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2010-05-26Make AgentComplete message more consistent.mmichelson1-1/+2
At times, the "Member" field was not specified during the event. It's there now. (closes issue #15638) Reported by: elbriga Patches: patchAppQueueAgentComplete.diff uploaded by elbriga (license 482) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266004 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25Don't mark the cdr records of unanswered queue calls with "NOANSWER". This ↵mnicholson1-9/+0
restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@265610 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Don't hang up on a queue caller if the file we attempt to play does not exist.mmichelson1-0/+4
This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@265089 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Set quieted flag when receiving a dtmf tone during playback in speechbackground.mnicholson1-0/+1
(closes issue #16966) Reported by: asackheim git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264334 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-18Modify directory name reading to be interrupted with operator or pound escape.jpeeler1-0/+15
In the case of accidentally entering the wrong first three letters for the reading, users could be very frustrated if the name listing is very long. This allows interrupting the reading by pressing 0 or #. 0 will attempt to execute a configured operator (o) extension and # will exit and proceed in the dialplan. ABE-2200 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263769 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-12fixes app_meetme dsp errordvossel1-5/+4
We attempted to detect silence after translating a frame from signed linear. This caused a flooding of errors. To resolve this the code to detect silence was moved before the translation. (closes issue #17133) Reported by: jsdyer git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@262662 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-11Fix issue #17302 a slightly different way (mad props to Qwell)tilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@262321 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Only allow the operator key to be accepted after leaving a voicemail.jpeeler1-1/+4
Or rather disallow the operator key from being accepted when not offered, such as after finishing a recording from within the mailbox options menu. ABE-2121 SWP-1267 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261735 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Revert 261698, code in trunk leads me to believe unadvertised options are ↵jpeeler1-1/+19
supported. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261699 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Remove some hidden broken code in the voicemail mailbox options menu.jpeeler1-19/+1
After finishing a recording from within the mailbox options menu, pressing 0 exhibited strange behavior with operator=yes turned on. Pressing 0 was not even advertised as an option and the options from the vm-saveoper prompt: "Press 1 to accept this recording. Otherwise, please continue to hold" did not function correctly. While this of course could be fixed, it didn't really seem to make sense even if it was working properly. ABE-2121 SWP-1267 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261698 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Voicemail transfer to operator should occur immediately, not after main menu.jpeeler1-7/+8
There were two scenarios in the advanced options that while using the operator=yes and review=yes options, the transfer occurred only after exiting the main menu (after sending a reply or leaving a message for an extension). Now after the audio is processed for the reply or message the transfer occurs immediately as expected. ABE-2107 ABE-2108 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260923 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Do not play goodbye prompt after timeout of message review.jpeeler1-2/+2
ABE-2124 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259664 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Fix looping forever when no input received in certain voicemail menu scenarios.jpeeler1-2/+10
Specifically, prompting for an extension (when leaving or forwarding a message) or when prompting for a digit (when saving a message or changing folders). ABE-2122 SWP-1268 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@258432 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-20Play correct prompt when voicemail store failure occurs after attempted forward.jpeeler1-29/+55
If a user's mailbox was full and a message was attempted to be forwarded to said box, warnings on the console would indicate failure. However, the played prompt was that of success (vm-msgsaved). Now storage failure is taken into account and the correct prompt (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@258029 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19make app_voicemail compile with IMAP_STORAGEjpeeler1-1/+9
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257856 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-16Make the mixmonitor thread process audio frames fasterdhubbard1-5/+6
Mantis issue 17078 reports MixMonitor recordings have shorter durations than the call duration. This was because the mixmonitor thread was not processing frames from the audiohook fast enough. The mixmonitor thread would slowly fall behind the most recent audio frame and when the channel hangs up, the mixmonitor thread would exit without processing the same number of frames as the channel; leaving the mixmonitor recording shorter than actual call duration. This revision fixes this issue by moving the ast_audiohook_trigger_wait() and the subsequent audiohook.status check into the block where the ast_audiohook_read_frame() function returns NULL. (closes issue #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review: https://reviewboard.asterisk.org/r/611/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257686 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-14When forwarding a message, ensure that prepending works correctly.tilghman1-42/+25
This is a regression in 1.4, only. (closes issue #17103) Reported by: mglazer Patches: 20100408__issue17103.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257266 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-31Ensure line terminators in email are consistent.tilghman1-12/+29
Fixes an issue with certain Mail Transport Agents, where attachments are not interpreted correctly. (closes issue #16557) Reported by: jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14) Tested by: ebroad, zktech Reviewboard: https://reviewboard.asterisk.org/r/544/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@255591 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-31Add documentation clarifying when 't' and 'T' can be used.lmadsen1-2/+4
(closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@255503 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Resolve a number of FreeBSD build issues.russell2-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253631 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-18Typo found while fixing issue #16961.lmadsen1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253349 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-09Use ast_strlen_zero to avoid a crash when a Dial() string isn't passed to ↵seanbright1-1/+1
ParkAndAnnounce (closes issue #16731) Reported by: sebele67 Patches: issue16731_20100129.diff uploaded by seanbright (license 71) Tested by: sebele67 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@251410 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02revert ability to exit echo appalecdavis1-17/+10
caused a regression, as only supported VOICE, not VIDEO etc. Left in small formatting change. (issue #16880) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249946 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fixes ability to exit echo appalecdavis1-10/+18
when called from a ISDN channel, null frames prevent '#' exit. Now only echo back VOICE and DTMF frames (issue #16880) Reported by: alecdavis Patches: based on echo_exit_1-6-1.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249845 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Fix crash in app_voicemail related to message counting.seanbright1-1/+1
We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *' causing a segfault. (closes issue #16921) Reported by: whardier Patches: 20100301_issue16921.patch uploaded by seanbright (license 71) Tested by: whardier git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249671 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-24Make deletion of temporary greetings work properly with IMAP_STORAGEjpeeler1-4/+6
This same patch was merged in 220833, but was skipped in this branch erroneously. (closes issue #16170) Reported by: francesco_r git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248668 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-22Don't log to debug unless debug is turned onoej1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248268 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-17Make sure that when autofill is disabled that callers not in the front of ↵mmichelson1-3/+5
the queue cannot place calls. (closes issue #16834) Reported by: kebl0155 Patches: app_queue_no_autofill.v1.patch uploaded by kebl0155 (license 356) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247168 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-10fixes random deadlock in app_queue with use_weight during reloaddvossel1-6/+8
(closes issue #16677) Reported by: tim_ringenbach Patches: app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@246115 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-01Backup and restore original textfile, for prosthesis (gerund of prepend).tilghman1-6/+30
Also, fix menuselect such that changing voicemail build options correctly causes rebuild. (closes issue #16415) Reported by: tomo1657 Patches: prepention.patch uploaded by tomo1657 (license 484) (with modifications by me to backport to 1.4) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@244242 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Revert 243570, I should have looked at this closer. Will reopen the issue, butjpeeler1-1/+1
am leaving the review closed as the change was pointless. (issue #16488) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@243691 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Extend announcement URL used with Queue from 80 chars to PATH_MAX.jpeeler1-1/+1
(closes issue #16488) Reported by: syspert Patches: soundfilelen.pacth-2 uploaded by syspert (license 938) Review: https://reviewboard.asterisk.org/r/475/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@243570 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-15Disallow leaving more than maxmsg voicemails.tilghman1-5/+72
This is a possibility because our previous method assumed that no messages are left in parallel, which is not a safe assumption. Due to the vmu structure duplication, it was necessary to track in-process messages via a separate structure. If at some point, we switch vmu to an ao2-reference-counted structure, which would eliminate the prior noted duplication of structures, then we could incorporate this new in-process structure directly into vmu. (closes issue #16271) Reported by: sohosys Patches: 20100108__issue16271.diff.txt uploaded by tilghman (license 14) 20100108__issue16271__trunk.diff.txt uploaded by tilghman (license 14) 20100108__issue16271__1.6.0.diff.txt uploaded by tilghman (license 14) Tested by: jsutton git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@240414 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-13add silence gen to wait appsdvossel2-0/+17
