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2011-07-21Merged revisions 329200 via svnmerge from rmudgett1-3/+5
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329200 | rmudgett | 2011-07-21 12:32:02 -0500 (Thu, 21 Jul 2011) | 24 lines Merged revisions 329199 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines Update PickupChan documentation. The PickupChan uses the ampersand as the argument separator. Was documented as: PickupChan(channel[,channel2[,...][,options]]) Fixed documentation to: PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) This is a continuation of ASTERISK-17494 for v1.8 and later. (closes issue ASTERISK-18144) Reported by: Erik Smith Patches: pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith Tested by: Erik Smith ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@329201 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-19Merged revisions 328771 via svnmerge from kmoore1-9/+7
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines Merged revisions 328770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines MeetMe requests a PIN twice in some circumstances If a call to MeetMe includes both the dynamic(D) and always request PIN(P) options, MeetMe will ask for the PIN two times: once for creating the conference and once for entering the conference. This behavior was introduced in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN entry for joining a conference. (closes AST-601) Review: https://reviewboard.asterisk.org/r/1305/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328772 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-18Merged revisions 328664 via svnmerge from markm1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines Merged revisions 328663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines app_dial may double free a channel datastore When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it. (closes issue ASTERISK-17917) Reported by: Mark Murawski Tested by: Mark Murawski ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328665 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-15Merged revisions 328451 via svnmerge from lmadsen1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) | 1 line Build app_macro by default because things depend on it. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328459 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-15Merged revisions 328329 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines Make hint watcher callback take const strings for context and exten parameters. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328344 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-14Merged revisions 328247 via svnmerge from lmadsen75-3/+254
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-13Merged revisions 328120 via svnmerge from dvossel1-5/+19
https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines Preserve sample rate quality of wideband mixmonitor recordings. MixMonitor has the ability to record in any file format Asterisk supports, but the quality of wideband audio is not preserved. This is because regardless of the sample rate the call is being recorded in, the audio is always downsampled to 8khz and then upsampled to whatever wideband format it is being written as. This patch resolves this by requesting the audio from the audiohook in the signed linear format closest to the sample rate of the format we are writing. This fix is only possible for Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband audio. Review: https://reviewboard.asterisk.org/r/1314/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328121 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-12Merged revisions 327890 via svnmerge from mnicholson1-4/+5
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue, 12 Jul 2011) | 2 lines search in the current context for 'a' and 'o' instead of 'default' ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327891 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-12Merged revisions 327852 via svnmerge from mjordan1-2/+30
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) | 12 lines Added additional checks for mailbox / password beginning with '*' character A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character. (closes issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327856 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-12Segfault on shutdown when confbridge is activekmoore1-0/+7
When undergoing a shutdown and channels are kicked out of a bridge, a segfault occurs because ConfBridge tries to play sounds on the bridge after the underlying channels have been blown away due to the shutdown. (closes ASTERISK-18040) Review: https://reviewboard.asterisk.org/r/1283/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327748 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Updates confbridge.conf video documentation and adds dtmf action for ↵dvossel3-14/+32
releasing video src. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326782 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-05Merged revisions 326411 via svnmerge from tilghman3-5/+5
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326412 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Video support for ConfBridge.dvossel4-23/+177
Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325931 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Merged revisions 325877 via svnmerge from mjordan1-3/+21
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30 Jun 2011) | 9 lines Patched voicemail user option for emailbody / emailsubject Incorporated changes per ASTERISK-16795; updated unit tests to check for vmu->emailbody / vmu->emailsubject (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325900 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325614 via svnmerge from rmudgett1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325614 | rmudgett | 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011) | 5 lines Fixed some error exit cleanup in app_queue.c. * Fixed error exit cleanup in app_queue.c copy_rules() and reload_queue_rules(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325616 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325610 via svnmerge from rmudgett1-1/+10
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) | 18 lines Response to QueueRule manager command does not contain ActionID if it was specified. * Add ActionID support as documented for the QueueRule AMI action. * Remove documentation for ActionID with the Queues AMI action. The output does not follow normal AMI response output and there is no place to put an ActionID header. (closes issue AST-602) Reported by: Vlad Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325611 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325537 via svnmerge from mnicholson1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines don't do native/remote bridging if a framehook is active on the channel ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325538 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Commit "distrotech" app_queue changes to Trunkirroot1-94/+266
* Added general option negative_penalty_invalid default off. when set members are seen as invalid/logged out when there penalty is negative. for realtime members when set remove from queue will set penalty to -1. * Added queue option autopausedelay when autopause is enabled it will be delayed for this number of seconds since last successful call if there was no prior call the agent will be autopaused immediately. * Added member option ignorebusy this when set and ringinuse is not will allow per member control of multiple calls as ringinuse does for the Queue. - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty. (closes issue ASTERISK-17421) (closes issue ASTERISK-17391) Reported by: irroot Tested by: irroot, jrose Review: https://reviewboard.asterisk.org/r/1119/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325483 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23ConfBridge: redundant code cleanupkmoore1-3/+0
There is no reason to clean up features twice. Review: https://reviewboard.asterisk.org/r/1279/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324709 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-21Fixes issue with channel write format being incorrectly restored when MOH is ↵dvossel1-3/+3
used in confbridge. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324422 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-21ConfBridge does not handle hangup properlykmoore1-25/+52
When playing back a prompt to a channel, confbridge neglects to check for hangup events causing lockup condititions for hangups that occur before actually joining the conference. This change ensures that the user is removed from the conference in the event of a premature hangup. Review: https://reviewboard.asterisk.org/r/1277/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324304 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-17Merged revisions 324176 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) | 2 lines Fix typo in documentation. Pointed out by Vlad Povorozniuc ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324177 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15CONFBRIDGE_INFO function to get conference datakmoore1-0/+87
Added the CONFBRIDGE_INFO dialplan function to get information about a conference bridge including locked status and number of parties, admins, and marked users. Review: https://reviewboard.asterisk.org/r/1271/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323517 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13Config inheritance doesn't work with ConfBridge() menu definitionskmoore1-1/+11
Current behavior in ConfBridge menu definitions is that first definition takes precedence, even in templated situations. This change allows inheritance and overriding to work as expected so that the last definition takes precedence. (closes ASTERISK-17986) Review: https://reviewboard.asterisk.org/r/1267/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323272 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13MOH for only user not working with ConfBridgekmoore2-4/+17
This adds the playing_moh flag to the conference_bridge_user struct that signifies when MOH should be playing so code doesn't have to guess whether MOH is playing. This change also adds the necessary checking to ensure that MOH continues playing for a single user in a conference after the join sound is played when configured to do so. (closes ASTERISK-17988) Review: https://reviewboard.asterisk.org/r/1263/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323107 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13ConfBridge: Use of bridge or user profiles that don't existkmoore1-2/+10
Bridge and user profiles are not checked for existence before use. The lack of a fully formed bridge profile can cause a segfault when sounds are accessed. This change ensures that bridge and user profiles exist prior to usage attempts. Review: https://reviewboard.asterisk.org/r/1264/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323106 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Merged revisions 322749 via svnmerge from rmudgett1-18/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines Remove potential deadlock in call pickup race. Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322750 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-08Merged revisions 322484 via svnmerge from rmudgett1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) | 15 lines Ring all queue with more than 255 agents will cause crash. 1. Create a ring-all queue with 500 permanent agents. 2. Call it. 3. Asterisk will crash. The watchers array in app_queue.c has a hard limit of 255. Bounds checking is not done on this array. No sane person should put 255 people in a ring-all queue, but we should not crash anyway. * Added bounds checking to the watchers array. JIRA AST-464 JIRA SWP-2903 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322485 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-06Remove Unused Var Warning rt_handle_member_recordirroot1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322128 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-06Refactor rt_handle_member_recordirroot1-14/+11
Review: https://reviewboard.asterisk.org/r/1172 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322111 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-01Merged revisions 321537 via svnmerge from bbryant1-6/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01 Jun 2011) | 8 lines This patch fixes an issue with using the wrong voicemail folders with greetings. (closes issue #17871) Reported by: edhorton Patches: digium_bug_17871_2 uploaded by fhackenberger (license 592) Tested by: edhorton, fhackenberger ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321538 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-27Merged revisions 321337 via svnmerge from rmudgett1-3/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert -r321331 and -r321332. ........ r321337 | rmudgett | 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321338 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-27Merged revisions 321330 via svnmerge from rmudgett1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. The trunk(v1.10) version will remove the unused options position. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321331 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-25Merged revisions 320823 via svnmerge from rmudgett4-3/+43
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320825 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 320237 via svnmerge from rmudgett1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320237 | rmudgett | 2011-05-20 15:49:03 -0500 (Fri, 20 May 2011) | 27 lines Merged revisions 320236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500 (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) | 13 lines The meetme CLI command completion leaves conferences mutex locked. When issuing a meetme kick CLI command and an invalid (non-existent) conference number is specified, pressing Tab leaves the conferences mutex locked and, therefore, all conferences deadlock. Add missing unlock. (closes issue #19336) Reported by: zvision Patches: app_meetme.diff uploaded by zvision (license 798) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320238 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 320162 via svnmerge from jrose1-1/+15
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May 2011) | 15 lines Fixes an imapfolder related crash imapfolders being set in the general section of voicemail would cause the inbox folder name to change. Since sound file names are made based on the names of the folders, this would cause the audio related to that folder name to change and if Asterisk attempted to play it, the channel would instantly hang up when the audio file couldn't be found. This patch searches for the name of the folder first to leave existing behavior in tact and if that fails, it uses the normal inbox name to get the sound file instead. (closes issue #16104) Reported by: blkline Review: https://reviewboard.asterisk.org/r/1215/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320178 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 320007 via svnmerge from rmudgett1-23/+24
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) | 2 lines Change some variable names to make pickup code easier to understand. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320013 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 319997 via svnmerge from rmudgett1-68/+60
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) | 25 lines Crash when using directed pickup applications. The directed pickup applications can cause a crash if the pickup was successful because the dialplan keeps executing. This patch does the following: * Completes the channel masquerade on a successful pickup before the application returns. The channel is now guaranteed a zombie and must not continue executing the dialplan. * Changes the return value of the directed pickup applications to return zero if the pickup failed and nonzero(-1) if the pickup succeeded. * Made some code optimizations that no longer require re-checking the pickup channel to see if it is still available to pickup. (closes issue #19310) Reported by: remiq Patches: issue19310_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, remiq, rmudgett Review: https://reviewboard.asterisk.org/r/1221/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319998 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-18Merged revisions 319529 via svnmerge from twilson1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319529 | twilson | 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) | 24 lines 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: http://svn.digium.com/svn/asterisk/trunk@319530 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-17Merged revisions 319367 via svnmerge from lmadsen1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319367 | lmadsen | 2011-05-17 07:53:50 -0500 (Tue, 17 May 2011) | 10 lines 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: http://svn.digium.com/svn/asterisk/trunk@319368 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-12Merged revisions 318671 via svnmerge from alecdavis1-61/+64
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ 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: http://svn.digium.com/svn/asterisk/trunk@318672 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317969 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines 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: http://svn.digium.com/svn/asterisk/trunk@317970 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317967 via svnmerge from russell1-6/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) | 2 lines Fix some more "set but unused" compiler warnings. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317968 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Add the Uniqueid header to Userevent.russell1-1/+6
(closes issue #16962) Reported by: jlpedrosa Patches: patch.diff uploaded by jlpedrosa (license 1002) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317915 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317584 via svnmerge from twilson1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317584 | twilson | 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) | 20 lines 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: http://svn.digium.com/svn/asterisk/trunk@317596 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317427 via svnmerge from russell1-2/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317427 | russell | 2011-05-05 16:58:45 -0500 (Thu, 05 May 2011) | 7 lines 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: http://svn.digium.com/svn/asterisk/trunk@317428 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317336 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317336 | russell | 2011-05-05 14:55:58 -0500 (Thu, 05 May 2011) | 7 lines 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: http://svn.digium.com/svn/asterisk/trunk@317337 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Merged revisions 316831 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) | 9 lines 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: http://svn.digium.com/svn/asterisk/trunk@316832 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Merged revisions 316709 via svnmerge from seanbright1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316709 | seanbright | 2011-05-04 12:15:32 -0400 (Wed, 04 May 2011) | 22 lines 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: http://svn.digium.com/svn/asterisk/trunk@316711 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Merged revisions 316650 via svnmerge from dvossel1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316650 | dvossel | 2011-05-04 09:25:03 -0500 (Wed, 04 May 2011) | 15 lines 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: http://svn.digium.com/svn/asterisk/trunk@316657 f38db490-d61c-443f-a65b-d21fe96a405b
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