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2010-06-04Comment out a rule that likes to break the chan_h323 build randomly.russell1-2/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@268053 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-04Build menuselect with the build environment's compiler, not the host ↵tilghman2-1/+3
(target)'s compiler. (closes issue #17464) Reported by: pprindeville Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@268050 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-04As-fixiate the build processtilghman2-4/+8
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@267971 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-04Make the default install path appear to be /usr on Linux, instead of /usr/local.tilghman3-825/+1067
Also, reorganize the options, so that they're more alphabetical. (closes issue #17013) Reported by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@267759 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-031.4 version of the dead PLC code fix.mmichelson9-217/+8
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@267539 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add CLI command that blocks until Asterisk has fully booted.russell1-0/+19
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@267183 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Cleanup error/warning messages in AEL2 parserpabelanger1-2/+2
(closes issue #16684) Reported by: Silmaril Patches: patch_ael2_logmsg.diff uploaded by Silmaril (license 979) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@267009 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Backport live_ast script from 1.6.2 branchpabelanger1-0/+261
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266924 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Prevent CLI prompt from distorting output of lines shorter than the prompt.tilghman1-1/+1
Uses the VT100 method of clearing the line from the cursor position to the end of the line: Esc-0K (closes issue #17160) Reported by: coolmig Patches: 20100531__issue17160.diff.txt uploaded by tilghman (license 14) Tested by: coolmig git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266585 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Fix formatting issue with previous patch.pabelanger1-4/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266580 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Missing fallback to audio fax feature when T.38 re-INVITE failedpabelanger1-14/+26
When a T.38 re-INVITE failed with an 488 or 606 answer, we should fallback to audio fax by send a re-re-INVITE without T.38. The function is backported from 1.6 asterisk. (closes issue #16795) Reported by: vrban (closes issue #16692) Reported by: vrban Patches: t38_fallback_to_audio_v3.patch uploaded by vrban (license 756) Tested by: lmadsen, vrban, haggard https://reviewboard.asterisk.org/r/514/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266579 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-30Reverting patch and reopening issue #16784, as patch breaks color display.tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266437 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Use sigaction for signals which should persist past the initial trigger, not ↵tilghman2-20/+38
signal. If you call signal() in a Solaris signal handler, instead of just resetting the signal handler, it causes the signal to refire, because the signal is not marked as handled prior to the signal handler being called. This effectively causes Solaris to immediately exceed the threadstack in recursive signal handlers and crash. (closes issue #17000) Reported by: rmcgilvr Patches: 20100526__issue17000.diff.txt uploaded by tilghman (license 14) Tested by: rmcgilvr git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266142 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26add dahdi_func_write to zap_tech structuredvossel1-0/+1
This was supposed to be committed with r263292, the back-port of teh DAHDI buffer policy dial string option git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266140 f38db490-d61c-443f-a65b-d21fe96a405b
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
2010-05-26Not finding rows in the DB does not rise to the level of a warning.tilghman1-1/+1
(closes issue #17062) Reported by: drookie Patches: 20100525__issue17062.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@265910 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25fixes build issue with zapteldvossel1-0/+2
(closes issue #17394) Reported by: aragon Patches: half_buffer_fix.diff uploaded by dvossel (license 671) Tested by: aragon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@265613 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
2010-05-25Merged revisions 265320,265467 via svnmerge from twilson5-0/+29
https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines Add the FullyBooted AMI event It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ ........ r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line Merge the rest of the FullyBooted patch ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@265570 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24fixes segfault when using generic plcdvossel1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@265365 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.mmichelson2-1/+5
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
2010-05-21Fix grammatical error in comment.mmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264999 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Allow ast_safe_sleep to defer specific frames until after the sleep has ↵mmichelson3-27/+73
concluded. From reviewboard Background: A Digium customer discovered a somewhat odd bug. The setup is that parties A and B are bridged, and party A places party B on hold. While party B is listening to hold music, he mashes a bunch of DTMF. Party A takes party B off hold while this is happening, but party B continues to hear hold music. I could reproduce this about 1 in 5 times. The issue: When DTMF features are enabled and a user presses keys, the channel that the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read from the channel during the sleep, the frame is dropped. Thus the unhold indication is never made to the channel that was originally placed on hold. The fix: Originally, I discussed with Kevin possible ways of fixing the specific problem reported. However, we determined that the same type of problem could happen in other situations where ast_safe_sleep() is used. Using autoservice as a model, I modified ast_safe_sleep_conditional() to defer specific frame types so they can be re-queued once the sleep has finished. I made a common function for determining if a frame should be deferred so that there are not two identical switch blocks to maintain. Review: https://reviewboard.asterisk.org/r/674/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264996 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20ast_callerid_parse() had a path that left name uninitialized.rmudgett1-0/+2
Several callers of ast_callerid_parse() do not initialize the name parameter before calling thus there is the potential to use an uninitialized pointer. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264820 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-201.4 version of PLC fix.mmichelson4-5/+118
Analogous to trunk revision 264452, but without the change to chan_sip since it is not necessary in this branch. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264541 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
2010-05-19Internal timing is now on by default, if you're using DAHDI 2.3 or above.tilghman3-6317/+19559
The reason for ensuring DAHDI 2.3 or above is that this version ensures that a timer is always available, whereas in previous versions, it was possible for DAHDI to be loaded, but have no drivers to actually generate timing. If internal_timing was turned on in this circumstance, a complete lack of audio would result. This is the reason why internal_timing was not on by default. However, now that DAHDI ensures the availability of a timer, there is no reason for this setting to be off (and in fact, it solves a great many initial user problems). (closes issue #15932) Reported by: dimas Patches: 20100519__issue15932.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264248 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19fix incorrectly typed indications for [nz] stutter and dialrecallalecdavis1-2/+2
(closes issue #17359) Reported by: alecdavis Patches: bug17359.diff.txt uploaded by alecdavis (license 585) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264056 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Because progress is called multiple times, across several frames, we must ↵tilghman1-51/+62
persist states when detecting multitone sequences. (closes issue #16749) Reported by: dant Patches: dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by: dant git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263949 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
2010-05-17Fix logic error when checking for a devstate provider.mmichelson1-1/+1
When using strsep, if one of the list of specified separators is not found, it is the first parameter to strsep which is now NULL, not the pointer returned by strsep. This issue isn't especially severe in that the worst it is likely to do is waste some cycles when a device with no '/' and no ':' is passed to ast_device_state. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263639 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Remove arbitrary size limitation for hints.mmichelson1-2/+2
(closes issue #17257) Reported by: tim_ringenbach Patches: hints_crash_fix.diff uploaded by tim ringenbach (license 540) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263637 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Manager cookies are not compatible with RFC2109.lmadsen1-1/+1
The Version field in the cookies we're setting contain quotes around the version number which is not compatible with RFC2109 and breaks some implementations. (closes issue #17231) Reported by: ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by ecarruda (license 559) Tested by: ecarruda, russell git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263456 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Update link to new version of core sounds.lmadsen1-1/+1
The latest version of the core sounds files 1.4.19 now includes the missing queue-minute sound file which is called by app_queue but which has been missing. (closes issue #17123) Reported by: n8ideas git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263374 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17backport of DAHDI buffer policy dial string optiondvossel1-21/+100
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263292 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-13Fix internal timing not working with Zapteljpeeler2-1/+2
dahdi_compat.h was not being included in channel.c when used with Zaptel and wasn't in file.c at all. (closes issue #15250) Reported by: mneuhauser Patches: dahdi_compat.patch uploaded by mneuhauser (license 425) Tested by: IgorG git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263112 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
2010-05-11Use a less silly method for modifying a flex-generated file.qwell1-2/+6
The sed syntax that was used wasn't actually valid, causing some versions to choke. This is the method that is used in 1.6.x+ for similar changes. (closes issue #16696) Reported by: bklang Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested by: qwell git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@262421 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-11Fix issue #17302 a slightly different way (mad props to Qwell)tilghman2-3/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@262321 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-10Allow compilation on Mac OS X 10.4 (Tiger)tilghman1-0/+3
(closes issue #17297) Reported by: jcovert Patches: 20100506__issue17297.diff.txt uploaded by tilghman (license 14) (closes issue #17302) Reported by: jcovert git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@262151 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
2010-05-06Use the versioned MOH tarballs, now that we have them.qwell1-2/+7
This makes for more reproducibility. Prompted by a discussion in #asterisk-dev git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261608 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05Registration fix for SIP realtime.pabelanger1-3/+7
Make sure realtime fields are not empty. (closes issue #17266) Reported by: Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis, sberney Review: https://reviewboard.asterisk.org/r/643/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261274 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Add a tiny corner case to the previous committilghman1-1/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261094 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Protect against overflow, when calculating how long to wait for a frame.tilghman1-1/+1
(closes issue #17128) Reported by: under Patches: d.diff uploaded by under (license 914) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261093 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
2010-05-04Fix FILTER() examples to work in 1.4tringenbach1-4/+4
Review: https://reviewboard.asterisk.org/r/644/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260887 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Fix fallout from removing from configure script. Pointed out by philipp64 ↵qwell1-1/+1
on #asterisk-dev git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260801 f38db490-d61c-443f-a65b-d21fe96a405b
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