mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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1.6.0.8
1.6.0.9
1.6.1-beta1
1.6.1-beta2
1.6.1-beta3
1.6.1-beta4
1.6.1-rc1
1.6.1.0
1.6.1.0-rc2
1.6.1.0-rc3
1.6.1.0-rc4
1.6.1.0-rc5
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1.6.1.10-rc1
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10.5.0-digiumphones
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10.5.1-digiumphones
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10.5.2-digiumphones
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10.7.0-digiumphones
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10.7.1-digiumphones
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10.8.0-digiumphones
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${ noResults }
6473 Commits (bdd785d31cc1abfdbe33c1a8fc7ca67c04293687)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
bdd785d31c |
various files - fix some alerts raised by lgtm code analysis
This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25 |
6 years ago |
|
|
391aafb971 |
Merge "pbx: deadlock when outgoing dialed channel hangs up too quickly"
|
6 years ago |
|
|
bf6f27388d |
pbx: deadlock when outgoing dialed channel hangs up too quickly
Here's the basic scenario that occurred when executing an AMI fast originate while at the same time something else locks the channels container, and also wants a lock on the dialed channel: 1. pbx_outgoing_attempt obtains a lock on a dialed channel 2. concurrently another thread obtains a lock on the channels container, and subsequently requests a lock on the dialed channel. It waits on #1. For instance, "core show channel <dialed channel" 3. the outgoing call does not fail, but ends before the pbx_outgoing_attempt function exits 4. pbx_outgoing_attempt function exits, the outgoing structure destructs, and attempts to hang up the dialed channel 5. hang up tries to obtain the channels container lock, but can't due to #2. 6. Asterisk is deadlocked. The solution was to allow the pbx_outgoing_exec function to "steal" ownership of the dialed channel, and handle hanging it up. The channel now is either hung up prior to it being potentially locked by the initiating thread, or if locked the hang up takes place in a different thread, thus alleviating the deadlock. ASTERISK-28561 patches: iliketrains.diff submitted by Joshua Colp (license 5000) Change-Id: I51b42b92dde8f2215b69bb509e28667ee3a3853a |
6 years ago |
|
|
c0efe19cec |
serializer: move/add asterisk serializer pool functionality
Serializer pools have previously existed in Asterisk. However, for the most part the code has been duplicated across modules. This patch abstracts the code into an 'ast_serializer_pool' object. As well the code is now centralized in serializer.c/h. In addition serializer pools can now optionally be monitored by a shutdown group. This will prevent the pool from being destroyed until all serializers have completed. Change-Id: Ib1e906144b90ffd4d5ed9826f0b719ca9c6d2971 |
6 years ago |
|
|
d73482e6d1 |
Merge "stasis_state: Create internal stasis_state_proxy object."
|
6 years ago |
|
|
729b286d59 |
stasis: Pass bumped topic_all reference to proxy_dtor.
This avoids use of the global variable and ensures topic_all remains active until all topics are freed. ASTERISK-28553 patches: ASTERISK-28553.patch by coreyfarrell (license 5909) Change-Id: I9a8cd8977f3c3a6aa00783f8336d2cfb9c2820f1 |
6 years ago |
|
|
57da20ddac |
Merge "taskprocessor.c: Added "like" support to 'core show taskprocessors'"
|
6 years ago |
|
|
702019fc80 |
pbx: Prevent Realtime switch crash on invalid priority
pbx_extension_helper takes two 'context' arguments. One (con) is a pointer directly to a 'struct ast_context' and the other (context) is the name of the context. In all cases, one of these arguments is NULL and the other is non-NULL. Functions that are ultimately called by pbx_extension_helper expect that 'context' will be non-NULL, so we set it unconditionally on entry into this function. ASTERISK-28534 #close Change-Id: Ifbbc5e71440afd80efd441f7a9d72e8b10b6f47d |
6 years ago |
|
|
4c3655ecfd |
taskprocessor.c: Added "like" support to 'core show taskprocessors'
Added "like" support for 'core show taskprocessors'. Now you can specify a specific set of taskprocessors (or just one) by adding the keyword "like" to the above command, followed by your search criteria. Change-Id: I021e740201e9ba487204b5451e46feb0e3222464 |
6 years ago |
|
|
f156391176 |
Merge "core: Fix ABI mismatch of ao2_global_obj."
|
6 years ago |
|
|
2d4abb9906 |
Merge "taskprocessor.c: Add CLI commands to reset taskprocessor stats."
|
6 years ago |
|
|
f2367e56c0 |
Merge "core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option."
|
6 years ago |
|
|
f7045cefd9 |
stasis_state: Create internal stasis_state_proxy object.
This improves the way which stasis_state reference counting works. Since manager->states holds onto the proxy object instead of the real object this allows stasis_state objects to be freed when appropriate without use of a special state_remove function. Additionally each distinct eid associated with the state holds a reference to the state to prevent early release and potentially allow easier debug of leaks. Change-Id: I400e0db4b9afa3d5cb4ac7dad60907897e73f9a9 |
6 years ago |
|
|
4de1e6d0e6 |
taskprocessor.c: Add CLI commands to reset taskprocessor stats.
Added two new CLI commands to reset stats for taskprocessors. You can
reset stats for a single, specific taskprocessor ('core reset
taskprocessor <taskprocessor>'), or you can reset all taskprocessors
('core reset taskprocessors'). These commands will reset the counter for
the number of tasks processed as well as the max queue size.
Change-Id: Iaf17fc4ae29396ab0c6ac92408fc7bdc2f12362d
|
6 years ago |
|
|
e034e7f51d |
Merge "astmm.c: Display backtrace with memory show allocations"
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6 years ago |
|
|
ffbad8ed83 |
Merge "stasis: refcounter.py can incorrectly report skewed objects."
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6 years ago |
|
|
725e991faf
|
core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option.
Previous to this patch passing a NULL tag to ao2_alloc or ao2_ref based functions would result in the reference not being logged under REF_DEBUG. This could sometimes cause inaccurate logging if NULL was accidentally passed to a reference action. Now reference logging is only disabled by option passed to the allocation method. Change-Id: I3c17d867d901d53f9fcd512bef4d52e342637b54 |
6 years ago |
|
|
e82f2f6e82 |
astmm.c: Display backtrace with memory show allocations
You can currently capture backtraces of memory allocations but they only get displayed when you stop asterisk and the atexit hooks are enabled. Now, if memory backtrace is on and you issue a "memory show allocations" CLI command for a specific file, then a backtrace will show for each allocation that occurred after you turned "memory backtrace on". The backtrace display is shown only when a specific file's allocations are displayed to prevent a massive CLI dump of every file's allocations. Change-Id: Ic657afc1fc6ec7205e16eb36a97a611d235a2b4f |
6 years ago |
|
|
a4142c8437
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core: Fix ABI mismatch of ao2_global_obj.
astobj2.c declares DEBUG_THREADS_LOOSE_ABI to avoid overhead of debug threads tracking information in the internal structures of astobj2. Unfortunately this means that ao2_global_obj contains the statically allocated debug threads tracking fields which are used by initialization and cleanup but main/astobj2.c believed those fields and associated space did not exist. Change-Id: Icef41ad97d88a8c1d1515e034ec8133cab3b1527 |
6 years ago |
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ca608d2575 |
stasis: refcounter.py can incorrectly report skewed objects.
It is possible for topic->name to be NULL, this causes the allocation reference to not be logged. Use the name variable instead which has been verified to be a non-empty. Change-Id: I3d0031d03c8356e4808f00cdf2d5428712575883 |
6 years ago |
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3dfbc05c53 |
stasis: Fix leaks
* Release reference returned by cache_remove * state_alloc unconditionally bumped state_topic even when it was locally allocated. Change-Id: I51101bf7d07b8dc8ce8fc46b6cb31fbbd213fbc7 |
6 years ago |
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e79a3b428a |
Merge "func_jitterbuffer: Add audio/video sync support."
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6 years ago |
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7298a785ad |
func_jitterbuffer: Add audio/video sync support.
This change adds support to the JITTERBUFFER dialplan function for audio and video synchronization. When enabled the RTCP SR report is used to produce an NTP timestamp for both the audio and video streams. Using this information the video frames are queued until their NTP timestamp is equal to or behind the NTP timestamp of the audio. The audio jitterbuffer acts as the leader deciding when to shrink/grow the jitterbuffer when adaptive is in use. For both adaptive and fixed the video buffer follows the size of the audio jitterbuffer. ASTERISK-28533 Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492 |
6 years ago |
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c18983207d |
core: Add H.265/HEVC passthrough support
This change adds H.265/HEVC as a known codec and creates a cached "h265" media format for use. Note that RFC 7798 section 7.2 also describes additional SDP parameters. Handling of these is not yet supported. ASTERISK-28512 Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2 |
6 years ago |
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32ce6e9a06 |
channels: Allow updating variable value
When modifying an already defined variable in some channel drivers they add a new variable with the same name to the list, but that value is never used, only the first one found. Introduce ast_variable_list_replace() and use it where appropriate. ASTERISK-23756 #close Patches: setvar-multiplie.patch submitted by Michael Goryainov Change-Id: Ie1897a96c82b8945e752733612ee963686f32839 |
6 years ago |
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0e56643d9f |
res_rtp: Add unit tests for RTCP stats.
Added unit tests for RTCP video stats. These tests include NACK, REMB, FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR tests are currently disabled due to a bug. We expect to receive a compound packet, but the code sends this out as a single packet, which the browser accepts, but makes Asterisk upset. While writing these tests, I noticed an issue with NACK as well. Where it is handling a received NACK request, it was reading in only the first 8 bits of following packets that were also lost. This has been changed to the correct value of 16 bits. Also made a minor fix to the data buffer unit test. Change-Id: I56107c7411003a247589bbb6086d25c54719901b |
6 years ago |
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1e9714a050 |
AST-2019-005 - translate: Don't assume all frames will have a src.
This change removes the assumption that a frame will always have a src set on it. This assumption is incorrect. Given a scenario where an RTP packet is received with no payload the resulting audio frame will have no samples. If this frame goes through a signed linear translation path an interpolated frame can be created (if generic packet loss concealment is enabled) that has minimal data on it, including no src. If this frame is given to a translation path a crash will occur due to the lack of src. ASTERISK-28499 Change-Id: I024d10dd98207eb8a6b35b59880bcdf1090538f8 |
7 years ago |
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9e015713cc |
dns_core: Create new API ast_dns_resolve_ipv6_and_ipv4
The new function takes in a pointer to an ast_sockaddr structure, a hostname and an optional port and then dispatches parallel "AAAA" and "A" record queries. If an "AAAA" record is returned, it's parsed into the ast_sockaddr structure along with the port if it was supplied. If no "AAAA" record was returned, the first "A" record returned (if any) is parsed instead. This is a synchronous call. If you need asynchronous lookups, use ast_dns_query_set_resolve_async and roll your own. Change-Id: I194b0b0e73da94b35cc35263a868ffac3a8d0a95 |
7 years ago |
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64906c4c9b |
audiohook.c: Substitute silence for unavailable audio frames
There are 4 scenarios to consider when capturing audio from a channel with an audiohook: 1. There is no rx and no tx audio, so return nothing. 2. There is rx but no tx audio, so return rx. 3. There is tx but no rx audio, so return tx. 4. There is rx and tx audio, so mix them and return. The file passed as the primary argument to MixMonitor will be written to in scenarios 2, 3, and 4. However, if you pass the r() and t() options to MixMonitor, a frame will only be written to the r() file if there was rx audio and a frame will only be written to the t() file if there was tx audio. If you subsequently take the r() and t() files and try to mix them, the sides of the conversation will 'drift' and be non-representative of the user experience. This patch adds a new 'S' option to MixMonitor that injects a frame of silence on either the r() side or the t() side of the channel so that when later mixed, there is no such drift. Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e |
7 years ago |
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bcc0b85da8 |
Merge "srtp: Fix possible race condition, and add NULL checks"
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7 years ago |
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b859dd450a |
Merge "cdr / cel: Use event time at event creation instead of processing."
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7 years ago |
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b805e1237d |
srtp: Fix possible race condition, and add NULL checks
Somehow it's possible for the srtp session object to be NULL even though the Asterisk srtp object itself is valid. When this happened it would cause a crash down in the srtp code when attempting to protect or unprotect data. After looking at the code there is at least one spot that makes this situation possible. If Asterisk fails to unprotect the data, and after several retries it still can't then the srtp->session gets freed, and set to NULL while still leaving the Asterisk srtp object around. However, according to the original issue reporter this does not appear to be their situation since they found no errors logged stating the above happened (which Asterisk does for that situation). An issue was found however, where a possible race condition could occur between the pjsip incoming negotiation, and the receiving of RTP packets. Both places could attempt to create/setup srtp for the same rtp instance at the same time. This potentially could be the cause of the problem as well. Given the above this patch adds locking around srtp setup for a given rtp, or rtcp instance. NULL checks for the session have also been added within the protect and unprotect functions as a precaution. These checks should at least stop Asterisk from crashing if it gets in this situation again. This patch also fixes one other issue noticed during investigation. When doing a replace the old object was freed before creating the replacement. If the new replacement object failed to create then the rtp/rtcp instance would now point to freed srtp data which could potentially cause a crash as well when the next attempt to reference it was made. This is now fixed so the old srtp object is kept upon replacement failure. Lastly, more logging has been added to help diagnose future issues. ASTERISK-28472 Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc |
7 years ago |
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261646c1c4 |
cdr / cel: Use event time at event creation instead of processing.
When updating times on CDR or CEL records using the time at which it is done can result in times being incorrect if the system is heavily loaded and stasis message processing is delayed. This change instead makes it so CDR and CEL use the time at which the stasis messages that drive the systems are created. This allows them to be backed up while still producing correct records. ASTERISK-28498 Change-Id: I6829227e67aefa318efe5e183a94d4a1b4e8500a |
7 years ago |
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4b14b86114 |
Merge "various modules: json integer overflow"
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7 years ago |
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97fd008562 |
Merge "main/udptl.c: correctly handle udptl sequence wrap around"
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7 years ago |
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3656c42cb0 |
various modules: json integer overflow
There were still a few places in the code that could overflow when "packing"
a json object with a value outside the base type integer's range. For instance:
unsigned int value = INT_MAX + 1
ast_json_pack("{s: i}", value);
would result in a negative number being "packed". In those situations this patch
alters those values to a ast_json_int_t, which widens the value up to a long or
long long.
ASTERISK-28480
Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1
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7 years ago |
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ae2ddaa496 |
Merge "manager: Send fewer packets"
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7 years ago |
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084901d548 |
main/udptl.c: correctly handle udptl sequence wrap around
incorrect handling of UDPTL squence number wrap arounds causes loss of packets every time the wrap around occurs ASTERISK-28483 #close Change-Id: I33caeb2bf13c574a1ebb81714b58907091d64234 |
7 years ago |
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5f66fb5139 |
manager: Send fewer packets
The functions that build manager message headers do so in a way that results in a single messages being split across multiple packets. While this doesn't matter to the remote end, it makes network captures noisier and harder to follow, and also means additional system calls. With this patch, we build up more of the message content into the TLS buffer before flushing to the network. This change is completely internal to the manager code and does not affect any of the existing API's consumers. Change-Id: I50128b0769060ca5272dbbb5e60242d131eaddf9 |
7 years ago |
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8e44d823c1 |
loader.c: Fix possible SEGV when a module fails to register
When a module fails to register itself (usually a coding error in the module), dlerror() can return NULL. We weren't checking for that in load_dlopen() before trying to strdup the error message so a SEGV was thrown. dlerror() is now surrounded with an S_OR so we don't SEGV. Change-Id: Ie0fb9316f08a321434f3f85aecf3c7d2ede8b956 |
7 years ago |
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5167555f8e |
Merge "sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread"
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7 years ago |
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54c1645c29 |
Merge "manager: Log AMI actions"
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7 years ago |
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3c6f11992b |
sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread
When fixing ASTERISK~24212, a change was done so a scheduled callback could not be removed while it was running. The caller of ast_sched_del would have to wait. However, when the caller of ast_sched_del is the callback itself (however wrong this might be), this new check would cause a deadlock: it would wait forever for itself. This changeset introduces an additional check: if ast_sched_del is called by the callback itself, it is immediately rejected (along with an ERROR log and a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the after-ast_sched_del-refcall function is only run if ast_sched_del returned success. This should fix the following spurious race condition found in chan_sip: - thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE) - thread 2: run sip_poke_peer_now - thread 2: blank out sched-ID (too soon!) - thread 1: set sched-ID (too late!) - thread 2: try to delete the currently running sched-ID After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and other madness) should occur. (Thanks Richard Mudgett for reviewing/improving this "scary" change.) Note that this change does not fix the observed race condition: unlocked access to peer->pokeexpire (and potentially other scheduled items in chan_sip), causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the deadlock go away. And in the observed case, it will not have adverse affects (like memory leaks) because the scheduled item is removed through a different path. ASTERISK-28282 Change-Id: Ic26777fa0732725e6ca7010df17af77a012aa856 |
7 years ago |
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ba25038fd5 |
manager: Log AMI actions
When manager debugging is turned on, this patch makes it so incoming AMI actions are now also logged. Change-Id: I8047524510e7ac97d99482b2448f8e368f29cd47 |
7 years ago |
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b31ac83900 |
mwi: Update the MWI core to use stasis_state API
** Note ** This patch is meant to be the minimum needed in order for the MWI core to use the now underlying stasis_state module. As such it does not completely remove its reliance on the stasis_cache. Doing so has allowed current consumers to not have to change, and update those code paths for this patch. When time allows, subsequent patches can/will be made to those consumers to take advantage of some of the new MWI API included here. Thus, eventually and ultimately removing MWI dependency on the stasis_cache. ** End Note ** This patch makes it so the MWI core now takes advantage of the new stasis_state API. Consumers of MWI should no longer need to depend upon stasis topic pooling, and the stasis cache directly. Similar functionality and implementation details have now been pushed into the stasis_state module. However, all MWI state should be accessed via the MWI API itself. As such a few new methods, and constructs have been added to the MWI core that facilitate consumer publishing, subscribing, and iterating over MWI state data. * ast_mwi_subscriber * Created via ast_mwi_add_subscriber, a subscriber subscribes to a given mailbox in order to receive updates about the given mailbox. Adding a subscriber will create the underlying topic, and associated state data if those do not already exist for it. The topic, and last known state data is guaranteed to exist for the lifetime of the subscriber. * ast_mwi_publisher * Before publishing to a particular topic a publisher should be created. This can be achieved by using ast_mwi_add_publisher. Publishing to a mailbox should then be done using one of the MWI publish functions. This ensures the message is published to the appropriate topic, and the last known state is maintained. * ast_mwi_observer * Add an observer in order to watch for particular MWI module related events. For instance if a submodule needs to know when a subscription is added to any mailbox an observer can be added to watch for that. * other * Urgent message count is now part of the published MWI state object. Also state can be iterated over using defined callbacks. ASTERISK-28442 Change-Id: I93f935f9090cd5ddff6d4bc80ff90703c05cf776 |
7 years ago |
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83c6ebbae8 |
stasis_state: Make unsubscribes NULL tolerant
Regular stasis unsubscribes can handle NULL subscription objects. This patch makes it so stasis state unsubscribes handles NULL's as well. ASTERISK-28442 Change-Id: Ic3648e8df043a85b77cff085e9ff10356028e479 |
7 years ago |
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99addaff69 |
Merge "stasis_state: Add new stasis_state module"
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7 years ago |
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363bafc29e |
stasis_state: Add new stasis_state module
This new module describes an API that can be thought of as a combination of stasis topic pools, and caching. Except, hopefully done in a more efficient and less memory "leaky" manner. The API defines methods, and data structures for managing, and tracking published message state through stasis. By adding a subscriber or publisher, consumers can more easily track the lifetime of the contained state. For instance, when no more publishers and/or subscribers have need of the topic, and associated state its data is removed from the managed container. * stasis_state_manager * The manager stores and well, manages state data. Each state is an association of a unique stasis topic, and the last known published stasis message on that topic. There is only ever one managed state object per topic. For each topic all messages are forwarded to an "all" topic also maintained by the manager. * stasis_state_subscriber * Topic and state can be created, or referenced within the manager by adding a stasis_state_subscriber. When adding a subscriber if no state currently exists new managed state is immediately created. If managed state already exists then a new subscriber is created referencing that state. The managed state is guaranteed to live throughout the subscriber's lifetime. State is only removed from the manager when no other entities require it. * stasis_state_publisher * Topic and state can be created, or referenced within the manager by also adding a stasis_state_publisher. When adding a publisher if no state currently exists new managed state is created. If managed state already exists then a new publisher is created referencing that state. The managed state is guaranteed to live throughout the publisher's lifetime. State is only removed from the manager when no other entities require it. * stasis_state_observer * Some modules may wish to watch for, and react to managed state events. By registering a state observer, and implementing handlers for the desired callbacks those modules can do so. * other * Callbacks also exist that allow consumers to iterate over all, or some of the managed state. ASTERISK-28442 Change-Id: I7a4a06685a96e511da9f5bd23f9601642d7bd8e5 |
7 years ago |
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c2ffb004aa |
tcptls.c: Add peer hostname and port to some error messages
Where possble, hostname and port has been added to error messages, mostly on the server side. ASTERISK-26006 Reported by: Oleksandr Natalenko Change-Id: Iff4f897277bc36ce8c5b493b71d0a4a7b74e62f0 |
7 years ago |
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3a51cdad18 |
Merge "translate.c do not log WARNING on empty audio frame"
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7 years ago |