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3891 Commits (0219829eef97b7faadd9b73b20a5278ac6ce9dd1)
Author | SHA1 | Message | Date |
---|---|---|---|
|
249f4b9022 |
Merged revisions 319528 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines Merged revisions 319527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines Fix app_dial ring groups Revert part of r315643. We need to remove the datastore here as well. The code in bridging code will catch anything that app_dial might miss. (closes issue #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff uploaded by elguero (license 37) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
c23377d8f2 |
Don't create [general] voicemail context when using users.conf
Prior to this patch, app_voicemail would create a [general] context when parsing users.conf. (closes issue #18891) Reported by: pdugas Patches: app_voicemail-ignore-general.patch uploaded by pdugas (license 1222) app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71) Tested by: pdugas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319367 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
87d80af96c |
Fix directed group pickup feature code *8 with pickupsounds enabled
Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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10e40660c7 |
Use the right variable to print the time in a debug message.
The original patch also increased some buffer sizes, but that was already done in this version. (closes issue #17034) Reported by: sysreq Patches: asterisk-issue-17034.patch uploaded by sysreq (license 1009) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317969 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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91c8e4d297 |
Fix some more "set but unused" compiler warnings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6cf3280dd6 |
Merged revisions 317575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines Merged revisions 317574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines Re-fix queue round-robin This part of the change for r315596 was incorrect. No bridge occurs when doing a roundrobin dial and no one answers, so this code shouldn't have been removed. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317584 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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bbf748d856 |
Fix potential memory leak, and use of uninitialized memory.
(closes issue #16476) Reported by: junky Patches: M16476.diff uploaded by junky (license 177) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317427 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a3d1ff1140 |
Increase buffer size to be PATH_MAX for a path.
(closes issue #19239) Reported by: byronclark Patches: queue_announce_length.patch uploaded by byronclark (license 1200) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317336 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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63579a892c |
Wait for leader with Music On Hold allows crosstalk between participants.
Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits. (closes issue #18418) Reported by: MrHanMan Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6c3ea80a35 |
Merged revisions 316708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316708 | seanbright | 2011-05-04 12:10:59 -0400 (Wed, 04 May 2011) | 15 lines Merged revisions 316707 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May 2011) | 8 lines If sox fails when processing a voicemail, don't delete the original file. (closes issue #18111) Reported by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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eaf8673a16 |
Merged revisions 316644 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011) | 9 lines Fixes one-way-audio when chanspy activated with the 'o' option (closes issue #18382) Reported by: jkister Patches: 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt uploaded by malin (license ) Tested by: firstsip, Greenlightcrm, malin, wdoekes, boroda, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316650 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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aa43b12c24 |
Merged revisions 316475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May 2011) | 10 lines Honor the C option to MeetMe when L is passed. This fixes a case that r304773 and friends missed. (closes issue #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff uploaded by var (license 1227) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316476 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1a8df4dc53 |
Resolve another warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a82f1bb995 |
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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734ca12381 |
Merged revisions 315643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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06223e643b |
Add missing set of name valid flag when dialing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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db02ef3704 |
Merged revisions 314202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011) | 7 lines Update seconds to milliseconds in ast_verb output. (closes issue #19084) Reported by: smurfix Patches: app_dial.patch uploaded by smurfix (license 547) Tested by: lmadsen, smurfix ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314203 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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bc620cd281 |
Unclear code in app_dial.c.
Make code formatting clear. (closes issue #19134) Reported by: oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314068 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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42882cd3bc |
Bring the dumpchan application inline with "core show channel".
* Added fields that are in "core show channel" to dumpchan output. * Fixed reuse of formatbuf before the previous string stored there was used by snprintf. All output strings now have their own buffer. * Adjusted the buffer sizes to not be so abusive of the stack now that there are more buffers. Change requested by oej. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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dde33a1e01 |
Frames from the inbound channel should go to all outbound channels in app_dial.c.
In app_dial.c:wait_for_answer() frames from the inbound channel should be sent to all outbound channels instead of only if there is just one outbound channel. Control frames like AST_CONTROL_CONNECTED_LINE need to be passed to all of the the outbound channels. This can happen if a blond transfer is done by a remote switch on the inbound channel. JIRA AST-443 JIRA SWP-2730 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313369 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6dc376082d |
Backport a restructuring change from trunk to make the next change stand out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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8fe6967f1d |
app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICE
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313002 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f6f5340777 |
Merged revisions 312762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r312762 | jrose | 2011-04-05 09:11:36 -0500 (Tue, 05 Apr 2011) | 1 line Backporting trunk change to add verbosity to 'L' option in meetme ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312765 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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62e679f784 |
Merged revisions 312210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines Merged revisions 312174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines voicemail: get real last_message_index and count_messages, ODBC resequence change last_message_index to read the max msgnum stored in the database change count_messages to actually count the number of messages. last_message_index change: This fixed overwriting of the last message if msgnum=0 was missing. Previously every incoming message would overwrite msgnum=1. count_messages change: allows us to detect when requencing is required in opneA_mailbox. resequence enabled for ODBC storage: Assists with fixing up corrupt databases with gaps, but only when a user actively opens there mailboxes. (closes issue #18692,#18582,#19032) Reported by: elguero Patches: based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37) Tested by: elguero, nivek, alecdavis Review: https://reviewboard.asterisk.org/r/1153/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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83aeb52dd0 |
Merged revisions 312103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines Merged revisions 312070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines app_voicemail: close_mailbox needs to respect additional messages while mailbox is open. close_mailbox leave gaps in message sequence if messages are deleted and new messages arrive during this time, this is because the shuffle down to slot 0, only shuffles the number of pre-existing messages when mailbox is opened, ignoring new arrivals. Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals. Happens on filebased or ODBC storage. (issues #19032,#18582,#18692,#18998) Reported by: alecdavis,tootai,afosorio Review: https://reviewboard.asterisk.org/r/1153/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312117 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0a186e3f4f |
Cross-reference VoiceMail() and VoiceMailMain() in the xml docs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311751 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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51ce432d07 |
This patch fixes a bug with MeetMe behavior where the 'P' option for always
prompting for a pin is ignored for the first caller. (closes issue #18070) Reported by: mav3rick Review: https://reviewboard.asterisk.org/r/1132/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a00e99ec56 |
Merged revisions 311496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011) | 2 lines Fixes memory leak in MeetMe AMI action ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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93601856b6 |
Merged revision 310986 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines Dial() o option broke when connected line feature added. The patch restores the o option behavior and adds the ability to specify the CallerID. The Dial o and f options are complementary to each other. The o option stores the CallerID on the outgoing channel as the channel's CallerID. The f option forces the CallerID sent by the outgoing channel. o(x) - The argument 'x' is optional. If not present, then specify that the CallerID that was present on the *calling* channel be stored as the CallerID on the *called* channel. This was the behavior of Asterisk 1.0 and earlier. If present, then specify the CallerID stored on the *called* channel. Note that o(${CALLERID(all)}) is similar to option o without parameters. f(x) - The argument 'x' is optional and its presence changes the behavior of this option. If not present, then force the outgoing CallerID on a call-forward or deflection to the dialplan extension for this Dial() using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If present, then force the outgoing CallerID to 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA SWP-3096 .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ef01ba5ff2 |
This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy. (closes issue #18742) Reported by: jkister Tested by: jkister, jcovert, jrose Review: http://reviewboard.digium.internal/r/106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311197 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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15641c348e |
Merged revisions 310141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. (closes issue #18295) Reported by: pruiz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310142 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4ad0ddf5e3 |
Merged revisions 309857 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309857 | jrose | 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | 8 lines Bug fix for MixMonitor involving filenames with '.' not in the extension Closes issue #18391) Reported by: pabelanger Patches: bugfix.patch uploaded by jrose (license 1225) Tested by: jrose ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d5e1774082 |
Merged revisions 309356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines fix small memory leak fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309403 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c8ef3e081b |
Merged revisions 308007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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227c620866 |
Don't crash when forcing caller id.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ff43beaa2d |
Calling a gosub routine defined in AEL from Dial/Queue ceased to work.
A bug in AEL did not distinguish between the "s" extension generated by AEL and an "s" extension that was required to exist by the chan_dahdi (or another channel) that was not supplied with a starting extension. Therefore, AEL made incorrect assumptions about what commands were permissable in the context. This was fixed by making AEL generate a different extension name. However, Dial and Queue make additional assumptions about the name of the default gosub extension. Therefore, they needed to be brought into line with a "macro" rendered by AEL (as a gosub), without breaking traditional dialplans written without the aid of AEL. Related to (issue #18480) Reported by: nivek (closes issue #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt uploaded by tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) Tested by: kkm git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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49c4800686 |
Merged revisions 306966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) | 1 line fix this line again ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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dad67ad1a4 |
Merged revisions 306961 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines Backup file storing message duration is not used with IMAP_STORAGE, remove code. The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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59502582b3 |
Merged revisions 306865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) | 1 line make this safer and fully correct, pointed out by Steve Davis ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e8bd6696b5 |
Merged revisions 306346 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines Don't fallthrough to 'unknown' in the 'ringing' case. This could cause improper exits from the queue. (closes issue #18499) Reported by: zaltar Patches: app_queue.patch uploaded by zaltar (license 1148) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cee1db213b |
Don't send redirecting updates to the caller if the dialplan forked the call.
Each fork in the dial could be redirected and confuse the caller. For ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN redirects calls in sequence not in parallel. * Also fixed a formatting inconsistency in app_dial.c and make a warning message more useful about what frame type could not be written. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306324 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a785544090 |
Merged revisions 305889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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69e83f1a72 |
Replacing doc/* and asterisk.pdf with wiki links
Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1a5122534c |
Merged revisions 305253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305254 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b27fc05f06 |
Merged revisions 304978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600 (Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31 Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304985 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b7d7fc94c2 |
Add Function and Application Relationships to documentation
Add and extend the see-also sections to the documentation for applications and functions in an effort to expand the online documentation of the wiki. Also check for and update any links to moved documentation in the doc folder. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b0c9f29c72 |
Merged revisions 304776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan 2011) | 15 lines If we fail to allocate our announcement objects, make sure we don't leak objects. The majority of this patch was committed already in r304726 and r304729. (issue #18225) Reported by: kenji (issue #18444) Reported by: junky (closes issue #18343) Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz (license 834) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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07bbfff4eb |
Merged revisions 304773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan 2011) | 9 lines When we pass the S() or L() options to MeetMe, make sure that we honor C as well. Without this patch, if the user was kicked from the conference via the S() or L() mechanism, we would just hang up on them even if we also passed C (continue in dialplan when kicked). With this patch we honor the C flag in those cases. (closes issue #17317) Reported by: var ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304774 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4ba774c116 |
Merged revisions 304729 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan 2011) | 15 lines Make sure that we unref the correct object when ejecting the most recent caller. Currently, when we kick the last user to enter, we decrement our own reference count which results in a crash when we kick another user or when we exit the conference ourselves. This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in 1.6.2. (closes issue #18225) Reported by: kenji Patches: issue18225.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304730 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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05116e68f4 |
Merged revisions 304726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan 2011) | 9 lines Fix user reference leak in MeetMe. We were unlinking the user from the conferences user container, but not decrementing the reference count of the user as well, resulting in a leak. (closes issue #18444) Reported by: junky Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304727 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |