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${ noResults }
21608 Commits (bac5a51e2121a63549941f3a9ce7f7a4c50454ca)
| Author | SHA1 | Message | Date |
|---|---|---|---|
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|
bac5a51e21 |
Merged revisions 332830 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332830 | rmudgett | 2011-08-22 13:32:09 -0500 (Mon, 22 Aug 2011) | 15 lines Merged revisions 332816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011) | 8 lines Memory leaks in realtime_multi_xxx() when database access returns error. * Fix realtime_multi_pgsql() configuration memory leak when the database access returns an error. * Fix realtime_multi_odbc() configuration category use after free when the database access returns an error. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
|
b8748f4c00 |
Merged revisions 332761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332761 | rmudgett | 2011-08-22 12:05:35 -0500 (Mon, 22 Aug 2011) | 22 lines Merged revisions 332759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011) | 15 lines Memory leak reading realtime database variable list. Calling ast_load_realtime() can leak the last list node if the read list only contains empty variable value items. * Fixed list filter loop in ast_load_realtime() to delete the list node immediately instead of the next time through the loop. The next time through the loop may not happen if the node to delete is the last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265) Patches: jira_asterisk_18265_v1.8_config.patch (license #5621) patch uploaded by rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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901e275c4c |
Add option for logging congested calls as CONGESTION instead of NO_ANSWER in CDR
This patch adds a CDR option to cdr.conf that will allow CDR files to log calls ending with congestion in a way that is unique from other unanswered calls. (closes issue ASTERISK-14842) Reported by: Alec Davis Patches: cdr_congestion.diff.txt (License #5546) patch uploaded by Alec Davis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332760 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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91d3a7d3a1 |
Merged revisions 332756 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332756 | mnicholson | 2011-08-22 11:29:45 -0500 (Mon, 22 Aug 2011) | 4 lines add a way to disable and/or modify the gateway timeout ASTERISK-18219 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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3a50145436 |
Merged revisions 332700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332700 | pabelanger | 2011-08-21 10:33:23 -0400 (Sun, 21 Aug 2011) | 12 lines Merged revisions 332699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug 2011) | 5 lines Fix outgoing calls in chan_gtalk (closes issue ASTERISK-18301) Reported by: az1324 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332701 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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3e89d62884 |
Merged revisions 332654 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332654 | kmoore | 2011-08-19 14:59:34 -0500 (Fri, 19 Aug 2011) | 8 lines Make CONFBRIDGE_INFO behave more nicely CONFBRIDGE_INFO doesn't behave as well in edge cases as MEETME_INFO. With this patch, CONFBRIDGE_INFO should behave in a much more reasonable manner when presented with invalid conferences and keywords. Review: https://reviewboard.asterisk.org/r/1359/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0f92716dbb |
Fix infinite loop releasing the same memory in ldap_loadentry().
* Fixed memory leak of vars in ldap_loadentry(). * Fixed potential NULL ptr dereference of vars in ldap_loadentry(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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d2af16a86c |
Merged revisions 332560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332560 | twilson | 2011-08-18 16:34:04 -0500 (Thu, 18 Aug 2011) | 12 lines Merged revisions 332559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 5 lines Fix possible error on stringification of IPv4-mapped addrs The FreeBSD netsock2 test has been failing for a while. We were pasing sa->len to getnameinfo instead of sa_tmp->len. ASTERISK-18289 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332561 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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75bb8797f5 |
Merged revisions 332504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332504 | kmoore | 2011-08-18 14:29:15 -0500 (Thu, 18 Aug 2011) | 15 lines Merged revisions 332503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) | 8 lines CRC4 in "dahdi show status" gives wrong impression to T1 users Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in more situations without confusing users, especially since T1 lines use CRC6 instead of CRC4. (closes issue AST-471) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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ee277c8891 |
Merged revisions 332447 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332447 | tilghman | 2011-08-18 09:48:40 -0500 (Thu, 18 Aug 2011) | 9 lines Merged revisions 332446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18 Aug 2011) | 2 lines Move BETTER_BACKTRACES out of development mode, as it's useful when DEBUG_THREADS is enabled. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332448 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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318f0f5514 |
Merged revisions 332369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332369 | tilghman | 2011-08-17 14:24:59 -0500 (Wed, 17 Aug 2011) | 17 lines Merged revisions 332355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 10 lines Re-add support for spaces in pathnames, including now spaces in DESTDIR. This was initially added to 1.8 prior to release, primarily to support the standard paths on Mac OS X, but was partially reverted recently in Subversion, due to the lack of support for spaces in DESTDIR. This commit restores support for the standard paths on Mac OS X, and also includes support for spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by: pabelanger Review: https://reviewboard.asterisk.org/r/1326/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332388 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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c38cb95863 |
Merged revisions 332321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332321 | twilson | 2011-08-17 13:09:49 -0500 (Wed, 17 Aug 2011) | 17 lines Merged revisions 332320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011) | 10 lines Don't read from a disarmed or invalid timerfd Numerous isues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. This patch adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18166, ASTERISK-18197, AST-486, AST-495, AST-507 and possibly others. Review: https://reviewboard.asterisk.org/r/1361/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332337 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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265102faf8 |
Merged revisions 332265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332265 | rmudgett | 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) | 33 lines Merged revisions 332264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) | 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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606ca8ee8c |
Merged revisions 332177 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332177 | pabelanger | 2011-08-16 16:11:49 -0400 (Tue, 16 Aug 2011) | 11 lines Merged revisions 332176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug 2011) | 4 lines Flag test modules as 'core' Review: https://reviewboard.asterisk.org/r/1369/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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269082f035 |
Merged revisions 332119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332119 | jrose | 2011-08-16 12:45:38 -0500 (Tue, 16 Aug 2011) | 23 lines Merged revisions 332118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) | 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox setting in sip.conf would only result in updates being sent on whichever mailbox triggered the mwi event. Now all of them get counted regardless. Also fixes a bug involving parsing of the mailbox option in sip.conf so that trailing and leading spaces before/after commas are trimmed. (closes issue ASTERISK-18067) Reported by: aragon (closes issue ASTERISK-15479) Reported by: Ben Winslow Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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3ad6dccac8 |
Merged revisions 332101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332101 | rmudgett | 2011-08-16 12:17:28 -0500 (Tue, 16 Aug 2011) | 140 lines Merged revisions 332100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332117 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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ea5262eeb5 |
Merged revisions 332042 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332042 | mnicholson | 2011-08-16 10:20:48 -0500 (Tue, 16 Aug 2011) | 2 lines fix a code comment AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332044 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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052ece39ee |
Merged revisions 332029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332029 | mnicholson | 2011-08-16 10:17:16 -0500 (Tue, 16 Aug 2011) | 2 lines Moved notes about 'storesipcause' to UPGRADE.txt from CHANGES AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332033 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 332027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332027 | mnicholson | 2011-08-16 10:08:40 -0500 (Tue, 16 Aug 2011) | 9 lines Merged revisions 332026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4403a89f50 |
Formatting changes while working with DTMF...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 332022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines In 10 and trunk this option is disabled by default. Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331956 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331956 | rmudgett | 2011-08-15 12:35:03 -0500 (Mon, 15 Aug 2011) | 20 lines Merged revisions 331955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) | 13 lines Fix some minor chan_dahdi config load issues. * Address chan_dahdi.conf dahdichan option todo item about needing line number. * Make ignore_failed_channels option also apply to dahdichan option. * Don't attempt to create a default pseudo channel if the chan_dahdi.conf channel/channels option is not allowed. * Add a similar check for dahdichan in normal chan_dahdi.conf sections as is done in users.conf. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331894 | pabelanger | 2011-08-15 11:22:45 -0400 (Mon, 15 Aug 2011) | 12 lines Merged revisions 331886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug 2011) | 5 lines Fix noisy message when briding channels (closes issue ASTERISK-18270) Reported by: Federico Alves ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331868 | dvossel | 2011-08-15 10:14:13 -0500 (Mon, 15 Aug 2011) | 12 lines Merged revisions 331867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) | 6 lines Fixes locking inversion issues present in the handling of the sip REFER method. (closes issue ASTERISK-18082) Reported by: James Van Vleet ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Formatting guideline fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331775 | mnicholson | 2011-08-12 14:03:31 -0500 (Fri, 12 Aug 2011) | 17 lines Merged revisions 331774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug 2011) | 11 lines Unlock the channel before calling update_queue. Holding the channel lock when calling update_queue which attempts to lock the queue lock can cause a deadlock. This deadlock involves the following chain: 1. hold chan lock -> wait queue lock 2. hold queue lock -> wait agent list lock 3. hold agent list lock -> wait chan list lock 4. hold chan list lock -> wait chan lock ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331772 | rmudgett | 2011-08-12 13:59:45 -0500 (Fri, 12 Aug 2011) | 15 lines Merged revisions 331771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) | 8 lines Suppress warning message when using DAHDITransfer or DAHDIHangup. * The fake event should only be processed by the channel that currently owns the private and not the associated call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331644 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331644 | jrose | 2011-08-12 11:18:57 -0500 (Fri, 12 Aug 2011) | 9 lines Merged revisions 331635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) | 1 line Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331715 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331715 | rmudgett | 2011-08-12 12:54:47 -0500 (Fri, 12 Aug 2011) | 29 lines Merged revisions 331714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) | 22 lines AMI actions DAHDIHangup and DAHDITransfer have no effect. The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI channel. These two AMI actions are highly specialized to analog channels and appear to make the channel behave like a jack port for headsets. * Made the faked DAHDI event get processed before a normal media stream read in dahdi_read() instead of trying to trigger an exception read by setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long ago that changed how AST_FLAG_EXCEPTION is processed in the core. Unfortunately, the faked DAHDI events no longer worked when that happened. * Updated the DAHDI AMI action documentation for the following actions: DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and DAHDIRestart. * Made use sscanf() instead of atoi() for better error checking of the DAHDIChannel header string. JIRA AST-620 JIRA SWP-3616 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331716 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331659 | twilson | 2011-08-12 11:31:21 -0500 (Fri, 12 Aug 2011) | 11 lines Merged revisions 331658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12 Aug 2011) | 4 lines Fix netsock2 multiple zero-expansion test Remove erroneous single bracket. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331660 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331654 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331654 | kmoore | 2011-08-12 11:21:37 -0500 (Fri, 12 Aug 2011) | 19 lines Merged revisions 331649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) | 12 lines Logger does not warn of failure to open logging channels Currently, logger only prints an error message to stderr when it fails to open a logger channel where many users will not see it because the logger lock is held. The alternative provided by this patch is to log the error to all attached consoles in the hopes that it will be easier to see. Additionally, this patch prevents the failed logger channel from being added to the list where it would silently fail on each call to the Asterisk logger. (closes issue ASTERISK-16231) Review: https://reviewboard.asterisk.org/r/1338 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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1a8069abe2 |
Merged revisions 331579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331579 | qwell | 2011-08-11 16:54:54 -0500 (Thu, 11 Aug 2011) | 13 lines Merged revisions 331578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) | 6 lines Use proper values for 64-bit option flags. Also, reusing bits es no bueno, so change the value of a duplicate. (issue ASTERISK-18239) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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e12184cf95 |
Merged revisions 331576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331576 | rmudgett | 2011-08-11 16:42:21 -0500 (Thu, 11 Aug 2011) | 16 lines Merged revisions 331575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines Segfault in shell_helper in func_shell.c. The return value of popen() was not checked for failure to open. (closes issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael Myles Tested by: rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331577 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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a6ea606a78 |
Merged revisions 331518 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331518 | kmoore | 2011-08-10 17:23:49 -0500 (Wed, 10 Aug 2011) | 17 lines Merged revisions 331517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed. (closes issue ASTERISK-18091) Review: https://reviewboard.asterisk.org/r/1332/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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9d785ca5f3 |
Merged revisions 331462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331462 | rmudgett | 2011-08-10 15:41:35 -0500 (Wed, 10 Aug 2011) | 37 lines Merged revisions 331461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011) | 30 lines Output of queue log not started until logger reloaded. ASTERISK-15863 caused a regression with queue logging. The output of the queue log is not started until the logger configuration is reloaded. * Queue log initialization is completely delayed until the first message is posted to the queue log system. Including the initial opening of the queue log file. * Fixed rotate_file() ROTATE strategy to give the file just rotated out to the configured exec function after rotate. Just like the other strategies. * Fixed logger reload to always post the queue reload entry instead of just if there is a queue log file. * Refactored some code to eliminate some redundancy and to reduce stack utilization. (closes issue ASTERISK-17036) JIRA SWP-2952 Reported by: Juan Carlos Valero Patches: jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett (closes issue ASTERISK-18208) Reported by: Christian Pinedo Review: https://reviewboard.asterisk.org/r/1333/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331420 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331420 | rmudgett | 2011-08-10 14:07:53 -0500 (Wed, 10 Aug 2011) | 2 lines Make sure feature_request_and_dial() initializes outstate if passed in. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331421 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331418 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011) | 6 lines Revert -r318141. It was a band-aid that only partially fixed parking. A better fix is on reviewboard review 1358. (issue ASTERISK-17374) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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dc9513a69d |
