mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
5636 Commits (d3ca0e0bc49feee43a4940e983e6bbd08ae68b29)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
59df60efb9 |
Merge "audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor."
|
9 years ago |
|
|
06c9966608 |
Merge "res_pjsip_messaging: Check URI type before dereferencing"
|
9 years ago |
|
|
9f64980e60 |
Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references."
|
9 years ago |
|
|
6b4b87787c |
res_pjsip_messaging: Check URI type before dereferencing
We aren't validating that the URI we just parsed is a SIP/SIPS one before trying to access the user, host, and port members of a possibly uninitialized structure. Also update the MessageSend documentation to indicate what 'from' formats are accepted. ASTERISK-26484 #close Reported by: Vinod Dharashive Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30 |
9 years ago |
|
|
25016a74f8 |
audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor.
Fixed a bug in function "ast_audiohook_write_frame" that checked the variable other_factory_samples and only flushed the factories, so they would be in sync, when other_factory_samples > 0. When there is not any rtp incoming the variable other_factory_samples will be 0, and although the result of "our_factory_ms - other_factory_ms" may be very large, this led to the record file not syncing. ASTERISK-26875 #close Reported-by: Aaron An Tested-by: Aaron An Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22 |
9 years ago |
|
|
fc71c18a9b |
thread safety: Don't use getprotobyname()
POSIX does not require getprotobyname() to be thread safe and some implementations use static memory which causes issues when multiple threads are used. Further, our usage of it today is just to ultimately get IPPROTO_TCP for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48 |
9 years ago |
|
|
76e64f5589 |
Merge "RFC sdp: Initial SDP creation"
|
9 years ago |
|
|
3f30ce1272 |
Merge "pbx.c: Fix crash from malformed exten pattern."
|
9 years ago |
|
|
c87e7dd9ec |
autochan/mixmonitor/chanspy: Fix unsafe channel locking and references.
Dereferencing struct ast_autochan.chan without first calling ast_autochan_channel_lock() is unsafe because the pointer could change at any time due to a masquerade. Unfortunately, ast_autochan_channel_lock() itself uses struct ast_autochan.chan unsafely and can result in a deadlock if the original channel happens to get destroyed after a masquerade in addition to the pointer getting changed. The problem is more likely to happen with v11 and earlier because masquerades are used to optimize out local channels on those versions. However, it could still happen on newer versions if the channel is executing a dialplan application when the channel is transferred or redirected. In this situation a masquerade still must be used. * Added a lock to struct ast_autochan to safely be able to use ast_autochan.chan while trying to get the channel lock in ast_autochan_channel_lock(). The locking order is the channel lock then the autochan lock. Locking in the other direction requires deadlock avoidance. * Fix unsafe ast_autochan.chan usages in app_mixmonitor.c. * Fix unsafe ast_autochan.chan usages in app_chanspy.c. * app_chanspy.c: Removed unused autochan parameter from next_channel(). ASTERISK-26867 Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592 |
9 years ago |
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3fe1d8afba |
Merge "core: Add stream topology changing primitives with tests."
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9 years ago |
|
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f997090877 |
pbx.c: Fix crash from malformed exten pattern.
