added to the container. (closes issue
+ #16993) Reported by: dmitri Patches:
+ res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001)
+ moh_crash2.diff uploaded by dvossel (license 671) Tested by:
+ dmitri
+
+2010-06-09 13:17 +0000 [r269238] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
+ dial by name in chan_dahdi * chan_dahdi supports dialing
+ configuring and dialing by device file name.
+ DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 .
+ Likewise it may appear in chan_dahdi.conf as 'channel =>
+ span-name!local!1'. * A new options for chan_dahdi.conf:
+ 'ignore_failed_channels'. Boolean. False by default. If set,
+ chan_dahdi will ignore failed 'channel' entries. Handy for the
+ above name-based syntax as it does not depend on initialization
+ order. * have my_pri_make_cc_dialstring() only manupulate
+ dial-strings of group (gGrR) dialing, which make it lsightly more
+ complicated. https://reviewboard.asterisk.org/r/535/
+
+2010-06-09 10:55 +0000 [r269187-269205] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/install_prereq: Add libjack-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libpopt-dev, libical-dev, and
+ libspandsp-dev to install_prereq.
+
+ * contrib/scripts/install_prereq: Add libnewt-dev to
+ install-prereq.
+
+ * contrib/scripts/install_prereq: Add libopenais-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add an "install-unpackaged"
+ command to install_prereq for installing unpackaged dependencies
+ (such as NBS and libresample).
+
+ * contrib/scripts/install_prereq: Add libcurl to install_prereq.
+
+ * contrib/scripts/install_prereq: Add freetds-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libradiusclient-ng-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libbluetooth-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libmysqlclient-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libgtk2.0-dev to the packages
+ list for install_prereq.
+
+2010-06-08 23:48 +0000 [r269153] Bradley Latus <brad.latus@gmail.com>
+
+ * configs/cdr_custom.conf.sample, configs/cdr_tds.conf.sample,
+ cdr/cdr_sqlite.c, configs/cdr_sqlite3_custom.conf.sample,
+ funcs/func_cdr.c, configs/cdr_syslog.conf.sample, UPGRADE.txt,
+ cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, cdr/cdr_pgsql.c,
+ CHANGES, cdr/cdr_odbc.c, cdr/cdr_tds.c,
+ configs/cdr_odbc.conf.sample: Add High Resolution Times to CDRs
+ for Asterisk People expressed an interest in having access to the
+ exact length of calls to a finer degree than seconds. See the
+ CHANGES and UPGRADE.txt for usage also updated the sample configs
+ to note the change. Patch by snuffy. (closes issue #16559)
+ Reported by: cianmaher Tested by: cianmaher, snuffy Review:
+ https://reviewboard.asterisk.org/r/461/
+
+2010-06-08 22:45 +0000 [r269119] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/localtime.h: Fix build on Mac OS X (and maybe
+ FreeBSD, too)
+
+2010-06-08 18:50 +0000 [r269083] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_fax.c: Don't pass null to manager_event() (closes issue
+ #17087) Reported by: bklang Patches: app-fax-null-sprintf1.diff
+ uploaded by mnicholson (license 96) Tested by: bklang
+
+2010-06-08 15:41 +0000 [r269008] Russell Bryant <russell@digium.com>
+
+ * Makefile.rules: Ensure CONFIG_FLAGS makes it into the build rules
+ when doing out of tree builds. (closes issue #16685) Reported by:
+ pprindeville
+
+2010-06-08 15:39 +0000 [r269007] Sean Bright <sean@malleable.com>
+
+ * /, cdr/cdr_tds.c: Merged revisions 269006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269006 | seanbright | 2010-06-08 11:28:49 -0400 (Tue, 08 Jun
+ 2010) | 11 lines Reduce startup time for cdr_tds with large CDR
+ tables. Since we are just checking for table existence, add a
+ WHERE clause that will return no rows but will raise an error if
+ the table doesn't exist. (closes issue #17380) Reported by:
+ kkwong Patches: issue17380-01.patch uploaded by seanbright
+ (license 71) Tested by: kkwong ........
+
+2010-06-08 15:23 +0000 [r268969-268988] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Update note in sip.conf.sample. Update
+ note in sip.conf.sample about externip and externhost with STUN.
+ (closes issue #16323) Reported by: klaus3000 Patches:
+ sip.conf.sample-patch.txt uploaded by klaus3000 (license 65)
+
+ * apps/app_meetme.c, main/ccss.c, include/asterisk/data.h,
+ res/res_jabber.c, res/res_config_sqlite.c,
+ include/asterisk/callerid.h, channels/chan_dahdi.c,
+ include/asterisk/bridging_technology.h,
+ include/asterisk/doxyref.h, include/asterisk/event.h,
+ include/asterisk/astmm.h, main/ast_expr2f.c, main/features.c,
+ include/asterisk/timing.h, include/asterisk/rtp_engine.h,
+ include/asterisk/ccss.h, include/asterisk/threadstorage.h,
+ include/asterisk/xml.h, main/pbx.c, channels/chan_sip.c,
+ include/asterisk/astobj2.h, include/asterisk/channel.h,
+ include/asterisk/calendar.h, include/asterisk/manager.h,
+ include/asterisk/features.h, include/asterisk/logger.h,
+ include/asterisk/http.h, channels/sig_pri.h,
+ include/asterisk/app.h, main/audiohook.c, include/asterisk/pbx.h,
+ include/asterisk/dnsmgr.h, include/asterisk/smdi.h,
+ apps/app_voicemail.c: Fix some doxygen warnings. (closes issue
+ #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded
+ by snuffy (license 35) Tested by: russell
+
+2010-06-08 06:57 +0000 [r268896-268933] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_sqlite.c: Release list lock before returning on
+ error.
+
+ * utils/extconf.c: Fix trunk build on Mac OS X.
+
+2010-06-08 05:29 +0000 [r268894] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/sdp_crypto.c (added), res/res_rtp_asterisk.c,
+ main/global_datastores.c, main/rtp_engine.c,
+ include/asterisk/res_srtp.h (added), channels/sip/srtp.c (added),
+ channels/chan_sip.c, include/asterisk/autoconfig.h.in,
+ res/res_srtp.exports.in (added), configure.ac, CHANGES,
+ channels/chan_iax2.c, res/res_srtp.c (added), main/channel.c,
+ build_tools/menuselect-deps.in, main/asterisk.exports.in,
+ configure, funcs/func_channel.c,
+ channels/sip/dialplan_functions.c,
+ channels/sip/include/sdp_crypto.h (added),
+ doc/tex/secure-calls.tex (added),
+ include/asterisk/global_datastores.h, channels/sip/include/srtp.h
+ (added), makeopts.in, include/asterisk/rtp_engine.h,
+ include/asterisk/frame.h, doc/tex/asterisk.tex,
+ channels/sip/include/sip.h: Add SRTP support for Asterisk After 5
+ years in mantis and over a year on reviewboard, SRTP support is
+ finally being comitted. This includes generic CHANNEL dialplan
+ functions that work for getting the status of whether a call has
+ secure media or signaling as defined by the underlying channel
+ technology and for setting whether or not a new channel being
+ bridged to a calling channel should have secure signaling or
+ media. See doc/tex/secure-calls.tex for examples. Original patch
+ by mikma, updated for trunk and revised by me. (closes issue
+ #5413) Reported by: mikma Tested by: twilson, notthematrix,
+ hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/
+
+2010-06-08 00:45 +0000 [r268857] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sip/dialplan_functions.c: Make SIP tests compile again.
+
+2010-06-07 22:56 +0000 [r268817-268818] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Use the mailbox destructor function,
+ instead.
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: Mailbox list
+ would previously grow at each reload, containing duplicates.
+ Also, optimize the allocation of mailboxes to avoid additional
+ memory structures. (closes issue #16320) Reported by: Marquis
+ Patches: 20100525__issue16320.diff.txt uploaded by tilghman
+ (license 14)
+
+2010-06-07 20:04 +0000 [r268774] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h
+ (added), channels/Makefile, channels/sig_pri.c,
+ channels/sig_ss7.c (added): Extract sig_ss7 out of chan_dahdi.
+ Extract the SS7 specific code out of chan_dahdi like what was
+ done to ISDN/PRI and analog signaling. The new SS7 structures
+ were modeled on sig_pri. The changes to sig_pri are an
+ enhancement and a bug fix made possible because SS7 was
+ extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable
+ should have been set unconditionally in
+ sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability
+ interaction in dahdi_new() fixed because of SS7 extraction. 3)
+ Module ref count error in dahdi_new() if startpbx failed to start
+ the PBX for some reason. Review:
+ https://reviewboard.asterisk.org/r/661/
+
+2010-06-07 19:52 +0000 [r268773] Tilghman Lesher <tlesher@digium.com>
+
+ * main/rtp_engine.c, channels/chan_sip.c,
+ channels/sip/dialplan_functions.c, include/asterisk/rtp_engine.h:
+ Seems strange (and the code backs up) that if the max and min of
+ a statistic is expressed as a double, the last value would not
+ also need to be a double. (closes issue #15807) Reported by:
+ klaus3000
+
+2010-06-07 19:06 +0000 [r268734] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Moved AOC request code out of the middle of
+ code parsing the dialed number.
+
+2010-06-07 18:59 +0000 [r268731] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: Event well was going dry. (issue #17234)
+
+2010-06-07 17:34 +0000 [r268690] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/dsp.c: Set threshold for silence detection defaults to 256
+ (closes issue #15685) Reported by: david_s5 Patches:
+ dsp-silence-threshold-init.diff uploaded by dant (license 670)
+ issue15685.patch.v5 uploaded by pabelanger (license 224) Tested
+ by: danti Review: https://reviewboard.asterisk.org/r/670/
+
+2010-06-07 17:14 +0000 [r268653] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_smdi.c: Avoid unloading res_smdi twice. (closes issue
+ #17237) Reported by: pabelanger
+
+2010-06-07 15:51 +0000 [r268578] Richard Mudgett <rmudgett@digium.com>
+
+ * main/file.c: Suppress warning in waitstream_core(). Suppress the
+ warning about unexpected control subclass frames for
+ AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and
+ AST_CONTROL_AOC in file.c:waitstream_core().
+
+2010-06-06 05:29 +0000 [r268454-268534] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: Take advantage of variable
+ substitution already in the Makefile to specify the correct
+ location for files in init.d. (closes issue #16979) Reported by:
+ jw-asterisk (issue #15691) Reported by: itamarjp
+
+ * channels/chan_iax2.c: Finally track down and eliminate the
+ "FRACK! warnings from chan_iax2".
+
+ * main/dsp.c: 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 #17371) Reported by: alecdavis
+ (closes issue #17474) Reported by: kenner
+
+ * main/manager.c: Verify event is not NULL before attempting to
+ lower its usecount. (closes issue #17234) Reported by: mav3rick
+
+2010-06-05 05:23 +0000 [r268395-268417] Kevin P. Fleming <kpfleming@digium.com>
+
+ * CHANGES: Typo fix.
+
+ * CHANGES: Grammatical error fix.
+
+2010-06-05 02:51 +0000 [r268321] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 268320 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010)
+ | 3 lines Rest In Peace
+ http://www.outandaboutnewspaper.com/article/4061 ........
+
+2010-06-04 22:37 +0000 [r268205-268281] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes compile error from uninitialized
+ variable
+
+ * channels/chan_sip.c: RFC3261 compliant sip unreliable retransmit
+ timing + 'registerattempts' option tweak Changes. 1. RFC 3261
+ states in section 17.1.2.2 and 17.1.1.2 that retransmission
+ timers should initially be set to timer T1. T1 by default is
+ 500ms. Asterisk was starting the retransmission timers at T1*2
+ which shouldn't cause any problems, but is not RFC compliant. 2.
+ RFC 3261 states in section 17.1.2.2 that for a non-INVITE client
+ transaction, if the retransmit timer fires while in the
+ proceeding state that the request must be retransmitted. Asterisk
+ currently ack's requests for both INVITE and non-INVITE
+ transactions when a 1XX response is received, this patch changes
+ this for non-INVITE requests. 3. The 'registerattempts' option in
+ sip.conf is supposed to set how many registry attempts will be
+ made before giving up. When this option is set to 1, I would
+ expect only one registry attempt to be made before stopping
+ because of a failure, but instead two are made. In my opinion
+ this is not expected behavior. This option does not indicate that
+ these are re-attempts. The logic behind this option has been
+ changed to only attempt registers the exact number of times this
+ option is set to. If this option is 0, it still continues to
+ re-attempt the registration forever. Review:
+ https://reviewboard.asterisk.org/r/687/
+
+2010-06-04 20:42 +0000 [r267972-268127] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 268126 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r268126 | tilghman | 2010-06-04 15:41:24 -0500 (Fri, 04
+ Jun 2010) | 2 lines AC_CONFIG_SUBDIRS has a bad side-effect on
+ cross-compiles. ........
+
+ * Makefile, /, makeopts.in: Merged revisions 268050 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r268050 | tilghman | 2010-06-04 14:38:57 -0500 (Fri, 04
+ Jun 2010) | 6 lines Build menuselect with the build environment's
+ compiler, not the host (target)'s compiler. (closes issue #17464)
+ Reported by: pprindeville Tested by: tilghman ........
+
+ * /, configure, configure.ac, autoconf/libcurl.m4: Merged revisions
+ 267971 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010)
+ | 2 lines As-fixiate the build process ........
+
+2010-06-04 14:45 +0000 [r267928] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Incoming overlap dialing no longer works
+ after sig_pri extraction. The problem would manifest itself if
+ your dialplan matching could accept more digits to match than
+ were actually dialed. The time out waiting for overlap digits
+ disconnected the call instead of matching any accumulated digits
+ to the dialplan. Accidental conversion of a break out of loop as
+ a break out of switch. (closes issue #17401) Reported by:
+ avalentin Patches: issue17401_digit_timeout.patch uploaded by
+ rmudgett (license 664) Tested by: avalentin, rmudgett
+
+2010-06-04 03:20 +0000 [r267877] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/slin.h: As signed linear audio data is accessed
+ as 16-bit values, certain processors require the values to be
+ aligned in memory. (closes issue #16912) Reported by:
+ michaelevdokimov Patches: asterisk.patch uploaded by
+ michaelevdokimov (license 997) Tested by: michaelevdokimov
+
+2010-06-04 03:11 +0000 [r267863] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Send an ACK for every final response
+ received for an INVITE From issue ABE-2247. RFC 3261 compliance
+ for sections 13.2.24 and 17.1.1.2. Review:
+ https://reviewboard.asterisk.org/r/692/
+
+2010-06-04 02:58 +0000 [r267775-267862] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/slin.h: As signed linear audio data is accessed
+ as 16-bit values, certain processors require the values to be
+ aligned in memory. (closes issue #16912) Reported by:
+ michaelevdokimov
+
+ * configure, autoconf/ast_ext_lib.m4: If there's a default, turn it
+ on, even when the option isn't specified.
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 267759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010)
+ | 7 lines Make the default install path appear to be /usr on
+ Linux, instead of /usr/local. Also, reorganize the options, so
+ that they're more alphabetical. (closes issue #17013) Reported
+ by: klaus3000 ........
+
+2010-06-03 20:41 +0000 [r267714] Russell Bryant <russell@digium.com>
+
+ * main/ccss.c: Remove a LOG_WARNING. This came up when using the
+ sample configs, and just indicates expected behavior.
+
+2010-06-03 19:46 +0000 [r267669] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Handle OOM errors more gracefully. (closes
+ issue #17084) Reported by: falves11 Patches:
+ issue17084_162_A.diff uploaded by falves11 (license 374) Tested
+ by: falves11
+
+2010-06-03 18:53 +0000 [r267624] Leif Madsen <lmadsen@digium.com>
+
+ * UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGE for CDR
+ functionality changes. Updated the UPGRADE.txt and CHANGES file
+ stating that CDR records will not be explicity written unless
+ cdr.conf exists and is configured. (closes issue #17373) Reported
+ by: wdoekes Tested by: pabelanger
+
+2010-06-03 18:38 +0000 [r267622] Richard Mudgett <rmudgett@digium.com>
+
+ * codecs/codec_dahdi.c: Make compile again.
+
+2010-06-03 17:31 +0000 [r267537] Russell Bryant <russell@digium.com>
+
+ * channels/chan_usbradio.c: Don't stop Asterisk if chan_usbradio
+ isn't configured.
+
+2010-06-03 17:09 +0000 [r267492] Mark Michelson <mmichelson@digium.com>
+
+ * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c,
+ codecs/codec_alaw.c, main/translate.c, codecs/codec_g726.c,
+ codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c,
+ include/asterisk/translate.h: Remove unnecessary code relating to
+ PLC. The logic for handling generic PLC is now handled in
+ ast_write in channel.c instead of in translation code. Review:
+ https://reviewboard.asterisk.org/r/683/
+
+2010-06-03 17:05 +0000 [r267445-267490] Russell Bryant <russell@digium.com>
+
+ * channels/chan_usbradio.c: Remove a line that was killing Asterisk
+ on startup.
+
+ * channels/h323/Makefile.in: Comment out a rule that likes to run
+ implicitly unnecessarily, breaking builds
+
+2010-06-03 00:02 +0000 [r267399] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
+ channels/sig_pri.c: Add ETSI Message Waiting Indication (MWI)
+ support. Add the ability to report waiting messages to ISDN
+ endpoints (phones). Relevant specification: EN 300 650 and EN 300
+ 745 Review: https://reviewboard.asterisk.org/r/599/
+
+2010-06-02 22:46 +0000 [r267352] Russell Bryant <russell@digium.com>
+
+ * channels/Makefile, channels/h323/Makefile.in: try to fix some
+ random chan_h323 compilation failures After some debugging, the
+ random chan_h323 build failures appear to be due to complications
+ introduced by some chan_h323 specific build stuff getting
+ triggered during a clean. Simplify this by moving the h323 clean
+ commands down into channels/makefile.
+
+2010-06-02 22:28 +0000 [r267350] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, include/asterisk/channel.h, CHANGES,
+ channels/sig_pri.c: Add ETSI Malicious Call ID support. Add the
+ ability to report malicious callers as an AMI event in the call
+ event class. Relevant specification: EN 300 180 Review:
+ https://reviewboard.asterisk.org/r/576/
+
+2010-06-02 21:44 +0000 [r267303-267305] Russell Bryant <russell@digium.com>
+
+ * utils/extconf.c: Fix a build error on mac.
+
+ * main/Makefile: Ensure the -Wno-strict-aliasing flag makes it,
+ even if ASTCFLAGS has been specified. When ASTCFLAGS was
+ specified with the make command, Makefile.rules was using the
+ specified value from the command line and not the one here,
+ making it so this flag would go missing.
+
+2010-06-02 21:05 +0000 [r267261] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
+ channels/sig_pri.c: Add ETSI Call Waiting support. Add the
+ ability to announce a call to an endpoint when there are no B
+ channels available. A call waiting call is a SETUP message with
+ no B channel selected. Relevant specification: EN 300 056, EN 300
+ 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan
+ function now supports the "no_media_path" option. * Returns "0"
+ if there is a B channel associated with the call. * Returns "1"
+ if no B channel is associated with the call. The call is either
+ on hold or is a call waiting call. If you are going to allow
+ incoming call waiting calls then you need to use
+ CHANNEL(no_media_path) do determine if you must drop a call to
+ accept the new call. Review:
+ https://reviewboard.asterisk.org/r/568/
+
+2010-06-02 19:33 +0000 [r267181] David Vossel <dvossel@digium.com>
+
+ * CHANGES, doc/advice_of_charge.txt: Update CHANGES and aoc help
+ doc to reflect AOC additions
+
+2010-06-02 18:53 +0000 [r267138] Russell Bryant <russell@digium.com>
+
+ * main/cli.c: Add a CLI command that blocks until Asterisk has
+ fully booted. Review: https://reviewboard.asterisk.org/r/684/
+
+2010-06-02 18:13 +0000 [r267097] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Prevent use of uninitialized values. Two
+ struct sockaddr_ins are created when applying directmedia host
+ access rules. The addresses of these are passed to the RTP engine
+ to be filled in. However, the RTP engine inspects the fields of
+ the structs before actually taking action. This inspection caused
+ valgrind to be a bit unhappy.
+
+2010-06-02 18:10 +0000 [r267096] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, configs/chan_dahdi.conf.sample,
+ include/asterisk/aoc.h (added), channels/chan_sip.c,
+ configs/manager.conf.sample, main/aoc.c (added),
+ apps/app_queue.c, channels/sig_pri.c, doc/advice_of_charge.txt
+ (added), main/channel.c, channels/sig_pri.h,
+ channels/chan_dahdi.c, main/manager.c, main/features.c,
+ tests/test_aoc.c (added), configs/sip.conf.sample,
+ include/asterisk/frame.h, main/asterisk.c,
+ channels/sip/include/sip.h: Generic Advice of Charge. Asterisk
+ Generic AOC Representation - Generic AOC encode/decode routines.
+ (Generic AOC must be encoded to be passed on the wire in the
+ AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent
+ generic encoded AOC data - Manager events for AOC-S, AOC-D, and
+ AOC-E messages Asterisk App Support - app_dial AOC-S pass-through
+ support on call setup - app_queue AOC-S pass-through support on
+ call setup AOC Unit Tests - AOC Unit Tests for encode/decode
+ routines - AOC Unit Test for manager event representation. SIP
+ AOC Support - Pass-through of generic AOC-D and AOC-E messages to
+ snom phones via the snom AOC specification. - Creation of
+ chan_sip page3 flags for the addition of the new
+ 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively
+ supports AOC pass-through through the use of the new
+ AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC
+ Pass-through support - 'aoc_enable' chan_dahdi.conf option for
+ independently enabling pass-through of AOC-S, AOC-D, AOC-E. -
+ 'aoce_delayhangup' option for retrieving AOC-E on disconnect. -
+ DAHDI A() dial string option for requesting AOC services. example
+ usage: ;requests AOC-S, AOC-D, and AOC-E on call setup
+ exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review:
+ https://reviewboard.asterisk.org/r/552/
+
+2010-06-02 17:57 +0000 [r267093] Russell Bryant <russell@digium.com>
+
+ * apps/app_voicemail.c: Silence a compiler warning.
+
+2010-06-02 17:29 +0000 [r267065] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/slin.h: Fix infinite loop when loading codec
+ speex This changes the sample slinear frame data to contain
+ non-zero data so that translation calculations for speex works
+ when preprocessing and VAD is turned on. The encoder expects
+ samples to be returned, but when attempted with the mentioned two
+ options and silent sample frames everything was discarded.
+ (closes issue #17240) Reported by: seandarcy Review:
+ https://reviewboard.asterisk.org/r/682/
+
+2010-06-02 17:25 +0000 [r267041] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, main/ast_expr2.y: Merged revisions 267009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun
+ 2010) | 7 lines Cleanup error/warning messages in AEL2 parser
+ (closes issue #16684) Reported by: Silmaril Patches:
+ patch_ael2_logmsg.diff uploaded by Silmaril (license 979)
+ ........
+
+2010-06-02 17:13 +0000 [r266926-267008] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, configs/manager.conf.sample, CHANGES,
+ channels/sig_pri.c, include/asterisk/manager.h: Add ETSI Advice
+ Of Charge (AOC) event reporting. This feature generates AMI
+ events in the new aoc event class from the events passed up by
+ libpri. Review: https://reviewboard.asterisk.org/r/537/
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
+ channels/sig_pri.c: Add ETSI Explicit Call Transfer (ECT)
+ support. Added ability to send and receive ETSI Explicit Call
+ Transfer (ECT) messages to eliminate tromboned calls. Note:
+ Asterisk already supported initiating the transfer of calls to
+ eliminate tromboned calls to libpri so there was nothing to do
+ for the asterisk portion. Review:
+ https://reviewboard.asterisk.org/r/520/
+
+2010-06-02 13:32 +0000 [r266877] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/bridging.c: pthread_join to assure the thread is really gone
+ (closes issue #15465) Reported by: fnordian Patches:
+ bridging.patch uploaded by fnordian (license 110) Tested by:
+ lmadsen, fnordian, peterh Review:
+ https://reviewboard.asterisk.org/r/679/
+
+2010-06-01 22:14 +0000 [r266832] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_exchange.c: Use the correct ical.h file
+
+2010-06-01 21:28 +0000 [r266828] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, tests/test_locale.c
+ (added), configure.ac, configs/voicemail.conf.sample,
+ include/asterisk/localtime.h, main/stdtime/localtime.c, CHANGES,
+ apps/app_voicemail.c: Support setting locale per-mailbox (changes
+ date/time languages for email, pager messages). (closes issue
+ #14333) Reported by: klaus3000 Patches:
+ 20090515__issue14333.diff.txt uploaded by tilghman (license 14)
+ app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000
+
+2010-06-01 21:12 +0000 [r266786] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, UPGRADE.txt: Set app and appdata fields when a
+ Dial is redirected (closes issue #17204) Reported by: one47
+ Tested by: twilson, one47
+
+2010-06-01 18:02 +0000 [r266592-266735] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_smdi.c: Don't register functions until the last possible
+ point, so they're not unloaded unnecessarily. (closes issue
+ #15996) Reported by: junky Patches: sdmi_wait.diff uploaded by
+ junky (license 177)
+
+ * main/manager.c: Eliminate stale manager events after a set
+ interval, even if AMI clients don't query for them. Actions (or
+ failures to act) by external clients should not cause memory
+ leaks in Asterisk, especially when those continued leaks could
+ cause Asterisk to misbehave later. (closes issue #17234) Reported
+ by: mav3rick Patches: 20100510__issue17234.diff.txt uploaded by
+ tilghman (license 14) 20100517__issue17234__trunk.diff.txt
+ uploaded by tilghman (license 14) Tested by: mav3rick, davidw
+ (closes issue #17365) Reported by: davidw
+
+ * /, main/asterisk.c: 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 ........
+
+2010-05-30 20:18 +0000 [r266438-266522] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c: Needs to be wrapped in <para>
+
+ * contrib/init.d/rc.debian.asterisk, /: 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 22:54 +0000 [r266386] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_icalendar.c, configure, configure.ac,
+ res/res_calendar_caldav.c: Fix ical library handling (again)
+ Newer versions of libical (which we require) store the header
+ file in a libical/ subfolder and include an ical.h file that does
+ a #warning for deprecation and then #includes <libical/ical.h>.
+ Since we now test for libical/ical.h, we can change the #includes
+ back to <libical/ical.h> and remove the test which specifically
+ adds /usr/include/libical as an include directory.
+
+2010-05-28 22:50 +0000 [r266337-266385] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c, UPGRADE.txt, main/asterisk.c: Setup environment
+ variables for the benefit of child processes and disallow
+ changing them. (closes issue #14899) Reported by: jmls Patches:
+ 20090916__issue14899.diff.txt uploaded by tilghman (license 14)
+ Tested by: jmls
+
+ * main/asterisk.c: Only report swap on platforms which can examine
+ those statistics
+
+2010-05-28 17:55 +0000 [r266292] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: 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
+
+2010-05-28 17:34 +0000 [r266289] Terry Wilson <twilson@digium.com>
+
+ * configure, configure.ac, makeopts.in: More build fixes for
+ ical/neon and res_calendar_ews
+
+2010-05-27 20:08 +0000 [r266240] Jeff Peeler <jpeeler@digium.com>
+
+ * pbx/pbx_realtime.c: fix compile error
+
+2010-05-27 19:25 +0000 [r266146-266238] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_realtime.c, CHANGES: Cache query results for one second.
+ Queries from the PBX core come in 3's. Caching avoids the
+ additional performance penalty from those two additional queries
+ hitting the database. (closes issue #16521) Reported by: tilghman
+ Patches: 20091229__issue16521.diff.txt uploaded by tilghman
+ (license 14) Tested by: Hubguru, tilghman
+
+ * /, main/logger.c, utils/extconf.c, main/asterisk.c: 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 20:17 +0000 [r266092-266098] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Remove redundant ast_conntected_line_free call.
+ This wouldn't cause any problems, but it's certainly not needed
+ either.
+
+ * res/res_musiconhold.c: Remove unrelated MOH change from previous
+ commit. Thanks Kevin!
+
+ * main/channel.c, res/res_musiconhold.c: Fix misspelling of macro
+ args.
+
+2010-05-26 19:46 +0000 [r266006-266090] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, main/app.c, channels/sip/config_parser.c,
+ channels/sip/include/sip.h: do all sip registry parsing before
+ transmit_register This patch breaks up every part of the sip
+ registry string during config parsing and removes all parsing
+ from transmit_register(). Thanks to Nick_Lewis for contributing
+ this patch! (closes issue #14331) Reported by: Nick_Lewis
+ Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis
+ (license 657) chan_sip.c.patch uploaded by Nick Lewis (license
+ 657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis
+ (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis
+ (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis
+ (license 657) nicklewispatch.diff uploaded by dvossel (license
+ 671) Tested by: Nick_Lewis, dvossel Review:
+ https://reviewboard.asterisk.org/r/628/
+
+ * channels/chan_sip.c: 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 16:23 +0000 [r265894-265923] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, /: 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) ........
+
+ * res/res_config_pgsql.c, configs/res_pgsql.conf.sample: 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 14:45 +0000 [r265842-265844] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: .......
+
+ * channels/chan_sip.c: 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 05:33 +0000 [r265793] Terry Wilson <twilson@digium.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_calendar_ews.c: Ensure that libneon > 0.29.0 is installed
+ for res_calendar_ews This uses a modified version of pabelanger's
+ patch that checks for NTLM support instead, which was added in
+ 0.29.0 which is what is required for res_calendar_ews. (closes
+ issue #17391) Reported by: loloski Patches: issue17391.patch.v2
+ uploaded by pabelanger (license 224) Tested by: twilson
+
+2010-05-26 00:29 +0000 [r265747] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c, res/res_calendar_caldav.c, res/res_calendar_ews.c:
+ 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 20:59 +0000 [r265698] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: 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 17:00 +0000 [r265611] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_queue.c: 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 ........
+
+2010-05-25 16:23 +0000 [r265608] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Memory leak in connected line data when SIP blond
+ transfer done. The handling of the control subclass
+ AST_CONTROL_READ_ACTION frame leaked connected line string memory
+ in __ast_read(). Also in __ast_read() the frame type switch
+ should not have had a case for AST_CONTROL_READ_ACTION.
+ AST_CONTROL_READ_ACTION is not a frame type.