SIP display-name needed to be empty for Avaya IP500
In order to address a compatability issue with certain features on certain devices which rely on display name content to change behavior, initreqprep in chan_sip.c has been changed to no longer substitute cid_number into the display name when cid_name isn't present. Instead, it will send no display name in that case. (closes issue ASTERISK-16198) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1341/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331371 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331316 | kmoore | 2011-08-10 08:48:41 -0500 (Wed, 10 Aug 2011) | 15 lines Merged revisions 331315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) | 8 lines AMI action ModuleReload returns Error if Module: missing or empty An empty string was not being checked for properly causing identification of the module to be reloaded to fail and return an Error with message "No such module." (closes issue AST-616) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331265 | rmudgett | 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) | 22 lines Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331147,331200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331147 | may | 2011-08-09 20:16:55 +0400 (Tue, 09 Aug 2011) | 11 lines Merged revisions 331146 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4 lines move ast_cond_signal for admitted call after all data filled/freed clear all log channels by pointed number not only first free allocated callToken in ooh323_answer ........ ................ r331200 | may | 2011-08-09 20:36:39 +0400 (Tue, 09 Aug 2011) | 9 lines Setup IP proto version for call in GK mode Added additional check for IP semantics before parse destination by ast_parse_args due to it can parse numeric as IP. (closes issue ASTERISK-18218) Reported by: slesru Patch: ASTERISK-18218.patch ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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c3bd5892a6 |
Allow ENUM query functions to report lookup errors
The ENUM dialplan functions do not report DNS query errors properly. It is useful to differentiate between failed query (e.g. non-existent domain) vs. no data records of the appropriate type. This is required to make overlapped dialing work. (closes issue ASTERISK-13769) Review: https://reviewboard.asterisk.org/r/1355/ Patch-by: Timo Teras git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331143 | qwell | 2011-08-09 10:59:54 -0500 (Tue, 09 Aug 2011) | 9 lines Merged revisions 331142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug 2011) | 1 line Regenerate asterisk man page from sgml. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331139 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331139 | qwell | 2011-08-09 10:50:07 -0500 (Tue, 09 Aug 2011) | 19 lines Merged revisions 306999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) | 12 lines Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331141 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) | 1 line Revert merge of r306999, due to merge conflict. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331140 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331097 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) | 5 lines Bump the AMI protocol version to 1.2 As a result of converting Unlink events that were missed in the AMI 1.1 update to Bridge events, the AMI protocol version is being incremented. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331041 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) | 6 lines Replace AMI Unlink events with Bridge events A previous update converted some of the Link and Unlink events to Bridge events, but a couple of Unlink events were missed. This patch rectifies the situation. (closes issues ASTERISK-17455) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 331039 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331039 | kmoore | 2011-08-08 15:53:30 -0500 (Mon, 08 Aug 2011) | 18 lines Merged revisions 331038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) | 11 lines In-queue MOH stops after a periodic announcement If the seek value is past the end of file when resuming G.722 MOH, MOH will cease to function for the duration of the MOH session through all starts and stops until saved state is cleared. Adjusting the code to guarantee a single valid read (which is already assumed) fixes the bug. (closes issue ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/ Tested-by: Jonathan Rose <jrose@digium.com> ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331040 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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0f5ef2c781 |
Log queue member name when state_interface is set for ADDMEMBER and REMOVEMEMBER events
app_queue logs the events ADDMEMBER and REMOVEMEMBER with the agent field set to the interface value rather than the membername value when a member is added with a state_interface value set. However all other member related queue events are logged with the membername when a state_interface is set. This patch makes these fields optionally more consistent and correct. (closes issue ASTERISK-14769) Review: https://reviewboard.asterisk.org/r/1286 Patch-by: Jamuel Starkey Tested-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331037 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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d1a0938c99 |
app_queue: Add StateInterface to output of "queue show" and "QueueStatus"
This patch adds the state_interface of the queue member struct to the output of "queue show" (CLI command) and "QueueStatus" (AMI action) when displaying relevant queue member information. For the AMI event message the variable StateInterface has been added. (closes issue ASTERISK-18071) Review: https://reviewboard.asterisk.org/r/1300/ Patch-by: Jamuel Starkey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331000 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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