Forgetting to indicate an exten is a pattern can cause a crash if the "pattern" has a character set range. e.g., "9999[3-5]" The crash is due to a buffer overwrite because the '-' exten eye-candy wasn't removed as expected and overran the allocated space. The buffer overwrite is fixed two ways in this patch. 1) Fix ext_strncpy() to distinguish between pattern and non-pattern extens. Now '-' characters are removed when they are eye-candy and not when they are part of a pattern character set. Since the function is private to pbx.c, the return value now returns the number of bytes written to the destination buffer instead of the strlen() of the final buffer so the callers that care don't need to add one. 2) Fix callers to ext_strncpy() to supply the correct available buffer size of the destination buffer. ASTERISK-26668 Change-Id: I555d97411140e47e0522684062d174fbe32aa84a |
9 years ago |
|
|
8470c2bdea |
RFC sdp: Initial SDP creation
* Added additional fields to ast_sdp_options. * Re-organized ast_sdp. * Updated field names to correspond to RFC4566 terminology. * Created allocs/frees for SDP children. * Created getters/setters for SDP children where appropriate. * Added ast_sdp_create_from_state. * Refactored res_sdp_translator_pjmedia for changes. Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48 |
9 years ago |
|
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b03b72717f |
main/stasis_cache: Demote the ERROR message when removing a nonexistent item
This patch demotes the ERROR message that is displayed when a nonexistent item is removed from the Stasis cache. The genesis of this demotion is due to chan_sip's realtime peers and their interaction with Asterisk's core ast_endpoint code, but ostensibly it could happen from other channel drivers as well. Since Mark Michelson already did an excellent job of explaining on this issue, it is quoted here for posterity: "Internally, when a realtime peer is retrieved, Asterisk creates an ast_endpoint structure. When that peer is destroyed, the ast_endpoint is destroyed as well. Part of the destruction of the ast_endpoint involves clearing the Stasis cache of all information about that endpoint. The problem here is that the act of creating the ast_endpoint is not enough to actually put any information in the Stasis cache. Instead, something has to happen, such as a state change, in order for the Stasis cache to have any information about that endpoint. When a device registers, chan_sip creates an ast_endpoint structure, processes the REGISTER, and then destroys the ast_endpoint. When the ast_endpoint is destroyed, there is nothing to destroy in the Stasis cache, so an error message is emitted. When you use rtcachefriends, ast_endpoint structures persist for the lifetime of the module and so you do not see this error message." ASTERISK-25237 #close Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70 |
9 years ago |
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35cfd2c0cc |
media_cache: Prefer ast_file_is_readable() over access()
Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def |
9 years ago |
|
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3ed05badb9 |
core: Add stream topology changing primitives with tests.
This change adds a few things to facilitate stream topology changing: 1. Control frame types have been added for use by the channel driver to notify the application that the channel wants to change the stream topology or that a stream topology change has been accepted. They are also used by the indicate interface to the channel that the application uses to indicate it wants to do the same. 2. Legacy behavior has been adopted in ast_read() such that if a channel requests a stream topology change it is denied automatically and the current stream topology is preserved if the application is not capable of handling streams. Tests have also been written which confirm the multistream and non-multistream behavior. ASTERISK-26839 Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9 |
9 years ago |
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272259a2c6 |
Saynumber is trying to get "and" from "digits/" subfolder
* say.c Changed 'digits/and' to 'vm-and' for en_GB ASTERISK-26598 #close Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe |
9 years ago |
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c9296b23d1 |
core: Cleanup ast_get_hint() usage.
* manager.c:manager_state_cb() Fix potential use of uninitialized hint[] if a hint does not exist for the requested extension. Ran into this when developing a testsuite test. The AMI event ExtensionStatus came out with the hint header value containing garbage. The AMI event PresenceStatus also had the same issue. * manager.c:action_extensionstate() no need to completely initialize the hint[]. Only initialize the first element. * pbx.c:ast_add_hint() Remove unnecessary assignment. * chan_sip.c: Eliminate an unneeded hint[] local variable. We only care about the return value of ast_get_hint() there. Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b |
9 years ago |
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fb11f038a3 |
Merge "stream: Unit tests for stream read and tweaks framework"
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9 years ago |
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0560c32375 |
stream: Unit tests for stream read and tweaks framework
* Removed the AST_CHAN_TP_MULTISTREAM tech property. We now rely on read_stream being set to indicate a multi stream channel. * Added ast_channel_is_multistream convenience function. * Fixed issue where stream and default_stream weren't being set on a frame retrieved from the queue. * Now testing for NULL being returned from the driver's read or read_stream callback. * Fixed issue where the dropnondefault code was crashing on a NULL f. * Now enforcing that if either read_stream or write_stream are set when ast_channel_tech_set is called that BOTH are set. * Added the unit tests. ASTERISK-26816 Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2 |
9 years ago |
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9c55a71798 |
SDP: Add initial SDP state machine.
This introduces and documents the various states in the state machine. This also introduces API functions that induce state changes, and places TODO comments telling what needs to be done in addition to what is already there. Those TODOs will be replaced with real code in upcoming changes. Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed |
9 years ago |
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26bf1846e2 |
Merge "media_cache: Mark cache entry stale if cache file is removed"
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9 years ago |
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60e9e4fcc0 |
media_cache: Mark cache entry stale if cache file is removed
In the event that a cache file is removed out from under us, we should treat the cache entry as stale and force a refresh. ASTERISK-26774 #close Reported by: Igor Gamayunov Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52 |
9 years ago |
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4692a32ed7 |
build: Warn if asterisk is installed in both 32 and 64 bit sys dirs
... and clean them both up on uninstall. We've fixed the issue where 'make install' was installing to /usr/lib on 64-bit systems that use /usr/lib64. Now we need to clean up the remnants in /usr/lib. * 'make install' now prints a warning if DESTDIR/ASTLIBDIR contains 'lib64' and libasterisk* shared libraries or modules are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'. * 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'. ASTERISK-26705 Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f |
9 years ago |
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df22d297a6 |
Merge "channel: Add ast_read_stream function for reading frames from all streams."
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9 years ago |
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6ac33bfe3e |
Merge "Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix."
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9 years ago |
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f943ead12e |
Merge "build: Execute ldconfig to build cache. (take two)"
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9 years ago |
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461577b23b |
Merge "channel: Add support for writing to a specific stream."
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9 years ago |
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c07c6714f2 |
channel: Add ast_read_stream function for reading frames from all streams.
This change introduces an ast_read_stream function and callback in the channel technology which allows reading frames from all streams and not just the default streams. The stream number has also been added to frames. This is to allow the case where frames are queued onto the channel instead of being read directly from the driver. This change does impose a restriction on reading though: a chain of frames can only contain frames from the same stream. ASTERISK-26816 Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce |
9 years ago |
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b0067bcf2c |
build: Execute ldconfig to build cache. (take two)
On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory. To make matters worse, options were being passed
to ldconfig on both Linux and FreeBSD that actually prevented
the rebuild of the cache.
* Fedora has a /usr/share/config.site that automatically tells
autoconf to use /usr/lib64 but CentOS does not. This logic was
copied to configure.ac and modified so systems like Ubuntu,
which still use /usr/lib for 64-bit systems, aren't affected.
Now that we have them in the correct directory...
In order for the system loader to find libasteriskssl and
libasteriskpj, one of 3 things has to happen...
- The linker cache must be rebuilt including the directory
where the libasterisk* libraries were installed. Only root
can rebuild the cache. This was busted.
- We have to link the asterisk binary with an rpath pointing
to the directrory where the libasterisk* libraries were
installed. This makes things very complicated and will happen
over the collective dead bodies of everyone who's had to
package a distribution with an rpath.
- Finally, you can start asterisk with LD_LIBRARY_PATH set to the
directrory where the libasterisk* libraries were installed.
There are no other options. So...
* The invokation of ldconfig has been moved from main/Makefile
to ASTTOPDIR/Makefile, the options have been removed, and
DESTDIR/ASTLIBDIR appended. If you aren't root, you will be
warned after the "Asterisk Installation Compete" banner that
you must re-run 'make install' as root, manually run
'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with
LD_LIBRARY_PATH.
ASTERISK-26705
Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982
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9 years ago |
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6cc890b880 |
channel: Add support for writing to a specific stream.
This change adds an ast_write_stream function which allows writing a frame to a specific media stream. It also moves ast_write() to using this underneath by writing media frames provided to it to the default streams of the channel. Existing functionality (such as audiohooks, framehooks, etc) are limited to being applied to the default stream only. Unit tests have also been added which test the behavior of both non-multistream and multistream channels to confirm that the write() and write_stream() callbacks are invoked appropriately. ASTERISK-26793 Change-Id: I4df20d1b65bd4d787fce0b4b478e19d2dfea245c |
9 years ago |
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094c26aa68 |
Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix.
Adds binaural synthesis to bridge_softmix (via convolution using libfftw3). Binaural synthesis is conducted at 48kHz. For a conference, only one spatial representation is rendered. The default rendering is applied for mono-capable channels. ASTERISK-26292 Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf |
9 years ago |
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9ad1df71b3 |
Merge "Revert "build: Execute ldconfig to build cache.""
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9 years ago |
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6c22d4b320 |
Merge "core: Show streams in "core show channel"."
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9 years ago |
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ced73d5b79 |
Revert "build: Execute ldconfig to build cache."
This reverts commit
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9 years ago |
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10302fa63f |
Merge "Add initial SDP state code."
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9 years ago |
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f58aefba5b |
core: Show streams in "core show channel".
The "core show channel" CLI command will now output the streams present on the channel with their details. ASTERISK-26811 Change-Id: I9c95b57aa09415005f0677a1949a0feb07e4987a |
9 years ago |
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a738772edd |
Add initial SDP state code.
This establishes the basic allocation/destruction of an SDP state object, plus some of the simpler getter methods involved. Subsequent tasks will deal with adding a state machine, creating SDPs from capabilities and options, and merging SDPs into a joint SDP. Change-Id: Ie3757ce186f04b65e9d1883f5aace53f24e53709 |
9 years ago |
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16b0bb39c1 |
Merge changes from topic 'sdp_state_beginnings'
* changes: Add SDP translator and PJMEDIA implementation. Add initial SDP options. |
9 years ago |
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28c8e4f58f |
build: Execute ldconfig to build cache.
On some platforms a multiarch approach is used for libraries. The build system does not take this into account and still places libraries into the lib directory if no --libdir is specified to configure. On initial startup this results in libasteriskssl.so not being found, as it is not in the multiarch lib directory. This change does the minimally invasive thing and executes ldconfig so that the libraries in the lib directory are found and their location cached. By doing so Asterisk starts up fine. If DESTDIR is specified, however, the old logic is executed as the install process may not have permission to alter the ldconfig cache. ASTERISK-26705 Change-Id: If4eca46ac510c6fea5568256280ffdb3888d7bb4 |
9 years ago |
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8dde33d184 |
Merge "tcptls.c: Add some missing allocation failure checks."
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9 years ago |
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496a7b0b4c |
Merge "Revert "build: Execute ldconfig to build cache.""
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9 years ago |
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7739b0b3ae |
Revert "build: Execute ldconfig to build cache."
This reverts commit
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9 years ago |
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c227745bc7 |
Merge "Remove extra ast_iostream_close() calls."
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9 years ago |
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0b427f9b59 |
tcptls.c: Add some missing allocation failure checks.
* Fix tcptls_session ref and fd leak in ast_tcptls_server_root(). Change-Id: I0ddf01cd3c10d3b6666d7bf68d4e206a37f4fbdb |
9 years ago |
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dbc3598014 |
Remove extra ast_iostream_close() calls.
When AMI encounters an error at the beginning of a session, it would explicitly call ast_iostream_close() on its tcptls session's iostream. It then would jump to a label where it would shut down the tcptls session instance. The tcptls session instance would again attempt to close the iostream. Under normal circumstances, this might go by unnoticed. However, when MALLOC_DEBUG is enabled, all fields on the iostream get set to 0xdeaddead when the iostream is freed. Thus a second call to ast_iostream_close() after the iostream has been freed would reslt in an attempt to call SSL_shutdown on 0xdeaddead, which would crash and burn horribly. The fix here is to not directly close the iostream from the dangerous scenarios. The specific scenarios are: * Exceeding the configured authlimit * Failing to build a mansession on a new connection Change-Id: I908f98d516afd5a263bd36b072221008a4731acd |
9 years ago |
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5a130b2e17 |
Add SDP translator and PJMEDIA implementation.
This creates the following: * Asterisk's internal representation of an SDP * An API for translating SDPs from one format to another * An implementation of a translator for PJMEDIA Change-Id: Ie2ecd3cbebe76756577be9b133e84d2ee356d46b |
9 years ago |
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8af6342555 |
Add initial SDP options.
This is step one of adding an SDP API: defining some configurable settings for SDPs. This is based on options that are currently supported in Asterisk. Change-Id: I1ede91aafed403b12a9ccdfb91a88389baa7e5d7 |
9 years ago |
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8851c3e088 |
build: Execute ldconfig to build cache.
On some platforms a multiarch approach is used for libraries. The build system does not take this into account and still places libraries into the lib directory if no --libdir is specified to configure. On initial startup this results in libasteriskssl.so not being found, as it is not in the multiarch lib directory. This change does the minimally invasive thing and executes ldconfig so that the libraries in the lib directory are found and their location cached. By doing so Asterisk starts up fine. ASTERISK-26705 Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519 |
9 years ago |
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ab34e46b3a |
Merge "stream: Rename creates/destroys to allocs/frees"
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9 years ago |
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f8f513d363 |
stream: Rename creates/destroys to allocs/frees
To be consistent with sdp implementation. Change-Id: I714e300939b4188f58ca66ce9d1e84b287009500 |
9 years ago |