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3696 Commits (9172658d422a7469f689f7aace29e3548d40aaa2)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Richard Mudgett
|
9172658d42 |
Merged revisions 318734 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r318734 | rmudgett | 2011-05-12 20:09:40 -0500 (Thu, 12 May 2011) | 43 lines Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 * The applicable fixes for v1.4 are the SIP deadlock and the in progress masquerade check for multiple parties trying to pickup the same call. issue18654_v1.4.patch uploaded by rmudgett (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch uploaded by rmudgett (license 664) ........ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Terry Wilson
|
30e8e7dedb |
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/1.6.2@317575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Sean Bright
|
ad0b87f72b |
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/1.6.2@316708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
David Vossel
|
1ef5452cd8 |
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/1.6.2@316644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Sean Bright
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1e2ce928a5 |
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/1.6.2@316475 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Terry Wilson
|
445ed7b6ee |
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/1.6.2@315643 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Leif Madsen
|
370816bdac |
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/1.6.2@314202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Alec L Davis
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360614263c |
app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICE
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@313004 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jonathan Rose
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f24755ffd7 |
Backporting trunk change to add verbosity to 'L' option in meetme
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@312762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Alec L Davis
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77b3624bf4 |
fix up bad merge46 extra 2 yuck: labels
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@312213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Alec L Davis
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c1063ef5dc |
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/1.6.2@312210 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Alec L Davis
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26af32c516 |
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/1.6.2@312103 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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David Vossel
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c781e46225 |
Fixes memory leak in MeetMe AMI action
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@311496 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Tilghman Lesher
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40d0500d80 |
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/1.6.2@310141 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Jonathan Rose
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eab4ccbd5b |
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/1.6.2@309857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
David Ruggles
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305b0f87d4 |
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/1.6.2@309356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Leif Madsen
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16487a72c4 |
Update PickupChan documentation.
The PickupChan uses the ampersand as the argument separator. (closes issue #18905) Reported by: vmikhnevych Tested by: vmikhnevych git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@309348 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Terry Wilson
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29469758df |
Merge missing bugfix for issue #11583
This is the combination of two commits that made it into 1.4, 1.6.0, 1.6.1, and trunk (and therefor 1.8) but that was missed for 1.6.2. ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@308677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jason Parker
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a98d65b5db |
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/1.6.2@308007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jeff Peeler
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d5fef6ab68 |
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/1.6.2@306966 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jeff Peeler
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06bdfcedd9 |
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/1.6.2@306961 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jeff Peeler
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de3f1d266b |
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/1.6.2@306865 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jason Parker
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3d510c9a32 |
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/1.6.2@306346 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Richard Mudgett
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050a3c39ec |
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/1.6.2@305889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jason Parker
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1dbde4531e |
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/1.6.2@305253 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
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01ad35cc4a |
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/1.6.2@304978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Sean Bright
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da82320fdd |
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/1.6.2@304776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Sean Bright
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1b79e47b27 |
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/1.6.2@304773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Sean Bright
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4ed3d869d9 |
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/1.6.2@304729 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Sean Bright
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ab0f37345e |
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/1.6.2@304726 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Sean Bright
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dc2a91d357 |
Revert part of the previous commit that snuck in.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Sean Bright
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b1cf1879bc |
Don't leak references if we can't create a pseudo channel for mixing in MeetMe.
If there was a problem allocating a pseudo channel when building our meetme, we weren't destroying our user container or destroying the mutexes that we created. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304659 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jeff Peeler
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5369a22c18 |
Merged revisions 303676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines Fix voicemail sequencing for file based storage. A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Russell Bryant
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0f8989a993 |
Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Leif Madsen
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01dfc39eb4 |
Fix changes to L() flag in Dial().
Tony Mountifield pointed out an error I had in my patch. I was a bit too aggressive on changing 'seconds' to 'milliseconds'. So I decided to do some additioanl testing and have no changed just the appropriate lines. One line says milliseconds, and the other says seconds. Probably should change this to be either just seconds or milliseconds, but I've spent too much time on this already :) (issue #18264) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303273 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jeff Peeler
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347d4b4bdb |
Merged revisions 303007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303008 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
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9aee0b5e0b |
Resolve a compiler warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Leif Madsen
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094efca065 |
Option L() is milliseconds, not seconds.
> Change the verbose output of option L() to say milliseconds and not seconds > as the value is in milliseconds. > > (closes issue #18264) > Reported by: jacco > Patches: > app_dial_patch.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Sean Bright
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39a167a12a |
Support greetingsfolder as documented in voicemail.conf.sample.
(closes issue #17870) Reported by: edhorton Patches: __20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Paul Belanger
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5e7a60d065 |
Indicate log level argument for Log() is not optional
(closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301176 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jason Parker
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15bf424dcb |
Initialize useropts/adminopts in case there is no column in the realtime DB.
(closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jeff Peeler
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1a3a5b1694 |
Fix regression causing forwarding voicemails to not work with file storage.
I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jeff Peeler
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5068f61abd |
Merged revisions 300918 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played at the correct time. Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Paul Belanger
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1aa72d01e0 |
Documentation typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jeff Peeler
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41ffe44640 |
Merged revisions 298683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 Dec 2010) | 2 lines After recording only silence for a voicemail prepending, restore backup files. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@298684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jeff Peeler
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43c0271f4d |
Merged revisions 298596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines Fix improper hangup when doing an attended transfer to queue. Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@298597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
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07fb2027c1 |
Merged revisions 297689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@297713 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
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2b8d1baccf |
Merged revisions 297228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines Add "DAHDI" to a couple of app_meetme error messages. This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@297229 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jeff Peeler
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97e335005a |
Merged revisions 296868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) | 4 lines Properly restore backup information file when hanging up during message prepending. ABE-2654 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@296869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
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130924af07 |
18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision).
(closes issue #18369) Reported by: tnakonz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@296466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |