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${ noResults }
1810 Commits (bdfda9ea23b3323c72cedd5733d94e6f6539f607)
Author | SHA1 | Message | Date |
---|---|---|---|
|
c6cb67b941 |
Resolve issue with negative vs non-negative length parameters.
(closes issue #14245) Reported by: dveiga git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168719 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
7c6d9c7235 |
Add option to hide console connect messages
(closes issue #14222) Reported by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded by jamesgolovich (license 176) Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168585 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
ef6ad2b53c |
Merged revisions 168561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
fd3cb90841 |
Some platforms (notably, the BSDs) have a more efficient implementation called
closefrom(3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
4a9e8078b9 |
When using ast_str with a non-ast_str-enabled API, we need to update the buffer
or otherwise, we cannot use ast_str_strlen(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
e1f941d7f6 |
Mostly just whitespace, but also convert 'CVS' to 'SVN' in a couple
places and fix a few typos I found in the CODING_GUIDELINES. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
aa905e347e |
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
c2999a8366 |
Introduce ast_careful_fwrite() and use in AMI to prevent partial writes.
This patch introduces a function to do careful writes on a file stream which will handle timeouts and partial writes. It is currently used in AMI to address the issue that has been reported. However, there are probably a few other places where this could be used. (closes issue #13546) Reported by: srt Tested by: russell http://reviewboard.digium.com/r/104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166282 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
894c91afe0 |
Make a note about formatting the review URL in commit messages
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166219 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
9733b30ff0 |
Adding a new dialplan function AUDIOHOOK_INHERIT
This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
775033301a |
Add configuration support for half_full DAHDI buffer policy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
79fe1aa0c6 |
Disable some automatic links generated by doxygen.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165886 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
2450098785 |
Introduce commit message formatting guidelines.
This documents the recommended outline to use for commit message. It also covers information on special tags that can be used in commit messages, as well as the template functionality that is available on bugs.digium.com. Review: http://reviewboard.digium.com/r/96/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
50a25ac847 |
Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.
This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165723 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
344a37f2a7 |
Remove duplicate code from the ast_str API. We now use __AST_STR_* to
access 'struct ast_str' members, but this must only be used inside the API implementation. (closes issue #14098) Reported by: eliel Patches: ast_str.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
27cbfc1bd5 |
Add timezone to the possible fields in a timespec.
(closes issue #14028) Reported by: mostyn Patches: timezone-v2.patch uploaded by mostyn (license 398) (with additional code guideline fixes and a memory leak fix by me - license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164976 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
d2d96b10ac |
introduce 'core show sysinfo' for systems that dont have the Linux-ish sysinfo stuff but do have sysctl.
(closes issue #13433) Reported by: mvanbaak Patches: 2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license 7) with two free calls replaced with ast_free based on feedback on reviewboard Review: http://reviewboard.digium.com/r/91/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
556b082522 |
Merged revisions 164736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was being used within the context of the thread local data destructors. We would go off and allocate more thread local data while the pthread lib was in the middle of destroying it all. This led to a memory leak. Another issue was an invalid argument being provided to the the object_add API call. (closes issue #13678) Reported by: ys Tested by: Russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164737 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
225b056239 |
Merged revisions 164422 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec 2008) | 3 lines Add the deadlock note to ast_spawn_extension as well ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164423 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c855c2c381 |
Merged revisions 164416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) | 4 lines Add notes to autoservice and pbx doxygen regarding a potential deadlock scenario so that it is avoided in the future ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
42e26ee700 |
Revert ast_str opacity in chan_sip for now, since something wasn't quite right
in the merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164417 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
203e224bcb |
I was getting this warning during a compile
on a 64-bit machine running ubuntu server 8.10, and gcc-4.3.2: [CXXi] chan_vpb.ii -> chan_vpb.oo cc1plus: warnings being treated as errors In file included from /home/murf/asterisk/trunk/include/asterisk/utils.h:671, from chan_vpb.cc:46: /home/murf/asterisk/trunk/include/asterisk/strings.h: In function ‘char* ast_str_truncate(ast_str*, ssize_t)’: /home/murf/asterisk/trunk/include/asterisk/strings.h:479: error: comparison between signed and unsigned integer expressions make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2 which this fix silences git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
808a5fda59 |
Fix a couple more build issues related to ast_str_opaque
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164309 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
8be6bc5f67 |
Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret.
(closes issue #14073) Reported by: seandarcy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164257 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
c31cbd7f1a |
Don't pass a negative to an unsigned type and expect things to work correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c8223fc957 |
Merge ast_str_opaque branch (discontinue usage of ast_str internals)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
7fcac067b2 |
Merged revisions 163448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c8c8995b70 |
add tab completion for 'core set debug X filename.c'
(closes issue #13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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96ae957f35 |
it does help if the compiler attribute syntax is correct
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162488 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d0bc22b3e8 |
Add some additional Asterisk project developer documentation.
After the nightly update of the documentation on asterisk.org, I'll post an update to asterisk-dev with a pointer to the changes. This covers some release branch and commit policy information. None of this should be a surprise, since it's just documenting what we have already been doing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162418 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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179667088b |
Merged revisions 162413 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines Remove the test_for_thread_safety() function completely. The test is not valid. Besides, if we actually suspected that recursive mutexes were not working, we would get a ton of LOG_ERROR messages when DEBUG_THREADS is turned on. (inspired by a discussion on the asterisk-dev list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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67cb0526b7 |
Merged revisions 162013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches: 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme, murf This crash was the result of a few small errors that would combine in 64-bit land to result in a crash. 32-bit land might have seen these combine to mysteriously drop the args to an application call, in certain circumstances. Also, in trying to find this bug, I spotted a situation in the flex input, where, in passing back a 'word' to the parser, it would allocate a buffer larger than necessary. I changed the usage in such situations, so that strdup was not used, but rather, an ast_malloc, followed by ast_copy_string. I removed a field from the pval struct, in u2, that was never getting used, and set in one spot in the code. I believe it was an artifact of a previous fix to make switch cases work invisibly with extens. And, for goto's I removed a '!' from before a strcmp, that has been there since the initial merging of AEL2, that might prevent the proper target of a goto from being found. This was pretty harmless on its own, as it would just louse up a consistency check for users. Many thanks to ckjohnsonme for providing a simplified and complete set of information about the bug, that helped considerably in finding and fixing the problem. Now, to get aelparse up and running again in trunk, and out of its "horribly broken" state, so I can run the regression suite! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162079 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fbb542055f |
Merged revisions 161426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r161426 | seanbright | 2008-12-05 16:02:20 -0500 (Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned int). (closes issue #14006) Reported by: alphaque Patches: astobj2.h-patch uploaded by alphaque (license 259) (Slightly modified by seanbright) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161427 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
3d4c0cd421 |
Merged revisions 160207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160208 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
033bffd32f |
Introduce CLI permissions.
Based on cli_permissions.conf configuration file, we are able to permit or deny cli commands based on some patterns and the local user and group running rasterisk. (Sorry if I missed some of the testers). Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue #11123) Reported by: eliel Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160062 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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887e28d7aa |
incorporates r159808 from branches/1.4:
------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
e14dfcbedc |
improve handling of API calls provided by loaded modules through use of some GCC features; this makes app_stack's usage of AGI APIs even cleaner, and will allow it to work 'as expected' either with or without res_agi being loaded
reviewed at http://reviewboard.digium.com/r/62 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
bb80c835e0 |
Add an option, waitfordialtone, for UK analog lines which do not end a call
until the originating line hangs up. (closes issue #12382) Reported by: one47 Patches: zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463) Tested by: fleed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ac296a4ad3 |
Merged revisions 159025 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) | 3 lines System call ioperm is non-portable, so check for its existence in autoconf. (Closes issue #13863) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fd8caa1778 |
This is basically a complete rollback of r155401, as it was determined that
it would be best to maintain API compatibility. Instead, this commit introduces ao2_callback_data() which is functionally identical to ao2_callback() except that it allows you to pass arbitrary data to the callback. Reviewed by Mark Michelson via ReviewBoard: http://reviewboard.digium.com/r/64 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
f3d086256c |
Added EVENT_FLAG_AGI and used it for manager calls in res_agi.c
(closes issue #13873) Reported by: fnordian Patches: ami_agievent.patch uploaded by fnordian (license 110) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158876 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6fb1f86054 |
Merged revisions 158539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158540 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3a9c27459e |
Merged revisions 158072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158133 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0e5367f7eb |
Merge the changes from the res_timing_timerfd branch.
This provides a new timing interface. In order to use it, you must be running a Linux with a kernel version of 2.6.25 or newer and glibc 2.8 or newer. This timing interface is a good alternative if a timing source is necessary (e.g. for IAX trunking) but DAHDI is otherwise unnecessary for the system. For now, this commit contains the actual work done in the res_timing_timerfd branch. There are no notices in the README or CHANGES files yet, but they will be added in my next commit. The timing API of Asterisk also needs to have a bit of work done with regards to choosing which timing interface to use. This commit makes the choice a build-time decision, by only allowing one of the timer interfaces to be chosen in menuselect. It would be preferable if the choice could be made at run-time, however. The preferred timing interface could be loaded and tested, and if it does not work, choice number two may be used instead. That sort of thing. That is beyond the scope of work in this branch though. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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81a16aa982 |
make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines
also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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afb571ba8f |
Starting with a change to ensure that ast_verbose() preserves ABI compatibility
in 1.6.1 (as compared to 1.6.0 and versions of 1.4), this change also deprecates the use of Asterisk with FreeBSD 4, given the central use of va_copy in core functions. va_copy() is C99, anyway, and we already require C99 for other purposes, so this isn't really a big change anyway. This change also simplifies some of the core ast_str_* functions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d91f1df3e0 |
Merged revisions 157305 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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cf6c66de65 |
Fix some refcounting in app_queue.c and change the
hashing used by app_queue.c to be case-insensitive. This is accomplished by adding a new case-insensitive hashing function. This was necessary to prevent bad refcount errors (and potential crashes) which would occur due to the fact that queues were initially read from the config file in a case-sensitive manner. Then, when a user issued a CLI command or manager action, we allowed for case-insensitive input and used that input to directly try to find the queue in the hash table. The result was either that we could not find a queue that was input or worse, we would end up hashing to a completely bogus value based on the input. This commit resolves the problem presented in issue #13703. However, that issue was reported against 1.6.0. Since this fix introduces a behavior change, I am electing to not place this same fix in to the 1.6.0 or 1.6.1 branches, and instead will opt for a change which does not change behavior. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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df6b78b742 |
Remove trailing whitespaces
using ':%s/\s\+$//' pointed by seanbright on #asterisk-dev git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156535 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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86f900b201 |
This commit does two things:
- Add CLI aliases module to asterisk. - Remove all deprecated CLI commands from the code Initial work done by file. Junk-Y and lmadsen did a lot of work and testing to get the list of deprecated commands into the configuration file. Deprecated CLI commands are now handled by this new module, see cli_aliases.conf for more info about that. ok russellb@ via reviewboard (closes issue #13735) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |