mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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10.7.0-digiumphones
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10.8.0
10.8.0-digiumphones
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${ noResults }
3037 Commits (3b2b004d699b8cc7b808f62536bb2bc4db8b4e0e)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
af66b0f3f7 |
res_pjsip_outbound_registration.c: Add missing line endings to CLI commands
Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7 |
11 years ago |
|
|
3f0708e5fe |
res_pjsip_outbound_registration.c: Eliminate simple RAII_VAR() usage.
Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e |
11 years ago |
|
|
9ceb848242 |
res_pjsip_outbound_registration.c: Misc code cleanups.
* Break some long lines. * Fix doxygen comment. Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305 |
11 years ago |
|
|
096b27d9d2 |
res_pjsip_outbound_registration.c: Fix whitespace conflict potential.
Change-Id: I82e6e388e3688aebe0783f16c9e0800a747584b5 |
11 years ago |
|
|
bd77ace25a |
Merge "Resolve race conditions involving Stasis bridges."
|
11 years ago |
|
|
d7a1e84a1e |
Resolve race conditions involving Stasis bridges.
This resolves two observed race conditions. First, a bit of background on what the Stasis application does: 1a Creates a stasis_app_control structure. This structure is linked into a global container and can be looked up using a channel's unique ID. 2a Puts the channel in an event loop. The event loop can exit either because the stasis_app_control structure has been marked done, or because of some other factor, such as a hangup. In the event loop, the stasis_app_control determines if any specific ARI commands need to be run on the channel and will run them from this thread. 3a Checks if the channel is bridged. If the channel is bridged, then ast_bridge_depart() is called since channels that are added to Stasis bridges are always imparted as departable. 4a Unlink the stasis_app_control from the container. When an ARI command is received by Asterisk, the following occurs 1b A thread is spawned to handle the HTTP request 2b The stasis_app_control(s) that corresponds to the channel(s) in the request is/are retrieved. If the stasis_app_control cannot be retrieved, then it is assumed that the channel in question has exited the Stasis app or perhaps was never in Stasis in the first place. 3b A command is queued onto the stasis_app_control, and the channel's event loop thread is signaled to run the command. 4b While most ARI commands do nothing further, some, such as adding or removing channels from a bridge, will block until the command they issued has been completed by the channel's event loop. The first race condition that is solved by this patch involves a crash that can occur due to faulty detection of the channel's bridged status in step 3a. What can happen is that in step 2a, the event loop may run the ast_bridge_impart() function to asynchronously place the channel into a bridge, then immediately exit the event loop because the channel has hung up. In step 3a, we would detect that the channel was not bridged and would not call ast_bridge_depart(). The reason that the channel did not appear to be bridged was that the depart_thread that is spawned by ast_bridge_impart() had not yet started. That is the thread where the channel is marked as being bridged. Since we did not call ast_bridge_depart(), the Stasis application would exit, and then the channel would be destroyed Then the depart_thread would start up and try to manipulate the destroyed channel, causing a crash. The fix for this is to switch from using ast_channel_is_bridged() to checking the NULLity of ast_channel_internal_bridge_channel() to determine if ast_bridge_depart() needs to be called. The channel's internal bridge_channel is set when ast_bridge_impart() is called and is NULLed by the call to ast_bridge_depart(). If the channel's internal bridge_channel is non-NULL, then the channel must have been imparted into the bridge and needs to be departed, even if the actual bridging operation has not yet started. By departing the channel when necessary, the thread that is running the Stasis application will block until the bridge gives the okay that the depart_thread has exited. The second race condition that is solved by this patch involves a leak of HTTP handler threads. The problem was that step 2b would successfully retrieve a stasis_app_control structure. Then step 2a would exit the channel from the event loop due to a hangup. Steps 3a and 4a would execute, and then finally steps 3b and 4b would. The problem is that at step 4b, when attempting to add a channel to a bridge, the thread would block forever since the channel would never execute the queued command since it was finished with the event loop. This meant that the HTTP handling thread would be leaked, along with any references that thread may have owned (in my case, I was seeing bridges leaked). The fix for this is to hone in better on when the channel has exited the event loop. The stasis_app_control structure has an is_done field that is now set at each point where the channel may exit the event loop. If step 2b retrieves a valid stasis_app_control structure but the control is marked as done, then the attempted operation exits immediately since there will be nothing to service the attempted command. ASTERISK-25091 #close Reported by Ilya Trikoz Change-Id: If66265b73b4c9f8f58599124d777fedc54576628 |
11 years ago |
|
|
9668a1acb5 |
res_sorcery_memory_cache: Remove 'prefetch' option.
To prevent confusion I am removing the prefetch option until such time as it is implemented. All other functionality, however, has been implemented. ASTERISK-25067 Change-Id: I9ce6aa3e5c6c5bc3c5baa8ff90fa036d73939895 |
11 years ago |
|
|
8c0b917032 |
Merge "Parking: Add documentation for AMI ParkedCallSwap event."
|
11 years ago |
|
|
59552c2d08 |
Parking: Add documentation for AMI ParkedCallSwap event.
This event was added some time ago in order to clarify when a channel took the place of another channel in a parking lot. However, there was no XML documentation added for the event. This patch adds the XML documentation. ASTERISK-24900 #close Reported by Rusty Newton Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac |
11 years ago |
|
|
93ac45d3bd |
res_pjsip: Add option to force G.726 to be treated as AAL2 packed.
Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615 |
11 years ago |
|
|
30cd559345 |
DNS: Need to use the same serializer for a pjproject SIP transaction.
All send/receive processing for a SIP transaction needs to be done under the same threadpool serializer to prevent reentrancy problems inside pjproject when using an external DNS resolver to process messages for the transaction. * Add threadpool API call to get the current serializer associated with the worker thread. * Pick a serializer from a pool of default serializers if the caller of res_pjsip.c:ast_sip_push_task() does not provide one. This is a simple way to ensure that all outgoing SIP request messages are processed under a serializer. Otherwise, any place where a pushed task is done that would result in an outgoing out-of-dialog request would need to be modified to supply a serializer. Serializers from the default serializer pool are picked in a round robin sequence for simplicity. A side effect is that the default serializer pool will limit the growth of the thread pool from random tasks. This is not necessarily a bad thing. * Made pjsip_resolver.c use the requesting thread's serializer to execute the async callback. * Made pjsip_distributor.c save the thread's serializer name on the outgoing request tdata struct so the response can be processed under the same serializer. ASTERISK-25115 #close Reported by: John Bigelow Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a |
11 years ago |
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b23f33e7e5 |
DNS: Fix some corner cases.
* Fix query_set destruction before we are done kicking the queries off. * Fixed no queries requested handling. * Add empty queries request unit test. * Added missing allocation check in ast_dns_query_set_add(). * Made initial pjsip resolving query vector slightly larger. ASTERISK-25115 Reported by: John Bigelow Change-Id: Ie8be8347d0992e93946d72b6e7b1299727b038f2 |
11 years ago |
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ae589da466 |
DNS: Remove trailing newline from summary and descriptions.
Those trailing newlines mess up test formatting. Change-Id: I5e3f3a55b82c9d7acb9661201d4993d1958f1185 |
11 years ago |
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83bc9d366d |
pjsip_resolver.c: Fix debug code to only execute at acceptable debug level.
Change-Id: I1716c93d6e097ad28128ecb9e806aac7a4180c8a |
11 years ago |
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07f5f45e5a |
res_pjsip_transport_websocket: Fix use-after-free bugs.
This patch fixes use-after-free bugs caught by AddressSanitizer. 1. PJSIP transport manager may decide to destroy transport on its own. For example, when the contact registered via websocket has not renewed its registration in time. The transport was destoyed, but the websocket listener thread was still active until the socket closes, and then tried to call transport_shutdown on transport that has been freed. Also, the transport destructor accessed wstransport->rdata.tp_info.pool right after freeing memory that contained wstransport itself. This patch converts transport to an ao2 object, allowing it to be refcounted, so that it is available until both websocket listener and pjsip transport manager are finished with it. 2. The websocket listener deletes the last reference on websocket session when the tcp connection is closed, and it gets destroyed, but the transport manager may still use it, for example when disconnect happens in the middle of a SIP transaction. A new reference to websocket session has been added that is released with the transport to prevent this. ASTERISK-25096 #close Reported by: Josh Kitchens ASTERISK-24963 #close Reported by: Badalian Vyacheslav Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b |
11 years ago |
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8785d0ccbf |
Merge "test_sorcery_memory_cache_thrash: Add unit tests for thrashing the memory cache."
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11 years ago |
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5788c6db67 |
Merge "res_sorcery_memory_cache: Implement expire_on_reload option."
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11 years ago |
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9fca378b36 |
Fixes for OS X
* Add some type casting so tv_usec can really be a long, instead of some strange platform specific type. * Add some .dylib style files to .gitignore. * Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer versions of GCC, when compiling the Homebrew formula for Asterisk, are not properly passing the -Xlinker options to the linker. Given that -Wl, does exactly the [same thing][], and does it properly, this patch changes the -Xlinker options to use -Wl, instead. [reasons unknown]: http://bit.ly/1SUbEYx [same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd |
11 years ago |
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128fe4cee8 |
res_sorcery_memory_cache: Implement expire_on_reload option.
This change implements the expire_on_reload option for memory caches. If enabled and a reload is performed all objects within the cache will be expired and the cache emptied. ASTERISK-25067 Reported by: Matt Jordan Change-Id: Id46aa1957d660556700e689e195eed57c989b85e |
11 years ago |
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028edae82e |
test_sorcery_memory_cache_thrash: Add unit tests for thrashing the memory cache.
This change adds a CLI command which can perform memory cache thrashing as well as unit tests which perform thrashing under the following configurations: 1. Low number of unique objects that go stale after 1 second 2. Low number of unique objects that expire after 1 second 3. Low number of unique objects which are constantly updated 4. Large number of unique objects which exceed a defined cache size 5. Large number of unique objects which exceed a defined cache size that also expire and go stale rapidly 6. Large number of unique objects which expire and go stale rapidly 7. Large number of unique objects For all of the above there are a large number of threads constantly attempting to retrieve random objects and each test runs for a few seconds. ASTERISK-25067 Reported by: Matt Jordan Change-Id: I8c8ceff977332c80ed4a31f10d694d48552b2f78 |
11 years ago |
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86c79314f1 |
Merge "res_sorcery_memory_cache: Add test event when a refresh occurs."
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11 years ago |
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269fbff366 |
Merge "Remove const cast from leaf functions."
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11 years ago |
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19de2bbc5f |
res_sorcery_memory_cache: Add test event when a refresh occurs.
This change adds a testsuite event for when a refresh occurs. This is useful as it provides a guaranteed mechanism of knowing when it has occurred instead of waiting an arbitrary amount of time. ASTERISK-25067 Reported by: Matt Jordan Change-Id: Iaa6b8d2d6bab7f99ee08e1c8908b8272a8987e65 |
11 years ago |
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92ccffd9e6 |
res_pjsip: Prevent access of NULL channels.
It is possible to receive incoming requests or responses after the channel on an ast_sip_session has been destroyed and NULLed out. Handlers of these sorts of requests or responses need to be prepared for the possibility that the channel is NULL or else they could cause a crash. While several places have been amended to deal with NULL channels, there were still a couple of places that needed updating. res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to return early if there is no channel on the session. res_pjsip_session.c: When handling a 302 response, we need to stop the redirecting attempt if there is no channel on the session. ASTERISK-25148 #close reported by Mark Michelson Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9 |
11 years ago |
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d355ee7ff3 |
res_pjsip/location: Fix ref leak in contact_apply_handler
contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status to force the creation of a contact_status object whenever a new contact is added but it didn't unref the returned object. Added an ao2_cleanup(status) to plug the leak. ASTERISK-25141 Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40 Reported-by: Corey Farrell |
11 years ago |
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6d8dc9bb5c |
res_pjsip: Remove outgoing authentication code no longer needed.
Associated with ASTERISK-25131 Change-Id: Iefa3b2066cfd8b108a90d2dd4a64d92c3a195d33 |
11 years ago |
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00a47ffc7e |
res_pjsip_session: Fix cherry pick to master compile error.
ASTERISK-25131 Reported by: Richard Mudgett Change-Id: I87c9c96ae4a8fe2bc8a0ddea6958a2ad9cefd8e3 |
11 years ago |
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9472bbaa95 |
Remove const cast from leaf functions.
app_control_register_rule and app_control_unregister_rule lock/unlock the queue, which is a mutating operation according to the ao2_lock/_unlock prototype. Depending on the specific (implicit) casts in SCOPED_LOCK and RAII_VAR, the compiler may warn or not. As the only callers of those functions do not have the const, get consistent results by just dropping it. Change-Id: Ib9e6296155a39bc5d627142a3828180c3cfe8fbb |
11 years ago |
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af420ba4ae |
Merge "res_pjsip_session: Fix in-dialog authentication."
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11 years ago |
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3906175426 |
Merge "res_sorcery_memory_cache: Add CLI commands and AMI actions."
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11 years ago |
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34bb5ca97c |
Merge "res_sorcery_memory_cache: Add support for refreshing stale objects."
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11 years ago |
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5cdcae5240 |
res_pjsip_session: Fix in-dialog authentication.
When the remote peer requires authentication for in-dialog requests then re-INVITEs to the peer cause the call to be disconnected and other in-dialog requests to the peer like MESSAGE just don't go through. * Made session_inv_on_tsx_state_changed() handle in-dialog authentication for re-INVITEs and other methods. Initial INVITEs cannot be handled here because the INVITE transaction must be restarted earlier. * Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in preparation for removing the file. The generic outbound authentication code did not work as well as anticipated. * Created outbound_invite_auth() to only handle initial outbound INVITEs. Re-INVITEs cannot be handled here. The re-INVITE transaction is still in progress and the PJSIP library cannot handle the overlapping INVITE transactions. Other method types should not be handled here as this code only works on outgoing calls and we need to handle incoming and outgoing calls. ASTERISK-25131 #close Reported by: Richard Mudgett Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0 |
11 years ago |
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9f1939ee27 |
pjsip_configuration: Fix leak in persistent_endpoint_update_state.
The loop to find the first available contact of an endpoint grabbed contact from the iterator, then checked for offline state. This caused the first contact after the state was found to leak a reference. ASTERISK-25141 Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08 |
11 years ago |
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bef000dd7c |
res_pjsip/location: Fix memory leak in permanent_uri_handler
When permanent_uri_handler was creating the contact status object for each contact, it wasn't unreffing it at the end of the loop. ASTERISK-25141 #close Reported-by: Corey Farrell Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12 |
11 years ago |
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dfc45254d1 |
res_sorcery_memory_cache: Add CLI commands and AMI actions.
This change adds the following CLI commands and AMI actions: sorcery memory cache show sorcery memory cache dump sorcery memory cache expire sorcery memory cache stale SorceryMemoryCacheExpire SorceryMemoryCacheExpireObject SorceryMemoryCacheStale SorceryMemoryCacheStaleObject These allow both examination and manipulation of sorcery memory caches from external sources. Cached objects can be explicitly expired from a cache or marked as stale. If expired they are immediately removed. If marked as stale they will be background refreshed when next retrieved. ASTERISK-25067 Reported by Matt Jordan Change-Id: I68e03cfd8c34b5e07f4b6ee4fd93a3f4a00a3d9e |
11 years ago |
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9159abb158 |
Merge "res/res_config_pgsql.c: Use PQescapeStringConn for escaping names."
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11 years ago |
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2e54e7227c |
res_sorcery_memory_cache: Add support for refreshing stale objects.
This change introduces a check of object_lifetime_stale when retrieving cached objects. If the amount of time the object has been in the cache exceeds the lifetime, then a task is scheduled to update the cached object based on an object retrieved from other sorcery wizards instead. To prevent the cached object from being retrieved during a refresh, thread-local storage is used to mark the thread as being a stale object update. This results in the cache returning no object, leading to sorcery querying other wizards for the object instead. A test has been added for stale objects as well. This test ensures that stale objects are retrieved the same as freshly-cached objects. The test also ensures that after an object is stale, changes in the backend are reflected in the cache, to include if the object has been deleted from the backend. ASTERISK-25067 Reported by Matt Jordan Change-Id: I9bd7c049adf6939bfe2899f393c2bfbbf412d217 |
11 years ago |
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b8ac683822 |
res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes
Add a new ContactStatus AMI event. Publish the following status/state changes: Created Removed Reachable Unreachable Unknown Contact URI, new status/state, aor and endpoint names, and the last qualify rtt result are included in the event. ASTERISK-25114 #close Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> |
11 years ago |
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95b186a174 |
res/res_config_pgsql.c: Use PQescapeStringConn for escaping names.
Use function PQescapeStringConn for escaping the name of the table and schema instead of doing it manually. ASTERISK-25132 #close Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> Change-Id: I302a263f7210d20925f14716b508b081998b7608 |
11 years ago |
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e1a64e021b |
Merge "Stasis: Fix unsafe use of stasis_unsubscribe in modules."
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11 years ago |
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70d54ab6c4 |
res_pjsip_transport_websocket: Fix crash on receiving large SIP packets
Incoming SIP packets larger than PJSIP_MAX_PKT_LEN were themselves truncated before passing to pjsip_tpmgr_receive_packet, but the length was passed unaltered, thus causing memory corruption and segfault. ASTERISK-25122 #close Change-Id: I608a6b6b7f229eacc33a0a7d771d18e27e5b08ab |
11 years ago |
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50044fdc15 |
Stasis: Fix unsafe use of stasis_unsubscribe in modules.
Many uses of stasis_unsubscribe in modules can be reached through unload. These have been switched to stasis_unsubscribe_and_join. Some subscription callbacks do nothing, for these I've created a noop callback function in stasis.c. This is used by some modules that monitor MWI topics in order to enable cache, since the callback does not become invalid after dlclose it is safe to use stasis_unsubscribe on these, even during module unload. ASTERISK-25121 #close Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c |
11 years ago |
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f66c41e668 |
res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS
In addition to specifying lists of 'presence' and 'message-summary', users can also create lists of type 'dialog'. These should be treated in the same fashion as 'presence'. Change-Id: I583bb69cd9f88b0b29bf09ddaddeac4e84189f6e |
11 years ago |
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ad7192a8fd |
res/res_pjsip_exten_state: Fix confusing NOTICE message
When a SUBSCRIBE request is made to a dialplan hint that doesn't exist, the current NOTICE message informing users of this swaps the context and extension parameters. This can cause a bit of confusion. Thanks to CptBurger in #asterisk for helping to point this out. Change-Id: Ie584d1a58ae217385c87a450ca25b55ca0e36e43 |
11 years ago |
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f7dc49b1f0 |
Merge "res_sorcery_memory_cache: Add support for object_lifetime_maximum."
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11 years ago |
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9cffcca5f9 |
res/ari: Register Stasis application on WebSocket attempt
Prior to this patch, when a WebSocket connection is made, ARI would not be informed of the connection until after the WebSocket layer had accepted the connection. This created a brief race condition where the ARI client would be notified that it was connected, a channel would be sent into the Stasis dialplan application, but ARI would not yet have registered the Stasis application presented in the HTTP request that established the WebSocket. This patch resolves this issue by doing the following: * When a WebSocket attempt is made, a callback is made into the ARI application layer, which verifies and registers the apps presented in the HTTP request. Because we do not yet have a WebSocket, we cannot have an event session for the corresponding applications. Some defensive checks were thus added to make the application objects tolerant to a NULL event session. * When a WebSocket connection is made, the registered application is updated with the newly created event session that wraps the WebSocket connection. ASTERISK-24988 #close Reported by: Joshua Colp Change-Id: Ia5dc60dc2b6bee76cd5aff0f69dd53b36e83f636 |
11 years ago |
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d7086a27b4 |
Merge "res_sorcery_memory_cache: Add support for maximum_objects."
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11 years ago |
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5aa1c30b31 |
Merge "res_pjsip: Refactor endpt_send_transaction (qualify_timeout)"
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11 years ago |
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242306ade3 |
Merge "res_pjsip_outbound_registration: Check request URI for line."
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11 years ago |
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29ef6571cb |
res_pjsip: Refactor endpt_send_transaction (qualify_timeout)
This patch refactors the transaction timeout processing to eliminate calling the lower level public pjsip functions and reverts to calling pjsip_endpt_send_request again. This is the result of me noticing a possible incompatibility with pjproject-2.4 which was causing contact status flapping. The original version of this feature used the lower level calls to get access to the tsx structure in order to cancel the transaction when our own timer expires. Since we no longer have that access, if our own timer expires before the pjsip timer, we call the callbacks and just let the pjsip transaction take it's own course. When the transaction ends, it discovers the callbacks have already been run and just cleans itself up. A few messages in pjsip_configuration were also added/cleaned up. ASTERISK-25105 #close Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> |
11 years ago |