+
+2010-05-25 08:31 +0000 [r265525] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * addons/ooh323c/src/oochannels.c: Typos: 'succesful' (lintian)
+
+2010-05-24 22:21 +0000 [r265467] Terry Wilson <twilson@digium.com>
+
+ * doc/manager_1_1.txt, main/manager.c, main/asterisk.c: Merge the
+ rest of the FullyBooted patch
+
+2010-05-24 22:16 +0000 [r265449-265453] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_senddtmf.c: Allow SendDTMF to play digits to a specified
+ channel. Patch supplied by reporter was modified to use
+ autoservice and prevent a potential channel ref leak but is
+ otherwise as the reporter uploaded it. (closes issue #17182)
+ Reported by: rcasas Patches: app_senddtmf.c.patch_trunk uploaded
+ by rcasas (license 641)
+
+ * channels/h323/ast_h323.cxx: 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: 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 20:08 +0000 [r265367] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_rpt.c: Make app_rpt.c able to compile again.
+
+2010-05-24 19:42 +0000 [r265366] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: reverses incorrect logic introduced by
+ r243200 The decoding of the replace_id did not need to be broken
+ up in this instance. This was brought to my attention again
+ because it caused a segfault when the from or to tags were not
+ present in the "Replaces" header.
+
+2010-05-24 19:06 +0000 [r265317-265320] Terry Wilson <twilson@digium.com>
+
+ * doc/tex/manager.tex: 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/
+
+ * CREDITS, configs/calendar.conf.sample, CHANGES,
+ res/res_calendar_ews.c (added), res/res_calendar.c: Calendaring
+ support for Exchange Server 2007+ via EWS This commit adds
+ support for calendaring with Exchange Server 2007+ via Exchange
+ Web Services. Full write support and for querying attendees. Many
+ thanks to Jan Kaláb for the feature. (closes issue #17022)
+ Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel
+ (license 1008) Tested by: pitel, twilson Review:
+ https://reviewboard.asterisk.org/r/557/ Review:
+ https://reviewboard.asterisk.org/r/668/
+
+2010-05-24 18:19 +0000 [r265316] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: 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)
+
+2010-05-24 16:10 +0000 [r265273] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: fixes segfault when using generic plc
+
+2010-05-23 18:23 +0000 [r265227] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: small changes to avoiding 'freeing unused
+ memory...'
+
+2010-05-21 22:46 +0000 [r265174] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Channel initialization failure causes crashes.
+ __ast_channel_alloc_ap() has several points in the initialization
+ of a new channel structure where it could fail. Since the channel
+ structure is now an ao2 object, the destructor callback needs to
+ be able to handle clean up when the structure setup is
+ incomplete. Problems corrected: 1) Failing to setup the alertpipe
+ would not unreference the structure but free it directly. Doing
+ this to an ao2_object is very bad. 2) File descriptors need to be
+ initialized to -1 before a construction failure could occur so
+ the destructor will not close unopened descriptors. 3) The
+ destructor needs to check that the string field has been
+ initialized before using any string field values. Crashes
+ expected. 4) The destructor should not notify devstate if the
+ device name is empty. It is a waste of cycles and a couple ERROR
+ log messages are generated. Review:
+ https://reviewboard.asterisk.org/r/675/
+
+2010-05-21 21:08 +0000 [r264953-265090] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/file.h, /, apps/app_queue.c: 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: Be sure to set the sin_family on the proxy
+ when allocating. (closes issue #17157) Reported by: stuarth
+
+ * /, include/asterisk/channel.h: 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 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/ ........
+
+ * res/res_fax.c, include/asterisk/res_fax.h,
+ res/res_fax.exports.in, res/res_fax_spandsp.c: Log spandsp's fax
+ debug output to the FAX logger level. Review:
+ https://reviewboard.asterisk.org/r/658
+
+2010-05-21 01:00 +0000 [r264905] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Take dup'd code for directmedia ACLs and
+ make utility func The same code was repeated in lots of different
+ places, so I made a utility fuction for it. This should make the
+ merge in the v6-new branch easier.
+
+2010-05-20 23:29 +0000 [r264828] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/callerid.c: 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 22:23 +0000 [r264752-264779] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: 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: Error message fix. (closes issue #17356)
+ Reported by: kenner Patches: app_stack.c.diff uploaded by kenner
+ (license 1040)
+
+2010-05-20 20:49 +0000 [r264669-264711] Richard Mudgett <rmudgett@digium.com>
+
+ * main/ccss.c: Avoid crash in generic CC agent init if caller name
+ or number is NULL.
+
+ * apps/app_dial.c, apps/app_queue.c: Dial and queue connected line
+ update macro not always run when expected. The connected line
+ update macro would not get run if the connected line number
+ string was empty. The number could be empty if the connected line
+ update did not update a number but the name. It should be run if
+ there was an AST_CONTROL_CONNECTED_LINE frame received for
+ pending dials and queues. Renamed and added some more comments
+ for some confusing identifiers directly connected to the related
+ code. Also fixed a memory leak in app_queue. Review:
+ https://reviewboard.asterisk.org/r/669/
+
+2010-05-20 17:54 +0000 [r264626] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: Add support for direct media ACLs
+ directmediapermit/directmediadeny support to restrict which peers
+ can do directmedia based on ip address. In some networks not all
+ phones are fully routed, i.e. not all phones can ping each other.
+ This patch adds a way to restrict directmedia for certain peers
+ between certain networks. (closes issue #16645) Reported by:
+ raarts Patches: directmediapermit.patch uploaded by raarts
+ (license 937) Tested by: raarts Review:
+ https://reviewboard.asterisk.org/r/467/
+
+2010-05-20 15:30 +0000 [r264497-264540] Kevin P. Fleming <kpfleming@digium.com>
+
+ * addons/ooh323c/src/h323, addons/ooh323c/src: Ignore pre-processed
+ source files generated during DONT_OPTIMIZE dev-mode builds.
+
+ * main/logger.c: Correct 'all logger levels' patch to work
+ properly. Nick Lewis pointed out that the patch as committed
+ wouldn't actually include dynamic logger levels, which was missed
+ by the other reviewers. Thanks!
+
+2010-05-19 21:29 +0000 [r264452] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, channels/chan_sip.c, include/asterisk/_private.h,
+ include/asterisk/options.h, main/asterisk.c, main/loader.c: 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 20:30 +0000 [r264400] David Vossel <dvossel@digium.com>
+
+ * main/udptl.c: 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 20:26 +0000 [r264335-264379] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c: Cast an unsigned int to a signed int when comparing
+ it with 0. (AST-377)
+
+ * /, apps/app_speech_utils.c: 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 19:21 +0000 [r264331] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: 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 17:48 +0000 [r264204-264249] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/options.h: 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: 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 15:39 +0000 [r264161] Leif Madsen <lmadsen@digium.com>
+
+ * main/cli.c: Fix compilation problem with previous commit. (issue
+ #16009)
+
+2010-05-19 15:29 +0000 [r264160] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/logger.c, configs/logger.conf.sample: Add ability for logger
+ channels to include *all* levels. Now that Asterisk modules can
+ dynamically create and destroy logger levels on demand, it's
+ useful to be able to configure a logger channel (console, file,
+ whatever) to be able to accept log messages from *all* levels,
+ even levels created dynamically. This patch adds support for
+ this, by allowing the '*' level name to be used in logger.conf.
+ Review: https://reviewboard.asterisk.org/r/663/
+
+2010-05-19 15:12 +0000 [r264117] Leif Madsen <lmadsen@digium.com>
+
+ * CHANGES, main/cli.c: Add ability to hangup all channels from the
+ CLI. Added the keyword 'all' to the 'channel hangup request' CLI
+ command so that you can request all channels to be hungup without
+ having to restart Asterisk. (closes issue #16009) Reported by:
+ moy Patches: hangup-all-rev-221688.patch uploaded by moy (license
+ 222) Tested by: moy, russell
+
+2010-05-19 14:38 +0000 [r264114] David Vossel <dvossel@digium.com>
+
+ * res/res_rtp_asterisk.c: 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 08:09 +0000 [r264031] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * configs/indications.conf.sample: 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 [r263905-263950] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/dsp.c: 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 ........
+
+ * configure, configure.ac, build_tools/sha1sum-sh (added),
+ makeopts.in, sounds/Makefile: Add an sha1sum-workalike for
+ platforms which don't have it (like Mac OS X)
+
+2010-05-18 22:48 +0000 [r263904] David Vossel <dvossel@digium.com>
+
+ * main/strings.c: 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 21:09 +0000 [r263860] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Be sure to heap-allocate the redirecting to
+ tag so as not to cause crashiness.
+
+2010-05-18 20:49 +0000 [r263858] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_timing_kqueue.c: Make happy green color come back
+
+2010-05-18 20:09 +0000 [r263810] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix memory leaks in redirecting structures
+ in chan_sip.c Thanks to Richard for pointing this out.
+
+2010-05-18 19:30 +0000 [r263807-263808] Jeff Peeler <jpeeler@digium.com>
+
+ * CHANGES: put changes with the correct version
+
+ * /, CHANGES, apps/app_directory.c: 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 23:49 +0000 [r263724] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ makeopts.in, sounds/Makefile, autoconf/ast_ext_lib.m4: Cache
+ sound tarfiles in a common directory, such that a clean reinstall
+ does not force a re-download of the tarballs. (closes issue
+ #15370) Reported by: pprindeville Patches:
+ asterisk-trunk-bugid15370.patch uploaded by pprindeville (license
+ 347) Tested by: pprindeville, tilghman, seanbright
+
+2010-05-17 22:08 +0000 [r263640] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/devicestate.c: 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 19:31 +0000 [r263589] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: 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
+
+2010-05-17 15:36 +0000 [r263541] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, channels/chan_local.c, main/rtp_engine.c,
+ channels/chan_sip.c, include/asterisk/channel.h,
+ configs/misdn.conf.sample, apps/app_queue.c,
+ funcs/func_redirecting.c, channels/misdn_config.c,
+ main/channel.c, main/dial.c, channels/chan_dahdi.c,
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h, main/features.c,
+ funcs/func_connectedline.c, include/asterisk/frame.h,
+ funcs/func_callerid.c, channels/sip/include/sip.h: Enhancements
+ to connected line and redirecting work. From reviewboard: Digium
+ has a commercial customer who has made extensive use of the
+ connected party and redirecting information present in later
+ versions of Asterisk Business Edition and which is to be in the
+ upcoming 1.8 release. Through their use of the feature, new
+ problems and solutions have come about. This patch adds several
+ enhancements to maximize usage of the connected party and
+ redirecting information functionality. First, Asterisk trunk
+ already had connected line interception macros. These macros
+ allow you to manipulate connected line information before it was
+ sent out to its target. This patch adds the same feature except
+ for redirecting information instead. Second, the ast_callerid and
+ ast_party_id structures have been enhanced to provide a "tag."
+ This tag can be set with func_callerid, func_connectedline,
+ func_redirecting, and in the case of DAHDI, mISDN, and SIP
+ channels, can be set in a configuration file. The idea behind the
+ callerid tag is that it can be set to whatever value the
+ administrator likes. Later, when running connected line and
+ redirecting macros, the admin can read the tag off the
+ appropriate structure to determine what action to take. You can
+ think of this sort of like a channel variable, except that
+ instead of having the variable associated with a channel, the
+ variable is associated with a specific identity within Asterisk.
+ Third, app_dial has two new options, s and u. The s option lets a
+ dialplan writer force a specific caller ID tag to be placed on
+ the outgoing channel. The u option allows the dialplan writer to
+ force a specific calling presentation value on the outgoing
+ channel. Fourth, there is a new control frame subclass called
+ AST_CONTROL_READ_ACTION added. This was added to correct a very
+ specific situation. In the case of SIP semi-attended (blond)
+ transfers, the party being transferred would not have the
+ opportunity to run a connected line interception macro to
+ possibly alter the transfer target's connected line information.
+ The issue here was that during a blond transfer, the SIP transfer
+ code has no bridged channel on which to queue the connected line
+ update. The way this was corrected was to add this new control
+ frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on
+ the channel on which the connected line interception macro should
+ be run. When ast_read is called to read the frame, ast_read
+ responds by calling a callback function associated with the
+ specific read action the control frame describes. In this case,
+ the action taken is to run the connected line interception macro
+ on the transferee's channel. Review:
+ https://reviewboard.asterisk.org/r/652/
+
+2010-05-17 15:14 +0000 [r263375-263460] Leif Madsen <lmadsen@digium.com>
+
+ * main/manager.c: Missing newlines added to Set-Cookie line in
+ manager.c Sean Bright pointed out that we lost a set of newline
+ characters in commit 190349 on a line I had recently changed. Yay
+ for code review on commits. (issue #17231, #10961)
+
+ * main/manager.c, /: 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 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 13:05 +0000 [r263294] David Vossel <dvossel@digium.com>
+
+ * CHANGES: Update CHANGES to reflect DAHDI buffer dialstring option
+ backport to 1.6.2
+
+2010-05-16 16:31 +0000 [r263250] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/scripts/live_ast: live_ast: add commands 'rsync' and
+ 'gen-live-asterisk' This adds the following two commands to
+ live_ast: * rsync [user]@host directory Copy over all generated
+ files to <directory> at remote host. Would allow running live_ast
+ there. Hence allows separating a build machine from a test
+ machine. * gen-live-asteris: regenerate live/asterisk . Useful if
+ copying over files to a different directory.
+
+2010-05-16 11:14 +0000 [r263208] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astobj2.c: Improve some very confusing structure names in
+ astobj2.c As pointed out by 'akshayb' on #asterisk-dev, the code
+ here called a list of bucket entries a 'bucket', and the entries
+ within the bucket were called 'bucket_list'. This made the code
+ very hard to understand without reading all of it... so I've
+ renamed 'bucket_list' to 'bucket_entry' to clarify the purpose of
+ the structure.
+
+2010-05-14 18:53 +0000 [r263151] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fix iax_frame double free Very unfortunate
+ things happen if we add an iax_frame to the frame queue and let
+ go of the lock before scheduling the frame's transmit... There is
+ a race condition that exists where the frame can be removed from
+ the frame_queue and freed before the transmit is scheduled if we
+ do not hold on to that lock. This results in a freed frame being
+ scheduled for transmit later.
+
+2010-05-13 22:01 +0000 [r263069] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix inverted logic in cli command: ss7 set
+ debug on/off
+
+2010-05-13 20:25 +0000 [r263028] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * configure, configure.ac: Remove "untested" feature PRI_VERSION
+ Nobody seems to actually test PRI_VERSION. It is only useful for
+ failing PRI support in chan_dahdi.
+
+2010-05-13 17:49 +0000 [r262940-262987] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_timing_kqueue.c: For FreeBSD
+
+ * res/res_timing_kqueue.c: Hmmm, probably should have read the
+ manpage more thoroughly.
+
+2010-05-13 15:36 +0000 [r262895-262897] Russell Bryant <russell@digium.com>
+
+ * channels/chan_console.c: Fix an off by one error that causes a
+ crash. Thanks to Raymond Burke for pointing it out.
+
+ * main/stdtime/localtime.c: Fix build on linux.
+
+ * pbx/pbx_spool.c: Fix build on linux.
+
+2010-05-13 05:37 +0000 [r262852] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, pbx/pbx_spool.c, tests/test_time.c,
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ main/stdtime/localtime.c, res/res_timing_kqueue.c (added): Add
+ kqueue(2) implementation to Asterisk in various places. This will
+ save a considerable amount of CPU on the BSDs, including Mac OS
+ X, as it eliminates several places in the code that we previously
+ used a busy loop. Additionally, this adds a res_timing interface,
+ using kqueue timers. Review:
+ https://reviewboard.asterisk.org/r/543/
+
+2010-05-12 19:59 +0000 [r262800] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/loader.c, main/cli.c: 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 [r262796-262798] Leif Madsen <lmadsen@digium.com>
+
+ * res/ael/pval.c: 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: Remove unnecessary WARNING message in ael/pval.c
+ (closes issue #17145) Reported by: okrief
+
+2010-05-12 18:01 +0000 [r262744] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_meetme.c: 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 17:57 +0000 [r262661-262743] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Don't crash when destroying chan_dahdi
+ pseudo channels. Must do a deep copy of the cc_params in
+ duplicate_pseudo(). Otherwise, when the duplicate pseudo channel
+ is destroyed, it frees the original pseudo channel cc_params. The
+ original pseudo channel is then left with a dangling pointer for
+ when the next duplicated pseudo channel is created.
+
+ * channels/chan_misdn.c: Merged revisions 262657,262660 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed,
+ 12 May 2010) | 4 lines Forgot some conditionals around the
+ callrerouting facility help text. JIRA ABE-2223 ..........
+ r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010)
+ | 22 lines Add mISDN Call rerouting facility for point-to-point
+ ISDN lines (exchange line) In the case of ISDN
+ point-to-multipoint (multidevice) you can use the mISDN "facility
+ calldeflect" application for call diversions from external (PSTN)
+ to external (PSTN). In that case this is the only way to get rid
+ of the two call legs to the PBX and let the calling number at the
+ C party become the number of the A party. In the case of ISDN
+ point-to-point (exchange line) the call deflection facility may
+ not be used. Instead a call rerouting facility has to be used.
+ This patch for chan_misdn.c is an extension to realize this
+ service (facility rerouting application). It can accept either
+ spelling: "callrerouting" or "callrerouteing". The patch is
+ tested towards Deutsche Telekom and requires a modified version
+ of mISDN from Digium, Inc. Patches:
+ misdn_rerouteing_corrected.patch (Slightly modified.) JIRA
+ ABE-2223
+
+2010-05-12 16:23 +0000 [r262656] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_privacy.c: 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
+
+2010-05-12 01:00 +0000 [r262613] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_sip.c, include/asterisk/cli.h: Convert to
+ AST_CLI_YESNO and AST_CLI_ONOFF Clean up chan_sip.c to use new
+ AST_CLI functions (closes issue #17287) Reported by: pabelanger
+ Patches: issue17287.patch uploaded by pabelanger (license 224)
+ Tested by: russell
+
+2010-05-11 23:18 +0000 [r262569] Richard Mudgett <rmudgett@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: Dialing an invalid extension causes
+ incomplete hangup sequence. Revision -r1489 of the libpri 1.4
+ branch corrected a deviation from Q.931 Section 5.3.2. However,
+ this resulted in an unexpected behaviour change to the upper
+ layer (Asterisk). This change uses pri_hangup_fix_enable() to
+ follow Q.931 Section 5.3.2 call hangup better if the version of
+ libpri supports it. (issue #17104) Reported by: shawkris Tested
+ by: rmudgett
+
+2010-05-11 21:25 +0000 [r262513] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/causes.h: 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 19:57 +0000 [r262422] Jason Parker <jparker@digium.com>
+
+ * /, res/Makefile: 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 19:40 +0000 [r262414-262419] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * pbx/pbx_config.c: 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: 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 17:23 +0000 [r262330] Tilghman Lesher <tlesher@digium.com>
+
+ * /, Makefile.rules, apps/app_voicemail.c: 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) ........
+
+2010-05-11 16:43 +0000 [r262299] Jason Parker <jparker@digium.com>
+
+ * bootstrap.sh: Allow bootstrap script to work on Solaris. As
+ usual, the way they do things is different, so we need to account
+ for that. automake is versioned ala BSD/Linux, but autoconf is
+ not. We don't actually need to specify a version there, since
+ AC_PREREQ will cover it for us. Things will fail pretty loudly if
+ AC_PREREQ isn't met. (closes issue #16341) Reported by: bklang
+ Patches: opensolaris_bootstrap.sh uploaded by bklang (license
+ 919)
+
+2010-05-10 19:06 +0000 [r262236-262240] David Vossel <dvossel@digium.com>
+
+ * apps/app_directed_pickup.c: 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: 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 16:36 +0000 [r262152] Tilghman Lesher <tlesher@digium.com>
+
+ * /, Makefile.rules: 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 02:14 +0000 [r262048-262102] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_c_define_check.m4, configure,
+ include/asterisk/autoconfig.h.in, autoconf/ast_ext_lib.m4,
+ autoconf/ast_c_compile_check.m4: 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: Use CPPFLAGS to pass PTHREAD_CFLAGS for
+ vpb only
+
+2010-05-07 23:54 +0000 [r262005] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * UPGRADE.txt, apps/app_voicemail.c: VoicemailMain and
+ VMauthenticate, allow escape to the 'a' extension when a single
+ '*' is entered Where a site uses VoicemailMain(mailbox) the users
+ have to be at their own extension to clear their voicemail, they
+ have no way of escaping VoicemailMain to allow entry of new
+ boxnumber. This patch, allows a site to include to 'a' priority
+ in the VoicemailMain context, to allow an escape. If the 'a'
+ priority doesn't exist in the context that VoicemailMain was
+ called from then it acts as the old behaviour. Reported by:
+ alecdavis Tested by: alecdavis Patch vm_a_extension.diff2.txt
+ uploaded by alecdavis (license 585) Review:
+ https://reviewboard.asterisk.org/r/489/
+
+2010-05-07 22:09 +0000 [r261913-261964] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/ooh323c/src/ooh323.c: Fix build on Linux
+
+ * funcs/func_odbc.c: 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: 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 16:05 +0000 [r261867] Leif Madsen <lmadsen@digium.com>
+
+ * UPGRADE-1.6.txt: Update UPGRADE-1.6.txt stating insecure=very has
+ been removed. (closes issue #17282) Reported by: stuarth Tested
+ by: stuarth
+
+2010-05-07 15:33 +0000 [r261866] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_pri.c: Fix deadlock in sig_pri when hanging up. The
+ pri_dchannel thread currently violates locking order by locking
+ the private and then attempting to queue a frame, which needs to
+ lock the channel. Queueing a frame is unneccesary though and is
+ actually a regression since sig_pri. All the places that
+ currently use ast_softhangup_nolock now will just set the
+ softhangup value directly as before. (closes issue #17216)
+ Reported by: lmsteffan Patches: bug17216.patch uploaded by
+ jpeeler (license 325)
+
+2010-05-06 23:41 +0000 [r261822] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Some code optimizations. * Made more places
+ use pri_queue_control() instead of pri_queue_frame() and a local
+ frame variable. * Made pri_queue_frame() use
+ sig_pri_lock_owner(). pri_queue_frame() no longer releases the
+ libpri access lock unless it is required. * Made the
+ pri_queue_frame() and pri_queue_control() parameter list similar
+ to sig_pri_lock_owner().
+
+2010-05-06 20:11 +0000 [r261736] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: 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 17:06 +0000 [r261609] Jason Parker <jparker@digium.com>
+
+ * /, sounds/Makefile: 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 15:39 +0000 [r261560] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/sip/include/sip.h: 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
+
+2010-05-06 14:15 +0000 [r261496-261500] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c: Add test case for removing random elements
+ from a heap. I modified the original patch for trunk to use the
+ unit test API. (issue #17277) Reported by: cappucinoking Patches:
+ test_heap.diff uploaded by cappucinoking (license 1036) Tested
+ by: cappucinoking, russell
+
+ * main/heap.c: 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 07:27 +0000 [r261451] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: When failing to configure, don't destroy
+ 'cfg' twice Fixes a crash when some config section had an
+ incorrect channel config.
+
+2010-05-05 22:22 +0000 [r261405] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Avoid a crash on SS7 channels.
+
+2010-05-05 20:48 +0000 [r261364] Russell Bryant <russell@digium.com>
+
+ * Makefile, configs/asterisk.conf.sample: Restore previous
+ asterisk.conf syntax, where the directories aren't commented out.
+ This fixes some breakage in the test suite, that uses the
+ contents of asterisk.conf to discover the install layout on the
+ system.
+
+2010-05-05 19:13 +0000 [r261316] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes sip native transfer The Refer-To
+ header field containing the Replaces header in the URI was not
+ being decoded properly. This caused invalid parsing between the
+ caller id field and the domain resulting in a failed transfer.
+ (closes issue #17284) Reported by: dvossel
+
+2010-05-05 18:43 +0000 [r261314] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: 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/ ........
+
+2010-05-05 18:28 +0000 [r261313] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/dialplan_functions.c: Prevent unnecessary warnings
+ when getting rtpsource or rtpdest. If a recognized media type was
+ present, but the media type was not enabled for the channel, then
+ a warning would be emitted. For instance, attempting to get
+ CHANNEL(rtpsource,video) on a call with no video would cause a
+ warning message to appear. With this change, the warning will
+ only appear if the stream argument is not recognized as being a
+ media type that can be specified.
+
+2010-05-05 15:42 +0000 [r261124-261232] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_queue.c: 'queue reset stats' erroneously clears
+ wrapuptime configuration. Resets each member's lastcall to 0 now.
+ (closes issue #17262) Reported by: rain Patches:
+ wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
+ by: rain
+
+ * main/manager.c, include/asterisk/cli.h, CHANGES,
+ include/asterisk/manager.h: New 'manager show settings' CLI
+ command. See the CHANGES file for more details. (closes issue
+ #16343) Reported by: pabelanger Patches: issue16343.patch.v5
+ uploaded by pabelanger (license 224) Tested by: pabelanger,
+ tilghman, lmadsen Review: https://reviewboard.asterisk.org/r/630/
+
+ * Makefile, configs/asterisk.conf.sample (added): New static
+ asterisk.conf.sample file. This simply moves the functionality
+ from the Makefile (cleaning it up) into an external
+ asterisk.conf.samples file. Also updates formatting (easier to
+ read) and grammar changes to asterisk.conf.samples. (closes issue
+ #17027) Reported by: pabelanger Patches:
+ 0017027.asterisk.conf.v6.patch uploaded by pabelanger (license
+ 224) Tested by: qwell, lmadsen, pabelanger, chappell Review:
+ https://reviewboard.asterisk.org/r/616/
+
+2010-05-04 23:51 +0000 [r261095] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: 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 22:46 +0000 [r261051] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add new
+ possible value to autopause option to allow members to be
+ autopaused in all queues. See the CHANGES file and
+ queues.conf.sample for more details. (closes issue #17008)
+ Reported by: jlpedrosa Patches: queues.autopause_en_review.diff
+ uploaded by jlpedrosa (license 1002) Review:
+ https://reviewboard.asterisk.org/r/581/
+
+2010-05-04 21:10 +0000 [r261007] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h, channels/sig_pri.c: The inalarm flag is
+ not passed up from the sig_analog and sig_pri submodules. The CLI
+ "dahdi show channel" command was not correctly reporting the
+ InAlarm status. The inalarm flag is now consistently passed
+ between chan_dahdi and submodules.
+
+2010-05-04 18:51 +0000 [r260924] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: 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 15:49 +0000 [r260802] Jason Parker <jparker@digium.com>
+
+ * /, build_tools/make_build_h: 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 ........
+
+2010-05-03 22:13 +0000 [r260757] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c, CHANGES: Add new admin features to meetme:
+ Roll call, eject all, mute all, record in-conf This patch adds
+ the following in-conference admin DTMF features: *81 - Roll call
+ (or simply user count if INTROUSER isn't enabled) *82 - Eject all
+ non-admins *83 - Mute/unmute all non-admins *84 - Start recording
+ the conference on the fly FWIW, this code uses newly recorded
+ prompts. (closes issue #16379) Reported by: rfinnie Patches:
+ meetme-enhancements-232771-v1.patch uploaded by rfinnie (license
+ 940) modified slightly by me
+
+2010-05-03 17:06 +0000 [r260663] 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 14:58 +0000 [r260570] Leif Madsen <lmadsen@digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt: 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. ........
+
+2010-05-02 02:52 +0000 [r260521] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/data.c, include/asterisk/data.h: Avoid making AstData depend
+ on libxml2 to compile. We have some functions inside the AstData
+ API to get the tree in XML form, but it is not required at the
+ moment to compile asterisk and we can disable that part of the
+ API if we don't have libxml2 support.
+
+2010-04-30 22:36 +0000 [r260437] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: 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 22:24 +0000 [r260435] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7,
+ and MFCR2 users. Created SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS
+ SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS SIG_MFCR2_MAX_CHANNELS
+ Also fixed the declaration of pollers[] in mfcr2_monitor(). It
+ was dimensioned to the number of bytes in struct
+ dahdi_mfcr2.pvts[] and not to the same dimension of the struct
+ dahdi_mfcr2.pvts[].
+
+2010-04-30 20:11 +0000 [r260344-260346] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: 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
+ ........
+
+ * apps/app_queue.c: Fix logic reversal error when queue callers
+ join the queue. When a specific position is specified for the
+ queue, the idea was that the caller cannot be placed ahead of
+ higher-priority callers. Unfortunately, the logic was reversed so
+ that the caller could ONLY be placed ahead of higher priority
+ callers. Discovered while writing a unit test.
+
+2010-04-30 06:19 +0000 [r260280-260292] Tilghman Lesher <tlesher@digium.com>
+
+ * main/strcompat.c: 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. Also introduce an additional
+ optimization, in the form of using the CLOEXEC flag to close
+ descriptors at the right time. (closes issue #17223) Reported by:
+ dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
+ tilghman (license 14) Tested by: dbackeberg
+
+ * configs/extensions.conf.sample: 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 22:44 +0000 [r260231] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: 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 18:15 +0000 [r260148] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Pattern match fail.
+
+2010-04-29 15:33 +0000 [r260050] David Vossel <dvossel@digium.com>
+
+ * /, include/asterisk/audiohook.h, main/audiohook.c: 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/ ........
+
+2010-04-29 00:35 +0000 [r260007] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/extconf.h: Fix comment.
+
+2010-04-28 22:34 +0000 [r259957] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: 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 21:20 +0000 [r259870] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, channels/chan_local.c, /: 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 21:08 +0000 [r259853] Leif Madsen <lmadsen@digium.com>
+
+ * /, config.guess: 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 20:32 +0000 [r259760-259848] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: 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 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 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 17:18 +0000 [r259672] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: 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 22:47 +0000 [r259587-259617] Jason Parker <jparker@digium.com>
+
+ * res/res_agi.c: Fix compile on systems without
+ HAVE_NULLSAFE_PRINTF defined.
+
+ * channels/sip/dialplan_functions.c: Be more explicit about field
+ naming in a test.
+
+2010-04-27 22:18 +0000 [r259538] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: 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 22:11 +0000 [r259533] Mark Michelson <mmichelson@digium.com>
+
+ * main/ccss.c: Shuffle some casts to make builds on bamboo happier.
+
+2010-04-27 21:49 +0000 [r259527] Leif Madsen <lmadsen@digium.com>
+
+ * /, sounds/Makefile: 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 21:18 +0000 [r259439-259451] Jason Parker <jparker@digium.com>
+
+ * /: Block 259441 instead of recording it as merged.
+
+ * /: Recorded merge of revisions 259441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259441 | qwell | 2010-04-27 16:15:46 -0500 (Tue, 27 Apr 2010) |
+ 1 line Add gar to the check for AR for those silly OSes (Solaris)
+ that don't have ar. ........
+
+ * main/editline/configure, main/editline/Makefile.in,
+ main/editline/configure.in: 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.
+
+2010-04-27 21:10 +0000 [r259438] Leif Madsen <lmadsen@digium.com>
+
+ * include/asterisk/doxygen/mantisworkflow.h: Update the Mantis
+ Workflow document in doxygen. (closes issue #17175) Reported by:
+ lmadsen Patches: Bug_Tracker_Workflow.v2.txt uploaded by
+ pabelanger (license 224) Tested by: pabelanger, lmadsen
+
+2010-04-27 19:52 +0000 [r259357] Mark Michelson <mmichelson@digium.com>
+
+ * main/ccss.c: Change cc_ref and cc_unref from macros to inline
+ functions. The hope is that Solaris won't be as whiny after this
+ change.
+
+2010-04-27 19:31 +0000 [r259353] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: 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 18:29 +0000 [r259229-259307] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: 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 ........
+
+ * channels/chan_dahdi.c: Re-fix dahdi_request() iflist locking
+ since CCSS merged.
+
+2010-04-27 15:25 +0000 [r259189] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/etc_default_asterisk (added): Add missing file
+ (pointed out by TheDavidFactor on #asterisk-dev) referenced by
+ revision 239231.
+
+2010-04-26 21:45 +0000 [r259023-259105] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: 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 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 19:05 +0000 [r258974] David Ruggles <thedavidfactor@gmail.com>
+
+ * contrib/valgrind.supp: Line 24 missed in compatibility fix in
+ revision 233577 added a "fun:" prefix line 24
+
+2010-04-26 15:59 +0000 [r258934] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_sip.c: 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
+
+2010-04-26 14:18 +0000 [r258896] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, include/asterisk/res_fax.h, res/res_fax_spandsp.c:
+ Update res_fax and res_fax_spandsp to be compatible with Fax For
+ Asterisk 1.2. The fax session initilization code for T.38 faxes
+ has been rewritten. T.38 session initialization was removed from
+ generic_fax_exec, and split into two different code paths for
+ receive and send. Also the 'z' option (to send a T.38 reinvite if
+ we do not receive one) was added to sendfax. In the output of
+ 'fax show sessions', the 'Type' column has been renamed to 'Tech'
+ and replaced with a new 'Tech' column that will report 'G.711' or
+ 'T.38'. Control of ECM defaults has been added to res_fax A 'fax
+ show settings' CLI command has been added. Support of the new
+ AST_T38_REQUEST_PARMS control method request to handle channels
+ that have already received a T.38 reinvite before the FAX
+ application is start has been added. Support for the 'fax show
+ settings' command has been added to res_fax_spandsp and handling
+ of the ECM flag has been slightly altered.
+
+2010-04-25 18:51 +0000 [r258838-258855] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: additional checking related to issue 17186
+
+ * addons/chan_ooh323.c: Don't pass zero length callerid to ooh323
+ stack Don't pass zero callerid string to ooh323 stack because it
+ can't encode this properly and can't generate setup message.
+ (closes issue #17186) Reported by: vmikhelson Patches:
+ zero_callerid_num.patch uploaded by may213 (license 454) Tested
+ by: may213
+
+2010-04-25 18:12 +0000 [r258776] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_monitor.c: 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 22:19 +0000 [r258685] Jason Parker <jparker@digium.com>
+
+ * utils/extconf.c: Add another random function that does nothing to
+ make the utils/ dir happy.
+
+2010-04-22 22:11 +0000 [r258675] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c: Fix previous commit.
+
+2010-04-22 22:10 +0000 [r258673-258674] Jason Parker <jparker@digium.com>
+
+ * utils/Makefile, utils/extconf.c: Make utils/ stuff *actually*
+ compile this time.
+
+ * utils/Makefile, utils/extconf.c: Let utils/ dir compile when
+ DEBUG_THREADS is not enabled.
+
+2010-04-22 21:57 +0000 [r258671] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, main/channel.c, /, main/features.c: 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
+
+2010-04-22 21:06 +0000 [r258632] Russell Bryant <russell@digium.com>
+
+ * tests/test_event.c, main/event.c: 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/
+
+2010-04-22 20:04 +0000 [r258595] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_voicemail.c: Pass interactive = 0 and fix a compile
+ error.
+
+2010-04-22 19:08 +0000 [r258557] Jason Parker <jparker@digium.com>
+
+ * main/lock.c (added), include/asterisk/res_odbc.h,
+ include/asterisk/astobj2.h, main/heap.c, include/asterisk/lock.h,
+ main/astobj2.c, res/res_odbc.c, include/asterisk/heap.h: Remove
+ ABI differences that occured when compiling with DEBUG_THREADS.
+ "Bad Things" would happen if Asterisk was compiled with
+ DEBUG_THREADS, but a loaded module was not (or vice versa). This
+ also immensely simplifies the lock code, since there are no
+ longer 2 separate versions of them. Review:
+ https://reviewboard.asterisk.org/r/508/
+
+2010-04-22 18:07 +0000 [r258517] Eliel C. Sardanons <eliels@gmail.com>
+
+ * doc/manager_1_1.txt, main/channel.c, include/asterisk/doxyref.h,
+ include/asterisk/xml.h, main/data.c (added), main/xml.c,
+ include/asterisk/channel.h, include/asterisk/_private.h,
+ include/asterisk/data.h (added), CHANGES, apps/app_queue.c,
+ main/asterisk.c, apps/app_voicemail.c: Asterisk data retrieval
+ API. This module implements an abstraction for retrieving and
+ exporting asterisk data. Developed by: Brett Bryant
+ <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY)
+ <eliels@gmail.com> For the Google Summer of code 2009 Project.
+ Documentation can be found in doxygen format and inside the
+ header include/asterisk/data.h Review:
+ https://reviewboard.asterisk.org/r/275/
+
+2010-04-22 17:36 +0000 [r258515] Russell Bryant <russell@digium.com>
+
+ * doc/tex/channelvariables.tex: Add MEETMEBOOKID from r256019.
+
+2010-04-21 21:56 +0000 [r258433] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: 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 19:45 +0000 [r258351-258387] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/asterisk.tex: Missed this when reverting the bad version
+ change in asterisk.tex.
+
+ * doc/tex/asterisk.tex: Fix change in asterisk.tex that got merged
+ in after testing. (issue #17220)
+
+ * Makefile, doc/tex/security-events.tex, configure,
+ include/asterisk/autoconfig.h.in, doc/tex/Makefile, configure.ac,
+ doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
+ build_tools/prep_tarball, doc/tex/localchannel.tex,
+ doc/tex/enum.tex, makeopts.in, doc/tex/asterisk.tex,
+ doc/tex/cel-doc.tex: 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 19:07 +0000 [r258345] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_callcompletion.c: Add small documentation update to
+ func_callcompletion.c. This directs users to documents which can
+ help explain the concepts and configuration options settable with
+ the function.
+
+2010-04-21 19:02 +0000 [r258344] Leif Madsen <lmadsen@digium.com>
+
+ * UPGRADE.txt, CHANGES, channels/chan_iax2.c: IAXpeers output now
+ matches SIPpeers format for manager (AMI). (closes issue #17100)
+ Reported by: secesh Tested by: pabelanger Review:
+ https://reviewboard.asterisk.org/r/594/
+
+2010-04-21 18:13 +0000 [r258305] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: 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)
+
+2010-04-21 13:26 +0000 [r258265] Leif Madsen <lmadsen@digium.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ res/res_calendar_caldav.c: Fix the \brief description in the
+ res_calendar_*.c files.
+
+2010-04-21 13:24 +0000 [r258190-258256] Julian Lyndon-Smith <julian@dotr.com>
+
+ * doc/manager_1_1.txt: fix whitespace issue
+
+ * doc/manager_1_1.txt, doc/tex/manager.tex: Added NEW ACTIONS entry
+ for new MixMonitorMute AMI command. Added State and Direction
+ variables for new MixMonitorMute AMI command.
+
+ * CHANGES: Added CHANGES entry for new MixMonitorMute AMI command.
+
+ * main/frame.c, include/asterisk/audiohook.h, main/audiohook.c,
+ include/asterisk/frame.h, apps/app_mixmonitor.c,
+ res/res_mutestream.c: Added MixMonitorMute manager command Added
+ a new manager command to mute/unmute MixMonitor audio on a
+ channel. Added a new feature to audiohooks so that you can mute
+ either read / write (or both) types of frames - this allows for
+ MixMonitor to mute either side of the conversation without
+ affecting the conversation itself. (closes issue #16740) Reported
+ by: jmls Review: https://reviewboard.asterisk.org/r/487/
+
+2010-04-20 19:02 +0000 [r258106-258149] Leif Madsen <lmadsen@digium.com>
+
+ * configs/cli_aliases.conf.sample: Add 'soft hangup' alias per
+ Steve Johnson on asterisk-users.
+
+ * configs/extensions.conf.sample: 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
+
+ * contrib/scripts/sip-friends.sql: 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
+
+2010-04-20 17:06 +0000 [r258065] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: 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 12:38 +0000 [r257988] Leif Madsen <lmadsen@digium.com>
+
+ * formats/format_pcm.c: Update supported file extensions in
+ doxygen. Updated the doxygen \arg line after looking at the file
+ for some other Asterisk documentation and noticing they weren't
+ up to date. Thanks to seanbright for looking at the code for me
+ :)
+
+2010-04-19 21:57 +0000 [r257947-257949] Jason Parker <jparker@digium.com>
+
+ * main/indications.c: Change log message to match severity.
+
+ * main/indications.c: 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 19:23 +0000 [r257883] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Bad merge fix
+
+2010-04-19 18:42 +0000 [r257851] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_srv.c: Commit compromise I suggested on review 608.
+ This allows for multiple SRV queries to be done from the dialplan
+ for the same service on a single call while still allowing one to
+ bypass the call to SRVQUERY if they so please. Taking action
+ since no comments had been left for a while. This can easily be
+ reverted if needed. External tests still pass.
+
+2010-04-19 17:57 +0000 [r257810] Terry Wilson <twilson@digium.com>
+
+ * main/features.c: 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 17:25 +0000 [r257768] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/cdr_odbc.conf.sample: Removing unused configuration
+ parameters
+
+2010-04-16 21:22 +0000 [r257713] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * /, apps/app_mixmonitor.c: 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/ ........
+
+2010-04-16 19:50 +0000 [r257646] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Make sure to fail a monitor if we receive a
+ negative response for a CC SUBSCRIBE.
+
+2010-04-16 19:25 +0000 [r257642] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * channels/chan_dahdi.c: Enable PRI SERVICE message support in
+ chan_dahdi for the 'national' switchtype Revision 1072 of libpri
+ added SERVICE message support for the 'national' switchtype. The
+ attached patch enables the use of 'pri service' CLI commands on
+ dahdi channels that are configured for the 'national' switchtype.
+ (closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch
+ uploaded by dhubbard (license 733) Tested by: elguero, dhubbard
+ Review: https://reviewboard.asterisk.org/r/612/
+
+2010-04-15 21:26 +0000 [r257493-257560] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, /, tests/test_app.c, main/app.c: 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 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 19:41 +0000 [r257343-257427] Leif Madsen <lmadsen@digium.com>
+
+ * /, doc/backtrace.txt: 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 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 22:57 +0000 [r257262] Tilghman Lesher <tlesher@digium.com>
+
+ * main/features.c, configs/features.conf.sample: 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 19:17 +0000 [r257191] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: 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 18:10 +0000 [r257146] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /, configs/manager.conf.sample: 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 16:33 +0000 [r257065] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: 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
+
+2010-04-13 16:18 +0000 [r256985-257032] Mark Michelson <mmichelson@digium.com>
+
+ * configs/sip.conf.sample: Update sample dialstrings in
+ sip.conf.sample file.
+
+ * funcs/func_srv.c: Address Russell's comments on func_srv from
+ reviewboard. * Change copyright date * Place channel in
+ autoservice when doing SRV lookup * Get rid of trailing
+ whitespace * Change logic in load_module function
+
+ * main/ccss.c: Fix issue where recall would not happen when it
+ should. Specifically, the situation would happen when multiple
+ callers would request CC for a single generically-monitored
+ device. If the monitored device became available but the caller
+ did not answer the recall, then there was nothing that would poke
+ the CC core to let it know that it should attempt to recall
+ someone else instead. After careful consideration, I came to the
+ conclusion that the only area of Asterisk that needed to be
+ touched was the generic CC monitor. All other types of CC would
+ require something outside of Asterisk to invoke a recall for a
+ separate device. This was accomplished by changing the generic
+ monitor destructor to poke other generic monitor instances if the
+ device is currently available and the specific instance was
+ currently not suspended. In order to not accidentally trigger
+ recalls at bad times, the fit_for_recall flag was also added to
+ the generic_monitor_instance_list struct. This gets set as soon
+ as a monitored device becomes available. It gets cleared if a
+ CCNR request triggers the creation of a new generic monitor
+ instance. By doing this, we don't accidentally try to recall a
+ device when the monitored device was being monitored for CCNR and
+ never actually became available for recall in the first place.
+ This error was discovered by Steve Pitts during in-house testing
+ at Digium.
+
+2010-04-12 17:29 +0000 [r256860-256901] Leif Madsen <lmadsen@digium.com>
+
+ * /, doc/HOWTO_collect_debug_information.txt (added): 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: Remove silly debug message that is not
+ useful. (issue #17159)
+
+2010-04-12 14:47 +0000 [r256823] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: gives channel reference before unlocking it
+ and using setvar helper. To guarantee the channel is valid when
+ calling setvar on the MASTER_CHANNEL dialplan function, a channel
+ reference must be taken before unlocking. Thanks to russell for
+ pointing out the error.
+
+2010-04-12 14:39 +0000 [r256821] Leif Madsen <lmadsen@digium.com>
+
+ * main/logger.c: 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)
+
+2010-04-12 02:19 +0000 [r256745-256783] Russell Bryant <russell@digium.com>
+
+ * tests/test_substitution.c: test_substitution expects func_curl to
+ be present to work.
+
+ * tests/test_pbx.c: Add ASTERISK_FILE_VERSION() macro
+
+2010-04-10 08:33 +0000 [r256704] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/scripts/safe_asterisk.8, doc/asterisk.8,
+ contrib/scripts/autosupport.8, contrib/scripts/astgenkey.8: fix
+ hyphen vs. minus in man pages In troff '-' is used for a hyphen.
+ A minus is denoted by '\-' . This is normally also used for a
+ dash. This patch converts all '-'-s that are minuses or dashes to
+ '\-'.
+
+2010-04-09 22:20 +0000 [r256646-256661] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, main/ccss.c: Remove status_response
+ callbacks where they are not needed.
+
+ * channels/chan_local.c: Prevent crash when originating a call to a
+ local channel. Call completion code tries to grab the call
+ completion parameters from the requesting channel during
+ local_request. When originating a call to a local channel,
+ however, this channel is NULL. This was causing an issue for me
+ when trying to run a test script.
+
+2010-04-09 19:46 +0000 [r256569-256608] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/CCSS_architecture.pdf (added): Merge CCSS architecture
+ document from CCSS branch.
+
+ * channels/sig_pri.h, configure, include/asterisk/autoconfig.h.in:
+ Remove PRI CCSS BUGBUG message and update configure script.
+
+2010-04-09 16:04 +0000 [r256485-256530] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/reqresp_parser.c, channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h: Add routines for parsing
+ SIP URIs consistently. From the original issue report opened by
+ Nick Lewis: Many sip headers in many sip methods contain the ABNF
+ structure name-andor-addr = name-addr / addr-spec Examples
+ include the to-header, from-header, contact-header,
+ replyto-header At the moment chan_sip.c makes various different
+ attempts to parse this name-andor-addr structure for each header
+ type and for each sip method with sometimes limited degrees of
+ success. I recommend that this name-andor-addr structure be
+ parsed by a dedicated function and that it be used irrespective
+ of the specific method or header that contains the
+ name-andor-addr structure Nick has also included unit tests for
+ verifying these routines as well, so...heck yeah. (closes issue
+ #16708) Reported by: Nick_Lewis Patches:
+ reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis
+ (license 657 Review: https://reviewboard.asterisk.org/r/549
+
+ * channels/chan_sip.c, tests/test_gosub.c, funcs/func_srv.c: Fix
+ some compiler errors that popped up after the CCSS merge.
+
+ * apps/app_dial.c, configs/chan_dahdi.conf.sample,
+ include/asterisk/devicestate.h, include/asterisk/xml.h,
+ channels/chan_local.c, doc/tex/ccss.tex (added), main/ccss.c
+ (added), channels/chan_sip.c, configure.ac, main/xml.c,
+ include/asterisk/channel.h, configs/manager.conf.sample,
+ include/asterisk/channelstate.h (added),
+ include/asterisk/manager.h, CHANGES, channels/sig_pri.c,
+ channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c,
+ main/manager.c, funcs/func_callcompletion.c (added),
+ channels/sig_analog.c, channels/sig_analog.h,
+ configs/ccss.conf.sample (added), include/asterisk/rtp_engine.h,
+ include/asterisk/frame.h, include/asterisk/ccss.h (added),
+ doc/tex/asterisk.tex, main/asterisk.c,
+ channels/sip/include/sip.h: Merge Call completion support into
+ trunk. From Reviewboard: CCSS stands for Call Completion
+ Supplementary Services. An admittedly out-of-date overview of the
+ architecture can be found in the file doc/CCSS_architecture.pdf
+ in the CCSS branch. Off the top of my head, the big differences
+ between what is implemented and what is in the document are as
+ follows: 1. We did not end up modifying the Hangup application at
+ all. 2. The document states that a single call completion monitor
+ may be used across multiple calls to the same device. This proved
+ to not be such a good idea when implementing protocol-specific
+ monitors, and so we ended up using one monitor per-device
+ per-call. 3. There are some configuration options which were
+ conceived after the document was written. These are documented in
+ the ccss.conf.sample that is on this review request. For some
+ basic understanding of terminology used throughout this code, see
+ the ccss.tex document that is on this review. This implements
+ CCBS and CCNR in several flavors. First up is a "generic"
+ implementation, which can work over any channel technology
+ provided that the channel technology can accurately report device
+ state. Call completion is requested using the dialplan
+ application CallCompletionRequest and can be canceled using
+ CallCompletionCancel. Device state subscriptions are used in
+ order to monitor the state of called parties. Next, there is a
+ SIP-specific implementation of call completion. This method uses
+ the methods outlined in draft-ietf-bliss-call-completion-06 to
+ implement call completion using SIP signaling. There are a few
+ things to note here: * The agent/monitor terminology used
+ throughout Asterisk sometimes is the reverse of what is defined
+ in the referenced draft. * Implementation of the draft required
+ support for SIP PUBLISH. I attempted to write this in a
+ generic-enough fashion such that if someone were to want to write
+ PUBLISH support for other event packages, such as dialog-state or
+ presence, most of the effort would be in writing callbacks
+ specific to the event package. * A subportion of supporting
+ PUBLISH reception was that we had to implement a PIDF parser. The
+ PIDF support added is a bit minimal. I first wrote a validation
+ routine to ensure that the PIDF document is formatted properly.
+ The rest of the PIDF reading is done in-line in the
+ call-completion-specific PUBLISH-handling code. In other words,
+ while there is PIDF support here, it is not in any state where it
+ could easily be applied to other event packages as is. Finally,
+ there are a variety of ISDN-related call completion protocols
+ supported. These were written by Richard Mudgett, and as such I
+ can't really say much about their implementation. There are notes
+ in the CHANGES file that indicate the ISDN protocols over which
+ call completion is supported. Review:
+ https://reviewboard.asterisk.org/r/523
+
+ * main/srv.c, channels/chan_sip.c, funcs/func_srv.c (added),
+ CHANGES, include/asterisk/srv.h: func_srv and explicit
+ specification of a remote IP for SIP. From Review Board: There
+ are two interrelated changes here. First, there is the
+ introduction of func_srv. This adds two new read-only dialplan
+ functions, SRVQUERY and SRVRESULT. They work very similarly to
+ the ENUMQUERY and ENUMRESULT functions, except that this allows
+ one to query SRV records instead. In order to facilitate this
+ work, I added a couple of new API calls to srv.h.
+ ast_srv_get_record_count tells the number of records returned by
+ an SRV lookup. This number is calculated at the time of the SRV
+ lookup. ast_srv_get_nth_record allows one to get a numbered SRV
+ record. Second, there is the modification to chan_sip that allows
+ one to specify a hostname or IP address (along with a port) to
+ send an outgoing INVITE to when dialing a SIP peer. This goes
+ hand-in-hand with func_srv. You can query SRV records and then
+ use the host and port from the results to dial via a specific
+ host instead of what is configured in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/608 SWP-1200
+
+2010-04-08 16:35 +0000 [r256428] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, Makefile.rules, build_tools/make_linker_version_script: Ensure
+ that linker version scripts (used for symbol export control)
+ always exist. Using wildcard matching in the Makefile is not
+ adequate to determine whether an export file should exist for a
+ module or not, so instead we'll just create one if the module
+ needs one, or copy the default one if it does not.
+
+2010-04-06 19:28 +0000 [r256370] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Mac OS X does not support comparing a
+ mutex to its initializer. Create a test for this.
+
+2010-04-06 14:42 +0000 [r256319] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes deadlock in chan_sip caused by usage
+ of MASTER_CHANNEL dialplan function (closes issue #16767)
+ Reported by: lmsteffan Patches: deadlock_16767v3.diff uploaded by
+ dvossel (license 671) Review:
+ https://reviewboard.asterisk.org/r/606/
+
+2010-04-06 00:39 +0000 [r256265] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: 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 ........
+
+2010-04-05 15:14 +0000 [r256161] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/localchannel.tex: Fix for localchannel.tex to allow PDFs
+ to be generated again.
+
+2010-04-03 02:12 +0000 [r256103-256104] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
+ include/asterisk/channel.h, main/cel.c, channels/sig_pri.c,
+ channels/chan_iax2.c, apps/app_queue.c, channels/chan_oss.c,
+ funcs/func_redirecting.c, main/channel.c, main/dial.c,
+ channels/chan_dahdi.c, channels/chan_misdn.c,
+ apps/app_dumpchan.c, res/res_agi.c, channels/chan_h323.c,
+ res/snmp/agent.c, apps/app_amd.c, funcs/func_callerid.c:
+ Consolidate ast_channel.cid.cid_rdnis into
+ ast_channel.redirecting.from.number. SWP-1229 ABE-2161 * Ensure
+ chan_local.c:local_call() will not leak cid.cid_dnid when
+ copying.
+
+ * apps/app_dial.c: Using the Dial application f option when the
+ call is forwarded will likely crash. Fix app_dial.c:do_forward()
+ OPT_FORCECLID setting cid.cid_num with a stack allocated string
+ instead of a heap allocated string.
+
+2010-04-02 23:55 +0000 [r256010-256019] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: 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 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 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 20:19 +0000 [r255952] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: 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
+
+2010-04-02 18:57 +0000 [r255906] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_ael_share.exports.in (added), codecs,
+ res/res_pktccops.exports.in (added), utils,
+ res/res_monitor.exports.in (added), Makefile.moddir_rules,
+ res/res_smdi.exports.in (added), Makefile.rules, cdr,
+ res/res_agi.exports.in (added), formats, main/asterisk.exports
+ (removed), res/res_odbc.exports (removed),
+ res/res_calendar.exports (removed), apps/app_voicemail.exports
+ (removed), bridges, res/res_odbc.exports.in (added),
+ main/asterisk.exports.in (added), apps/app_voicemail.exports.in
+ (added), res/res_calendar.exports.in (added),
+ res/res_features.exports (removed), res/res_fax.exports.in
+ (added), pbx, res/res_adsi.exports.in (added),
+ res/res_jabber.exports (removed), res/res_pktccops.exports
+ (removed), channels, res/res_jabber.exports.in (added),
+ main/Makefile, res/res_smdi.exports (removed), tests, apps, cel,
+ res/res_agi.exports (removed), addons, res/res_speech.exports
+ (removed), Makefile, funcs, res/res_speech.exports.in (added),
+ res/res_fax.exports (removed), main, res/res_adsi.exports
+ (removed), res/res_features.exports.in (added),
+ res/res_ael_share.exports (removed),
+ build_tools/make_linker_version_script (added), res,
+ res/res_monitor.exports (removed): Allow symbol export filtering
+ to work properly on platforms that have symbol prefixes. Some
+ platforms prefix externally-visible symbols in object files
+ generated from C sources (most commonly, '_' is the prefix). On
+ these platforms, the existing symbol export filtering process
+ ends up suppressing all the symbols that are supposed to be left
+ visible. This patch allows the prefix string to be supplied to
+ the top-level Makefile in the LINKER_SYMBOL_PREFIX variable, and
+ then generates the linker scripts as required to include the
+ prefix supplied.
+
+2010-04-02 06:45 +0000 [r255850-255851] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Ignore Redial softkey when no previous
+ dialed number is known (closes issue #17126) Reported by: wedhorn
+ Patches: skinny79xx_redial1.diff uploaded by wedhorn (license 30)
+
+ * channels/chan_skinny.c: Cleanup transmit_* functions Bulk lot of
+ generally trivial changes for cleaning up the transmit stuff.
+ Line state request has been modified for line only responses.
+ (closes issue #16994) Reported by: wedhorn Patches:
+ skinny-clean07.diff uploaded by wedhorn (license 30) Tested by:
+ wedhorn
+
+2010-04-01 18:16 +0000 [r255796] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/lock.h: Fix DEBUG_THREADS build on Darwin.
+ (closes issue #16828) Reported by: oej Patches:
+ 20100331__issue16828.diff.txt uploaded by tilghman (license 14)
+
+2010-04-01 16:09 +0000 [r255751] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/sip.conf.sample: Removed documentation of the non
+ existent 'both' option to 'faxdetect' in sip.conf
+
+2010-03-31 22:35 +0000 [r255701] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix improper comaparison of anonymous URI
+ when getting P-Asserted-Identity. There was a bug where we split
+ the URI on the @ sign and then attempted to compare to
+ "anonymous@anonymous.invalid" afterwards. This comparison could
+ never evaluate true. So now we keep a copy of the URI prior to
+ the split so that the comparison is valid.
+
+2010-03-31 19:13 +0000 [r255592] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: 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 17:48 +0000 [r255504] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_dial.c, /, configs/sip.conf.sample: 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 20:56 +0000 [r255323-255410] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_h323.c: 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 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. ........
+
+2010-03-29 14:07 +0000 [r255281] Jared Smith <jaredsmith@jaredsmith.net>
+
+ * apps/app_confbridge.c, CHANGES: This patch adds custom device
+ state handling for ConfBridge conferences, matching the devstate
+ handling of the MeetMe conferences. Review:
+ https://reviewboard.asterisk.org/r/572/ Closes issue #16972
+
+2010-03-29 05:10 +0000 [r255240] Russell Bryant <russell@digium.com>
+
+ * main/event.c: Remove a debugging log entry.
+
+2010-03-27 23:51 +0000 [r255199] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
+ addons/chan_ooh323.c, addons/ooh323c/src/ooh323.h,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c:
+ corrections in gk interface, small fixes in call clearing.
+
+2010-03-27 14:44 +0000 [r255158] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c: We need to inclde sys/wait.h on OpenBSD to
+ get WEXITSTATUS.
+
+2010-03-27 06:09 +0000 [r255117] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c: inotify support for pbx_spool This should give a
+ good speed boost, in that one particular thread isn't waking up
+ once a second to read directory contents. Reviewboard:
+ https://reviewboard.asterisk.org/r/137/
+
+2010-03-26 19:27 +0000 [r255021-255066] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: 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: 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
+
+2010-03-26 16:27 +0000 [r254976] Sean Bright <sean@malleable.com>
+
+ * contrib/scripts/live_ast: Work around a bug in dash on Ubuntu by
+ checking the number of arguments before shift'ing. Reported and
+ tested by pabelanger.
+
+2010-03-25 23:38 +0000 [r254931] Kevin P. Fleming <kpfleming@digium.com>
+
+ * addons/chan_ooh323.h, addons/ooh323c/src/ooasn1.h,
+ addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
+ addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/dlist.c, addons/ooh323c/src/eventHandler.c,
+ addons/ooh323c/src/ooCapability.c, addons/ooh323cDriver.c,
+ addons/mp3/interface.c, addons/ooh323cDriver.h,
+ addons/ooh323c/src/rtctype.c, addons/ooh323c/src/ooCalls.c,
+ addons/ooh323c/src/encode.c, addons/ooh323c/src/ooUtils.c,
+ addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooh323ep.c,
+ addons/ooh323c/src/ooports.c, addons/mp3/decode_ntom.c,
+ addons/ooh323c/src/memheap.c, addons/ooh323c/src/ooh323.c,
+ addons/ooh323c/src/ooh245.c, addons/mp3/common.c,
+ addons/ooh323c/src/decode.c, addons/ooh323c/src/context.c,
+ addons/ooh323c/src/perutil.c, addons/mp3/layer3.c,
+ addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooCmdChannel.c,
+ addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ootrace.c: Use "local" instead of "system"
+ header file inclusion. Now that these files are in the tree, they
+ should prefer the tree's local copy of all Asterisk headers over
+ any that may be installed.
+
+2010-03-25 21:39 +0000 [r254884] Russell Bryant <russell@digium.com>
+
+ * addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooSocket.h: Fix
+ a number of other build problems on Mac OS X.
+
+2010-03-25 20:41 +0000 [r254802] Jason Parker <jparker@digium.com>
+
+ * utils/Makefile, /: 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. ........
+
+2010-03-25 20:41 +0000 [r254718-254801] Russell Bryant <russell@digium.com>
+
+ * addons/chan_ooh323.h: Resolve compiler warning on FreeBSD.
+
+ * addons/ooh323c/src/ooh323.c, addons/Makefile,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ootrace.c: Fix
+ chan_ooh323 so it works on Mac OS X, as well.
+
+ * channels/chan_usbradio.c: chan_usbradio depends on alsa.
+
+2010-03-25 18:38 +0000 [r254636-254638] Kevin P. Fleming <kpfleming@digium.com>
+
+ * .cleancount: Bump cleancount due to ast_channel change.
+
+ * include/asterisk/channel.h: Remove no-longer-used (and unsafe)
+ field in ast_channel for linked lists. The ast_channel structure
+ had a field used for linking a channel into a linked list, but
+ now that ast_channel structures are ao2 objects, this is no
+ longer needed, and could be harmful as ao2 objects really
+ shouldn't ever be placed into linked lists (since those lists
+ don't assist with reference count management on the objects).
+
+ * addons/Makefile: Get chan_ooh323 building again after recent
+ build system changes.
+
+2010-03-25 17:52 +0000 [r254454-254557] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_acl.c (added): Add unit test for testing ACL
+ functionality. There are two unit tests contained here. 1.
+ "Invalid ACL" This attempts to read a bunch of badly formatted
+ ACL entries and add them to a host access rule. The goal of this
+ test is to be sure that all invalid entries are rejected as they
+ should be. 2. "ACL" This sets up four ACLs. One is a permit all,
+ one is a deny all, and the other two have specific rules about
+ which subnets are allowed and which are not. Then a set of test
+ addresses is used to determine whether we would allow those
+ addresses to access us when each ACL is applied. This test, by
+ the way, was what resulted in AST-2010-003's creation. Review:
+ https://reviewboard.asterisk.org/r/532
+
+ * include/asterisk/acl.h, /: 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 ........
+
+ * channels/sip/dialplan_functions.c: Add new rtpsource options to
+ the CHANNEL function. This adds rtpsource options analogous to
+ the rtpdest functions that already exist. In addition, this fixes
+ potential crashes which could result due to trying to read values
+ from nonexistent RTP streams.
+
+ * res/res_rtp_asterisk.c, /: 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 16:03 +0000 [r254453] Terry Wilson <twilson@digium.com>
+
+ * /, main/file.c: 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 15:27 +0000 [r254450] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, channels/chan_sip.c, res/res_fax.c,
+ configs/sip.conf.sample, include/asterisk/frame.h,
+ channels/sip/include/sip.h: 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/
+
+2010-03-25 15:21 +0000 [r254446] Leif Madsen <lmadsen@digium.com>
+
+ * res/res_agi.c: 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
+
+2010-03-25 10:09 +0000 [r254406] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: remove unneeded explicit channel in dahdi
+ ioctls This patch removes some cases where the channel number for
+ an ioctl was passed as a member in a struct rather then through
+ the file descriptor. The gain setting functions passed around a
+ channel which is always 0, and thus this parameter is simply
+ dropped. Review: https://reviewboard.asterisk.org/r/584/
+
+2010-03-24 21:10 +0000 [r254362] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Fix potential invalid reads that could occur in pbx.c
+ Here is a cut and paste of my review request for this change:
+ This past weekend, Russell ran our current suite of unit tests
+ for Asterisk under valgrind. The PBX pattern match test caused
+ valgrind to spew forth two invalid read errors. This patch
+ contains two changes that shut valgrind up and do not cause any
+ new memory leaks. Change 1: In
+ ast_context_remove_extension_callerid2, valgrind reported an
+ invalid read in the for loop close to the function's end.
+ Specifically, one of the the strcmp calls in the loop control was
+ reading invalid memory. This was because the caller of
+ ast_context_remove_extension_callerid2 (__ast_context destroy in
+ this case) passed as a parameter a shallow copy of an ast_exten's
+ exten field. This same ast_exten was what was destroyed inside
+ the for loop, thus any iterations of the for loop beyond the
+ destruction of the ast_exten would result in invalid reads. My
+ fix for this is to make a copy of the ast_exten's exten field and
+ pass the copy to ast_context_remove_extension_callerid2. In
+ addition, I have also acted similarly with the ast_exten's
+ matchcid field. Since in this case a NULL is handled quite
+ differently than an empty string, I needed to be a bit more
+ careful with its handling. Change 2: In __ast_context_destroy, we
+ iterated over a hashtab and called
+ ast_context_remove_extension_callerid2 on each item.
+ Specifically, the hashtab over which we were iterating was an
+ ast_exten's peer_table. Inside of
+ ast_context_remove_extension_callerid2, we could possibly destroy
+ this ast_exten, which also caused the hashtab to be freed.
+ Attempting to call ast_hashtab_end_traversal on the hashtab
+ iterator caused an invalid read to occur when trying to read the
+ iterator->tab->do_locking field since iterator->tab had already
+ been freed. My handling of this problem is a bit less
+ straightforward. With each iteration over the hashtab's contents,
+ we set a variable called "end_traversal" based on the return of
+ ast_context_remove_extension_callerid2. If 0 is ever returned,
+ then we know that the extension was found and destroyed. Because
+ of this, we cannot call ast_hashtab_end_traversal because we will
+ be guaranteeing a read of invalid memory. In such a case, we
+ forego calling ast_hashtab_end_traversal and instead call
+ ast_free on the hashtab iterator. Review:
+ https://reviewboard.asterisk.org/r/585
+
+2010-03-24 18:13 +0000 [r254277-254321] Jeff Peeler <jpeeler@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Allow configuration of minsecs and nextaftercmd per mailbox.
+ Previously only configurable globally. A unit test has also been
+ written to provide protection against parse failures for
+ supported mailbox options. (closes issue #16864) Reported by:
+ kobaz Patches: voicemail2.patch uploaded by kobaz (license 834)
+ Review: https://reviewboard.asterisk.org/r/555/
+
+ * /, res/res_monitor.c: 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 22:48 +0000 [r254162] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/asterisk.c: make 'core show settings' should show all
+ settable directories (closes issue #17086) Reported by: tzafrir
+ Patches: asterisk_extra_settings_dirs.diff uploaded by tzafrir
+ (license 46)
+
+2010-03-23 22:35 +0000 [r254159] Russell Bryant <russell@digium.com>
+
+ * main/test.c: Put test output for a failure in a CDATA section in
+ the XML results.
+
+2010-03-23 21:17 +0000 [r254050] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: 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/
+
+2010-03-23 20:52 +0000 [r254045] Sean Bright <sean@malleable.com>
+
+ * apps/app_queue.c: Remove unused structure member in app_queue.
+ (closes issue #15494) Reported by: makoto
+
+2010-03-23 19:19 +0000 [r254001] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * tests/Makefile: Change the name of the category 'TEST' to match
+ the name of the subdir
+
+2010-03-23 16:52 +0000 [r253958] Terry Wilson <twilson@digium.com>
+
+ * main/http.c: Don't act like an http write failed when it didn't
+ fwrite returns the number of items written, not the number of
+ bytes
+
+2010-03-23 14:22 +0000 [r253917] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/Makefile, include/asterisk/logger.h, main/Makefile,
+ Makefile.moddir_rules, pbx/Makefile, res/Makefile, CHANGES,
+ channels/Makefile, include/asterisk/options.h, main/cli.c: Change
+ per-file debug and verbose levels to be per-module, the way users
+ expect them to work. 'core set debug' and 'core set verbose' can
+ optionally change the level for a specific filename; however,
+ this is actually for a specific source file name, not the module
+ that source file is included in. With examples like chan_sip,
+ chan_iax2, chan_misdn and others consisting of multiple source
+ files, this will not lead to the behavior that users expect. If
+ they want to set the debug level for chan_sip, they want it set
+ for all of chan_sip, and not to have to also set it for
+ reqresp_parser and other files that comprise the chan_sip module.
+ This patch changes this functionality to be module-name based
+ instead of file-name based. To make this work, some Makefile
+ modifications were required to ensure that the AST_MODULE
+ definition is present in each object file produced for each
+ module as well. Review: https://reviewboard.asterisk.org/r/574/
+
+2010-03-22 20:32 +0000 [r253872] Mark Michelson <mmichelson@digium.com>
+
+ * main/asterisk.c: Initialize channels prior to loading "preload"
+ modules. We can have bad results when a module, upon being
+ loaded, attempts to reference the channels container if the
+ container hasn't yet been initialized. I saw this happen by
+ trying to preload pbx_config.so and having a hint defined which
+ referenced a non-existent SIP peer.
+
+2010-03-22 19:52 +0000 [r253800] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: 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 19:05 +0000 [r253712-253758] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/dbsep.cgi: Update query should be an UPDATE, not
+ a SELECT.
+
+ * contrib/scripts/dbsep.cgi: Return the list for later
+ manipulation. This fixes an issue with the update procedure.
+ Debugging with mmichelson.
+
+ * contrib/scripts/dbsep.cgi, configs/dbsep.conf.sample: Accomodate
+ equal signs in DSNs and add documentation, based upon
+ mmichelson's feedback.
+
+2010-03-20 16:50 +0000 [r253536-253579] Russell Bryant <russell@digium.com>
+
+ * funcs/func_strings.c: Fix memory corruption found by unit tests.
+ ast_str_reset() was being called on a potentially uninitialized
+ pointer. Valgrind is my hero, once again.
+
+ * cel/cel_pgsql.c, main/tcptls.c, main/manager.c, main/features.c,
+ main/test.c, cdr/cdr_pgsql.c, main/stdtime/localtime.c,
+ main/cel.c: Resolve more compiler warnings on FreeBSD.
+
+ * apps/app_voicemail.c: Include sys/wait.h on FreeBSD to get the
+ WEXITSTATUS() macro.
+
+ * apps/app_dial.c, apps/app_followme.c: Resolve compiler warnings
+ on FreeBSD.
+
+ * pbx/pbx_dundi.c: Resolve a compiler warning on FreeBSD.
+
+ * channels/chan_dahdi.c: Use SHRT_MAX instead of MAXSHORT. These
+ changes fix build issues I had with this module on FreeBSD.
+
+2010-03-19 07:37 +0000 [r253490] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/astobj2.c: 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 18:23 +0000 [r253357-253378] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Update comment to reflect new timeout value.
+
+ * main/asterisk.c: 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/
+
+2010-03-18 17:52 +0000 [r253345] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_userevent.c: 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
+
+2010-03-18 15:59 +0000 [r253261] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Prevent a crash when a buddy gets offline.
+ (closes issue #16760) Reported by: fiddur Patches: 248394.diff
+ uploaded by fiddur (license 678)i with modifications by me Tested
+ by: fiddur, phsultan
+
+2010-03-18 15:46 +0000 [r253256] Leif Madsen <lmadsen@digium.com>
+
+ * /, doc/tex/localchannel.tex: 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)
+
+2010-03-18 15:45 +0000 [r253255] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c: Just in case of a race, send the signal
+ on interrupt.
+
+2010-03-17 19:06 +0000 [r253205] Leif Madsen <lmadsen@digium.com>
+
+ * main/test.c: main/test.c reports erroneous CLI message. (closes
+ issue #17051) Reported by: Nick_Lewis
+
+2010-03-17 14:16 +0000 [r253113] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_gosub.c: Switch to using intptr_t, as suggested by
+ Kevin Fleming on the -dev list
+
+2010-03-17 00:40 +0000 [r253028-253032] Leif Madsen <lmadsen@digium.com>
+
+ * main/xmldoc.c: Fix a typo.
+
+ * configs/say.conf.sample: 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) ........
+
+2010-03-17 00:23 +0000 [r252976-253004] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_gosub.c: Argh.
+
+ * configure, include/asterisk/autoconfig.h.in, tests/test_gosub.c,
+ configure.ac: Fix bamboo compile error by calculating an integer
+ with the same size as a pointer.
+
+ * tests/test_gosub.c (added), apps/app_stack.c: 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 19:36 +0000 [r252849] Russell Bryant <russell@digium.com>
+
+ * tests/test_time.c: Re-enable test_time on non-Linux.
+
+2010-03-16 19:36 +0000 [r252848] Sean Bright <sean@malleable.com>
+
+ * res/res_clialiases.c: 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 19:34 +0000 [r252846] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_time.c, include/asterisk/localtime.h,
+ main/stdtime/localtime.c: Fix test_time on Mac OS X (and other
+ platforms without inotify) Reviewboard:
+ https://reviewboard.asterisk.org/r/554/
+
+2010-03-16 19:01 +0000 [r252767] Russell Bryant <russell@digium.com>
+
+ * utils/Makefile, /: 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 18:48 +0000 [r252762] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.ael.sample: 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) ........
+
+2010-03-16 18:40 +0000 [r252760] Tilghman Lesher <tlesher@digium.com>
+
+ * codecs/gsm/Makefile: OSARCH is not inherited to this directory
+
+2010-03-16 18:36 +0000 [r252759] Russell Bryant <russell@digium.com>
+
+ * tests/test_time.c: Disable this test on non-Linux for now.
+
+2010-03-15 22:48 +0000 [r252709] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_fax.c: Improve handling of values supplied to
+ FAXOPT(ecm). Previously, values that began with whitespace were
+ silently treated as 'no', and all non-'yes' values were also
+ treated as 'no'. Now the supplied value is specifically checked
+ for a 'yes' or 'no' (or equivalent) value, after skipping leading
+ whitespace. If the value is not valid, then a warning message is
+ generated.
+
+2010-03-15 22:14 +0000 [r252627] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Tell the RTP engine API about the initial
+ read and write format. Peer reviewed out-of-band by file.
+
+2010-03-15 21:55 +0000 [r252623] Sean Bright <sean@malleable.com>
+
+ * apps/app_meetme.c: Resolve a crash in SLATrunk when the specified
+ trunk doesn't exist. Reported by philipp64 in #asterisk-dev.
+
+2010-03-15 21:51 +0000 [r252619] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/org.asterisk.asterisk.plist, /: 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 20:52 +0000 [r252534] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/extensions.ael.sample: 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 16:27 +0000 [r252362-252488] Tilghman Lesher <tlesher@digium.com>
+
+ * codecs/gsm/Makefile: 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
+
+ * channels/chan_sip.c: 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)
+
+ * /: Recorded merge of revisions 252366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252366 | tilghman | 2010-03-14 20:39:00 -0500 (Sun, 14 Mar 2010)
+ | 2 lines Typo ........
+
+ * Makefile, contrib/init.d/org.asterisk.asterisk.plist (added), /,
+ main/asterisk.c: 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 17:43 +0000 [r252314] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, cel/cel_sqlite3_custom.c: 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
+
+2010-03-14 14:42 +0000 [r252277] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h,
+ configs/chan_ooh323.conf.sample, addons/ooh323c/src/ooh245.h,
+ addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ootypes.h,
+ addons/ooh323c/src/ooq931.c: generate roundtrip delay requests
+ and responses added response to roundtrip delay requests from
+ opposite side added roundtrip delay request sending to opposite
+ side after answer, added options for sending request (interval
+ between request and count of unreplied requests before forced
+ call hangup) (closes issue #16976) Reported by: vmikhelson
+ Patches: rtdr-1.6.0-2.patch uploaded by may213 (license 454)
+ Tested by: vmikhelson, may213
+
+2010-03-13 22:21 +0000 [r252229-252241] Russell Bryant <russell@digium.com>
+
+ * main/app.c: Resolve unit test failure that occurred on Mac OSX.
+ On Linux (glibc), regcomp() does not return an error for an empty
+ string. However, the version on OSX will return an error. The
+ test for channel group matching by regex now passes on the mac,
+ as well.
+
+ * tests/test_time.c: Resolve compiler warning by paying attention
+ to system() return value. This resolves the last compile failure
+ on bamboo.
+
+2010-03-12 23:18 +0000 [r252133] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_time.c (added): Test script to verify that timezone
+ cache is properly removed on zonefile alteration.
+
+2010-03-12 22:04 +0000 [r252089] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, res/res_rtp_asterisk.c, addons/chan_ooh323.c,
+ main/rtp_engine.c, channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_h323.c, configs/sip.conf.sample,
+ include/asterisk/frame.h, include/asterisk/rtp_engine.h,
+ channels/sip/include/sip.h, channels/chan_mgcp.c: 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 21:57 +0000 [r252088] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: add missing mfcr2_skip_category setting
+
+2010-03-12 19:43 +0000 [r251989] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: 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 19:40 +0000 [r251946-251987] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Merged revisions 251986 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 Mar 2010)
+ | 1 line Make chan_dahdi wakeup_sub() prototype not conditional.
+ ........
+
+ * channels/chan_dahdi.c: Doxegen this chan_dahdi lock.
+
+2010-03-11 21:07 +0000 [r251877-251884] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_exec.c: 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
+
+ * tests/test_stringfields.c: Fix tests on 32-bit systems.
+
+ * apps/app_system.c: 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 18:07 +0000 [r251821] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c: Minor tweaks and
+ comment updates to chan_dahdi.
+
+2010-03-11 07:03 +0000 [r251779] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_directory.c: Add supporting code for app-directory pause
+ option. Since 1.6.1 CLI help reports that option p(n) 'initial
+ pause' is available. Supporting code was never implemented.
+ (closes issue #16751) Reported by: alecdavis Patches:
+ directory_pause.trunk.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/481/
+
+2010-03-10 23:15 +0000 [r251736] Jeff Peeler <jpeeler@digium.com>
+
+ * tests/test_stringfields.c (added), main/utils.c: Add new unit
+ test for stringfields. (Copied from reviewboard) Tests the
+ following: 1. Basic allocation and setting of string fields. 2.
+ Shrinking a string field and re-expanding it. 3. Growing the last
+ allocation in a string field pool. 4. Setting a string to a large
+ value such that a new string field pool must be allocated. In
+ each part, we make sure that the string field is accurate (has
+ the correct value in it), make sure that the 2 bytes before the
+ string field has the correct capacity for the field, and for
+ tests 2-4, we make sure that the string field is where we expect
+ it to be in memory. Also tested: 5. Shrinking a string field and
+ partially re-expanding it. 6. Setting strings in such a way as to
+ create three separate string field pools and then removing the
+ middle pool. There is a bug fix in the init function, which
+ ensures the embedded_pool is set to NULL which is important for
+ stack allocated structures. Review:
+ https://reviewboard.asterisk.org/r/185/
+
+2010-03-10 20:54 +0000 [r251682] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Hmmm, apparently needed to be fixed in
+ trunk, too. (closes issue #16900) Reported by: bluecrow76
+ Patches: asterisk-1.6.2.4-func_strings.diff uploaded by
+ bluecrow76 (license 270)
+
+2010-03-10 20:53 +0000 [r251680] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_record.c: Be less ambiguous in Record() app docs. For
+ some reason the documentation for the 'k' application in trunk
+ and 1.6.2 is different than 1.6.0 and 1.6.1, so I'm setting them
+ all to match. The wording in 1.6.2 and trunk was ambiguous, so
+ you could interpret the wording the mean that recording would
+ continue upon hangup indefinitely, or you could interpret it to
+ mean that the recorded data would not be discarded upon hangup.
+ This change makes it clear we mean the latter, and not the
+ former. Came from a discussion in #asterisk on IRC.
+
+2010-03-10 20:51 +0000 [r251679] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Fix ParkAndAnnounce not respecting parking
+ options. The patch ensures that if a peer does not exist, parking
+ settings are read from the channel. A unit test has been written
+ to ensure proper operation for both standard parking and parking
+ using masquerades. (closes issue #16592) Reported by: mwyres
+ Patches: bug_16592.diff uploaded by snuffy (license 35) Review:
+ https://reviewboard.asterisk.org/r/539/
+
+2010-03-10 20:30 +0000 [r251677] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_substitution.c, funcs/func_strings.c: It's amazing
+ what writing a test will find. (issue #16900) Reported by:
+ bluecrow76
+
+2010-03-10 18:25 +0000 [r251631] Jeff Peeler <jpeeler@digium.com>
+
+ * main/abstract_jb.c: Fix jitterbuffer logging not creating
+ logfiles. Three changes made here: 1) Do not fail if a previous
+ log does not exist (in fact, this is probably expected). 2)
+ Ensure that the file descriptor to write to gets assigned
+ properly. I am at a loss as to why assigning safe_fd outside the
+ if fixes this, but it makes the if statement slightly less
+ complicated anyway. 3) Move up the failure message so that the
+ errno of the failure is not overwritten by fclose. (closes issue
+ #16917) Reported by: Artem
+
+2010-03-10 16:55 +0000 [r251538-251585] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h, channels/sig_pri.c: Simplified
+ dahdi_request() channel selection failed reason/cause code. Also
+ avoid potential crash because cause could be NULL.
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Reduce the amount of database access for
+ HAVE_PRI_SERVICE_MESSAGES. Rework HAVE_PRI_SERVICE_MESSAGES to
+ not use the active values directly from the database. Database
+ access is likely expensive. Database access now only happens on
+ initialization, destruction, and when the B channel is taken in
+ or out of service. This change is not related to call waiting but
+ it would cause the search for a call waiting interface to be very
+ expensive and slow down D channel message servicing.
+
+2010-03-09 20:30 +0000 [r251475] Tilghman Lesher <tlesher@digium.com>
+
+ * codecs/gsm/Makefile, Makefile.rules: Build system modifications
+ to ensure that Asterisk properly builds on Mac OS X 10.6. (closes
+ issue #16997) Reported by: jquinn Patches:
+ 20100309__issue16997__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: tilghman, russell
+
+2010-03-08 18:08 +0000 [r251310] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 251309 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r251309 | lmadsen | 2010-03-08 12:07:44 -0600 (Mon, 08 Mar 2010)
+ | 13 lines 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. (closes issue #16784) Reported by:
+ pabelanger Tested by: pabelanger, mnick, davidw, mutineer612
+ (closes issue #16887) Reported by: jlpedrosa Tested by:
+ jlpedrosa, mutineer612 ........
+
+2010-03-08 05:15 +0000 [r251262-251263] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/stdtime/localtime.c: Remove portions that weren't meant to
+ be committed for the OS X compat fix
+
+ * funcs/func_pitchshift.c, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
+ main/stdtime/localtime.c: Change needed to make Mac OS X 10.6
+ happy
+
+2010-03-07 14:53 +0000 [r251221-251222] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Clean transmit_* for start/stop media
+ transmission Small patch changing skinny_set_rtp_peer to use
+ transmit_stopmediatransmission and to use new
+ transmit_startmediatransmission. Basic testing on 30VIP's by
+ wedhorn Basic testing on 7960 by me (closes issue #16956)
+ Reported by: wedhorn Patches: skinny-clean05b.diff uploaded by
+ wedhorn (license 30) Tested by: wedhorn,mvanbaak
+
+ * channels/chan_skinny.c: Cleanup transmit_callstate handling Broke
+ the various functions included in transmit_callstate to their own
+ functions. Transmit_callstate now just transmits callstate.
+ Generally left the functionality as it was, which highlight some
+ minor code issues (eg multiple transmit_callstate's). I did
+ however revise the hint code usage of the old transmit_callstate
+ as it it not appropriate to put a device on hook based on the
+ change of a hinted device. (closes issue #16939) Reported by:
+ wedhorn Patches: skinny-clean04.diff uploaded by wedhorn (license
+ 30) Tested by: mvanbaak,wedhorn
+
+2010-03-07 00:45 +0000 [r251181] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooq931.c: small log issue from bug 0016664
+
+2010-03-06 14:16 +0000 [r251137] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix a crash in SIP blind transfer handling
+ found by an automated external test. The first real test added to
+ the external test suite found a pretty nasty crash that occurred
+ in Asterisk trunk. The crash was due to a race condition between
+ the REFER handling and channel destruction in the channel thread.
+ After the transfer has been completed, we go back to the
+ transferrer channel and try to lock it so we can fire off a CEL
+ event. However, there was no guarantee that the channel was still
+ around at that point since it's racing against the channel
+ thread. Since ast_channel is a reference counted object, the fix
+ is simple. The code unlocks the transferrer channel before
+ finally completing the transfer with an async goto. At this point
+ the channel thread is going to start call tear down and the
+ channel will eventually be destroyed. To ensure that the channel
+ is valid when we want to fire off the CEL event, increase the
+ channel's reference count.
+
+2010-03-05 21:51 +0000 [r251038-251087] David Vossel <dvossel@digium.com>
+
+ * funcs/func_pitchshift.c: fixes xml error in func_pitchshift
+
+ * funcs/func_pitchshift.c (added), CHANGES: PITCH_SHIFT dialplan
+ function The PITCH_SHIFT function can be used on a channel to
+ independently modify the pitch of both rx and tx audio streams.
+ Now you can improve your conference calls by assigning a random
+ pitch effect to everyone entering a meetme room, or just make
+ your day more interesting by making your co-workers sound funny.
+ These are just some of the numerious practical uses for this
+ function. Enjoy! https://reviewboard.asterisk.org/r/526/
+
+2010-03-05 19:32 +0000 [r251022] Russell Bryant <russell@digium.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
+ pbx/pbx_gtkconsole.c (removed): Remove pbx_gtkconsole and related
+ gtk1 checks. Review: https://reviewboard.asterisk.org/r/541/
+
+2010-03-05 19:10 +0000 [r250979] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_followme.c: Fix app_followme playing wrong sound files.
+ Fixes regression introduced in 140167 that uses the wrong
+ variable names. (closes issue #16930) Reported by: ianc Patches:
+ fix_reload_followme.diff uploaded by ianc (license 998)
+
+2010-03-05 05:03 +0000 [r250917] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix up some of chan_sip's usage of the RTP
+ engine API. The get_local_address() function for an RTP instance
+ was used when building an SDP, but the results were not honored.
+ The RTP engine activate() function was not being used once we
+ have determined that media will now flow.
+
+2010-03-05 04:37 +0000 [r250913] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Missing quote in ODBC query. (closes issue
+ #16953) Reported by: elguero Patches:
+ app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license
+ 37)
+
+2010-03-05 02:07 +0000 [r250871] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/rtp_engine.h: Fix up the ast_rtp_property enum.
+ The mis-placement of the latest entry meant that when it was set,
+ it was writing one index past the end of the properties array in
+ the ast_rtp_instance (which happened to be the local_address
+ field).
+
+2010-03-05 01:05 +0000 [r250787] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 250786 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r250786 | jpeeler | 2010-03-04 19:02:58 -0600 (Thu, 04
+ Mar 2010) | 9 lines Fix not being able to specify a URL in MOH
+ class directory. Don't attempt to chdir on a URL! (closes issue
+ #16875) Reported by: raarts Patches: moh-http.patch uploaded by
+ raarts (license 937) ........
+
+2010-03-04 20:12 +0000 [r250730] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_channel.c: Adjust XML for func_channel to indicate
+ that rtpdest can take a "text" argument.
+
+2010-03-03 21:28 +0000 [r250609-250614] Leif Madsen <lmadsen@digium.com>
+
+ * /: Recorded merge of revisions 250613 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250613 | lmadsen | 2010-03-03 16:28:02 -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. (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 ........
+
+ * doc/tex/localchannel.tex: 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 19:38 +0000 [r250565] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, channels/chan_dahdi.c, main/dial.c,
+ channels/chan_local.c, include/asterisk/channel.h,
+ apps/app_queue.c: Removed cdrflags from ast_channel structure.
+ Only chan_dahdi set a value in cdrflags. Everyone else just
+ copied it around the system. Noone cared about any value it may
+ have contained.
+
+2010-03-03 19:06 +0000 [r250481] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: 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 19:02 +0000 [r250395-250478] David Vossel <dvossel@digium.com>
+
+ * main/test.c: Changes 0ms to <1ms in cli END results during 'test
+ execute'
+
+ * /, channels/chan_iax2.c: 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 ........
+
+2010-03-03 17:37 +0000 [r250392] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
+ Add new config option to control AMI alarm event reporting in
+ chan_dahdi. New config parameter "reportalarms" added in
+ chan_dahdi.conf which supports the following possible values:
+ "channels": report each channel alarms (current behavior, default
+ for backward compatibility) "spans": report an "SpanAlarm" event
+ when the span of any configured channel is alarmed "all": report
+ channel and span alarms (aggregated behavior) "none": do not
+ report any alarms (closes issue #16709) Reported by: nahuelgreco
+ Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco
+ (license 162)
+
+2010-03-03 16:43 +0000 [r250303-250346] Tilghman Lesher <tlesher@digium.com>
+
+ * main/editline/configure: One more fix to editline
+
+ * main/editline/configure, main/editline/Makefile.in,
+ main/editline/sys.h, main/editline/configure.in: Eliminate
+ remaining libedit warnings (shown in bamboo)
+
+2010-03-03 15:39 +0000 [r250302] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, apps/app_fax.c, CHANGES, res/res_fax_spandsp.c:
+ Updated CHANGES file to mention res_fax and res_fax_spandsp. Also
+ fixed MODULEINFO depends and conflicts for app_fax, res_fax, and
+ res_fax_spandsp.
+
+2010-03-03 00:18 +0000 [r250235-250246] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes signed to unsigned int comparision
+ issue for FaxMaxDatagram value.
+
+ * main/test.c: fixes assumption that test failed if it did not pass
+ when generating results
+
+ * tests/test_utils.c: base64 unit test
+
+2010-03-02 23:22 +0000 [r250190-250213] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/res_fax.conf.sample (added), include/asterisk/res_fax.h
+ (added): Merge missed files from res_fax/res_fax_spandsp merge.
+
+ * res/res_fax.c (added), res/res_fax.exports (added),
+ include/asterisk/frame.h, res/res_fax_spandsp.c (added): Merge
+ res_fax and res_fax_spandsp.
+
+2010-03-02 21:58 +0000 [r250141] David Vossel <dvossel@digium.com>
+
+ * apps/app_directed_pickup.c, CHANGES: adds 'p' option to
+ PickupChan The 'p' option allows the PickupChan app to pickup a
+ ringing phone by looking for the first match to a partial channel
+ name rather than requiring a full match. (closes issue #16613)
+ Reported by: syspert Patches: pickipbycallid.patch uploaded by
+ syspert (license 938) pickupbycallerid_v2.patch uploaded by
+ dvossel (license 671) Tested by: dvossel, syspert
+
+2010-03-02 21:09 +0000 [r249950-250051] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/imapstorage.tex: 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 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 ........
+
+ * /: Recorded merge of revisions 250041 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250041 | lmadsen | 2010-03-02 15:45:37 -0500 (Tue, 02 Mar 2010)
+ | 4 lines Update documentation to not imply we support overriding
+ options. (issue #16855) Reported by: davidw ........
+
+ * doc/tex/configuration.tex: Update documentation to not imply we
+ support overriding options. (closes issue #16855) Reported by:
+ davidw
+
+ * apps/app_directory.c: Fix literal values wrapped in
+ documentation. (closes issue #16145) Reported by: tilghman
+
+2010-03-02 19:39 +0000 [r249947] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_echo.c: revert ability to exit echo app caused a
+ regression, as only supported VOICE, not VIDEO etc. (issue
+ #16880)
+
+2010-03-02 19:24 +0000 [r249912-249925] Leif Madsen <lmadsen@digium.com>
+
+ * main/features.c: 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: 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 19:08 +0000 [r249893] David Vossel <dvossel@digium.com>
+
+ * channels/chan_unistim.c, configs/chan_dahdi.conf.sample,
+ configs/console.conf.sample, channels/chan_local.c,
+ channels/chan_sip.c, configs/oss.conf.sample,
+ configs/usbradio.conf.sample, configs/misdn.conf.sample,
+ 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: 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 19:02 +0000 [r249892] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_osplookup.c, apps/app_confbridge.c, res/res_jabber.c:
+ Fix several XML documentation validate errors.
+
+2010-03-02 18:31 +0000 [r249889-249891] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c: fix build by checking result of symlink in
+ test_voicemail_vmsayname
+
+ * CHANGES, apps/app_voicemail.c: Add new application VMSayName for
+ use with voicemail. VMSayName that will play the recorded name of
+ the voicemail user if it exists, otherwise will play the mailbox
+ number. A unit test has been written to verify correct
+ functionality called test_voicemail_vmsayname. (closes issue
+ #14973) Reported by: ghjm Review:
+ https://reviewboard.asterisk.org/r/530/
+
+2010-03-02 07:38 +0000 [r249759-249801] 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.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis
+
+ * channels/chan_dahdi.c: fix asterisk setting of pritimers from
+ chan_dahdi.conf regression since sig_pri split. (issue #16909)
+ Reported by: alecdavis Patches: pritimer.asterisk.diff.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis
+
+2010-03-01 19:36 +0000 [r249672] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_voicemail.c: 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 19:33 +0000 [r249669-249670] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Cleanup display_*message functions. This
+ patch splits transmit_displaymessage into
+ transmit_clear_display_message and transmit_display_message which
+ better aligns with the skinny protocol. The new
+ transmit_display_message is not used in the current code, but
+ will be and so it is commented. Moved handle_datetime from this
+ function to onhook and offhook functions (display now properly
+ cleared at the end of a call on 30VIP). Removed skinny debug
+ messages from inline code as there's an ast_verb in
+ transmit_clear_display_message. Also, removed commentary that it
+ was a clear display as it is now apparent from the function name.
+ Split transmit_displaypromptmessage into display and clear.
+ (closes issue #16878) Reported by: wedhorn Patches:
+ skinny-clean02.diff uploaded by wedhorn (license 30)
+ skinny-clean03.diff uploaded by wedhorn (license 30)
+
+ * channels/chan_skinny.c: fix endianes issues in chan_skinny
+ (closes issue #16826) Reported by: PipoCanaja Patches:
+ chan_skinny.c_bigendianPatch_20100218.diff uploaded by PipoCanaja
+ (license 994) Tested by: wedhorn
+
+2010-03-01 18:36 +0000 [r249623] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Constify a bit of app_voicemail, to make
+ ODBC and IMAP compile once again.
+
+2010-03-01 17:11 +0000 [r249538] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: 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 20:50 +0000 [r249491] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Fix unit test that Alec Davis broke.
+ (closes issue #16927) Reported by: alecdavis
+
+2010-02-28 16:36 +0000 [r249449] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_voicemail.c: make unit test check for NULL folder, which
+ then defaults to INBOX previous test, gave false level of
+ assurance that code was healthy. (issue #16927) Reported by:
+ alecdavis Patches: based on app_voicemail_test.diff.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis
+
+2010-02-28 07:10 +0000 [r249405] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, apps/app_voicemail.c: Properly document
+ voicemail API documents. Also fix a crash reported via the -dev
+ list.
+
+2010-02-27 22:49 +0000 [r249320] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sig_pri.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 14:08 +0000 [r249235] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_iax2.c: 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 18:41 +0000 [r249187] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: 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. - Unit tests added to verify behavior in the
+ future. (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 18:41 +0000 [r249186] David Vossel <dvossel@digium.com>
+
+ * main/test.c: adds Time field to "test show results" cli command
+
+2010-02-26 17:13 +0000 [r249101-249105] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Send a manager event when the manager
+ BridgeAction command is used. (closes issue #16769) Reported by:
+ syspert Patches: bridgeaction.patch uploaded by syspert (license
+ 938)
+
+ * /, channels/chan_sip.c: 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) ........
+
+2010-02-26 08:45 +0000 [r249009-249058] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c, cdr/cdr_sqlite.c,
+ 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_tds.c, cdr/cdr_csv.c: formatting tweaks and
+ constification
+
+ * main/cdr.c: Trim trailing whitespace (to help reduce diff against
+ cdr-q branch)
+
+ * include/asterisk/cdr.h: Trim trailing whitespace, convert lists
+ of defines to enums
+
+ * cdr/cdr_sqlite.c: trivial formatting tweak (working on reducing
+ diff against trunk for cdr-q)
+
+ * cdr/cdr_sqlite3_custom.c: remove include
+
+ * cdr/cdr_csv.c: constification, remove include
+
+ * cdr/cdr_tds.c: Remove unnecessary includes, formatting tweak
+
+ * cdr/cdr_pgsql.c: constification and remove unnecessary include
+
+2010-02-25 23:09 +0000 [r248952] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_monitor.c: 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 22:41 +0000 [r248946] Mark Michelson <mmichelson@digium.com>
+
+ * main/acl.c: Fix incorrect ACL behavior when CIDR notation of "/0"
+ is used. AST-2010-003
+
+2010-02-25 21:22 +0000 [r248861] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/asterisk.c: 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 18:37 +0000 [r248793] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_monitor.c: 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 22:44 +0000 [r248584-248667] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/Makefile: Also kill the .i files, or else the build
+ process will not recreate them, when we change flags. Fixes a
+ weird symbol problem mmichelson was having in a group branch, but
+ also applies to trunk.
+
+ * /, main/logger.c, include/asterisk/term.h, main/term.c: 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
+ ........
+
+2010-02-24 06:39 +0000 [r248533-248534] Russell Bryant <russell@digium.com>
+
+ * funcs/func_strings.c: Remove unnecessary warning message, make a
+ couple of formatting tweaks
+
+ * tests/test_strings.c: Add ASTERISK_FILE_VERSION macro.
+
+2010-02-23 22:29 +0000 [r248489] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_strings.c (added): Unit test for ast_str API. Review:
+ https://reviewboard.asterisk.org/r/517
+
+2010-02-23 16:34 +0000 [r248397] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: 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
+ ........
+
+2010-02-22 20:19 +0000 [r248347] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Move the REF_DEBUG comment higher in the
+ include list. Uncommenting the REF_DEBUG definition where it was
+ in the source resulted in only a small part of the astobj2
+ references being logged to a file. Moving this up higher in the
+ include list causes all references to be logged as they should
+ be.
+
+2010-02-22 06:45 +0000 [r248225-248226] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/taskprocessor.h, main/taskprocessor.c: Minor
+ tweaks to comment blocks and includes. Fix the copyright lines,
+ tweak doxygen formatting, and remove some unnecessary includes.
+
+ * tests/test_devicestate.c: Tweak copyright and author lines.
+
+2010-02-21 12:09 +0000 [r248184] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Cleanup transmit_* functions, part 1
+ Break transmit_tone into transmit_start_tone and
+ transmit_stop_tone as per the skinny protocol. (closes issue
+ #16874) Reported by: wedhorn Patches: skinny-clean01.diff
+ uploaded by wedhorn (license 30)
+
+2010-02-20 22:37 +0000 [r248108] Olle Johansson <oej@edvina.net>
+
+ * res/res_rtp_asterisk.c: Improve support for RTCP reports without
+ report blocks
+
+2010-02-19 18:38 +0000 [r248003] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: mfcr2 issue 0016844 - Fix portability bit
+ fields and make mfcr2_immediate_accept work again, reported and
+ patched by korihor
+
+2010-02-19 17:40 +0000 [r247915] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: 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.
+
+2010-02-19 17:33 +0000 [r247914] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: 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
+ ................
+
+2010-02-18 23:13 +0000 [r247787-247841] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_speech.c: 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: 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 21:23 +0000 [r247736-247770] David Vossel <dvossel@digium.com>
+
+ * bridges/bridge_softmix.c: 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: fixes Queue with C option crash (closes issue
+ #16475) Reported by: okrief Patches: queue_crash.diff uploaded by
+ dvossel (license 671)
+
+2010-02-18 19:39 +0000 [r247652] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: 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 18:31 +0000 [r247609] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Fix placing ISDN calls on hold preventing native
+ bridging from being reexamined after a transfer. Consider the
+ following scenario: /-- B A == * == Network \-- C Party B calls
+ party A (EuroISDN BRI phone) Party A puts B on hold using the
+ HOLD/RETRIEVE messages. Party A calls party C. Party A puts C on
+ hold to talk with party B again. Party A transfers B to C by
+ hanging up. The call does not get the opportunity to get
+ re-transferred into the ISDN network by the native bridge because
+ native bridging is not being reexamined after the initial
+ transfer.
+
+2010-02-18 16:54 +0000 [r247503-247509] Leif Madsen <lmadsen@digium.com>
+
+ * /, README-SERIOUSLY.bestpractices.txt: 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
+ 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 16:34 +0000 [r247500] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * CHANGES, res/res_jabber.c: Add a new manager event for our
+ buddies status. The new JabberStatus event gives a concise view
+ of the status change to the AMI clients. Thanks fiddur! (closes
+ issue #16760) Reported by: fiddur Patches: 244498.2.diff uploaded
+ by fiddur (license 678) Tested by: fiddur, phsultan
+
+2010-02-18 04:20 +0000 [r247423] Russell Bryant <russell@digium.com>
+
+ * Makefile, /, sounds/Makefile: 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 22:44 +0000 [r247335-247381] Mark Michelson <mmichelson@digium.com>
+
+ * main/test.c: Fix a couple of bugs in test tab completion. 1. Add
+ missing unlock of lists. 2. Swap order of arguments to
+ test_cat_cmp in complete_test_name.
+
+ * main/test.c: Tab completion for test categories and names for
+ "test show registered" and "test execute" CLI commands.
+
+ * main/strings.c, include/asterisk/strings.h: 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...
+
+2010-02-17 19:51 +0000 [r247295] Jeff Peeler <jpeeler@digium.com>
+
+ * funcs/func_groupcount.c, tests/test_app.c (added), main/app.c,
+ CHANGES: Add support for GROUP_MATCH_COUNT regex matching on
+ category Current support for regex matching was previously only
+ available on the group. Also, error reporting for regex failures
+ has been added. In addition to this feature enhancement a unit
+ test has been written to check the regular expression logic to
+ ensure the count operation is working as expected. (closes issue
+ #16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by
+ kobaz (license 834) Review:
+ https://reviewboard.asterisk.org/r/503/
+
+2010-02-17 19:23 +0000 [r247248-247282] David Vossel <dvossel@digium.com>
+
+ * tests/test_devicestate.c: modified device2extension_test's
+ category
+
+ * tests/test_devicestate.c (added): unit test for combined device
+ state mapping and device to exten state mapping Review:
+ https://reviewboard.asterisk.org/r/516/
+
+ * main/features.c, CHANGES, configs/features.conf.sample: addition
+ of dynamic parkinglots feature This feature allows for
+ parkinglots to be created dynamically within the dialplan. Thanks
+ to all who were involved with getting this patch written and
+ tested! (closes issue #15135) Reported by: IgorG Patches:
+ features.dynamic_park.v3.diff uploaded by IgorG (license 20)
+ 2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7)
+ dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested
+ by: eliel, IgorG, acunningham, mvanbaak, zktech Review:
+ https://reviewboard.asterisk.org/r/352/
+
+2010-02-17 16:24 +0000 [r247169] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: 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.
+ ........
+
+2010-02-17 07:01 +0000 [r247124-247125] Tilghman Lesher <tlesher@digium.com>
+
+ * main/loader.c: RTP documentation states that you can pass NULL as
+ the module, so make sure that's really the case.
+
+ * channels/sip/include/dialog.h (added), channels/chan_sip.c,
+ channels/sip/include/config_parser.h,
+ channels/sip/include/globals.h (added),
+ channels/sip/dialplan_functions.c (added), channels/Makefile,
+ channels/sip/include/sip_utils.h,
+ channels/sip/include/dialplan_functions.h (added): Make all of
+ the various rtpqos parameters in this branch available from the
+ CHANNEL function. Also includes a test for retrieving rtpqos
+ parameters, including a NULL RTP driver. Additionally, some
+ further separation of the SIP internal API into headers was
+ necessary. (closes issue #16652) Reported by: kkm Patches:
+ 20100204__issue16652.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/501/
+
+2010-02-16 23:44 +0000 [r247076] Mark Michelson <mmichelson@digium.com>
+
+ * main/strings.c: 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.
+
+2010-02-16 22:58 +0000 [r247035] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c: generate
+ connected line info update from info in h.323 packets Tested by:
+ benngard
+
+2010-02-16 21:15 +0000 [r246985] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/strings.h: Add some clarifying documentation to
+ the ast_str_set and ast_str_append functions.
+
+2010-02-16 21:03 +0000 [r246980-246981] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c: swap openssl with OpenSSL in warning message.
+ (issue #16673)
+
+ * main/tcptls.c: 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)
+
+2010-02-16 18:29 +0000 [r246942] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_pbx.c (added): Add unit test for dialplan pattern
+ matching. This test works by reading input from arrays to build a
+ sample dialplan. From there, patterns are attempted to be matched
+ against said dialplan, with the expected match given. We then
+ search in our example dialplan to see if we find a match and if
+ what we find matches what we expected it to match. (closes issue
+ #16809) Reported by: lmadsen Tested by: mmichelson Review:
+ https://reviewboard.asterisk.org/r/504/
+
+2010-02-16 17:07 +0000 [r246899] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: 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
+
+2010-02-16 15:36 +0000 [r246710-246863] Tilghman Lesher <tlesher@digium.com>
+
+ * build_tools/cflags.xml, build_tools/cflags-devmode.xml: Revert
+ changes for now, pending discussion
+
+ * build_tools/cflags-devmode.xml: Add a few more targets for
+ DEBUG_THREADLOCALS
+
+ * build_tools/cflags.xml, channels/chan_usbradio.c,
+ build_tools/cflags-devmode.xml, main/strings.c,
+ apps/app_voicemail.c: Change the blanket rules to delete
+ .lastclean on all CFLAGS menuselect targets to be more
+ particular. This change builds upon the recent change to
+ menuselect to add 'touch_on_change' as an attribute of both
+ categories and members. This should allow only the most invasive
+ defines to cause a complete rebuild, while defines which only
+ affect a subset of modules will only cause a rebuild of that
+ smaller set.
+
+ * channels/chan_sip.c: Allow Timer B to be set on the peer, and
+ ensure SIP rules are followed (or warn) in comparison to Timer
+ T1. (closes issue #16643) Reported by: nahuelgreco Patches:
+ 20100204__issue16643.diff.txt uploaded by tilghman (license 14)
+ Tested by: oej
+
+ * Makefile, /: 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.)
+ ........
+
+2010-02-15 22:08 +0000 [r246669] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Restore triedtopribridge flag code removed
+ in -r211197. Ooops. Failed to note that we were inside a for loop
+ and pri_channel_bridge() needs to be executed only once.
+
+2010-02-15 21:37 +0000 [r246667] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/utils.xml: Instead of just automatically filtering out in
+ the Makefile, give an indication of dependencies in menuselect.
+
+2010-02-15 15:45 +0000 [r246627] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c,
+ channels/sip/include/sip_utils.h,
+ channels/sip/include/reqresp_parser.h: chan_sip parse code
+ refactoring plus two new unit tests Code Refactoring Changes -
+ read_to_parts() moved to reqresp_parser.c and has been renamed as
+ get_name_and_number() - get_in_brackets() moved to
+ reqresp_parser.c - find_closing_quotes() added to sip_utils.h
+ Logic Changes - get_name_and_number() now uses parse_uri() and
+ get_calleridname() for parsing. Before this change only names
+ within quotes were found, when names not within quotes are
+ possible. New Unit Tests -sip_get_name_and_number_test
+ -sip_get_in_brackets_test (closes issue #16707) Reported by:
+ Nick_Lewis Patches: issue16706.diff uploaded by dvossel (license
+ 671) Review: https://reviewboard.asterisk.org/r/499/
+
+2010-02-12 23:32 +0000 [r246420-246546] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: 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 ........
+
+ * channels/chan_sip.c: fixes areas where port should be removed
+ from domain during parsing A patch was committed recently that
+ converted duplicate uri parsing code to use the parse_uri
+ function. There were two instances where this conversion did not
+ mimic previous behavior exactly because the port was not being
+ parsed off the end of the domain. In order to do this, a dummy
+ pointer argument needs to be passed into parse_uri so it will
+ know it must parse out the port from the domain. If a port output
+ paramenter is not present, the domain is returned with the port
+ still attached.
+
+2010-02-12 08:30 +0000 [r246382] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c, UPGRADE.txt, CHANGES: Updated doc for OSP
+ lookup application.
+
+2010-02-11 21:57 +0000 [r246299-246338] David Vossel <dvossel@digium.com>
+
+ * tests/test_heap.c, tests/test_event.c,
+ channels/sip/reqresp_parser.c, channels/sip/config_parser.c:
+ fixes some test description formatting inconsistencies so log
+ file looks nice
+
+ * tests/test_astobj2.c (added), main/astobj2.c: astobj2 unit test
+ and bug fix A bug was discovered during the creation of the
+ astobj2 unit test. When OBJ_MULTIPLE | OBJ_UNLINK is used, the
+ objects being returned had a ref count issue. This patch resolves
+ that. Review: https://reviewboard.asterisk.org/r/496/
+
+2010-02-10 23:19 +0000 [r246260] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h, tests/test_event.c (added),
+ main/event.c: Add a test module for the event API, test_event.c.
+ This module includes a single test so far that creates events
+ using two different methods and does some verification on the
+ result to make sure the correct data can be retrieved from the
+ event that was created. One bug was found in the event API while
+ developing this test, which makes me happy. :-) Review:
+ https://reviewboard.asterisk.org/r/495/
+
+2010-02-10 23:13 +0000 [r246249] David Vossel <dvossel@digium.com>
+
+ * channels/sip/reqresp_parser.c,
+ channels/sip/include/reqresp_parser.h: additional parse_uri test
+ and documentation
+
+2010-02-10 21:55 +0000 [r246200-246208] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_pktccops.exports (added): res_pktccops needs to be able
+ to export a symbol for chan_mgcp (closes issue #16782) Reported
+ by: nahuelgreco Patches: res_pktccops.exports uploaded by
+ nahuelgreco (license 162)
+
+ * funcs/func_strings.c: Fussy compiler on another machine...
+
+ * funcs/func_strings.c: Fix weird issue with unit tests on
+ optimized build - turned out to be a signing issue.
+
+2010-02-10 17:49 +0000 [r246116] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_queue.c: 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 16:47 +0000 [r246070] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c: 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 16:01 +0000 [r245945-246030] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_agi.c: 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/
+
+ * funcs/func_strings.c: Enable warnings on atypical conditions for
+ the FILTER function (suggested by mmichelson on the -dev list).
+
+ * /, funcs/func_strings.c, configs/extensions.conf.sample: 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 23:32 +0000 [r245864] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/test.h, tests/test_sha1.c (removed),
+ include/asterisk/utils.h, tests/test_substitution.c,
+ tests/test_heap.c, tests/test_ast_format_str_reduce.c,
+ tests/test_skel.c, tests/test_utils.c, funcs/func_math.c,
+ channels/sip/reqresp_parser.c, main/test.c, tests/test_md5.c
+ (removed), channels/sip/config_parser.c, tests/test_sched.c:
+ Various updates to the unit test API. 1) It occurred to me that
+ the difference in usage between the error ast_str and the
+ ast_test_update_status() usage has turned out to be a bit
+ ambiguous in practice. In a lot of cases, the same message was
+ being sent to both. In other cases, it was only sent to one or
+ the other. My opinion now is that in every case, I think it makes
+ sense to do both; we should output it to the CLI as well as save
+ it off for logging purposes. This change results in most of the
+ changes in this diff, since it required changes to all existing
+ unit tests. It also allowed for some simplifications of unit test
+ API implementation code. 2) Update ast_test_status_update() to
+ include the file, function, and line number for the code
+ providing the update. 3) There are some formatting tweaks here
+ and there. Hopefully they aren't too distracting for code review
+ purposes. Reviewboard's diff viewer seems to do a pretty good job
+ of pointing out when something is a whitespace change. 4) I moved
+ the md5_test and sha1_test into the test_utils module. It seemed
+ like a better approach since these tests are so tiny. 5) I
+ changed the number of nodes used in heap_test_2 from 1 million to
+ 100 thousand. The only reason for this was to reduce the time it
+ took for this test to run. 6) Remove an unused function prototype
+ that was at the bottom of utils.h. 7) Simplify test_insert()
+ using the LIST_INSERT_SORTALPHA() macro. The one minor difference
+ in behavior is that it no longer checks for a test registered
+ with the same name. 8) Expand the code in test_alloc() to provide
+ specific error messages for each failure case, to clearly inform
+ developers if they forget to set the name, summary, description,
+ etc. 9) Tweak the output of the "test show registered" CLI
+ command. I swapped the name and category to have the category
+ first. It seemed more natural since that is the sort key. 10)
+ Don't output the status ast_str in the "test show results" CLI
+ command. This is going to tend to be pretty verbose, so just
+ leave that for the detailed test logs (test generate results).
+ Review: https://reviewboard.asterisk.org/r/493/
+
+2010-02-09 23:18 +0000 [r245793-245804] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fixes a merging error for the iaxs and
+ iaxsl off by one fix
+
+ * /, channels/chan_iax2.c: 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 18:06 +0000 [r245729] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_fax.c: 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 17:40 +0000 [r245727] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: 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 16:24 +0000 [r245680] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: 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 23:43 +0000 [r245597-245624] Russell Bryant <russell@digium.com>
+
+ * main/event.c: 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.
+
+ * tests/test_utils.c: UNREGISTER instead of REGISTER in
+ unload_module().
+
+ * main/pbx.c: Use memmove() instead of memcpy() for a case where
+ the buffers overlap. Once again, valgrind is freaking awesome.
+ That is all.
+
+ * channels/Makefile: Remove object files from the channels/sip/
+ directory on make clean.
+
+2010-02-08 22:31 +0000 [r245578] Tilghman Lesher <tlesher@digium.com>
+
+ * main/Makefile, channels/Makefile: 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 20:41 +0000 [r245497] Jason Parker <jparker@digium.com>
+
+ * /, main/ast_expr2f.c, main/ast_expr2.fl: 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?). ........
+
+2010-02-08 04:51 +0000 [r245268-245385] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/install_prereq: Add the libvpb-dev package as a
+ dependency.
+
+ * pbx/pbx_gtkconsole.c: Add a todo for pbx_gtkconsole for updating
+ to gtk2. This module needs to be converted to gtk2, or we will
+ eventually have to just remove it from the tree. gtk1 isn't even
+ packaged anymore in the distro I'm using. I suspect nobody uses
+ this and that nobody would notice if we removed it.
+
+ * contrib/scripts/install_prereq: Add more packages required for
+ building Asterisk modules.
+
+ * channels/chan_usbradio.c: Make chan_usbradio compile.
+
+ * tests/test_sha1.c (added): Add a SHA1 test module. Review:
+ https://reviewboard.asterisk.org/r/492/
+
+ * tests/test_md5.c: Remove unnecessary include, ast_md5_hash()
+ comes from utils.h.
+
+ * tests/test_md5.c (added): Add an MD5 test module. Review:
+ https://reviewboard.asterisk.org/r/491/
+
+ * tests/test_ast_format_str_reduce.c: Fix a couple of spelling
+ errors, and add format module dependencies.
+
+ * channels/sip/include/config_parser.h, channels/sip/include/sip.h,
+ channels/sip/include/sip_utils.h,
+ channels/sip/include/reqresp_parser.h: Tweak formatting and add
+ minor updates to some comments.
+
+ * main/test.c: Remove an extra space.
+
+2010-02-07 19:51 +0000 [r245230] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Remove parsing of constantssrc from
+ reload_config. This config option is already handled by the
+ function handle_common_options and it is unnecessary to parse the
+ value again.
+
+2010-02-06 14:43 +0000 [r245192] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: 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.
+
+2010-02-05 21:21 +0000 [r245147] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: fixes astobj2
+ unlinking of multiple objects when OBJ_MULTIPLE was disabled When
+ OBJ_MULTIPLE was off but OBJ_UNLINK was on, all the items in a
+ bucket were being unlinked instead of just the first match. This
+ fixes that. Review: https://reviewboard.asterisk.org/r/490/
+
+2010-02-05 19:26 +0000 [r245090] Jeff Peeler <jpeeler@digium.com>
+
+ * /, LICENSE, contrib/firmware (removed): 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 19:07 +0000 [r245046] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_ast_format_str_reduce.c, main/file.c: Merge tests that
+ verify the same thing. (Oops.)
+
+2010-02-05 18:12 +0000 [r245006] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: adds total call numbers available to 'iax2
+ show callnumber usage' cli output
+
+2010-02-05 17:20 +0000 [r244945] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ res/res_calendar_caldav.c: Fix crash on 32-bit for users not
+ using https (closes issue #16778) Reported by: pitel Patches:
+ diff.txt uploaded by twilson (license 396) Tested by: twilson,
+ pitel
+
+2010-02-05 17:05 +0000 [r244927] Sean Bright <sean@malleable.com>
+
+ * /, main/asterisk.c: 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. ........
+
+2010-02-05 16:59 +0000 [r244769-244924] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/config_parser.h,
+ channels/sip/config_parser.c: fixes issue with sip registry not
+ having correct default expiry default expiry was not being set
+ correctly for a registry object. Thanks to ebroad for reporting
+ the issue and testing the patch.
+
+ * main/astobj2.c: fixes memory leak in astobj2 test
+ ao2_iterator_destroy was not being used on the iterator during
+ the test. This resulted in the container never actually being
+ destroyed.
+
+ * channels/chan_sip.c: parse_moved_contact tries to parse
+ contact_name twice parse_moved_contact attempts to remove a
+ quoted string twice, and the first try wasn't even being done
+ correctly.
+
+2010-02-04 22:43 +0000 [r244728-244768] Tilghman Lesher <tlesher@digium.com>
+
+ * main/file.c: Try to make ast_format_str_reduce fail...
+
+ * include/asterisk/manager.h: Oops
+
+ * include/asterisk/manager.h: Define a small set of constant return
+ values
+
+2010-02-04 15:36 +0000 [r244688] David Vossel <dvossel@digium.com>
+
+ * main/test.c: fix truncated format string in 'test show
+ registered' When using the 'test show registered' cli command the
+ 'Test Results' category was truncating the last few characters
+ making it look like 'Test Resul'. I also expanded other parts of
+ the format to better represent how long function names and
+ categories will likely be.
+
+2010-02-04 00:12 +0000 [r244647] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sip: Add ignore *.i files property to the new
+ channels/sip directory.
+
+2010-02-03 20:48 +0000 [r244598] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c, CHANGES: Add some additional option support for
+ non-default parking lots. The options are: parkedcallparking,
+ parkedcallhangup, parkedcallrecording, and parkedcalltransfers.
+ Previously these options were only available for the default
+ parking lot. (closes issue #16641) Reported by: bluecrow76
+ Patches: asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76
+ (license 270)
+
+2010-02-03 20:33 +0000 [r244597] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/config_parser.h
+ (added), channels/sip/reqresp_parser.c (added), channels/sip
+ (added), channels/Makefile, channels/sip/config_parser.c (added),
+ channels/sip/include (added), channels/sip/include/sip.h (added),
+ channels/sip/include/sip_utils.h (added),
+ channels/sip/include/reqresp_parser.h (added): -----Changes -----
+ New files - channels/sip/sip.h – A new header for shared #define,
+ enum, and struct definitions. - channels/sip/include/sip_utils.h
+ – sip util functions shared among the all the sip APIs -
+ channels/sip/include/config_parser.h – sip config-parser API -
+ channels/sip/config_parser.c – Contains sip.conf parsing helper
+ functions with unit tests. -
+ channels/sip/include/reqresp_parser.h – sip request response
+ parser API - channels/sip/reqresp_parser.c – Contains sip request
+ and response parsing helper functions with unit tests. New Unit
+ Tests - sip_parse_uri_test - sip_parse_host_test -
+ sip_parse_register_line_test Code Refactoring - All reusable
+ #define, enum, and struct definitions were moved out of
+ chan_sip.c into sip.h. During this process formatting changes
+ were made to comments in both sip.h and chan_sip.c in order to
+ better adhere to the coding guidelines. - The beginnings of three
+ new sip APIs, sip-utils.h, config-parser.h, reqresp-parser.h
+ using existing chan_sip.c functions. - parse_uri() and
+ get_calleridname() were moved from chan_sip.c to request-parser.c
+ along with unit tests for both functions. - sip_parse_host() and
+ sip_parse_register_line() were moved from chan_sip.c to
+ config-parser.c along with unit tests for both functions. Changes
+ to parse_uri() -removal of the options parameter. It was never
+ used and did not behave correctly. -additional check for
+ [?header] field. When this field was present, the transport type
+ was not being set correctly. ----- Overview ----- This patch is
+ introduced with the hope that unit tests for all our sip parsing
+ functions will be written soon. chan_sip is a huge file, and with
+ the addition of each unit test chan_sip is going to grow larger
+ and harder to maintain. I'm proposing we begin refactoring
+ chan_sip, starting with the parsing functions. With each parsing
+ function we move into a separate helper file, a unit test should
+ accompany it. I've attempted to lay down the ground work for this
+ change by creating two new parser helper files (config-parser.c
+ and reqresp-parser.c) and moving all shared structs, enums, and
+ defines from chan_sip.c into a shared sip.h file. We can't verify
+ everything in Asterisk using unit tests, but string parsing is
+ one area where unit tests make the most sense. By beginning to
+ restructure the code in this way, chan_sip not only becomes less
+ bloated, but Asterisk as a whole will become more stable. Review:
+ https://reviewboard.asterisk.org/r/477/
+
+2010-02-03 19:26 +0000 [r244547] Mark Michelson <mmichelson@digium.com>
+
+ * main/sched.c: Initialize counters in ast_sched_report so that
+ resulting data is not bogus.
+
+2010-02-03 18:34 +0000 [r244505] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: 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 22:27 +0000 [r244443] David Vossel <dvossel@digium.com>
+
+ * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
+ 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 20:32 +0000 [r244071-244393] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, CHANGES: 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. Also, allow
+ GOSUB_RETVAL to be inherited back across a peer/master channel.
+ (closes issue #16687) Reported by: bklang Patches:
+ app_dial-preserve-gosub_retval.patch uploaded by bklang (license
+ 919) (with modifications) (closes issue #16686) Reported by:
+ bklang Patches: app_dial-respect-gosub_result.patch uploaded by
+ bklang (license 919) (with modifications)
+
+ * funcs/func_math.c: Correct some off-by-one errors, especially
+ when expressions don't contain expected spaces. Also include the
+ tests provided by the reporter, as regression tests. (closes
+ issue #16667) Reported by: wdoekes Patches:
+ astsvn-func_match-off-by-one.diff uploaded by wdoekes (license
+ 717)
+
+ * /, apps/app_voicemail.c: Merged revisions 244242 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r244242 | tilghman | 2010-02-01 17:13:44 -0600 (Mon, 01
+ Feb 2010) | 11 lines Backup and restore original textfile, for
+ prosthesis (gerund of prepend). Also, fix menuselect such that
+ changing voicemail build options correctly causes rebuild.
+ (closes issue #16415) Reported by: tomo1657 Patches:
+ prepention.patch uploaded by tomo1657 (license 484) (with
+ modifications by me to backport to 1.4) ........
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 244070
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010)
+ | 16 lines Revert previous chan_local fix (r236981) and fix
+ instead by destroying expired frames in the queue. (closes issue
+ #16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt
+ uploaded by tilghman (license 14)
+ 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman
+ (license 14) Tested by: kobaz, atis (closes issue #16581)
+ Reported by: ZX81 (closes issue #16681) Reported by: alexr1
+ ........
+
+2010-01-28 22:37 +0000 [r243986] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: Optimization to manager events. When potentially
+ sending manager events, return immediately if there are no
+ sessions or hooks. Also, avoid locking the hooks list if it is
+ empty. (issue #16455) Reported by: atis Patches:
+ manager_hooks_trunk.patch uploaded by atis (license 242)
+
+2010-01-28 20:00 +0000 [r243943] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/iax2-parser.c: Informational message, not an error.
+
+2010-01-28 18:35 +0000 [r243780-243860] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Add a missing line terminator for T.38 SDP.
+
+ * /, channels/chan_sip.c: Merged revisions 243779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010)
+ | 2 lines Fix a bogus third argument to ast_copy_string().
+ ........
+
+2010-01-27 20:37 +0000 [r243551-243693] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 243691 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010)
+ | 5 lines Revert 243570, I should have looked at this closer.
+ Will reopen the issue, but am leaving the review closed as the
+ change was pointless. (issue #16488) ........
+
+ * CHANGES: expand code based appreviation of AST_CONFIG_DIR to
+ configuration directory
+
+ * /, apps/app_queue.c: Merged revisions 243570 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010)
+ | 9 lines Extend announcement URL used with Queue from 80 chars
+ to PATH_MAX. (closes issue #16488) Reported by: syspert Patches:
+ soundfilelen.pacth-2 uploaded by syspert (license 938) Review:
+ https://reviewboard.asterisk.org/r/475/ ........
+
+ * Makefile, CHANGES, include/asterisk/options.h, main/asterisk.c,
+ main/loader.c: Add new option to asterisk.conf (lockconfdir) to
+ protect conf dir during reloads (closes issue #16358) Reported
+ by: raarts Patches: lockconfdir.diff uploaded by raarts (license
+ 937) modified by me
+
+2010-01-27 18:08 +0000 [r243487] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 243486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243486 | mmichelson | 2010-01-27 12:06:43 -0600 (Wed, 27 Jan
+ 2010) | 3 lines Use a safe list traversal while checking for
+ duplicate vars in pbx_builtin_setvar_helper. ........
+
+2010-01-27 17:32 +0000 [r243482] Russell Bryant <russell@digium.com>
+
+ * funcs/func_channel.c, channels/chan_iax2.c: Fix the ability to
+ specify an OSP token for an outbound IAX2 call. When this patch
+ was originally submitted, the code allowed for the token to be
+ set via a channel variable. I decided that a cleaner approach
+ would be to integrate it into the CHANNEL() function.
+ Unfortunately, that is not a suitable approach. It's not possible
+ to get the value set on the channel soon enough using that
+ method. So, go back to the simple channel variable method.
+ (closes issue #16711) Reported by: homesick Patches: iax-svn.diff
+ uploaded by homesick (license 91)
+
+2010-01-26 23:56 +0000 [r243391] David Vossel <dvossel@digium.com>
+
+ * /, main/features.c: Merged revisions 243390 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243390 | dvossel | 2010-01-26 17:55:49 -0600 (Tue, 26 Jan 2010)
+ | 9 lines fixes bug with channel receiving wrong privileges after
+ call parking (closes issue #16429) Reported by: Yasuhiro Konishi
+ Patches: features.c.diff uploaded by Yasuhiro Konishi (license
+ 947) Tested by: dvossel ........
+
+2010-01-26 20:49 +0000 [r243346] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_senddtmf.c: Code clean up in app_senddtmf Pushes code
+ clean up done in app_externalivr back into app_senddtmf Review:
+ https://reviewboard.asterisk.org/r/473/
+
+2010-01-26 18:20 +0000 [r243244-243266] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 243258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243258 | jpeeler | 2010-01-26 12:19:10 -0600 (Tue, 26 Jan 2010)
+ | 2 lines Remove unnecessary code in ast_read as issue 16058 has
+ been fully solved now. ........
+
+ * main/frame.c: Fix crash resulting from frames with invalid data
+ pointers. In ast_frdup the frame data union does not get set to
+ point to malloced memory if the datalen is zero, so make sure to
+ handle the same case in ast_frisolate appropriately. (closes
+ issue #16058) Reported by: atis Patches: bug16058-fix.patch
+ uploaded by jpeeler (license 325) Tested by: atis
+
+2010-01-26 17:40 +0000 [r243200-243242] David Vossel <dvossel@digium.com>
+
+ * main/test.c: modify 'test show registered' cli output format In
+ order to improve readability, the output from 'test show
+ registered' has been modified to truncate fields to fit within
+ the format output if they are over a certain length.
+
+ * include/asterisk/utils.h, channels/chan_sip.c, tests/test_utils.c
+ (added), main/test.c, main/utils.c: RFC compliant uri and
+ display-name encode/decode 1. URI Encoding This patch changes
+ ast_uri_encode()'s behavior when doreserved is enabled.
+ Previously when doreserved was enabled only a small set of
+ reserved characters were encoded. This set was comprised
+ primarily of the reserved characters defined in RFC3261 section
+ 25.1, but contained other characters as well. Rather than only
+ escaping the reserved set, doreserved now escapes all characters
+ not within the unreserved set as defined by RFC 3261 and RFC
+ 2396. Also, the 'doreserved' variable has been renamed to
+ 'do_special_char' in attempts to avoid confusion. When doreserve
+ is not enabled, the previous logic of only encoding the
+ characters <= 0X1F and > 0X7f remains, except for the '%'
+ character, which must always be encoded as it signifies a HEX
+ escaped character during the decode process. 2. URI Decoding:
+ Break up URI before decode. In chan_sip.c ast_uri_decode is
+ called on the entire URI instead of it's individual parts after
+ it is parsed. This is not good as ast_uri_decode can introduce
+ special characters back into the URI which can mess up parsing.
+ This patch resolves this by not decoding a URI until parsing is
+ completely done. There are many instances where we check to see
+ if pedantic checking is enabled before we decode a URI. In these
+ cases a new macro, SIP_PEDANTIC_DECODE, is used on the individual
+ parsed segments of the URI rather than constantly putting if
+ (pedantic) { decode() } checks everywhere in the code. In the
+ areas where ast_uri_decode is not dependent upon pedantic
+ checking this macro is not used, but decoding is still moved to
+ each individual part of the URI. The only behavior that should
+ change from this patch is the time at which decoding occurs.
+ Since I had to look over every place URI parsing occurs to create
+ this patch, I found several places where we use duplicate code
+ for parsing. To consolidate the code, those areas have updated to
+ use the parse_uri() function where possible. 3. SIP display-name
+ decoding according to RFC3261 section 25. To properly decode the
+ display-name portion of a FROM header, chan_sip's
+ get_calleridname() function required a complete re-write. More
+ information about this change can be found in the comments at the
+ beginning of this function. 4. Unit Tests. Unit tests for
+ ast_uri_encode, ast_uri_decode, and get_calleridname() have been
+ written. This involved the addition of the test_utils.c file for
+ testing the utils api. (closes issue #16299) Reported by: wdoekes
+ Patches: astsvn-16299-get_calleridname.diff uploaded by wdoekes
+ (license 717) get_calleridname_rewrite.diff uploaded by dvossel
+ (license 671) Tested by: wdoekes, dvossel, Nick_Lewis Review:
+ https://reviewboard.asterisk.org/r/469/
+
+2010-01-26 15:46 +0000 [r243118-243158] Russell Bryant <russell@digium.com>
+
+ * tests/test_substitution.c: Log the variable name being tested.
+
+ * tests/test_substitution.c: Update test_substitution to show
+ failures in the test log.
+
+ * funcs/func_aes.c: Update func_aes to its pre-ast_str_substitution
+ state. This change makes the AES tests in test_substitution.c
+ pass. We still need to work through what's going wrong in the
+ ast_str version.
+
+2010-01-26 01:56 +0000 [r242967-243077] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_substitution.c: Fixing last errors in the conversion,
+ though it appears that the AES_* functions are still broken.
+
+ * tests/test_substitution.c: Using a dummy channel causes CDR()
+ testing to fail.
+
+ * tests/test_substitution.c: Wish I had gotten to the review before
+ this got submitted, because there's failures we need to address.
+
+ * /, main/Makefile, res/Makefile: Merged revisions 242969 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242969 | tilghman | 2010-01-25 15:50:22 -0600 (Mon, 25 Jan 2010)
+ | 2 lines Err, and use the new menuselect define, too. ........
+
+ * build_tools/cflags.xml, /, build_tools/menuselect-deps.in,
+ configure, configure.ac: Merged revisions 242966 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r242966 | tilghman | 2010-01-25 15:36:33 -0600 (Mon, 25
+ Jan 2010) | 2 lines Only rebuild parsers by an option in
+ menuselect ........
+
+2010-01-25 21:32 +0000 [r242954-242965] Russell Bryant <russell@digium.com>
+
+ * tests/test_substitution.c, tests/test_heap.c,
+ tests/test_ast_format_str_reduce.c, tests/test_skel.c,
+ tests/test_sched.c: Make unit test modules depend on
+ TEST_FRAMEWORK instead of off by default.
+
+ * tests/test_substitution.c: Convert test_substitution module to
+ the unit test API. Review:
+ https://reviewboard.asterisk.org/r/474/
+
+2010-01-25 21:20 +0000 [r242933] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooCalls.c: small corrections in call clearing
+
+2010-01-25 21:13 +0000 [r242904-242919] Olle Johansson <oej@edvina.net>
+
+ * main/pbx.c, main/manager.c, include/asterisk/pbx.h: Change api
+ for pbx_builtin_setvar to actually return error code if a
+ function can't be written to. This patch removes code that was
+ duplicated from pbx.c to manager.c in order to prevent API change
+ in released versions of Asterisk. There are propably also other
+ places that would benefit from reading the return code and react
+ if a function returns error codes on writing a value into it.
+
+ * main/manager.c, /: Merged revisions 242850 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242850 | oej | 2010-01-25 21:03:38 +0100 (Mån, 25 Jan 2010) | 2
+ lines Report error when writing to functions returns error in AMI
+ setvar action ........
+
+2010-01-25 20:18 +0000 [r242857] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, main/Makefile, configure.ac, res/Makefile: Merged
+ revisions 242852 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242852 | tilghman | 2010-01-25 14:15:45 -0600 (Mon, 25 Jan 2010)
+ | 2 lines Restore FreeBSD to able-to-compile-ish-mode ........
+
+2010-01-25 18:01 +0000 [r242812] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar.c: Fix INTERNAL_OBJ error on stop when
+ calendars.conf missing Initialize the calendars container before
+ calling load_config and return FAILURE on allocation failure.
+ Also, use the AST_MODULE_LOAD_* values for return values. Thanks
+ to rmudgett for pointing out the error and the need to use the
+ defined values for return
+
+2010-01-25 05:45 +0000 [r242719-242729] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/Makefile, res/Makefile: Merged revisions 242728 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242728 | tilghman | 2010-01-24 23:42:22 -0600 (Sun, 24 Jan 2010)
+ | 2 lines Buildbot pointed out an error (thanks, buildbot!)
+ ........
+
+ * /, res/Makefile: Merged revisions 242723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242723 | tilghman | 2010-01-24 23:33:37 -0600 (Sun, 24 Jan 2010)
+ | 2 lines Oops, should have used CMD_PREFIX, not ECHO_PREFIX, for
+ the commands. ........
+
+ * /, main/Makefile: Merged revisions 242683 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242683 | tilghman | 2010-01-24 23:13:28 -0600 (Sun, 24 Jan 2010)
+ | 2 lines Make the build of the Asterisk expression parser match
+ that of the AEL parser. ........
+
+2010-01-24 22:42 +0000 [r242645] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooStackCmds.h,
+ addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooCmdChannel.c,
+ addons/ooh323c/src/ooStackCmds.c: AST_CONTROL_CONNECTED_LINE
+ frame type processing added to setup DisplayIE field incorrect
+ q.931 message order filtered on incoming calls (first msg must be
+ setup, next must be not setup)
+
+2010-01-24 21:49 +0000 [r242607] Sean Bright <sean@malleable.com>
+
+ * res/res_phoneprov.c: Instead of crashing, allocate our header
+ ast_str before we try to use it. (closes issue #16680) Reported
+ by: lmadsen Patches: issue16680_20100122.patch uploaded by
+ seanbright (license 71) Tested by: lmadsen
+
+2010-01-24 06:40 +0000 [r242521] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/Makefile, res/Makefile, makeopts.in: Merged revisions 242520
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010)
+ | 8 lines Only rebuild bison and flex source files on demand, if
+ bison and flex are detected by the configure script. Changed
+ after discussion on the -dev list about possible unnecessary
+ build failures, due to checkouts/untars causing these special
+ source files to possibly be newer than their resulting C files.
+ This should additionally ensure that nobody need learn about
+ extra Makefile arguments to ensure the proper files get rebuilt
+ when changes are made to these special source files. ........
+
+2010-01-22 21:45 +0000 [r242424] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/Makefile: Merged revisions 242423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242423 | tilghman | 2010-01-22 15:44:18 -0600 (Fri, 22 Jan 2010)
+ | 7 lines Rebuild from flex, bison sources when necessary. (issue
+ #14629) Reported by: Marquis Patches:
+ 20100121__issue14629.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2010-01-22 16:20 +0000 [r242357] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Add send DTMF feature to ExternalIVR app
+ Implemented a new command 'D' that allows client IVRs to send
+ DTMF digits to the channel. (closes issue #16615) Reported by:
+ thedavidfactor Review: https://reviewboard.asterisk.org/r/465/
+
+2010-01-22 15:09 +0000 [r242317] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_sched.c: The irony of not compile-testing a test
+ program before committing is killing me.
+
+2010-01-22 09:28 +0000 [r242227] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 242226 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3
+ lines Initialize notify_types to NULL ........
+
+2010-01-22 04:57 +0000 [r242184-242186] Russell Bryant <russell@digium.com>
+
+ * main/test.c: Update the doxygenification of some comments.
+
+ * tests/test_sched.c: Convert scheduler API entry order test to the
+ test API. Review: https://reviewboard.asterisk.org/r/470/
+
+ * tests/test_skel.c: Add test API usage example to test_skel.c.
+ Review: https://reviewboard.asterisk.org/r/471/
+
+2010-01-21 22:37 +0000 [r242092] Mark Michelson <mmichelson@digium.com>
+
+ * main/acl.c: Add missing argument to ast_calloc calls.
+
+2010-01-21 21:05 +0000 [r242043] Olle Johansson <oej@edvina.net>
+
+ * main/acl.c: Make sure we initialize the ast_ha structure with
+ ast_calloc
+
+2010-01-21 15:27 +0000 [r241938] Sean Bright <sean@malleable.com>
+
+ * /, configure, configure.ac: Merged revisions 241932 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r241932 | seanbright | 2010-01-21 10:25:46 -0500 (Thu,
+ 21 Jan 2010) | 5 lines Fix configure check for PTHREAD_ONCE_INIT
+ when manually adding -Wall to CFLAGS. (closes issue #16666)
+ Reported by: romain_proformatique ........
+
+2010-01-21 15:14 +0000 [r241896] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_vpb.cc: Formats are inconsistent between even
+ 32-bit and 64-bit Linux. Use casts to ensure both compile.
+
+2010-01-21 14:10 +0000 [r241855-241856] Russell Bryant <russell@digium.com>
+
+ * main/test.c: Point to a useful reference on the XML output
+ format.
+
+ * main/test.c: Modify test results XML format to match the JUnit
+ format. When this code was developed, we came up with our own XML
+ format for the test output. I have since started looking at
+ integration with other tools, namely continuous integration
+ frameworks, and this format seems to be supported across a number
+ of applications. With these changes in place, I was able to get
+ Atlassian Bamboo to interpret the test results.
+
+2010-01-21 05:54 +0000 [r241766] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_math.c: Merged revisions 241765 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r241765 | tilghman | 2010-01-20 23:53:17 -0600 (Wed, 20 Jan 2010)
+ | 2 lines Guard against division by zero. ........
+
+2010-01-20 21:14 +0000 [r241627-241714] David Vossel <dvossel@digium.com>
+
+ * res/res_rtp_asterisk.c: rtp timestamp to timeval calculation fix
+ The rtp timestamp to timeval calculation was only accurate for
+ 8kHz audio. This patch corrects this. Review:
+ https://reviewboard.asterisk.org/r/468/ SWP-648
+
+ * Makefile, /: Merged revisions 241626 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r241626 | dvossel | 2010-01-20 14:00:04 -0600 (Wed, 20 Jan 2010)
+ | 6 lines fixes parsing error in Makefile. Some echo lines were
+ missing "; . Thanks to jparker for pointing out the problem.
+ ........
+
+2010-01-20 17:49 +0000 [r241581] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/cdr.c: Add Calling and Called Subaddress to CDR record
+ Requires 'callingsubaddr' and 'calledsubaddr' fields in backend
+ cdr. (closes issue #16600) Reported by: alecdavis Patches:
+ cdr_subaddr.diff.txt uploaded by alecdavis (license 585) Tested
+ by: alecdavis Review: https://reviewboard.asterisk.org/r/460/
+
+2010-01-20 13:01 +0000 [r241503] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_vpb.cc: Fix up compile breakage from
+ ast_tvdiff_ms() API change.
+
+2010-01-20 08:18 +0000 [r241416] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/pbx.c, channels/sig_pri.c: Update CDR variables as pbx
+ starts Allows CDR variables added in cdr.c:set_one_cid to become
+ visable during the call, by executing ast_cdr_update() early in
+ __ast_pbx run. Reverts sig_pri changes in trunk that are specific
+ to isdn technology only. (closes issue #16638) Reported by:
+ alecdavis Patches: cdr_update.diff3.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2010-01-19 22:59 +0000 [r241366] Jeff Peeler <jpeeler@digium.com>
+
+ * main/pbx.c: Initialize data on the stack so that Park doesn't
+ interpret random arguments. passdata was only being set in
+ pbx_substitue_variables when arguments were passed. (closes issue
+ #16406) (closes issue #16586) Reported by: DLNoah Patches:
+ bug16586v2.patch uploaded by jpeeler (license 325) Tested by:
+ DLNoah
+
+2010-01-19 22:41 +0000 [r241364] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/janitor-projects.txt, apps/app_sendtext.c: Enable SendText to
+ send strings in encoded format. See
+ http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html
+
+2010-01-19 18:51 +0000 [r241314-241315] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_agent.c: small correction from 241314
+
+ * /, channels/chan_agent.c: Merged revisions 241227 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19
+ Jan 2010) | 13 lines Fix deadlock in agent_read by removing call
+ to agent_logoff. One must always lock the agents list lock before
+ the agent private. agent_read locks the private immediately, so
+ locking the agents list lock is not an option (which is what
+ agent_logoff requires). Because agent_read already has access to
+ the agent private all that is necessary is to do the required
+ hanging up that agent_logoff performed. (closes issue #16321)
+ Reported by: valon24 Patches: bug16321.patch uploaded by jpeeler
+ (license 325) ........
+
+2010-01-19 17:42 +0000 [r241230] Jason Parker <jparker@digium.com>
+
+ * Makefile: Allow parallel make (-j) to work properly. After some
+ back and forth with the reporter, we came up with the necessary
+ changes. (closes issue #16489) Reported by: Chainsaw Patches:
+ asterisk-1.6.2.1-parallel-make-minimal.patch uploaded by Chainsaw
+ (license 723) Tested by: Chainsaw, qwell
+
+2010-01-19 00:28 +0000 [r241188] Tilghman Lesher <tlesher@digium.com>
+
+ * main/srv.c, res/res_agi.c, CHANGES, include/asterisk/srv.h:
+ Create iterative method for querying SRV results, and use that
+ for finding AGI servers. (closes issue #14775) Reported by:
+ _brent_ Patches: 20091215__issue14775.diff.txt uploaded by
+ tilghman (license 14) hagi-5.patch uploaded by brent (license
+ 388) Tested by: _brent_ Reviewboard:
+ https://reviewboard.asterisk.org/r/378/
+
+2010-01-19 00:24 +0000 [r241187] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sig_pri.c: Update CDR variables before pbx starts
+ (overlap dial) Allows CDR variables added in cdr.c:set_one_cid to
+ become visable during the call. (issue #16638) Reported by:
+ alecdavis Patches: cdr_update.diff2.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2010-01-18 22:31 +0000 [r241143] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c,
+ main/features.c, pbx/pbx_dundi.c, main/enum.c,
+ include/asterisk/time.h, main/timing.c: Extend max call limit
+ duration from 24.8 days to 292+ million years. If the limit was
+ set past MAX_INT upon answering, the call was immediately hung up
+ due to overflow from the return of ast_tvdiff_ms (in
+ ast_check_hangup). The time calculation functions ast_tvdiff_sec
+ and ast_tvdiff_ms have been changed to return an int64_t to
+ prevent overflow. Also the reporter suggested adding a message
+ indicating the reason for the call hanging up. Given that the new
+ limit is so much higher, the message (which would only really be
+ useful in the overflow scenario) has been made a debug message
+ only. (closes issue #16006) Reported by: viraptor
+
+2010-01-18 22:03 +0000 [r241098] Jason Parker <jparker@digium.com>
+
+ * main/rtp_engine.c: Fix an RTP instance allocation failure on
+ Solaris. (closes issue #16543) Reported by: crjw Patches:
+ rtp_sin_family.patch uploaded by crjw (license 963) Tested by:
+ crjw, qwell
+
+2010-01-18 22:00 +0000 [r241097] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sig_pri.c: Update CDR variables before pbx starts Allows
+ CDR variables added in cdr.c:set_one_cid to become visable during
+ the call. (closes issue #16638) Reported by: alecdavis Patches:
+ cdr_update.diff.txt uploaded by alecdavis (license 585)
+
+2010-01-18 19:57 +0000 [r241016] Sean Bright <sean@malleable.com>
+
+ * /, main/config.c: Merged revisions 241015 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r241015 | seanbright | 2010-01-18 14:54:19 -0500 (Mon, 18 Jan
+ 2010) | 12 lines Plug a memory leak when reading configs with
+ their comments. While reading through configuration files with
+ the intent of returning their full contents (comments
+ specifically) we allocated some memory and then forgot to free
+ it. This doesn't fix 16554 but clears up a leak I had in the lab.
+ (issue #16554) Reported by: mav3rick Patches:
+ issue16554_20100118.patch uploaded by seanbright (license 71)
+ Tested by: seanbright ........
+
+2010-01-18 19:26 +0000 [r241012] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c, CHANGES: Make HASHes inheritable across
+ channel creation.
+
+2010-01-18 18:00 +0000 [r240973-240974] David Ruggles <thedavidfactor@gmail.com>
+
+ * UPGRADE.txt: ExternalIVR information for UPGRADE.txt added a
+ paragraph about the fixes and changes to the ExternalIVR
+ application.
+
+ * doc/externalivr.txt: Updated ExternalIVR documentation Rewrote a
+ large portion of the existing documentation and added information
+ about the TCP/IP socket interface
+
+2010-01-18 17:45 +0000 [r240971] David Vossel <dvossel@digium.com>
+
+ * Makefile, CHANGES: transmit_silence_during_record replaced by
+ transmit_silence In asterisk.conf, transmit_silence_during_record
+ has been removed in favor of using only the transmit_silence
+ option. The transmit_silence_during_record option remains a valid
+ option in asterisk.conf, but has been removed from the sample
+ config and noted in CHANGES.
+
+2010-01-18 17:41 +0000 [r240969] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Add notification of interrupted file Add
+ file information to data element of T event so the file
+ information is sent to the client when it is interrupted.
+ Previously only notification of pending files that were dropped
+ was sent (closes issue #16147) Reported by: thedavidfactor Tested
+ by: thedavidfactor Review:
+ https://reviewboard.asterisk.org/r/449/
+
+2010-01-18 16:45 +0000 [r240842-240887] David Vossel <dvossel@digium.com>
+
+ * Makefile: updated transmit_silence option documentation in
+ asterisk.conf This patch updates the transmit_silence option to
+ better document why the option exists, and what it affects.
+ Thanks to russell for providing the verbage for this update.
+
+ * apps/app_queue.c: fixes spelling error. s/memeber/member
+
+2010-01-17 19:45 +0000 [r240717] Sean Bright <sean@malleable.com>
+
+ * main/pbx.c: Avoid a crash on Solaris when running 'core show
+ functions.' (closes issue #16309) Reported by: asgaroth
+
+2010-01-16 00:54 +0000 [r240667] Sean Bright <sean@malleable.com>
+
+ * res/res_musiconhold.c: Get MoH building on OpenSolaris.
+
+2010-01-15 23:50 +0000 [r240629] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, main/asterisk.c: Err, oops, it was already the way I
+ intended.
+
+2010-01-15 23:09 +0000 [r240548-240552] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxygen/commits.h: Note where empty lines should
+ reside in commit messages.
+
+ * Makefile, /: Merged revisions 240547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r240547 | russell | 2010-01-15 17:06:11 -0600 (Fri, 15 Jan 2010)
+ | 2 lines Fix a spelling error in the asterisk.conf sample.
+ ........
+
+2010-01-15 22:07 +0000 [r240505] Sean Bright <sean@malleable.com>
+
+ * res/res_timing_timerfd.c: Clarify error message in
+ res_timing_timerfd.
+
+2010-01-15 21:42 +0000 [r240421-240500] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/astcanary.c: Oops, missed an include
+
+ * utils/astcanary.c, main/asterisk.c: The previous attempt at using
+ a pipe to guarantee astcanary shutdown did not work. We're
+ revisiting the previous patch, albeit with a method that
+ overcomes the prior criticism that it was not POSIX-compliant.
+ (closes issue #16602) Reported by: frawd Patches:
+ 20100114__issue16602.diff.txt uploaded by tilghman (license 14)
+ Tested by: frawd
+
+ * apps/app_directed_pickup.c, main/features.c,
+ include/asterisk/manager.h: Add pickup event to AMI. Also, fix
+ AMI documentation. (closes issue #16431) Reported by: syspert
+ Patches: 20100112__issue16431.diff.txt uploaded by tilghman
+ (license 14)
+
+2010-01-15 20:58 +0000 [r240420] Mark Michelson <mmichelson@digium.com>
+
+ * main/utils.c: Make sure to set owner_line, ownder_func, and
+ owner_file in ast_calloc_with_stringfields. Asterisk would crash
+ on startup if MALLOC_DEBUG were set in menuselect. This is
+ because the manager action UpdateConfig had to resize its string
+ field allocation to set the description. When the resize
+ occurred, ast_copy_string would crash because we were attempting
+ to copy a string from a NULL pointer. Setting the strings
+ initially makes the code much less crashy.
+
+2010-01-15 20:58 +0000 [r240415-240419] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Make sure that the limit is N, not N - 1.
+
+ * /, apps/app_voicemail.c: Merged revisions 240414 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15
+ Jan 2010) | 15 lines Disallow leaving more than maxmsg
+ voicemails. This is a possibility because our previous method
+ assumed that no messages are left in parallel, which is not a
+ safe assumption. Due to the vmu structure duplication, it was
+ necessary to track in-process messages via a separate structure.
+ If at some point, we switch vmu to an ao2-reference-counted
+ structure, which would eliminate the prior noted duplication of
+ structures, then we could incorporate this new in-process
+ structure directly into vmu. (closes issue #16271) Reported by:
+ sohosys Patches: 20100108__issue16271.diff.txt uploaded by
+ tilghman (license 14) 20100108__issue16271__trunk.diff.txt
+ uploaded by tilghman (license 14)
+ 20100108__issue16271__1.6.0.diff.txt uploaded by tilghman
+ (license 14) Tested by: jsutton ........
+
+2010-01-15 20:41 +0000 [r240411] Russell Bryant <russell@digium.com>
+
+ * main/event.c: Ensure payload type is properly checked when
+ comparing against cached events. (closes issue #16607) Reported
+ by: ddv2005 Patches: event.patch uploaded by ddv2005 (license
+ 769)
+
+2010-01-15 18:21 +0000 [r240368] Sean Bright <sean@malleable.com>
+
+ * main/pbx.c, main/manager.c, res/res_smdi.c, apps/app_meetme.c,
+ channels/chan_sip.c, cel/cel_tds.c, main/features.c,
+ res/res_phoneprov.c, cdr/cdr_tds.c, apps/app_jack.c: Convert a
+ few places to use ast_calloc_with_stringfields where applicable.
+
+2010-01-15 16:51 +0000 [r240329] Russell Bryant <russell@digium.com>
+
+ * configure: Update configure script for an OSP toolkit related
+ change.
+
+2010-01-15 16:28 +0000 [r240328] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/sip.conf.sample: Clarify RTP NAT handling a bit.
+
+2010-01-14 23:13 +0000 [r240226-240271] Sean Bright <sean@malleable.com>
+
+ * res/res_config_ldap.c: Plug a memory leak in res_config_ldap.
+ (closes issue #16257) Reported by: nito Patches:
+ issue16257_20100111.diff uploaded by seanbright (license 71)
+
+ * res/res_timing_timerfd.c: If we aren't running on a machine that
+ support CLOCK_MONOTONIC, don't load. Group developed and tested
+ by seanbright, Corydon76, Kobaz, and Amorsen.
+
+2010-01-14 18:03 +0000 [r240179] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: Fix broken call pickup The problem was the
+ OUTGOING flag was not getting set properly on the channel,
+ resulting in pickup failing as ast_read thought the call was
+ inbound. Refer to 170393 for a more verbose description as this
+ is the same exact change. (closes issue #16539) Reported by:
+ syspert Patches: bug16539.patch uploaded by jpeeler (license 325)
+ Tested by: syspert
+
+2010-01-14 17:34 +0000 [r240129-240175] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Similarly, ensure that matchcid is duplicated
+ correctly when merging contexts.
+
+ * main/pbx.c: Ensure that the callerid is NULL when the parent is
+ effectively NULL. This applies only to pattern-match hints, which
+ create exact-match hints on the fly.
+
+2010-01-14 16:14 +0000 [r240078] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c: This change fixes a few bugs in the way the far max
+ IFP was calculated that were introduced in r231692. (closes issue
+ #16497) Reported by: globalnetinc Patches:
+ udptl-max-ifp-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: globalnetinc
+
+2010-01-14 14:38 +0000 [r240039] Leif Madsen <lmadsen@digium.com>
+
+ * doc/building_queues.txt (added): Add documentation about how to
+ build queues. Add a how-to set of documentation about building
+ queues with Asterisk. This documentation is based on Asterisk
+ 1.6.2 but should work on most versions with minor modifications.
+ (closes issue #16237) Reported by: lmadsen Patches: Building
+ Queues (FINAL).txt uploaded by lmadsen (license 10) Tested by:
+ pdhales, lmadsen, cmdrwalrus
+
+2010-01-13 23:22 +0000 [r239920-239997] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Oops, another tag error
+
+ * main/pbx.c: Oops, missed a closing tag
+
+ * main/pbx.c, include/asterisk/pbx.h: Add the TESTTIME() dialplan
+ function, which permits testing GotoIfTime. Specifically, by
+ setting TESTTIME() to a particular date and time, you can test
+ whether a dialplan correctly branches as was intended. This was
+ developed after recent questions on the -users list on how to
+ test their holiday dialplan logic. (closes issue #16464) Reported
+ by: tilghman Patches: 20100112__issue16464.diff.txt uploaded by
+ tilghman (license 14) Review:
+ https://reviewboard.asterisk.org/r/458/
+
+ * main/ast_expr2f.c, main/ast_expr2.fl: Flex uses fwrite
+ incorrectly, which breaks the build. Providing a workaround.
+
+2010-01-13 19:48 +0000 [r239839] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 239838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r239838 | jpeeler | 2010-01-13 13:43:33 -0600 (Wed, 13 Jan 2010)
+ | 11 lines Fix regression for timed out parked call returning to
+ caller This issue seems to have been exposed by the fix in 160390
+ whereby using a masquerade prevented a crash. The new channel
+ used in the masquerade was not copying the macro information from
+ the old channel. (closes issue #15459) Reported by: djrodman
+ Patches: patch_15459.txt uploaded by mnick (license ) ........
+
+2010-01-13 19:31 +0000 [r239834] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.conf.sample: Add more examples to
+ extensions.conf showing how to use various functionality and
+ provide commonly useful features. (closes issue #16090) Reported
+ by: pprindeville Patches: extensions.conf-bugid16090.patch#3
+ uploaded by pprindeville (license 347) Tested by: tzafrir,
+ pprindeville, lmadsen
+
+2010-01-13 18:16 +0000 [r239797] Tilghman Lesher <tlesher@digium.com>
+
+ * main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Code
+ previously added to ast_expr2f.c warranted a change in the source
+ file ast_expr2.fl. Also, made a Makefile change to ensure that
+ the expression parser C source files get regenerated correctly,
+ when we need that to happen.
+
+2010-01-13 16:31 +0000 [r239712] David Vossel <dvossel@digium.com>
+
+ * Makefile, main/channel.c, apps/app_waitforring.c,
+ apps/app_waitforsilence.c: add silence gen to wait apps
+ asterisk.conf's 'transmit_silence' option existed before this
+ patch, but was limited to only generating silence while recording
+ and sending DTMF. Now enabling the transmit_silence option
+ generates silence during wait times as well. To achieve this,
+ ast_safe_sleep has been modified to generate silence anytime no
+ other generators are present and transmit_silence is enabled.
+ Wait apps not using ast_safe_sleep now generate silence when
+ transmit_silence is enabled as well. (closes issue #16524)
+ Reported by: kobaz (closes issue #16523) Reported by: kobaz
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/456/
+
+2010-01-13 10:45 +0000 [r239663-239665] Olle Johansson <oej@edvina.net>
+
+ * main/poll.c: MAX() moved to utils.h
+
+ * channels/chan_sip.c: SIP Show channelstats fix - use float
+ division to show proper stats (closes issue #15819) Reported by:
+ klaus3000 Patches: asterisk-sip-show-channelstats-trunk.txt
+ uploaded by klaus3000 (license 65) Tested by: klaus3000, oej This
+ patch is for trunk only and will be blocked in 1.6.2
+
+2010-01-13 07:02 +0000 [r239624-239625] TransNexus OSP Development <support@transnexus.com>
+
+ * doc/tex/channelvariables.tex: Updated channel variable list of
+ osplookup application.
+
+ * apps/app_osplookup.c: Updated XML doc for OSP.
+
+2010-01-12 19:58 +0000 [r239571] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Blank callerid and NULL callerid should not compare
+ equal. The second is the default state for matching CID in the
+ dialplan (no matching) while the first matches one particular
+ CallerID. This is a regression. (fixes AST-314, SWP-611)
+
+2010-01-12 18:55 +0000 [r239525] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/cdr.c: add Dialed Number Identifier (DNID) field to cdr
+ records. reviewboard link:
+ https://reviewboard.asterisk.org/r/455/ Reported by: alecdavis
+ Tested by: alecdavis Patch cdr_dnid.diff2.txt uploaded by
+ alecdavis (license 585)
+
+2010-01-12 18:22 +0000 [r239520] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Note that direct T.38 is not supported.
+ (closes issue #16411) Reported by: stanusr Patches:
+ __20091210-sip.conf.sample-documentation.txt uploaded by lmadsen
+ (license 10)
+
+2010-01-12 17:09 +0000 [r239473] Sean Bright <sean@malleable.com>
+
+ * res/res_config_ldap.c: Fix crash in res_config_ldap. We need to
+ allocate enough room for 2 pointers, not 2 characters. (closes
+ issue #16397) Reported by: bklang Patches: res_config_ldap.patch
+ uploaded by applsplatz (license 949) Tested by: applsplatz
+
+2010-01-12 16:14 +0000 [r239427] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes text support in sdp answer The code
+ that handled setting 'm=text' in the sdp was not executing in the
+ correct order. The check to see if text was needed came after the
+ check to add 'm=text' to the sdp, this resulted in 'm=text'
+ always being set to 0 because it looked like text was never
+ required. (closes issue #16457) Reported by: peterj Patches:
+ textportinsdp.diff uploaded by peterj (license 951)
+ issue16457.diff uploaded by dvossel (license 671) Tested by:
+ peterj
+
+2010-01-12 07:48 +0000 [r239389] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/astmm.h: Adding Tilghman's documentation from
+ asterisk-dev to the actual file.
+
+2010-01-12 03:21 +0000 [r239152-239308] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/scripts/safe_asterisk: Merged revisions 239307 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r239307 | tilghman | 2010-01-11 21:18:36 -0600 (Mon, 11 Jan 2010)
+ | 8 lines Portability and other fixes for the safe_asterisk
+ script (closes issue #16416) Reported by: bklang Patches:
+ safe_asterisk-compat-1.patch uploaded by bklang (license 919)
+ 20100106__issue16416__trunk.diff.txt uploaded by tilghman
+ (license 14) Tested by: bklang ........
+
+ * contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk,
+ contrib/init.d/rc.archlinux.asterisk,
+ contrib/init.d/rc.suse.asterisk: Add LSB headers to init scripts.
+ (closes issue #14864) Reported by: lathama Patches:
+ lsb-init-info-debian.diff uploaded by pkempgen (license 169)
+
+ * res/res_pktccops.c: Socket level option is SOL_SOCKET, not
+ SO_SOCKET. (issue #16580)
+
+ * Makefile, contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.suse.asterisk: Permit more options in the
+ Makefile as to startup options (closes issue #16454) Reported by:
+ syspert Patches: 20091228__issue16454__3.diff.txt uploaded by
+ tilghman (license 14) Tested by: syspert
+
+ * Makefile: Including bundle1.o breaks Tiger and Leopard (issue
+ #16449)
+
+ * addons/cdr_mysql.c, configs/cdr_mysql.conf.sample: Permit dates
+ and times to be stored in timezones other than the default
+ (typically, UTC) (closes issue #16401) Reported by: lordmortis
+
+2010-01-11 16:41 +0000 [r239111-239114] Sean Bright <sean@malleable.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ res/res_calendar_caldav.c, res/res_clialiases.c: Pass NULL for
+ the ao2_callback function pointer instead of duplicating cb_true.
+
+ * main/astobj2.c: Fix ao2_callback when both OBJ_MULTIPLE and
+ OBJ_NODATA are passed. There is an issue which only affects trunk
+ and the new ao2_callback OBJ_MULTIPLE implementation. When both
+ OBJ_MULTIPLE and OBJ_NODATA are passed, only the first object is
+ visited, regardless of what is returned by the specified
+ callback. This causes a problem when we are clearing a container,
+ i.e.: ao2_callback(container, OBJ_UNLINK | OBJ_NODATA |
+ OBJ_MULTIPLE, NULL, NULL); Only unlinks the first object. This
+ patch resolves this. (closes issue #16564) Reported by: pj
+ Patches: issue16564_20100111.diff uploaded by seanbright (license
+ 71) Tested by: pj, seanbright Review:
+ https://reviewboard.asterisk.org/r/457/
+
+ * main/test.c: Fix spelling of 'category.'
+
+2010-01-10 19:37 +0000 [r239074] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/chan_ooh323.c, main/frame.c, channels/chan_iax2.c:
+ According to POSIX, the capital L modifier applies only to
+ floating point types. Fixes a crash on Solaris. (closes issue
+ #16572) Reported by: crjw Patches: frame_changes.patch uploaded
+ by crjw (license 963) Plus several others found and fixed by me
+
+2010-01-10 17:53 +0000 [r239037] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooq931.h, addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooq931.c: add docallbacks flag in q931decode
+ function because when we decode received q931 packet we must do
+ callbacks and when we print sended q931 packet we must not.
+
+2010-01-10 06:56 +0000 [r239000] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, main/asterisk.c: It's been long enough -- make the
+ behavior introduced in 1.6 the default.
+
+2010-01-09 01:08 +0000 [r238916] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 238915 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238915 | tilghman | 2010-01-08 18:57:58 -0600 (Fri, 08 Jan 2010)
+ | 6 lines -1 is interpreted as an error, intead of the maximum
+ mask. (closes issue #16241) Reported by: vnovy Patches:
+ manager.c.patch uploaded by vnovy (license 922) ........
+
+2010-01-08 23:30 +0000 [r238835] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 238834 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238834 | jpeeler | 2010-01-08 17:28:37 -0600 (Fri, 08 Jan 2010)
+ | 4 lines Stop a crash when no peer is passed to masq_park_call.
+ (distantly related to issue #16406) ........
+
+2010-01-08 22:54 +0000 [r238754-238795] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Add the class actually used in the
+ MusicOnHold start event. (closes issue #16499) Reported by:
+ syspert Patches: mohclass.patch uploaded by syspert (license 938)
+
+ * res/res_agi.c: Initialize variables that we attempt to free
+ later. (closes issue #16302) Reported by: yahsyn Patches:
+ 20091124__issue16302.diff.txt uploaded by tilghman (license 14)
+ Tested by: yahsyn
+
+2010-01-08 21:04 +0000 [r238716] Matthew Nicholson <mnicholson@digium.com>
+
+ * tests/test_ast_format_str_reduce.c (added): Added a test for
+ ast_format_reduce_str(). (related to issue #16560)
+
+2010-01-08 19:39 +0000 [r238635] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: fixes
+ AUDIOHOOK_INHERIT regression During the process of removing an
+ audiohook from one channel and attaching it to another the
+ audiohook's status is updated to DONE and then back to whatever
+ it was previously. Typically updating the status after setting it
+ to DONE is not a good idea because DONE can trigger unrecoverable
+ audiohook destruction events... because of this a conditional
+ check was added to audiohook_update_status to explicitly prevent
+ the audiohook from ever changing after being set to DONE. It was
+ this check that prevented audiohook inherit from work properly
+ though. Now ast_audiohook_move_by_source is treated as a special
+ exception, as the audiohook must be returned to its previous
+ status after attaching it to the new channel. This is only a safe
+ operation because the audiohook's lock is held the entire time,
+ otherwise this could cause trouble. (closes issue #16522)
+ Reported by: corruptor
+
+2010-01-08 19:32 +0000 [r238630] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/file.c: Merged revisions 238629 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238629 | mnicholson | 2010-01-08 13:20:44 -0600 (Fri, 08 Jan
+ 2010) | 5 lines Properly calculate the remaining space in the
+ output string when reducing format strings. (closes issue #16560)
+ Reported by: goldwein ........
+
+2010-01-08 17:18 +0000 [r238583] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Stop trying to find a parking space after
+ traversing the parkinglot one time. (closes issue #16428)
+ Reported by: Yasuhiro Konishi
+
+2010-01-07 21:24 +0000 [r238527] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Fix using the wrong pointer type in
+ do_idle_thread().
+
+2010-01-07 20:42 +0000 [r238361-238492] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: fixes ast_transfer stall until hangup if called
+ with a channel that doesn't support transfers ast_transfer sets
+ res to 0 if there is no technology transfer function, but then
+ tests for it to be negative before deciding to do an early exit.
+ As a result, it will will wait for an AST_CONTROL_TRANSFER
+ message that will never come. (closes issue #16424) Reported by:
+ davidw Patches: Issue_16424_trunk_234134.patch uploaded by davidw
+ (license 780)
+
+ * /, channels/chan_iax2.c: 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: 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: 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 09:14 +0000 [r238313] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * configs/sip.conf.sample: Document the usefulness of explicit
+ udp:// in the register string
+
+2010-01-06 21:45 +0000 [r238231] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_cdr.c: 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 20:37 +0000 [r238134-238181] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: 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: 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 16:36 +0000 [r238091] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/test.h: fixes test.c compile issue when
+ TEST_FRAMEWORK is not enabled The ast_test_status_update()
+ function is defined in test.h. When TEST_FRAMEWORK is not enabled
+ a macro is defined as a no-op place holder for this function. The
+ macro did not contain the correct number of arguments. This
+ caused a compile error. Much thanks to wdoekes for reporting the
+ issue and supplying the patch!
+
+2010-01-06 15:35 +0000 [r238014] Sean Bright <sean@malleable.com>
+
+ * addons/format_mp3.c: Fix reading samples from format_mp3 after
+ ast_seekstream/ast_tellstream. There is a bug when using
+ ast_seekstream/ast_tellstream with format_mp3 in that the file
+ read position is not reset before attempting to read samples. So
+ when we seek to determine the maximum size of the file (as in
+ res_agi's STREAM FILE) we weren't then resetting the file pointer
+ so that we could properly read samples. This patch addresses that
+ (in a similar manner to format_wav.c). (closes issue #15224)
+ Reported by: rbd Patches: 20091230_addons_1.4_issue15224.diff
+ uploaded by seanbright (license 71) Tested by: rbd, seanbright
+ Review: https://reviewboard.asterisk.org/r/453
+
+2010-01-06 15:19 +0000 [r238010] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_mp3.c: 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 06:53 +0000 [r237968] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Whoa, duplicate setting (dead code).
+
+2010-01-05 23:08 +0000 [r237920] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c: 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
+
+2010-01-05 20:56 +0000 [r237882] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Mismerged a bit.
+
+2010-01-05 19:29 +0000 [r237839] David Vossel <dvossel@digium.com>
+
+ * main/pbx.c: 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 18:57 +0000 [r237804] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h, channels/sig_pri.c: Removed unused
+ parameters from analog_available() and sig_pri_available().
+
+2010-01-05 18:46 +0000 [r237802-237803] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, CHANGES: Add a missing part of the connected
+ line work into trunk. Part of the work done for connected line
+ was to add an optional argument to the 'f' option to allow for
+ the connected party information of the outgoing channel to be set
+ to the argument provided. This was overlooked during the merge of
+ the work to trunk and is being added back now. The CHANGES file
+ has also been updated to note this change.
+
+ * CHANGES: Spell "aficionado" like someone who isn't stupid.
+
+2010-01-05 17:26 +0000 [r237699-237749] Russell Bryant <russell@digium.com>
+
+ * main/utils.c: Fix build of utility apps that include utils.c.
+
+ * /, main/utils.c: 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 16:08 +0000 [r237656] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_mixmonitor.c: 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 21:48 +0000 [r237406-237574] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: 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 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: Oops, didn't compile (thanks, kpfleming)
+
+ * main/config.c: 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 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 16:39 +0000 [r237327] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c: 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 16:24 +0000 [r237323] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_agi.c: Fix timeout for AGI command speech recognize.
+ (closes issue #16297) Reported by: semond
+
+2010-01-04 16:20 +0000 [r237319] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: 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/ ........
+
+2010-01-04 07:55 +0000 [r237284] Olle Johansson <oej@edvina.net>
+
+ * res/res_pktccops.c, channels/chan_mgcp.c: - Disable res_pktccops
+ by default - Add dependency in chan_mgcp that was missing - Add a
+ small amount of doc to the source code
+
+2010-01-04 03:38 +0000 [r237250] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: 1. Added reporting operator names in
+ AuthReq. 2. Added retrieving operator names from AuthRsp and
+ exporting them.
+
+2010-01-02 16:35 +0000 [r237213] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: global_contact_ha was renamed in trunk
+
+2010-01-02 09:54 +0000 [r237136] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: 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 23:51 +0000 [r237098] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooCalls.c: small q931 processing and
+ signalling corrections don't decode UUIE from Q931StatusMessage
+ clean call without callIdentifier data don't start tcs/msd
+ exchange procedure after call proceeding received (closes issue
+ #16365) Reported by: benngard2 Tested by: may213, benngard2
+
+2009-12-30 22:30 +0000 [r237050] Jason Parker <jparker@digium.com>
+
+ * main/say.c, doc/lang/vietnamese.ods (added),
+ apps/app_voicemail.c: Add app_voicemail and say.c support for
+ Vietnamese. Also add an XXX comment that I'm baffled nobody has
+ ever complained about. We say "first message", and then we go
+ into language-specific stuff where we proceed to say..."first
+ message". (closes issue #15053) Reported by: dinhtrung Patches:
+ vietnamese.ods uploaded by dinhtrung (license 776)
+ app_voicemail.c.diff uploaded by dinhtrung (license 776) (closes
+ issue #15626) Reported by: dinhtrung Patches: say.c.diff uploaded
+ by dinhtrung (license 776)
+
+2009-12-30 21:59 +0000 [r236982] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: 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 21:09 +0000 [r236893-236902] Jeff Peeler <jpeeler@digium.com>
+
+ * utils/ael_main.c: One more LOW_MEMORY compile fix.
+
+ * channels/chan_sip.c, main/cli.c: Fix compiling with LOW_MEMORY.
+ Modified handle_verbose to be LOW_MEMORY aware, removed old RTP
+ related code in chan_sip. (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 17:53 +0000 [r236802-236847] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, cel/cel_adaptive_odbc.c: 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: 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
+
+2009-12-29 10:59 +0000 [r236756] TransNexus OSP Development <support@transnexus.com>
+
+ * configs/osp.conf.sample, apps/app_osplookup.c, configure.ac: 1.
+ Updated for OSP Toolkit 3.6.0. 2. Added service type ported
+ number query. 3. Formated code.
+
+2009-12-28 22:09 +0000 [r236713] Jason Parker <jparker@digium.com>
+
+ * main/ast_expr2.y, main/ast_expr2.c: 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 17:37 +0000 [r236667] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Use recommended option, not deprecated
+ option. (closes issue #16515) Reported by: ManChicken
+
+2009-12-28 15:22 +0000 [r236510-236613] Sean Bright <sean@malleable.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/threadstorage.h: 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 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 18:20 +0000 [r236434] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: 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 15:27 +0000 [r236358] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, sounds/Makefile: 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 19:17 +0000 [r236304-236312] David Vossel <dvossel@digium.com>
+
+ * CHANGES: Update CHANGES to reflect new QUEUE_MEMBER option,
+ "ready"
+
+ * apps/app_queue.c: QUEUE_MEMBER(..., ready) counts only ready
+ agents, not free agents wrapping up The QUEUE_MEMBER dialplan
+ function can return total members, logged-in members and "free"
+ members count. A member is counted as "free" immediately after
+ his call ends, even though its wrap-up time, if specified in
+ queues.conf, has not yet expired, and the queue will not actually
+ route a call to it. This Patch introduces a new "ready" option
+ that only counts free agents no longer in the wrap up time
+ period. (closes issue #16240) Reported by: kkm Patches:
+ appqueue-memberfun-readyoption-trunk.diff uploaded by kkm
+ (license 888) Tested by: kkm, dvossel
+
+ * CHANGES, apps/app_queue.c: update CHANGES to reflect new 'R'
+ app_queue option plus a minor optimization to the feature patch
+ (issue #16384)
+
+ * apps/app_queue.c: new parameter 'R' to the Queue application The
+ 'R' argument stops moh and indicates ringing once the agent is
+ ringing. This allows the person in the queue to know their call
+ is potentially about to be answered. (closes issue #16384)
+ Reported by: haakon Patches: new_app_queue.c.patch uploaded by
+ haakon (license 880) Tested by: haakon, loloski, dvossel
+
+2009-12-23 18:25 +0000 [r236183-236300] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c: 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 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
+ ........
+
+ * include/asterisk/test.h: Allow test_heap.c to compile when
+ AST_DEVMODE is true, but TEST_FRAMEWORK is false
+
+ * apps/app_voicemail.c: Actually use tmp for something (brings
+ trunk back into sync with 1.6 branches).
+
+2009-12-22 21:53 +0000 [r236027-236144] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fixes iax "can't compress subclass
+ 4294967295" error (closes issue #16456) Reported by: dvossel
+ Tested by: dvossel
+
+ * /, channels/chan_sip.c: 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) ........
+
+ * CHANGES: update CHANGES to reflect the addition of the test
+ framework
+
+ * include/asterisk/test.h (added), build_tools/cflags-devmode.xml,
+ tests/test_heap.c, main/test.c (added),
+ include/asterisk/_private.h, main/asterisk.c: Unit Test Framework
+ API The Unit Test Framework is a new API that manages
+ registration and execution of unit tests in Asterisk with the
+ purpose of verifying the operation of C functions. The Framework
+ consists of a single test manager accompanied by a list of
+ registered test functions defined within the code. A test is
+ defined, registered, and unregistered from the framework using a
+ set of macros which allow the test code to only be compiled
+ within asterisk when the TEST_FRAMEWORK flag is enabled in
+ menuselect. This allows the test code to exist in the same file
+ as the C functions it intends to verify. Registered tests may be
+ viewed and executed via a set of new CLI commands. CLI commands
+ are also present for generating and exporting test results into
+ xml and txt formats. For more information and use cases please
+ refer to the documentation provided at the beginning of the
+ test.h file. Review: https://reviewboard.asterisk.org/r/447/
+
+2009-12-21 19:54 +0000 [r235941] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_monitor.c: 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 18:51 +0000 [r235904] Kevin P. Fleming <kpfleming@digium.com>
+
+ * contrib/upstart/asterisk.upstart-0.3.9, include/asterisk/cel.h,
+ main/say.c, include/asterisk/channel.h,
+ include/asterisk/manager.h, channels/sig_pri.c,
+ include/asterisk/logger.h, include/asterisk/http.h,
+ include/asterisk/callerid.h, include/asterisk/syslog.h,
+ channels/chan_dahdi.c, include/asterisk/app.h,
+ include/asterisk/doxyref.h, include/asterisk/event.h,
+ channels/sig_analog.c, channels/chan_misdn.c,
+ contrib/upstart/asterisk.user.conf,
+ include/asterisk/rtp_engine.h,
+ include/asterisk/security_events.h,
+ include/asterisk/stringfields.h: Change all refererences to 1.6.3
+ to be 1.8, since that will be the next feature release
+
+2009-12-21 17:00 +0000 [r235822] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/features.c: 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 08:22 +0000 [r235740-235774] 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 0016389) Reported
+ by: alecdavis Tested by: alecdavis Patch
+ dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
+
+ * apps/app_dial.c, CHANGES: app_dial optional parameter to option
+ 'r' to allow play indication from indications.conf (closes issue
+ #14504) Reported by: alecdavis Tested by: alecdavis,jsmith Patch
+ app_dial.play_ring_indications.diff7.txt uploaded by alecdavis
+ (license 585)
+
+2009-12-18 22:51 +0000 [r235660] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h: 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 22:40 +0000 [r235573-235656] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: 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 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 23:21 +0000 [r235521] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: 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 17:19 +0000 [r235422] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 235421 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235421 | tilghman | 2009-12-17 11:17:51 -0600 (Thu, 17 Dec 2009)
+ | 8 lines Use context from which Macro is executed, not macro
+ context, if applicable. Also, ensure that the extension COULD
+ match, not just that it won't match more. (closes issue #16113)
+ Reported by: OrNix Patches: 20091216__issue16113.diff.txt
+ uploaded by tilghman (license 14) Tested by: OrNix ........
+
+2009-12-17 00:52 +0000 [r235342-235382] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Fix call forwarding
+ for analog phones. (closes issue #16440) Reported by: mmichelson
+
+ * configs/jabber.conf.sample, include/asterisk/jabber.h, CHANGES,
+ res/res_jabber.c: Add auth_policy option to jabber.conf for auto
+ user registration. The option is global and currently the
+ acceptable values as noted in the sample config are accept or
+ deny. (closes issue #15228) Reported by: lp0
+
+2009-12-16 05:24 +0000 [r235298] Jared Smith <jaredsmith@jaredsmith.net>
+
+ * /, configs/sip.conf.sample: Merged revisions 235181 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15
+ Dec 2009) | 4 lines Add a line showing that we can use CIDR
+ notation. patch by jsmith, after discussion with jtodd ........
+
+2009-12-16 00:31 +0000 [r235265] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, CHANGES: Enhance AMI redirect to allow channels
+ to be redirected to different places. New parameters
+ ExtraContext, ExtraExtension, and ExtraPriority have been added
+ to redirect the second channel to a different location.
+ Previously, it was only possible to redirect both channels to the
+ same place. (closes issue #15853) Reported by: haakon Patches:
+ trunk-manager.c.patch uploaded by haakon (license 880) Tested by:
+ jpeeler
+
+2009-12-15 23:51 +0000 [r235229] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/strings.h: Is it Friday yet?
+
+2009-12-15 23:41 +0000 [r235226] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: Change match criteria existence in
+ ast_channel_cmp_cb to use ast_strlen_zero. (closes issue #16161)
+ Reported by: may213 Patches: core-show-channel.patch uploaded by
+ may213 (license 454)
+
+2009-12-15 18:43 +0000 [r235132] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: reverse minor sip registration regression A
+ registration regression caused by a code tweak in (issue #14331)
+ and a bug fix in (issue #15539) caused some sip registration
+ config entries to be constructed incorrectly. Origially issue
+ #14331 contained the code tweak as well as a bug fix, but since
+ the issue was reported as a tweak the bug fix portion was moved
+ into issue #15539. Both the tweak and the bug fix contained minor
+ incorrect logic that resulted in some SIP registrations to fail.
+ (issue #14331) (issue #15539)
+
+2009-12-15 15:33 +0000 [r235053] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 235052 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235052 | tilghman | 2009-12-15 09:29:24 -0600 (Tue, 15 Dec 2009)
+ | 4 lines Mandatory argument checking (closes issue #16446)
+ Reported by: nicchap ........
+
+2009-12-15 14:35 +0000 [r235010] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: spandsp does in fact support V.17 modulation at
+ 14.4 kilobits per second, so we should generate T38MaxBitRate of
+ 14400 (even though that doesn't really affect the FAX
+ transmission much at all)
+
+2009-12-15 07:18 +0000 [r234855-234976] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_directory.c: Support option 'n', as applications like
+ Playback, Background etc. Suggested on asterisk-dev as trivial
+ application change. Reported by: alecdavis Tested by: alecdavis
+
+ * main/dsp.c: Whitespace.
+
+ * main/dsp.c: restarts busydetector (if enabled) when DTMF is
+ received. (closes issue #16389) Reported by: alecdavis Tested by:
+ alecdavis Patch dtmf_busydetector.diff.txt uploaded by alecdavis
+ (license 585)
+
+ * apps/app_directory.c: fixes escape to extensions 'o' and 'a', for
+ digits '0' and '*' (closes issue #16437) Reported by: alecdavis
+ Tested by: alecdavis Patch extension_o_a_fix.diff.txt uploaded by
+ alecdavis (license 585)
+
+ * apps/app_directory.c: ast_stream_and_wait(chan,dir-usingkeypad)
+ didn't capture the dialled DTMF. (closes issue #16409) Reported
+ by: alecdavis Tested by: alecdavis Patch bug_16409.diff.txt
+ uploaded by alecdavis (license 585)
+
+2009-12-14 23:16 +0000 [r234820] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Allow greetings-only mailboxes for Voicemail. (closes issue
+ #15132) Reported by: floletarmo Patches: voicemail_changes.patch
+ uploaded by floletarmo (license 784) (with some additional
+ changes by me)
+
+2009-12-14 21:32 +0000 [r234776] Jason Parker <jparker@digium.com>
+
+ * apps/app_readexten.c: Allow tonelist as argument to ReadExten.
+ ReadExten already supported playing a tonezone from
+ indications.conf. It now has the ability to use a tonelist like
+ 440+480/2000|0/4000 (closes issue #15185) Reported by: jcovert
+ Patches: app_readexten.c.patch uploaded by jcovert (license 551)
+ Tested by: qwell Patch modified by me, to maintain backwards
+ compatibility.
+
+2009-12-14 21:13 +0000 [r234700] Tilghman Lesher <tlesher@digium.com>
+
+ * /, build_tools/make_version_c, build_tools/make_version_h: Merged
+ revisions 234699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234699 | tilghman | 2009-12-14 15:09:56 -0600 (Mon, 14 Dec 2009)
+ | 5 lines Deal with the situation where .flavor exists but
+ .version does not. Also make the script slightly more portable,
+ in keeping with autoconf syntax. (closes issue #14737) Reported
+ by: davidw ........
+
+2009-12-14 17:19 +0000 [r234631] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/imapstorage.tex, /: Update IMAP build documentation.
+ Update the IMAP build documentation to show how to build on
+ 64-bit platforms. (issue #16433) Reported by: shrift Tested by:
+ lmadsen
+
+2009-12-14 16:08 +0000 [r234572] Sean Bright <sean@malleable.com>
+
+ * main/timing.c: The default rate for 'timing test' is actually
+ 50/sec, not 100/sec as advertised.
+
+2009-12-14 10:46 +0000 [r234526] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 234492 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8
+ lines Stop sending 183's after call hangup. There where still
+ cases where the 183 keep-alive mechanism would not stop sending
+ 183's even though the Asterisk server had sent a final reply to
+ the invite. EDVX-28 ........
+
+2009-12-13 09:41 +0000 [r234458] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Trim leading/trailing spaces from the filename, to
+ deal with common user error.
+
+2009-12-11 23:17 +0000 [r234380] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 234379 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009)
+ | 11 lines Fix talking detection status after conference user is
+ muted. This patch ensures that when a conference user is muted
+ that the accompanying AMI Meetme talking off event is sent. Also,
+ the meetme list output is updated to show the muted user as
+ unmonitored. (closes issue #16247) Reported by: dimas Patches:
+ v3-16247.patch uploaded by dimas (license 88) ........
+
+2009-12-10 21:01 +0000 [r234256] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 234255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234255 | qwell | 2009-12-10 14:58:09 -0600 (Thu, 10 Dec 2009) |
+ 9 lines Fix unselecting of menuselect options via GLOBAL_MAKEOPTS
+ and USER_MAKEOPTS. (closes issue #16296) Reported by: abelbeck
+ Patches: issue16296-20091210.diff uploaded by qwell (license 4)
+ (abelbeck described a fix, which I expanded upon) Tested by:
+ abelbeck, qwell, lmadsen ........
+
+2009-12-10 18:56 +0000 [r234210] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Missed a case that emits a WARNING where
+ none is warranted.
+
+2009-12-10 17:31 +0000 [r234173] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c, apps/app_page.c, main/app.c, CHANGES: Add
+ audio announcement option to app_page As described in the CHANGES
+ file: * MeetMe has a new option 'G' to play an announcement
+ before joining a conference. * Page has a new option 'A(x)' which
+ will playback an announcement simultaneously to all paged phones
+ (and optionally excluding the caller's one using the new option
+ 'n') before the call is bridged. To add the new option to meetme,
+ the conference flag options had to be extended to 64 bits.
+ (closes issue #14365) Reported by: dferrer Patches:
+ page_announce.patch uploaded by dferrer (license 525) modified by
+ me Review: https://reviewboard.asterisk.org/r/188/
+
+2009-12-10 16:24 +0000 [r234129] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 234095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009)
+ | 9 lines When we receive no response at all to our INVITE, allow
+ the channel to be destroyed. (closes issue #15627) Reported by:
+ falves11 Patches: 20091209__issue15627__1.6.0.diff.txt uploaded
+ by tilghman (license 14) 20091209__issue15627__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: falves11 Review:
+ https://reviewboard.asterisk.org/r/446/ (closes issue #15716)
+ Reported by: dant (closes issue #16270) Reported by: corruptor
+ (closes issue #15356) Reported by: falves11 (issue #16382)
+ Reported by: lftsy ........
+
+2009-12-09 23:35 +0000 [r233967-234055] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt, CHANGES: Move an entry from CHANGES to UPGRADE.txt.
+
+ * UPGRADE.txt, CHANGES: Move an entry from CHANGES that should be
+ in UPGRADE.txt.
+
+ * CHANGES: Provide a real description of LOCAL_PEEK().
+
+ * CHANGES: Remove a feature from CHANGES that was listed twice for
+ 1.6.2.
+
+ * CHANGES: Fix up the faxdetect entry in CHANGES. This feature was
+ listed as a 1.6.2 feature, even though it's in all 1.6.X
+ versions. The description of the feature was also no longer
+ accurate.
+
+ * CHANGES: Remove an entry from CHANGES that is already in
+ UPGRADE.txt (where it should be).
+
+2009-12-08 18:40 +0000 [r233718-233732] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/res_config_mysql.c: Typo pointed out on #asterisk-dev (by
+ atis_work)
+
+ * res/res_musiconhold.c: Find another ref leak and change how we
+ manage module references. (closes issue #16388, closes issue
+ #16279, closes issue #16390) 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 18:00 +0000 [r233692] Russell Bryant <russell@digium.com>
+
+ * formats/format_sln.c, formats/format_wav.c,
+ formats/format_ogg_vorbis.c, 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_g726.c, formats/format_siren14.c,
+ formats/format_jpeg.c, formats/format_gsm.c,
+ formats/format_g729.c: 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
+
+2009-12-07 23:28 +0000 [r233611] David Vossel <dvossel@digium.com>
+
+ * main/utils.c: fixes incorrect logic in ast_uri_encode issue
+ #16299
+
+2009-12-07 23:10 +0000 [r233577] Atis Lezdins <atis@iq-labs.net>
+
+ * contrib/valgrind.supp: 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 19:48 +0000 [r233545] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Fix TCP Client interface Fix a couple of
+ very minor bugs that prevent the socket client from working. The
+ wrong set of properties were used in one place and the size of
+ the address variable isn't set if the host name is an ip address.
+ Also includes a fix for a bug that was introduced previously.
+ (closes issue #16121) Reported by: thedavidfactor Tested by:
+ thedavidfactor Review: https://reviewboard.asterisk.org/r/439/
+
+2009-12-07 18:08 +0000 [r233472] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: 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 17:59 +0000 [r233468] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/jabber.h, CHANGES, res/res_jabber.c: Add
+ applications JabberJoin, JabberLeave, JabberSendGroup for XMPP
+ groupchat (closes issue #14352) Reported by: fiddur Patches:
+ trunk-14352-2.diff uploaded by phsultan (license 73) Tested by:
+ fiddur
+
+2009-12-07 16:14 +0000 [r233394] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: 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
+
+2009-12-06 07:01 +0000 [r233358] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/compat.h, main/strcompat.c, main/app.c: Move
+ implementation of closefrom(3) from app.c to strcompat.c
+
+2009-12-04 21:54 +0000 [r233280] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /: 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 21:06 +0000 [r233239] Tilghman Lesher <tlesher@digium.com>
+
+ * main/translate.c: Using the builtin function breaks OpenBSD 4.2
+ (closes issue #16395) Reported by: jtodd
+
+2009-12-04 20:21 +0000 [r233121-233235] David Vossel <dvossel@digium.com>
+
+ * CHANGES: update CHANGES file for .m3u support in Mp3Player
+ application
+
+ * apps/app_mp3.c: .m3u support for Mp3Player app (closes issue
+ #14823) Reported by: macli Patches: app_mp3.diff1 uploaded by
+ macli (license ) Tested by: macli, dvossel
+
+ * CHANGES: update CHANGES for new queue option,
+ penaltymemberslimit.
+
+ * apps/app_queue.c: changes penaltymemberslimit to use scanf for
+ config value parsing
+
+ * configs/queues.conf.sample, apps/app_queue.c: new queue option,
+ penaltymemberslimit, disregards penalty on too few queue members
+ when enabled (closes issue #14559) Reported by: fiddur Patches:
+ trunk-199584-1.diff uploaded by fiddur (license 678) Tested by:
+ fiddur, dvossel
+
+ * /, apps/app_voicemail.c: 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 17:18 +0000 [r233100] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: 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 17:15 +0000 [r233093] Matthias Nick <mnick@digium.com>
+
+ * pbx/pbx_config.c: 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 16:55 +0000 [r233059-233089] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Let's unlock the lines list after the
+ AST_LIST_TRAVERSE instead of inside it.
+
+ * channels/chan_skinny.c: Only assign line and device in
+ handle_transfer_button when we have a subchannel. (closes issue
+ #16040) Reported by: ebroad
+
+2009-12-04 16:08 +0000 [r233050] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/res_config_mysql.c: Update the mysql driver to always
+ return NULL columns, as this is needed for the realtime API to
+ work correctly. (closes issue #16138) Reported by: sohosys
+ Patches: 20091029__issue16138.diff.txt uploaded by tilghman
+ (license 14) Tested by: sohosys
+
+2009-12-04 15:38 +0000 [r233046] Matthias Nick <mnick@digium.com>
+
+ * /, main/dsp.c: 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 ........
+
+2009-12-04 05:26 +0000 [r232854-232982] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_pktccops.c: Buildbot complained
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_pktccops.c: OS X does not define MSG_NOSIGNAL, but it
+ does have a socket option SO_NOSIGPIPE. (closes issue #16178)
+ Reported by: oej
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
+ pagerdateformat, to allow shorter dates for SMS messages. (closes
+ issue #16263) Reported by: andrew Patches: pagerdate.patch
+ uploaded by andrew (license 240) (with a slight modification by
+ me)
+
+ * /, apps/app_voicemail.c: 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 20:26 +0000 [r232853] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c: jitterbuffer setup correction
+ correction of double pointer references from previous rev
+
+2009-12-03 08:47 +0000 [r232738-232771] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Replaced two deprecated functions of OSP
+ Toolkit.
+
+ * apps/app_osplookup.c: Added custom info support.
+
+2009-12-03 00:38 +0000 [r232700] Jeff Peeler <jpeeler@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Extend voicemail to allow IMAP folders to be specified per
+ mailbox. Previously only possible per context, new option called
+ imapfolder. (closes issue #14298) Reported by: jablko Patches:
+ patch-200906202 uploaded by jablko (license 675)
+
+2009-12-03 00:09 +0000 [r232660-232661] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Remove debugging line
+
+ * include/asterisk/astobj2.h, res/res_musiconhold.c: 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
+
+2009-12-02 23:27 +0000 [r232657] David Vossel <dvossel@digium.com>
+
+ * UPGRADE.txt, CHANGES: 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 22:17 +0000 [r232587] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: 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/
+
+2009-12-02 22:02 +0000 [r232582] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, /: 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) ........
+
+2009-12-02 21:37 +0000 [r232580] Matthew Nicholson <mnicholson@digium.com>
+
+ * addons/chan_mobile.c: Fix support for multiline SMS messages in
+ chan_mobile. (closes issue #16278) Reported by: Artem Patches:
+ multiline-sms-fix2.diff uploaded by mnicholson (license 96)
+ Tested by: Artem
+
+2009-12-02 21:32 +0000 [r232576] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: 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 21:13 +0000 [r232544] Matthew Nicholson <mnicholson@digium.com>
+
+ * addons/chan_mobile.c: Do something with the service indicator so
+ that asterisk does not attempt to use a chan_mobile endpoint that
+ does not have service. (closes issue #16132) Reported by: nikkk
+ Patches: service-indicator2.diff uploaded by mnicholson (license
+ 96) Tested by: nikkk
+
+2009-12-02 20:10 +0000 [r232442-232510] Joshua Colp <jcolp@digium.com>
+
+ * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add an 'X' option to
+ the asterisk application which enables #exec for configuration
+ files. This option can be used to enable #exec support in the
+ asterisk.conf configuration file. (closes issue #16260) Reported
+ by: atis Patches: exec_includes.patch uploaded by atis (license
+ 242)
+
+ * apps/app_record.c, CHANGES: Add an option to Record which enables
+ a mode where any DTMF digit will terminate recording. (closes
+ issue #15436) Reported by: Vince Patches: app_record.diff
+ uploaded by Vince (license 823) Tested by: dbrooks
+
+2009-12-02 17:18 +0000 [r232365] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Do not change the exten string field or
+ rebuild the contact header on an inbound sip_pvt if the outbound
+ call is redirected.
+
+2009-12-02 17:06 +0000 [r232356] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_amd.c: 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 17:00 +0000 [r232351] David Vossel <dvossel@digium.com>
+
+ * /, main/acl.c: 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 16:40 +0000 [r232345] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: 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 15:42 +0000 [r232269] David Vossel <dvossel@digium.com>
+
+ * funcs/func_groupcount.c, /: 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 14:54 +0000 [r232230] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: 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 03:26 +0000 [r232164] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac: So
+ apparently, some platforms don't have ffsll(3). The manpage lies;
+ it says that the function is in POSIX, but that's only for
+ ffs(3), not ffsll(3).
+
+2009-12-02 00:45 +0000 [r232091] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: 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 23:56 +0000 [r232008-232017] Russell Bryant <russell@digium.com>
+
+ * main/translate.c: Use __builtin_ffsll() from gcc instead of
+ ffssll() to fix a FreeBSD build error.
+
+ * funcs/func_lock.c: Fix a build error on FreeBSD.
+
+ * /, main/file.c: 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 21:54 +0000 [r231927] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: 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 21:20 +0000 [r231867] David Vossel <dvossel@digium.com>
+
+ * main/pbx.c, /: 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 20:27 +0000 [r231814-231850] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_rtp_asterisk.c, channels/chan_unistim.c,
+ main/rtp_engine.c, addons/chan_ooh323.c, channels/chan_sip.c,
+ res/res_adsi.c, addons/chan_ooh323.h,
+ include/asterisk/callerid.h, channels/chan_phone.c,
+ channels/chan_dahdi.c, channels/chan_skinny.c, main/callerid.c,
+ channels/chan_h323.c, addons/ooh323cDriver.c,
+ include/asterisk/rtp_engine.h, addons/ooh323cDriver.h: More
+ 32->64 bit codec conversions. In the process of swapping ULAW to
+ a place in the extended codec space, we found several unhandled
+ cases, where a 32-bit integer was still being used to handle a
+ codec field. Most of these have been fixed with this commit,
+ although there is at least one case (codec_dahdi) which depends
+ upon outside headers to be altered before a conversion can be
+ made. (Fixes AST-278, SWP-459)
+
+ * include/asterisk/mod_format.h: Formats need to be able to
+ represent all 64 codec bits.
+
+2009-12-01 15:47 +0000 [r231741] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/file.c: 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 21:47 +0000 [r231692] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
+ 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.
+
+2009-11-30 21:31 +0000 [r231616-231688] Matthew Nicholson <mnicholson@digium.com>
+
+ * include/asterisk/file.h, /, main/file.c, main/app.c,
+ apps/app_voicemail.c: 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/ ........
+
+ * include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c:
+ Reverted 231616
+
+ * include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c:
+ 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 20:44 +0000 [r231602] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: 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 18:55 +0000 [r231491-231556] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c: 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
+
+ * res/res_rtp_asterisk.c, /: 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 ........
+
+2009-11-30 16:53 +0000 [r231439] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.dynamics (added), Makefile.rules: Export dynamic
+ (weak-linked) symbols correctly. (closes issue #15193) Reported
+ by: eliel Patches: 20091111__issue15193.diff.txt uploaded by
+ tilghman (license 14)
+
+2009-11-30 16:29 +0000 [r231436] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where an immediate masquerade
+ would cause a queued unhold frame to get lost. Now we just
+ indicate unhold directly after the masquerade is complete. (issue
+ ABE-2011)
+
+2009-11-27 08:47 +0000 [r231401] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: 1. Modified exported variable names. 2.
+ Added destination port support. 3. Added new protocols. 4. Added
+ QoS.
+
+2009-11-26 02:09 +0000 [r231299-231369] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/CODING-GUIDELINES, main/asterisk.c: Reorder option flags.
+ Change guidelines so that example code is consistent with
+ guidelines
+
+ * main/channel.c, /: 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 15:42 +0000 [r231189] Matthew Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_lua.c: Load pbx_lua with global symbols to allow linking
+ with other lua libraries. Found by Maxim Litnitskiy.
+
+2009-11-24 20:31 +0000 [r231134] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: 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 18:50 +0000 [r231058-231095] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: 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
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Fix problem on digital channels due to digital flag not getting
+ set Changed areas in sig_pri to set the digital flag using a
+ callback that will also set the corresponding flag in chan_dahdi.
+ Modified dahdi_request slightly so that if a bearer is marked as
+ digital, that information is available when creating the new
+ channel. (closes issue #16151) Reported by: alecdavis Patch based
+ on bug_16151.diff.txt uploaded by alecdavis (license 585)
+
+2009-11-24 13:52 +0000 [r231025] Matthew Nicholson <mnicholson@digium.com>
+
+ * CHANGES: Updated CHANGES file to describe the new 'd' option to
+ app_followme added in r230964 (related to issue #14155) Reported
+ by: junky
+
+2009-11-24 04:58 +0000 [r230994] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, funcs/func_strings.c, CHANGES: Add
+ REPLACE & PASSTHRU functions, overhaul of func_strings, fix API
+ docs for the ast_get_encoded_* functions. * Add REPLACE function,
+ which searches a given variable for a set of characters and
+ replaces each with a given character. * Add PASSTHRU function,
+ which passes a literal string back, like a NoOp for functions.
+ Intent is to be able to specify a literal string to another
+ function that takes a variable name as an argument. * Let the
+ array manipulation functions work with dialplan functions, in
+ addition to variables. This allows the array manipulation
+ functions to modify ASTDB and ODBC backends, assuming the
+ func_odbc configuration has both read and write functions.
+ (closes issue #15223) Reported by: ajohnson Patches:
+ 20091112__issue15223.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen, tilghman
+
+2009-11-23 22:37 +0000 [r230964] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_followme.c: Add an option to app_followme to disable the
+ "please hold" announcement. (closes issue #14155) Reported by:
+ junky Patches: M14555-trunk.diff uploaded by junky (license 177)
+ (modified) Tested by: junky
+
+2009-11-23 15:45 +0000 [r230881] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: 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 15:34 +0000 [r230773-230877] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: 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 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 22:35 +0000 [r230726] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: 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 21:47 +0000 [r230697] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/unaligned.h: Revert code in error and include
+ the gcc suggested workaround for the original problem, while gcc
+ investigates.
+
+2009-11-20 21:01 +0000 [r230628] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: 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 17:28 +0000 [r230584] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt, apps/app_externalivr.c: Fix/Implement error
+ events for non-existing files also include a better cmd define
+ for S command Review: https://reviewboard.asterisk.org/r/430/
+
+2009-11-20 17:26 +0000 [r230509-230583] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: audiohook signal
+ trigger on every status change (issue #14618) Review:
+ https://reviewboard.asterisk.org/r/434/
+
+ * /, apps/app_mixmonitor.c: 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/ ........
+
+2009-11-19 14:53 +0000 [r230438] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Basic cleanup of ExternalIVR: cleaned up
+ argument parsing; implemented good coding practices where
+ applicable; replaced most notice level logging with verbose
+ logging; replaced warning messages that terminated with error
+ messages; fixed memory leak identified by russellb
+
+2009-11-16 16:40 +0000 [r230343-230381] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Fix another buglet in T.38 session teardown at
+ the end of FAX sessions.
+
+ * apps/app_fax.c: Ensure that only one end of a T.38 session
+ initiates teardown at completion.
+
+2009-11-16 01:49 +0000 [r230314] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: 1. Added SIP Diversion support. 2. Fixed
+ compile warning for UUID.
+
+2009-11-15 17:23 +0000 [r230247] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_iax2.c: 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) ........
+
+2009-11-15 07:53 +0000 [r230217] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/channel.h: Increase maximum length of language
+ buffers (closes issue #16217) Reported by: dsessions
+
+2009-11-13 22:00 +0000 [r230145] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: 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 ........
+
+2009-11-13 20:42 +0000 [r230111] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, channels/chan_sip.c, apps/app_meetme.c,
+ apps/app_fax.c, configs/manager.conf.sample,
+ res/res_musiconhold.c, include/asterisk/manager.h,
+ channels/chan_iax2.c, apps/app_queue.c, CHANGES,
+ res/res_monitor.c, main/cdr.c, main/channel.c, main/manager.c,
+ main/features.c, apps/app_minivm.c, apps/app_chanspy.c,
+ apps/app_voicemail.c: Display a list of channel variables in each
+ channel-oriented event. (Closes AST-33) Reviewboard:
+ https://reviewboard.asterisk.org/r/368/
+
+2009-11-13 19:44 +0000 [r229912-230039] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /: 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 ........
+
+ * UPGRADE.txt, apps/app_chanisavail.c, CHANGES: Store the cause
+ code that is returned when trying to create a channel in
+ ChanIsAvail in the AVAILCAUSECODE dialplan variable instead of
+ overwriting the device state in AVAILSTATUS. (closes issue
+ #14426) Reported by: macli
+
+ * /: 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
+ ........
+
+ * channels/chan_sip.c: Fix T.38 negotiation regression introduced
+ with the SDP parser changes.
+
+2009-11-13 10:53 +0000 [r229819-229871] Olle Johansson <oej@edvina.net>
+
+ * main/loader.c: Fixing trunk in a way so that it compiles again.
+ Thanks, Philippe :-)
+
+ * addons/cdr_mysql.c: If CDR logging is disabled, it's considered a
+ FAILURE
+
+ * configs/modules.conf.sample, CHANGES, main/asterisk.c,
+ main/loader.c: Add the capability to require a module to be
+ loaded, or else Asterisk exits. Review:
+ https://reviewboard.asterisk.org/r/426/
+
+2009-11-13 03:16 +0000 [r229788] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Added full number portability parameter
+ support.
+
+2009-11-12 23:43 +0000 [r229750-229754] Jason Parker <jparker@digium.com>
+
+ * configs/alsa.conf.sample: Update sample config for ALSA mute and
+ noaudiocapture
+
+ * channels/chan_alsa.c: Add mute functionality. Add config option
+ to not try to open capture device. Adds "console {mute|unmute}"
+ CLI command. Adds mute and noaudiocapture config options (will
+ update sample configs shortly). (closes issue #14673) Reported
+ by: Nick_Lewis Patches: chan_alsa.c-oneway3.patch uploaded by
+ Nick Lewis (license 657) Tested by: qwell
+
+ * channels/chan_oss.c: Fix mute toggling on OSS channels.
+
+2009-11-12 16:44 +0000 [r229670] David Vossel <dvossel@digium.com>
+
+ * funcs/func_audiohookinherit.c, /: 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
+ ........
+
+2009-11-12 13:54 +0000 [r229639] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Update sip.conf.sample. Just updating a
+ spelling error and some capitalization in a documentation update
+ that Olle added. May the Swenglish be with you.
+
+2009-11-12 10:24 +0000 [r229606-229607] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Clarification
+
+ * configs/sip.conf.sample: Clarify some security issues early in
+ the sample configuration
+
+2009-11-11 20:47 +0000 [r229568] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: 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 19:48 +0000 [r229460-229499] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: 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: 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
+
+2009-11-11 14:30 +0000 [r229431] Leif Madsen <lmadsen@digium.com>
+
+ * CHANGES: Update CHANGES file. Updating the CHANGES file after
+ noticing an email on the asterisk-dev mailing list from Russell.
+ (issue #15874)
+
+2009-11-10 22:14 +0000 [r229361] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: 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 22:01 +0000 [r229356] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: 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 21:22 +0000 [r229351] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c: 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 20:06 +0000 [r229282] Joshua Colp <jcolp@digium.com>
+
+ * /, codecs/codec_g726.c: 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 17:33 +0000 [r229228] David Ruggles <thedavidfactor@gmail.com>
+
+ * /, doc/externalivr.txt: 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 17:16 +0000 [r229168] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: 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 15:53 +0000 [r229102] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Reverted revision 201717. (closes issue
+ 0016175) Reported by: paul-tg
+
+2009-11-10 15:27 +0000 [r229093] David Vossel <dvossel@digium.com>
+
+ * res/res_config_pgsql.c: 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 11:16 +0000 [r229050] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema: 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 22:50 +0000 [r229015] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c: 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 17:17 +0000 [r228979] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/iax2-parser.c: Don't try to convert a 64-bit integer,
+ where only a 32-bit integer is stored. (closes issue #16194)
+ Reported by: habile
+
+2009-11-09 16:28 +0000 [r228947] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add the
+ 'relative-periodic-announce' option to app_queue to allow for
+ calculating the time of announcments from the end of the previous
+ announcment rather than from the beginning. (closes issue #15260)
+ Reported by: tonils
+
+2009-11-09 15:38 +0000 [r228897] Leif Madsen <lmadsen@digium.com>
+
+ * main/channel.c, /: 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 14:37 +0000 [r228858] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, include/asterisk/lock.h: 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 ........
+
+2009-11-09 07:37 +0000 [r228798] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/cdr_mysql.c, main/event.c, channels/chan_console.c,
+ res/res_pktccops.c, main/loader.c: Fix various problems detected
+ with Valgrind. * chan_console accessed pvts after deallocation. *
+ cdr_mysql stored a pointer that was freed by realloc() * The
+ module loader did not check usecount on shutdown, which led to
+ chan_iax2 reading a timer that was already unloaded. * The event
+ subsystem sometimes creates an event with no IEs. Due to a corner
+ condition, the code would read beyond the memory boundary. *
+ res_pktccops did not correctly check whether its monitor thread
+ was started. (closes issue #16062) Reported by: alexanderheinz
+ Patches: 20091109__issue16062.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman
+
+2009-11-07 17:02 +0000 [r228766] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/init.d/rc.debian.asterisk: Add LSB headers to the Debian
+ init.d script See also issue #14864 .
+
+2009-11-06 22:35 +0000 [r228693] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: 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 (closes issue #16045) Reported by:
+ bluecrow76 Patches: issue16045.diff uploaded by dvossel (license
+ 671) Tested by: bluecrow76, dvossel, habile ........
+
+2009-11-06 22:32 +0000 [r228691] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, CHANGES, channels/sig_pri.c: Created
+ standard location to add options to chan_dahdi for ISDN dialing.
+ Dial(DAHDI/g1[/extension[/options]]) Current options:
+ K(<keypad_digits>) R Reverse charging indication (Collect calls)
+ The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format
+ was variable and did not allow for the easy addition of more
+ options. The earlier 'C' prefix character for reverse charge
+ indiation would conflict with the a-d DTMF digits if ISDN uses
+ them.
+
+2009-11-06 22:07 +0000 [r228661] David Brooks <dbrooks@digium.com>
+
+ * tests/test_amihooks.c: ami_testhooks.c automatically registers
+ hook ami_testhooks.c was registering for AMI events upon module
+ load. Moved the registration to its own CLI command. Added CLI
+ command for unregistering the hook. Changed some of the wording,
+ removed unnecessary arguments/parameters. Reported by: rmudgett
+
+2009-11-06 22:02 +0000 [r228658-228659] Mark Michelson <mmichelson@digium.com>
+
+ * addons/chan_ooh323.c: Make compilation of chan_ooh323 disabled by
+ default. All addons modules should be disabled by default,
+ requiring the user to turn them on if desired. After all, these
+ are addons we're talking about here.
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c: Get
+ chan_ooh323 to compile with gcc 4.2. For some reason, the code
+ compiles just fine with later versions of GCC, but this one
+ requires some weird double casting in order to get rid of all
+ warnings. Whatever.
+
+2009-11-06 19:53 +0000 [r228621] Richard Mudgett <rmudgett@digium.com>
+
+ * main/frame.c: Fix compiler warning gcc 4.2.4 found
+
+2009-11-06 19:47 +0000 [r228620] Matthew Nicholson <mnicholson@digium.com>
+
+ * funcs/func_base64.c, /, main/utils.c: 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 19:38 +0000 [r228616] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_nbs.c, addons/chan_mobile.c: Missed these two
+ channel drivers on the codec_bits merge
+
+2009-11-06 18:37 +0000 [r228499-228548] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: 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)
+ ........
+
+ * doc/tex/localchannel.tex: Fix the localchannel.tex file.
+
+2009-11-06 17:22 +0000 [r228420-228441] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_ilbc.c: Fixes merging issue from 1.4, frame data is
+ held in data.ptr in trunk
+
+ * /, codecs/codec_ilbc.c: 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 16:42 +0000 [r228410] Joshua Colp <jcolp@digium.com>
+
+ * /, main/abstract_jb.c: 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 15:42 +0000 [r228268-228339] David Vossel <dvossel@digium.com>
+
+ * /, main/astfd.c: 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: 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 22:59 +0000 [r228233] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_cdr.c: Fix XML in func_cdr.c
+
+2009-11-05 22:12 +0000 [r228191-228196] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: Yet another error message in the dialplan
+ (thanks, rmudgett/russellb)
+
+ * apps/app_meetme.c: 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 21:23 +0000 [r228189] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c: 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 19:34 +0000 [r228145] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_misdn.c, /: 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 19:16 +0000 [r228080] Jason Parker <jparker@digium.com>
+
+ * channels/chan_vpb.cc, /: 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 17:26 +0000 [r228015-228049] Tilghman Lesher <tlesher@digium.com>
+
+ * main/frame.c: Rework codecs command to comply with the 64-bit
+ scheme
+
+ * apps/app_externalivr.c: Don't crash if no arguments are passed.
+ (closes issue #16119) Reported by: thedavidfactor
+
+2009-11-04 23:50 +0000 [r227914-227945] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_monitor.c: 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 ........
+
+ * addons/chan_ooh323.c: Update chan_ooh323 to support the expanded
+ codec bitfield from 227580.
+
+2009-11-04 22:10 +0000 [r227898] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.h,
+ addons/ooh323c/src/ooCmdChannel.h, addons/chan_ooh323.c,
+ addons/ooh323c/src/printHandler.h, addons/ooh323c/src/ooq931.h,
+