asterisk.conf's 'transmit_silence' option existed before this patch, but was limited to only generating silence while recording and sending DTMF. Now enabling the transmit_silence option generates silence during wait times as well. To achieve this, ast_safe_sleep has been modified to generate silence anytime no other generators are present and transmit_silence is enabled. Wait apps not using ast_safe_sleep now generate silence when transmit_silence is enabled as well. (closes issue 0016524) Reported by: kobaz (closes issue 0016523) Reported by: kobaz Tested by: dvossel Review: https://reviewboard.asterisk.org/r/456/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@239718 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-06Resolve a crash due to an ast_frame not being fully initialized.russell1-2/+4
(closes issue #16531) Reported by: john8675309 (closes SWP-615) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@238009 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-28Avoid a crash with large numbers of MeetMe conferences.seanbright1-30/+33
Similar to changes made to Queue(), when we have large numbers of conferences in meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and crash, so instead just use a single fixed buffer. (closes issue #16509) Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded by seanbright (license 71) Tested by: seanbright git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@236509 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-11Fix talking detection status after conference user is muted.jpeeler1-20/+38
This patch ensures that when a conference user is muted that the accompanying AMI Meetme talking off event is sent. Also, the meetme list output is updated to show the muted user as unmonitored. (closes issue #16247) Reported by: dimas Patches: v3-16247.patch uploaded by dimas (license 88) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@234379 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-10Update chan_spy documentation.lmadsen1-2/+2
Update the documentation in ChanSpy and ExtenSpy to reflect that only a single group can be specified to the g() option. (closes issue #16420) Reported by: diatonic git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@234094 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-04document and rename strip_control() in app_voicemaildvossel1-3/+9
(closes issue #16291) Reported by: wdoekes git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@233116 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Deprecate "cz" in favor of "cs".tilghman1-63/+75
Also, change the use of language codes so that language registers as a prefix, rather than an exact match. (closes issue #16272) Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232820 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02fixes app_queue ao2 errordvossel1-2/+2
(closes issue #16369) Reported by: vrban Patches: queue_issue_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232444 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Fix a bug where if you hung up very quickly after calling AMD it would ↵file1-0/+1
overwrite the AMDSTATUS of HANGUP with TOOLONG. (closes issue #16239) Reported by: CGMChris git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232355 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Remove duplicate entries from voicemail format lists. This prevents ↵mnicholson1-2/+11
app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231614 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30app_queue crashes randomly, often during call-transfersdvossel1-100/+114
In app_queue, it is possible for a call_queue to be destroyed while another object still holds a pointer to it. This patch converts call_queue objects to ao2 objects allowing them to be ref counted. This makes it safe for the queue_ent object in queue_exec() to reference it's parent call_queue even after it has left the queue. (closes issue #15686) Reported by: Hatrix Patches: v2_queue_ao2.diff uploaded by dvossel (license 671) Tested by: dvossel, aragon Review: https://reviewboard.asterisk.org/r/427/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231437 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-25fixes solaris segfault on dial with verbosity >= 3dvossel1-2/+2
(closes issue #16193) Reported by: asgaroth Patches: bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231235 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-19fixes MixMonitor thread not exiting when StopMixMonitor is useddvossel1-28/+70
(closes issue #16152) Reported by: AlexMS Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, AlexMS Review: https://reviewboard.asterisk.org/r/424/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@230508 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Document a limitation in the AVAILSTATUS variable from ChanIsAvail and providefile1-1/+6
a workaround for it that does not change existing behavior. (closes issue #14426) Reported by: macli git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@229965 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04This patch modifies the Dial application to monitor the calling channel for ↵mnicholson1-13/+62
hangups while playing back announcements. (closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227827 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02Fix a bug where the recorded privacy introduction file would not get removed ↵file1-14/+31
if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226889 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Fix documentation for ast_softhangup() and correct the misuse thereof.tilghman1-1/+1
(closes issue #16103) Reported by: majorbloodnok git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225105 f38db490-d61c-443f-a65b-d21fe96a405b
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