mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
1256 Commits (d5c19c4c984e792ef3e5849e97d1dde7d01f244b)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
a0e2fe78a8 |
Use ast_free since ast_variable_new uses ast_calloc
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@283834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
c5278fc111 |
Send a SRCCHANGE indication when we masquerade
Masquerading a channel means that the src of the audio is potentially changing, so send a SRCCHANGE so that RTP-based media streams can get a new SSRC generated to reflect the change. Original patch by addix (along with lots of testing--thanks!). (closes issue #17007) Reported by: addix Patches: 1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006) srcchange.diff uploaded by twilson (license 396) Tested by: addix, twilson Review: https://reviewboard.asterisk.org/r/862/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@282430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
e90fb11e61 |
Ensure SSRC is changed when media source is changed to resolve audio delay.
This change causes the SSRC to change right before the channels are bridged, which is what used to happen. It seems that fixes were made to attempt limiting SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC with this change. There are two other control frames sent in ast_channel_bridge that probably should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change up to the discretion of resolving issue #17007. For reference - old review implementing new control frame SRCCHANGE: https://reviewboard.asterisk.org/r/540 (closes issue #17404) Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler (license 325) Tested by: sdolloff git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@281911 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
b7cb184823 |
Change context lock back to a mutex, because functionality depends upon the lock being recursive.
(closes issue #17643) Reported by: zerohalo Patches: 20100726__issue17643.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@280982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
27c86f1e9a |
fixes issue with translator frame not getting freed
A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@280448 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
0ebd53bba8 |
remove empty audiohook write list on channel
If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@279945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
307d1b7888 |
Avoid race with consolethread on shutdown (on parallel processors).
(closes issue #17080) Reported by: sybasesql Patches: 20100721__issue17080.diff.txt uploaded by tilghman (license 14) Tested by: sybasesql git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@278981 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
f4f7637e62 |
Allow PLC to function properly when channels use SLIN for audio.
If a channel involved in a bridge was using SLIN audio, then translation paths were not guaranteed to be set up properly since in all likelihood the number of translation steps was only 1. This patch enforces the transcode_via_slin behavior if transcode_via_slin or generic_plc is enabled and one of the formats to make compatible is SLIN. AST-352 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@278618 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
8618bacfd8 |
Do not queue up DTMF frames while a call is on hold.
(Fixes ABE-2110) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@278167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
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b765278efe |
Off-by-one error
(closes issue #16506) Reported by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@278023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
9f51a79d88 |
Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE.
(closes issue #16035) Reported by: francesco_r Patches: pbx.c.patch uploaded by viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@277327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
7ac1e8e9c7 |
If variable gotten is not set, will segfault on Solaris.
(closes issue #17636) Reported by: bklang git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@277261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
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cc8a3986cf |
For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation.
AST-362 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@277247 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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4fbc487659 |
In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@276652 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cd054ab3b4 |
Change ast_write to not stop generator when called from ast_prod.
For SIP channels configured with the progressinband option on, the ringback was being immediately stopped. This problem was due to ast_prod being moved for a deadlock fix in 259858. Prodding the channel after setting up the generator triggered the check in ast_write to stop the generator. The fix here should write the frame the same as was done before the call to ast_prod was moved. (closes issue #17372) Reported by: tech_admin git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ea332c39bf |
fix tab-completion for unload command.
(closes issue #17536) Reported by: junky Patches: unload_vs_mod_unload.diff uploaded by junky (license 177) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275290 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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de1dd975e6 |
give a better error message when attempting to unload a module that is not loaded
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275182 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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4654483d6b |
don't unload modules that returned AST_MODULE_LOAD_DECLINE when they were loaded
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275143 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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938b6264ff |
Document that a leading and trailing slash is expected for test categories.
Also, emit a warning if a test is registered without one of these. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275021 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c9818a396f |
Ensure file.o is built correctly.
(related to issue #15250) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@274359 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5724d8f905 |
Fix problem with RFC 2833 DTMF not being accepted.
A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order. The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the seqno rolling over will not cause us to stop accepting DTMF. (closes issue #17571) Reported by: mdeneen Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license 60) Tested by: richardf, maxochoa, JJCinAZ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@274157 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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9fc9b42295 |
Remove extra line breaks from 'core show config mappings'
(closes issue #17583) Reported by: pabelanger Patches: issue17583.patch uploaded by pabelanger (license 224) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273884 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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01af082769 |
Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup.
(closes issue #17564) Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2a18c6627d |
Don't return a partially initialized datastore.
If memory allocation fails in ast_strdup(), don't return a partially initialized datastore. Bad things may happen. (related to ABE-2415) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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96e6915e83 |
_Really_ skip the channel... don't just retry for another 200 cycles.
(Closes issue SWP-1652, ABE-2240) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273057 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7258f41a2b |
Don't change ownership/group/permissions on run directory, if it already exists.
(closes issue #17076) Reported by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by tilghman (license 14) Tested by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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8f2e2863be |
Also trim trailing blanks on #includes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272922 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c0abc6a359 |
Change the way that we read include files, to accommodate for changes in GCC 4.4.
(closes issue #17472) Reported by: seandarcy Patches: config2.patch uploaded by nivan (license 1066) Tested by: nivan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272921 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0dbe3c9921 |
Backport unit test API to 1.4.
Review: https://reviewboard.asterisk.org/r/750/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e3ee8deffd |
Set sin_family in ast_get_ip_or_srv() and removed the 'last' member of the ast_dnsmgr_entry struct.
(closes issue #15827) Reported by: DennisD Patches: (modified) dnsmgr_15827.patch uploaded by chappell (license 8) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@271123 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0800ce3602 |
Variables have always been case-sensitive, so we should not be removing case-insensitive matches.
Bug reported via the -dev list. See http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@270583 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ba3f2b0d20 |
For SpeeX, 0 bits remaining is valid and does not need an emitted warning.
(closes issue #15762) Reported by: nblasgen Patches: issue15672.patch uploaded by pabelanger (license 224) Tested by: nblasgen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@269960 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7a7326f8e9 |
Fix potential crash when writing raw SLIN audio on a PLC-enabled channel.
The issue here was that the frame created when adjusting for PLC had no offset to its audio data. If this frame were translated to another format prior to being sent out an RTP socket, all went well because the translation code would put an appropriate offset into the frame. However, if the SLIN audio were not translated before being sent out the RTP socket, bad things would happen. Specifically, the ast_rtp_raw_write makes the assumption that the frame has at least enough of an offset that it can accommodate an RTP header. This was not the case. As such, data was being written prior to the allocation, likely corrupting the data the memory allocator had written. Thus when the time came to free the data, all hell broke loose. ....Well, Asterisk crashed at least. The fix was just what one would expect. Offset the data in the frame by a reasonable amount. The method I used is a bit odd since the data in the frame is 16 bit integers and not bytes. I left a big ol' comment about it. This can be improved on if someone is interested. I was more interested in getting the crash resolved. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@269821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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458abdf68c |
Ensure restartable system calls can restart (BSD signal semantics)
This eliminates the annoying <beep> on the console. (closes issue #17477) Reported by: jvandal Patches: 20100610__issue17477.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@269635 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2b04df5697 |
Fix Debian init script to not use -c.
When using the init script as-is currently, it could cause issues on Debian such as high CPU usage. This fix has worked for several people so I'm implementing the change. We now handle color displays properly. (closes issue #16784) Reported by: pabelanger Patches: 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14) Tested by: pabelanger, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@269334 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0d37564b31 |
Fix summary for "core waitfullybooted" CLI command. Thanks, mnicholson.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@268579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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9d65386480 |
Get rid of a warning that gets printed out when the console is configured without any logger levels
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@268203 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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bc766d5b69 |
1.4 version of the dead PLC code fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@267539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d4af1fbc81 |
Add CLI command that blocks until Asterisk has fully booted.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@267183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6585975951 |
Cleanup error/warning messages in AEL2 parser
(closes issue #16684) Reported by: Silmaril Patches: patch_ael2_logmsg.diff uploaded by Silmaril (license 979) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@267009 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f703a2d7fa |
Prevent CLI prompt from distorting output of lines shorter than the prompt.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@266585 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a9b91b8435 |
Use sigaction for signals which should persist past the initial trigger, not 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@266142 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7e9a71f5d8 |
Merged revisions 265320,265467 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@265570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6105e07933 |
fixes segfault when using generic plc
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@265365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b91ff0498e |
Allow ast_safe_sleep to defer specific frames until after the sleep has 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@264996 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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20f140f1c7 |
ast_callerid_parse() had a path that left name uninitialized.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@264820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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610c1dc6bf |
1.4 version of PLC fix.
Analogous to trunk revision 264452, but without the change to chan_sip since it is not necessary in this branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@264541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e7ef5288c9 |
Because progress is called multiple times, across several frames, we must 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@263949 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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760f0a8a8b |
Fix logic error when checking for a devstate provider.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@263639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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109a363733 |
Remove arbitrary size limitation for hints.
(closes issue #17257) Reported by: tim_ringenbach Patches: hints_crash_fix.diff uploaded by tim ringenbach (license 540) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@263637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |