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${ noResults }
1490 Commits (6477f207f1dd3a158ac53b0eb056a695d4884745)
| Author | SHA1 | Message | Date |
|---|---|---|---|
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0ee1f43b4a |
Merged revisions 107161 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) | 8 lines Fix another bug specifically related to asynchronous call origination. Once the PBX is started on the channel using ast_pbx_start(), then the ownership of the channel has been passed on to another thread. We can no longer access it in this code. If the channel gets hung up very quickly, it is possible that we could access a channel that has been free'd. (inspired by BE-386) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107162 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
2d95fb33bd |
Merged revisions 107158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) | 9 lines Fix some bugs related to originating calls. If the code failed to start a PBX on the channel (such as if you set a call limit based on the system's load average), then there were cases where a channel that has already been free'd using ast_hangup() got accessed. This caused weird memory corruption and crashes to occur. (fixes issue BE-386) (much debugging credit goes to twilson, final patch written by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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86fe5cc800 |
Merged revisions 107102 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008) | 2 lines Resolve a compiler warning. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107103 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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541039ce1b |
Merged revisions 107099 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008) | 3 lines Fix a race condition where the generator can go away (closes issue #12175, reported by edantie, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107100 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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dafa73bade |
way back in July, in r.75706, a fix was made ot the strftime usages, which was good, but in this case, the check for a nil time was accidentally removed, and now it is restored, to keep timevals like '1969-12-31 17:00:00' from showing up in the cdrs. No idea what databases will do with this. No bugs filed as yet, but it felt like a bug.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107019 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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af7e1964f2 |
Merged revisions 107016 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial. (closes issue #11516) Reported by: ys Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: anest, jcapp, dartvader ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107017 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8aa30ccf0b |
Merged revisions 106842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) | 5 lines Fix hardcoded grep in editline, were GNU grep is required. (closes issue #12124) Reported by: dmartin ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106843 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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733a740922 |
Merged revisions 106788 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4 lines Ignore source update control frame. (closes issue #12168) Reported by: plack ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106789 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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377e51c4d4 |
(closes issue #6002)
Reported by: rizzo Tested by: murf Proposal of the changes to be made, and then an announcement of how they were accomplished: http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html and: http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html Here is a recap, file by file, of what I have done: pbx/pbx_config.c pbx/pbx_ael.c All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set. Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it is just as necessary to have the TABLE available. This is because the list/table in question might not be the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global position when things are ready. We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing "find" and "create", as all existing usages used both in tandem anyway. pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and then call merge_contexts_and_delete, which will merge (now) existing contexts and priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will lock down the contexts, swap the lists and tables, and unlock (real quick), and then destroy the old dialplan. chan_sip.c chan_iax.c chan_skinny.c All the channel drivers that would add regcontexts now use the ast_context_find_or_create now. chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered. apps/app_meetme.c apps/app_dial.c apps/app_queue.c All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead. include/asterisk/pbx.h ast_context_create() is removed. Find_or_create_ is the new method. ast_context_find_or_create() interface gets the hashtab added. ast_merge_contexts_and_delete() gets the local hashtab arg added. ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking. ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael ast_hashtab_hash_contexts was in like fashion make public. include/asterisk/pval.h ast_compile_ael2() interface changed to include the local hashtab table ptr. main/features.c For the sake of the parking context, we use ast_context_find_or_create(). main/pbx.c I changed all the "tree" names to "table" instead. That's because the original implementation was based on binary trees. (had a free library). Then I moved to hashtabs. Now, the names move forward too. refcount field added to contexts, so you can keep track of how many modules wanted this context to exist. Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING. Added some calls to ast_verb(3,...) for debug messages Lots of little mods to ast_context_remove_extension2, which is now excersized in ways it was not previously; one definite bug fixed. find_or_create was upgraded to handle both local lists/tables as well as the globals. context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables ast_merge_contexts_and_delete() was heavily modified. ast_add_extension2() was also upgraded to handle changes. the context_destroy() code was re-engineered to handle the new way of doing things, by exten/prio instead of by context. res/ael/pval.c res/ael/ael.tab.c res/ael/ael.tab.h res/ael/ael.y res/ael/ael_lex.c res/ael/ael.flex utils/ael_main.c utils/extconf.c utils/conf2ael.c utils/Makefile Had to change the interface to ast_compile_ael2(), to include the hashtab ptr. This ended up involving several external apps. The main gotcha was I had to include lock.h and hashtab.h in several places. As a side note, I tested this stuff pretty thoroughly, I replicated the problems originally reported by Luigi, and made triply sure that reloads worked, and everything worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into trunk, that did not appear in my tests of bug6002. How's this for verbose commit messages? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a60f591c72 |
Merged revisions 106606 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) | 3 lines Properly initialize rtp->schedid (Closes issue #12154) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106607 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8718878490 |
Merged revisions 106552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines Safely use the strncat() function. (closes issue #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5ca5d97673 |
Merge changes from team/russell/g722-sillyness ...
Fix a number of other places where the number of samples in a G722 frame was not properly handled because of various reasons. main/rtp.c: - When a G722 frame is read from the smoother, the number of samples in the frame must be divided by 2 before being sent out over the network. Even though G722 is 16 kHz, an error in some previous spec has made it so that we have to list the number of samples such as if it was 8 kHz. main/file.c: - When scheduling the next time to expect a frame, take into account that the format of the file we're reading from may not be 8 kHz. codecs/codec_g722.c: - When converting from G722 to slinear, g722_decode() expects its samples parameter to be in the silly (real samples / 2) format. Make it so. - When converting from slinear to G722, properly set the number of samples in the frame to be the number of bytes of output * 2. formats/format_pcm.c: - This format module handles G722, among a number of other formats. However, the read() and seek() functions did not account for the fact that G722 has 2 samples per byte. (closes issue #12130, reported by rickross, patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106501 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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49ef635a6d |
Fix file playback in many cases.
(closes issue #12115) Reported by: pj Patches: v2-fileexists.patch uploaded by dimas (license 88) (with modifications by me) Tested by: dimas, qwell, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106439 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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924b7d3636 |
Merged revisions 106437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar 2008) | 8 lines Quell an annoying message that is likely to print every single time that ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial allocates the cdr for the channel, so it should be expected that the channel will have a cdr on it. Thanks to joetester on IRC for pointing this out ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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496adc6fc0 |
Merged revisions 106235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2ed30d47e8 |
Adding the Atxfer manager command. With this, you may initiate
an attended transfer over AMI (closes issue #10585) Reported by: ornati Patches: atxfer-trunk-r90428.diff uploaded by ornati (license 210) (with modifications from me) Tested by: putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106236 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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455f6137b4 |
Fix code up to what it was meant to be.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106110 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8a411ccf83 |
Create a centralized configuration option for silencethreshold
(closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a760a033e9 |
Merged revisions 105932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines Fix a bug that I just noticed in the RTP code. The calculation for setting the len field in an ast_frame of audio was wrong when G.722 is in use. The len field represents the number of ms of audio that the frame contains. It would have set the value to be twice what it should be. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105933 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6cf6059402 |
There are several places in manager.c where BUFSIZ is used for a buffer
which will contain nowhere near that amount of data. This makes these buffers more reasonably sized. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7007c565fe |
Fix minor misuses of snprintf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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cfc1df4c1a |
Whitespace changes only
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105840 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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96e04792bd |
add a destroy API call for a server instance
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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cc55483858 |
More public API name changes to use an appropriate ast_ prefix
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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efb1e30a38 |
Rename public object server_instance to ast_tcptls_server_instance
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3b070a815d |
Merged revisions 105676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 lines In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4de0d8368f |
Merged revisions 105674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines When a new source of audio comes in (such as music on hold) make sure the marker bit gets set. (closes issue #10355) Reported by: wdecarne Patches: 10355.diff uploaded by file (license 11) (closes issue #11491) Reported by: kanderson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7205c1f23c |
Make it so you don't have to cast away const in a couple places
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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83002989a9 |
remove unnecessary casts
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6d3b251588 |
- Add curly braces around the while loop
- Properly break out of the loop on error when an included context is not found git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105590 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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71173779dc |
Use ast_copy_string() instead of strncpy(), and use sizeof() instead of
a magic number git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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f7467c83e6 |
Merged revisions 105565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105565 | russell | 2008-03-03 10:01:50 -0600 (Mon, 03 Mar 2008) | 3 lines Update the copyright information for autoservice. Most of the code in this file now is stuff that I have written recently ... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8372a9bf08 |
3) In addition to merging the changes below, change trunk back to a regular
LIST instead of an RWLIST. The way this list works makes it such that a RWLIST provides no additional benefit. Also, a mutex is needed for use with the thread condition. Merged revisions 105563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines Merge in some changes from team/russell/autoservice-nochans-1.4 These changes fix up some dubious code that I came across while auditing what happens in the autoservice thread when there are no channels currently in autoservice. 1) Change it so that autoservice thread doesn't keep looping around calling ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition so that the thread properly goes to sleep and does not wake up until a channel is put into autoservice. This actually fixes an interesting bug, as well. If the autoservice thread is already running (almost always is the case), then when the thread goes from having 0 channels to have 1 channel to autoservice, that channel would have to wait for up to 1/2 of a second to have the first frame read from it. 2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no channels and no fds to poll() on, such as was the case with the previous code for the autoservice thread. In this case, the code would call alloca(0), and pass the result as the first argument to poll(). In this case, the 2nd argument to poll() specified that there were no fds, so this invalid pointer shouldn't actually get dereferenced, but, this code makes it explicit and ensures the pointers are NULL unless we have valid data to put there. (related to issue #12116) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105564 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b21ca22fb0 |
Merged revisions 105560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7 lines It is possible for no audio to pass between the current digit and next digit so expand logic that clears emulation to AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg Patches: v1-11911.patch uploaded by dimas (license 88) Tested by: tbsky ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105561 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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00641af7d2 |
Add support for 16KHz signed linear.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7d1891d5c3 |
Asterisk, when parking can drop rights a caller when a parking timeout occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue.
(closes issue #11520) Reported by: pliew Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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796780679e |
Merged revisions 105409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008) | 23 lines Fix a major bug in autoservice. There was a race condition in the handling of the list of channels in autoservice. The problem was that it was possible for a channel to get removed from autoservice and destroyed, while the autoservice thread was still messing with the channel. This led to memory corruption, and caused crashes. This explains multiple backtraces I have seen that have references to autoservice, but do to the nature of the issue (memory corruption), could cause crashes in a number of areas. (fixes the crash in BE-386) (closes issue #11694) (closes issue #11940) The following issues could be related. If you are the reporter of one of these, please update to include this fix and try again. (potentially fixes issue #11189) (potentially fixes issue #12107) (potentially fixes issue #11573) (potentially fixes issue #12008) (potentially fixes issue #11189) (potentially fixes issue #11993) (potentially fixes issue #11791) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7da06e6cb8 |
Merged revisions 105116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008) | 8 lines Fix a bug in the lock tracking code that was discovered by mmichelson. The issue is that if the lock history array was full, then the functions to mark a lock as acquired or not would adjust the stats for whatever lock is at the end of the array, which may not be itself. So, do a sanity check to make sure that we're updating lock info for the proper lock. (This explains the bizarre stats on lock #63 in BE-396, thanks Mark!) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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beb7a540c1 |
Merged revisions 104841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb 2008) | 17 lines Two fixes: 1. Make the list of ast_dial_channels a lockable list. This is because in some cases, the ast_dial may exist in multiple threads due to asynchronous execution of its application, and I found some cases where race conditions could exist. 2. Check in ast_dial_join to be sure that the channel still exists before attempting to lock it, since it could have gotten hung up but the is_running_app flag on the ast_dial_channel may not have been cleared yet. (closes issue #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by putnopvut (license 60) Tested by: jvandal ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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62c63a8412 |
Merged revisions 105005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb 2008) | 9 lines Make pbx_exec pass an empty string into applications, if we get NULL. This protects against possible segfaults in applications that may try to use data before checking length (ast_strdupa'ing it, for example) (closes issue #12100) Reported by: foxfire Patches: 12100-nullappargs.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7d564048ed |
Merged revisions 104868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008) | 7 lines Compatibility fix for PPC64 (closes issue #12081) Reported by: jcollie Patches: asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412) Tested by: jcollie, Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Merged revisions 104783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb 2008) | 4 lines Bump a couple of more buffers up by 2 so that annoying warnings aren't generated like crazy on every fileexists_core call. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104784 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Merged revisions 104704 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104704 | tilghman | 2008-02-27 12:15:10 -0600 (Wed, 27 Feb 2008) | 2 lines Ensure the session ID can't be 0. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104705 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 104665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2 lines Bump up the buffer by 2. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Merged revisions 104598 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104598 | qwell | 2008-02-27 11:26:55 -0600 (Wed, 27 Feb 2008) | 8 lines Inherit language from the transfering channel on a blind transfer. (closes issue #11682) Reported by: caio1982 Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982 (license 22) Tested by: caio1982, victoryure ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Merged revisions 104596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104596 | file | 2008-02-27 13:07:33 -0400 (Wed, 27 Feb 2008) | 4 lines Use the lock (which already existed, it just wasn't used) on the updaters list to protect the contents instead of the overall module list lock. (closes issue #12080) Reported by: ChaseVenters ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Merged revisions 104593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb 2008) | 8 lines fallback to standard English prompts properly when using new prompt directory layout (closes issue #11831) Reported by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license 20) (modified by me to improve code and conform rest of function to coding guidelines) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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dca12f4aa7 |
Fix T38 passthrough regression introduced by state changes.
(closes issue #12078) Reported by: dimas Patches: v1-12078.patch uploaded by dimas (license 88) (closes issue #12074) Reported by: Ivan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b0ba749ca1 |
I swear I compiled this ... *cough*
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0036cde5f3 |
Rename version.h to ast_version.h. Next, I will be re-adding version.h as an
automatically generated file like it used to be. This still needs to be there for modules that have to check it to compile against multiple asterisk versions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104244 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |