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${ noResults }
24831 Commits (f6593b4156b0daa0a74f1f04dc062a6b24acab6c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
2d1d5a98d5 |
Fix graceful shutdown crash.
The cleanup code for optional_api needs to happen after all of the optional API users and providers have unused/unprovided. Unfortunately, regsitering the atexit() handler at the beginning of main() isn't soon enough, since module destructors run after that. ........ Merged revisions 398149 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398150 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
be219c9ec9 |
New pjsip.conf.sample
(issue ASTERISK-22145) (closes issue ASTERISK-22145) Reported By: Matt Jordan Review: https://reviewboard.asterisk.org/r/2811/ ........ Merged revisions 398147 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398148 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
9bad1dabcf |
Add a reloadable option for sorcery type objects
Some configuration objects currently won't place nice if reloaded. Specifically, in this case the pjsip transport objects. Now when registering an object in sorcery one may specify that the object is allowed to be reloaded or not. If the object is set to not reload then upon reloading of the configuration the objects of that type will not be reloaded. The initially loaded objects of that type however will remain. While the transport objects will not longer be reloaded it is still possible for a user to configure an endpoint to an invalid transport. A couple of log messages were added to help diagnose this problem if it occurs. (closes issue ASTERISK-22382) Reported by: Rusty Newton (closes issue ASTERISK-22384) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2807/ ........ Merged revisions 398139 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398140 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
16b8d0cb5a |
Fix various memory leaks
main/config.c - cleanup cache fie includes res/res_security_log.c - unregister logger level channesl/chan_sip.c - cleanup io context and notify_types main/translator.c - cleanup at shutdown main/named_acl.c - cleanup cli commands main/indications.c - ast_get_indication_tone() unref default_tone_zone if used (closes issues ASTERISK-22378) Reported by: Corey Farrell Patches: config_shutdown.patch uploaded by coreyfarrell (license 5909) res_security_log.patch uploaded by coreyfarrell (license 5909) chan_sip-11.patch uploaded by coreyfarrell (license 5909) indications_refleak.patch uploaded by coreyfarrell (license 5909) named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license 5909) translate_shutdown.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 398102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398103 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398116 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398124 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
dbbb674be0 |
Update UPGRADE.txt file for Asterisk 12
This simply pulls in the changes that were breaking from the CHANGES file and updates a few other areas accordingly. It also removes the 10 => 11 notes, which are traditionally removed from each major version and stored in the appropriate UPGRADE-X.txt file. ........ Merged revisions 398100 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398101 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
0dfe658a5c |
features_config: Ignore parkinglots in features.conf instead of failing to load
Parkinglots are defined in res_features.conf now, but this patch fixes features_config so that features don't fail to load when parkinglots are present in features.conf Review: https://reviewboard.asterisk.org/r/2801/ ........ Merged revisions 398068 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398099 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
dcaa0cf659 |
features_config: Don't require features.conf to be present for Asterisk to load
(closes issue ASTERISK-22426) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2806/ ........ Merged revisions 398020 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
af1747ee6c |
Memory leak fix
ast_xmldoc_printable returns an allocated block that must be freed by the caller. Fixed manager.c and res_agi.c to stop leaking these results. (closes issue ASTERISK-22395) Reported by: Corey Farrell Patches: manager-leaks-12.patch uploaded by coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 398060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398061 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398062 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
9f1379686f |
test_substitution: Fix failing test.
Revert the -r392190 change. The original test was correct. The CDR code was actually returning an unititialized buffer. ........ Merged revisions 398025 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398026 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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b27160406c |
test_substituition: Fix failed test reporting to actually report failure.
You cannot put the "Testing <blah> pass/fail" on a single line before actually performing the test. Now any additional failure information is logged before the test pass/fail announcement. * Added an additional CDR(answer,u) test. ........ Merged revisions 398018 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398019 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398023 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
87bedd0695 |
Fix memory leaks
(closes issue ASTERISK-22368) Reported by: Corey Farrell Patches: issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes (license 5674) ........ Merged revisions 398004 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 398011 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398016 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398017 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
1d3d6e0661 |
Check return value on fwrite
........ Merged revisions 398000 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398002 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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9bed50db41 |
optional_api: Fix linking problems between modules that export global symbols
With the new work in Asterisk 12, there are some uses of the optional_api that are prone to failure. The details are rather involved, and captured on [the wiki][1]. This patch addresses the issue by removing almost all of the magic from the optional API implementation. Instead of relying on weak symbol resolution, a new optional_api.c module was added to Asterisk core. For modules providing an optional API, the pointer to the implementation function is registered with the core. For modules that use an optional API, a pointer to a stub function, along with a optional_ref function pointer are registered with the core. The optional_ref function pointers is set to the implementation function when it's provided, or the stub function when it's now. Since the implementation no longer relies on magic, it is now supported on all platforms. In the spirit of choice, an OPTIONAL_API flag was added, so we can disable the optional_api if needed (maybe it's buggy on some bizarre platform I haven't tested on) The AST_OPTIONAL_API*() macros themselves remained unchanged, so existing code could remain unchanged. But to help with debugging the optional_api, the patch limits the #include of optional API's to just the modules using the API. This also reduces resource waste maintaining optional_ref pointers that aren't used. Other changes made as a part of this patch: * The stubs for http_websocket that wrap system calls set errno to ENOSYS. * res_http_websocket now properly increments module use count. * In loader.c, the while() wrappers around dlclose() were removed. The while(!dlclose()) is actually an anti-pattern, which can lead to infinite loops if the module you're attempting to unload exports a symbol that was directly linked to. * The special handling of nonoptreq on systems without weak symbol support was removed, since we no longer rely on weak symbols for optional_api. [1]: https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue ASTERISK-22296) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2797/ ........ Merged revisions 397989 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397990 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
7f547872e4 |
ARI: Implement /recordings/stored API's
his patch implements the ARI API's for stored recordings. While the original task only specified deleting a recording, it was simple enough to implement the GET for all recordings, and for an individual recording. The recording playback operation was modified to use the same code for accessing the recording as the REST API, so that they will behave consistently. There were several problems with the api-docs that were also fixed, bringing the ARI spec in line with the implementation. There were some 'wishful thinking' fields on the stored recording model (duration and timestamp) that were removed, because I ended up not implementing a metadata file to go along with the recording to store such information. The GET /recordings/live operation was removed, since it's not really that useful to get a list of all recordings that are currently going on in the system. (At least, if we did that, we'd probably want to also list all of the current playbacks. Which seems weird.) (closes issue ASTERISK-21582) Review: https://reviewboard.asterisk.org/r/2693/ ........ Merged revisions 397985 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397988 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
51cd0ea28b |
Multiple revisions 397975-397976
........ r397975 | rmudgett | 2013-08-29 20:00:00 -0500 (Thu, 29 Aug 2013) | 1 line pbx.c: Make ast_str_substitute_variables_full() not mask variables. ........ r397976 | rmudgett | 2013-08-29 20:00:41 -0500 (Thu, 29 Aug 2013) | 1 line Revert last commit. ........ Merged revisions 397975-397976 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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bac9a478eb |
pbx.c: Make pbx_substitute_variables_helper_full() not mask variables.
........ Merged revisions 397977 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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eb955dad9a |
Sanitize XML output for PIDF bodies.
PJSIP's PIDF API does not replace angle brackets with their appropriate counterparts for XML. So we have to do it ourself. In this particular case, the problem had to do with attempting to place an unsanitized SIP URI into an XML node. Now we don't get a 488 from recipients of our PIDF NOTIFYs. ........ Merged revisions 397968 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397969 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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b5f61b5608 |
Fix method for creating activities string in PIDF bodies.
The previous method did not allocate enough space to create the entire string, but adjusted the string's slen value to be larger than the actual allocation. This resulted in garbled text in NOTIFY requests from Asterisk. This method allocates the proper amount of space first and then writes the content into the buffer. ........ Merged revisions 397960 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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d7b9a702d8 |
Verbose logging discrepancies
Refactored cases where a combination of ast_verbose/options_verbose were present. Also in general tried to eliminate, in as many places as possible, where the options_verbose global variable was being used. Refactored the way local and remote consoles handle verbose message logging in an attempt to solve the various discrepancies that sometimes would show between the two. (closes issue AST-1193) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2798/ ........ Merged revisions 397948 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 397958 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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8990924953 |
Fix when the subscription_terminated callback is called for subscription handlers.
The previous placement would result in the resubscribe() callback called instead of the subscription_terminated() callback being called when a subscription was ended via a SUBSCRIBE request. This would result in confusing PJSIP and having it throw an assertion. ........ Merged revisions 397955 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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de7ce39187 |
Fix a race condition where a canceled call was answered.
RFC 5407 section 3.1.2 details a scenario where a UAC sends a CANCEL at the same time that a UAS sends a 200 OK for the INVITE that the UAC is canceling. When this occurs, it is the role of the UAC to immediately send a BYE to terminate the call. This scenario was reproducible by have a Digium phone with two lines place a call to a second phone that forwarded the call to the second line on the original phone. The Digium phone, upon realizing that it was connecting to itself, would attempt to cancel the call. The timing of this happened to trigger the aforementioned race condition about 80% of the time. Asterisk was not doing its job of sending a BYE when receiving a 200 OK on a cancelled INVITE. The result was that the ast_channel structure was destroyed but the underlying SIP session, as well as the PJSIP inv_session and dialog, were still alive. Attempting to perform an action such as a transfer, once in this state, would result in Asterisk crashing. The circumstances are now detected properly and the session is ended as recommended in RFC 5407. (closes issue AST-1209) reported by John Bigelow ........ Merged revisions 397945 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397956 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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e1cfc18a78 |
Memory leaks fix
(closes ASTERISK-22376) Reported by: John Hardin Patches: memleak.patch uploaded by jhardin (license 6512) memleak2.patch uploaded by jhardin (license 6512) ........ Merged revisions 397946 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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449afdd9e8 |
Revert r394939 due to (numerous) objections
The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter and Tzafrir have pointed out numerous issues with the approach and have propsed an alternative in r/2757. Since it's not a time critical issue and is not worth holding up the release of 12 for it, I've gone ahead and reverted r394939 from 12/trunk and re-opened ASTERISK-21965. ........ Merged revisions 397938 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397939 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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417230848f |
Account for {} in Swagger notes
........ Merged revisions 397927 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397932 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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3f0148cd67 |
Recursively search for '.c' files when making documentation with 'make full'
Without this, documentation defined in sub-folders is ignored. Since having properly generated documentation is especially important in Asterisk 12 - not having it can cause a module to not load - 'make full' needs to look in all .c files. ........ Merged revisions 397924 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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0bc2a77365 |
Multiple revisions 397921-397922
........ r397921 | mmichelson | 2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) | 6 lines Resolve assumptions that bridge snapshots would be non-NULL for transfer stasis events. Attempting to transfer an unbridged call would result in crashes in either CEL code or in the conversion to AMI messages. ........ r397922 | mmichelson | 2013-08-29 10:42:29 -0500 (Thu, 29 Aug 2013) | 3 lines Remove extra debug message. ........ Merged revisions 397921-397922 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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72cf2779e8 |
Actually *add* the database schema management utilities
In r397874, the scripts were removed... but not replaced. Thanks to Michael Young for noticing this! ........ Merged revisions 397911 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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186db8fdaf |
Fix some uninitialized buffers for CDR handling valgrind found.
* Made ast_strftime_locale() ensure that the output buffer is initialized. The std library strftime() returns 0 and does not touch the buffer if it has an error. However, the function can also return 0 without an error. (closes issue ASTERISK-22412) Reported by: rmudgett ........ Merged revisions 397902 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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94e4733d89 |
Fixed problems with ast_cdr_serialize_variables().
* Fixed return value of ast_cdr_serialize_variables() on error. It needs to return 0 indicating no CDR variables found. * Made ast_cdr_serialize_variables() check the return value of cdr_object_format_property() and assert if nonzero. A member of the cdr_readonly_vars[] was not handled. * Removed unused elements from cdr_readonly_vars[]: total_duration, total_billsec, first_start, and first_answer. ........ Merged revisions 397900 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397901 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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87bf699dc9 |
Made the on/off in CLI "cdr set debug [on|off]" case insensitive.
........ Merged revisions 397898 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397899 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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ea095f6d1b |
Make CDR variable name chandling consistently case insensitive.
........ Merged revisions 397896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397897 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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7387282aa1 |
Make CDR code deal with channel names case insensitively.
........ Merged revisions 397894 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397895 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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5482fd21c8 |
Some CDR code optimization.
........ Merged revisions 397892 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397893 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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4ea0acbb44 |
Whitespace and curly braces.
........ Merged revisions 397885 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397886 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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6cc7319455 |
Improve detection of answer on SIP blind transfer.
A problem encountered during testing was that res_pjsip_refer would not ever send a NOTIFY with a 200 OK sipfrag. This is because the framehook that was supposed to send the NOTIFY would never be told that an answer had occurred. This happened for two reasons: 1) The transferee channel on which the framehook was on was already up. 2) Answers are rarely if ever written to channels. Rather, the ast_answer() or ast_raw_answer() function is used to answer channels. Thanks to a suggestion by Matt Jordan, the best way to detect that the call had been answered was to find out when the transferee channel joined a bridge. With stasis this is an easy task. So now, in addition to the framehook logic, there is a stasis subscription used to determine when the transferee has entered a bridge. Once it has entered, an appropriate NOTIFY is sent. ........ Merged revisions 397876 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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629f43d2b6 |
Add database schema management using Alembic
This patch replaces contrib/realtime/ with a new setup for managing the database schema required for database integration with Asterisk. In addition to initializing a database with the proper schema, alembic can do a database migration to assist with upgrading Asterisk in the future. Hopefully this helps make setting up and operating Asterisk with a database easier. With this the schema only needs to be maintained in one place instead of once per database. The schemas I have added here have a bit of improvement over the examples that were there before (some added consistency and added some missing indexes). Managing the schema in one place here also applies to all databases supported by SQLAlchemy. See contrib/ast-db-manage/README.md for more details. Review: https://reviewboard.asterisk.org/r/2731 patch by Russell Bryant (license 6300) ........ Merged revisions 397874 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397875 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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b7ec25ec2e |
Update CHANGES file for Asterisk 12
This updates the Asterisk 12 CHANGES file with the things that were missed during the development cycle. Review: https://reviewboard.asterisk.org/r/2795/ ........ Merged revisions 397870 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397872 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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b252c11aff |
pbx.c: Make ast_str_substitute_variables_full() not mask variables.
........ Merged revisions 397859 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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c7b7d98a37 |
ast_free() is null tollerant.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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b6a080d6a3 |
Match use of ast_free() with ast_calloc() and add some curly braces.
........ Merged revisions 397856 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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a181544299 |
Fix dialog matching in the SIP distributor.
Dialog matching is performed in the distributor for the sole purpose of retrieving an associated serializer so the request may be serialized. This patch fixes two problems. First, incoming CANCEL requests that had no to-tag (which really should be *all* CANCEL requests) would not match with a dialog. An earlier bug fix to deal with early CANCEL requests would result in the CANCEL being replied to with a 481. The fix for this is to find the matching INVITE transaction and get the dialog from that transaction. Second, no SIP responses were matching dialogs. This is because we were inverting the tags that we were passing into PJSIP's dialog finding function. This logic has been corrected by setting local and remote tag variables based on whether the incoming message is a request or response. ........ Merged revisions 397854 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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451993f4f5 |
ARI: WebSocket event cleanup
Stasis events (which get distributed over the ARI WebSocket) are created by subscribing to the channel_all_cached and bridge_all_cached topics, filtering out events for channels/bridges currently subscribed to. There are two issues with that. First was a race condition, where messages in-flight to the master subscribe-to-all-things topic would get sent out, even though the events happened before the channel was put into Stasis. Secondly, as the number of channels and bridges grow in the system, the work spent filtering messages becomes excessive. Since r395954, individual channels and bridges have caching topics, and can be subscribed to individually. This patch takes advantage, so that channels and bridges are subscribed to on demand, instead of filtering the global topics. The one case where filtering is still required is handling BridgeMerge messages, which are published directly to the bridge_all topic. Other than the change to how subscriptions work, this patch mostly just moves code around. Most of the work generating JSON objects from messages was moved to .to_json handlers on the message types. The callback functions handling app subscriptions were moved from res_stasis (b/c they were global to the model) to stasis/app.c (b/c they are local to the app now). (closes issue ASTERISK-21969) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2754/ ........ Merged revisions 397816 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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3540c7ac6e |
Made MALLOC_DEBUG less CPU intensive by default.
Storing a backtrace for each allocation in anticipation of a memory management problem is very CPU intensive. * Added the CLI "memory backtrace {on|off}" command to request that the backtrace be gathered only on request. The backtrace is off by default. (issue ASTERISK-22221) Reported by: Matt Jordan ........ Merged revisions 397809 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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c32f8a5ca9 |
AST-2013-005: Fix crash caused by invalid SDP
If the SIP channel driver processes an invalid SDP that defines media descriptions before connection information, it may attempt to reference the socket address information even though that information has not yet been set. This will cause a crash. This patch adds checks when handling the various media descriptions that ensures the media descriptions are handled only if we have connection information suitable for that media. Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing the solution to this problem. (closes issue ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches: issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674) ........ Merged revisions 397756 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397757 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 397758 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 397759 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397760 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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0472e14dee |
AST-2013-004: Fix crash when handling ACK on dialog that has no channel
A remote exploitable crash vulnerability exists in the SIP channel driver if an ACK with SDP is received after the channel has been terminated. The handling code incorrectly assumed that the channel would always be present. This patch adds a check such that the SDP will only be parsed and applied if Asterisk has a channel present that is associated with the dialog. Note that the patch being applied was modified only slightly from the patch provided by Walter Doekes of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches: issueA21064_fix.patch uploaded by wdoekes (License 5674) ........ Merged revisions 397710 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397711 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 397712 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 397713 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397753 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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868be02a2f |
Fix uninitialized value in struct ast_control_pvt_cause_code usage.
........ Merged revisions 397744 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 397745 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397746 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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24c56515b1 |
Better handle clearing the OUTGOING flag when a channel leaves a bridge
When a channel with the OUTGOING flag leaves a bridge, and it will survive being pulled from the bridge (either because it will execute dialplan, go into another bridge, or live in a friendly autoloop), we have to clear the OUTGOING flag. This is the signal to the CDR engine that this channel is no longer a second class citizen, i.e., it is not "dialed". The soft hangup flags are only half the picture. If a channel is being moved from one bridge to another, the soft hangup flags aren't set; however, the state of the bridge_channel will not be hung up. Since the channel does not have one of the two hang up states, that implies that the channel is still technically alive. This patch modifies the check so that it checks both the soft hangup flags as well as the bridge_channel state. If either suggests that the channel is going to persist, we clear the OUTGOING flag. ........ Merged revisions 397690 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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6c410d00d1 |
Fixed bucket.c for systems where tv_usec is not an unsigned long.
........ Merged revisions 397673 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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d647b4ae02 |
bridging: Fix a livelock with local channel optimization.
Use a better means of waking up the bridge channel thread. ........ Merged revisions 397650 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397651 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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13dbdd1ae7 |
chan_dahdi: Add some missing build cleanup.
........ Merged revisions 397643 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |