mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
30146 Commits (5dffbe11abc4e488c9ebb8822b9d169d8a2a45b2)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
5dffbe11ab |
CI: Initial commit for moving CI into source repo
Create tests/CI directory and add files used by Jenkins to build and test Asterisk. With this commit, Jenkins will run the Asterisk Unit Tests using the Jenkinsfile at tests/CI/unittests.jenkinsfile. Bash scripts to do the actual building and testing are also in the same directory. Output is placed in tests/CI/output so that directory has been added to .gitignore. Change-Id: I9448065465e6de2b878634510ace8fd1ef378608 |
8 years ago |
|
|
8273b8cbf3 |
test.c: Make output jUnit compatible
Separate "name" into "classname" and "name". Use '.' for classname separator instead of '/'. Prefix reserved words with '_'. Wrap output with a top-level "testsuites" element. Change-Id: Iec1a985eba1c478e5c1d65d5dfd95cb708442099 |
8 years ago |
|
|
3cccb60321 |
AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses.
When endpoint specific ACL rules block a SIP request they respond with a 403 forbidden. However, if an endpoint is not identified then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints. * Made endpoint specific ACL rules now respond with a 401 unauthorized which is the same as if an endpoint were not identified. The fix is accomplished by replacing the found endpoint with the artificial endpoint which always fails authentication. ASTERISK-27818 Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32 |
8 years ago |
|
|
f5bc8aeb9a |
app_sendtext: Allow content types other than text/plain
There was no real reason to limit the conteny type to text/plain other than that's what it was limited to before. Now any text/* content type will be allowed for channel drivers that don't support enhanced messaging and any type will be allowed for channel drivers that do support enhanced messaging. Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9 |
8 years ago |
|
|
613363adee |
Merge "app_sendtext: Enhance SendText to support Enhanced Messaging" into certified/13.21
|
8 years ago |
|
|
8f5fc3870d |
app_sendtext: Enhance SendText to support Enhanced Messaging
SendText now accepts new channel variables that can be used to override the To and From display names and set the Content-Type of a message. Since you can now set Content-Type, other text/* content types are now valid. Change-Id: I648b4574478119f95de09d9f08e9595831b02830 |
8 years ago |
|
|
c1deeb28c2 |
bridge_softmix: Forward TEXT frames
Core bridging and, more specifically, bridge_softmix have been
enhanced to relay received frames of type TEXT or TEXT_DATA to all
participants in a softmix bridge. res_pjsip_messaging and
chan_pjsip have been enhanced to take advantage of this so when
res_pjsip_messaging receives an in-dialog MESSAGE message from a
user in a conference call, it's relayed to all other participants
in the call.
res_pjsip_messaging already queues TEXT frames to the channel when
it receives an in-dialog MESSAGE from an endpoint and chan_pjsip
will send an MESSAGE when it gets a TEXT frame. On a normal
point-to-point call, the frames are forwarded between the two
correctly. bridge_softmix was not though so messages weren't
getting forwarded to conference bridge participants. Even if they
were, the bridging code had no way to tell the participants who
sent the message so it would look like it came from the bridge
itself.
* The TEXT frame type doesn't allow storage of any meta data, such
as sender, on the frame so a new TEXT_DATA frame type was added that
uses the new ast_msg_data structure as its payload. A channel
driver can queue a frame of that type when it receives a message
from outside. A channel driver can use it for sending messages
by implementing the new send_text_data channel tech callback and
setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech
properties. If set, the bridging/channel core will use it instead
of the original send_text callback and it will get the ast_msg_data
structure. Channel drivers aren't required to implement this. Even
if a TEXT_DATA enabled driver uses it for incoming messages, an
outgoing channel driver that doesn't will still have it's send_text
callback called with only the message text just as before.
* res_pjsip_messaging now creates a TEXT_DATA frame for incoming
in-dialog messages and sets the "from" to the display name in the
"From" header, or if that's empty, the caller id name from the
channel. This allows the chat client user to set a friendly name
for the chat.
* bridge_softmix now forwards TEXT and TEXT_DATA frames to all
participants (except the sender).
* A new function "ast_sendtext_data" was added to channel which
takes an ast_msg_data structure and calls a channel's
send_text_data callback, or if that's not defined, the original
send_text callback.
* bridge_channel now calls ast_sendtext_data for TEXT_DATA frame
types and ast_sendtext for TEXT frame types.
* chan_pjsip now uses the "from" name in the ast_msg_data structure
(if it exists) to set the "From" header display name on outgoing text
messages.
Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489
(cherry picked from commit
|
8 years ago |
|
|
72fb285d9b |
bridge_softmix: Forward TEXT frames
Core bridging and, more specifically, bridge_softmix have been
enhanced to relay received frames of type TEXT or TEXT_DATA to all
participants in a softmix bridge. res_pjsip_messaging and
chan_pjsip have been enhanced to take advantage of this so when
res_pjsip_messaging receives an in-dialog MESSAGE message from a
user in a conference call, it's relayed to all other participants
in the call.
res_pjsip_messaging already queues TEXT frames to the channel when
it receives an in-dialog MESSAGE from an endpoint and chan_pjsip
will send an MESSAGE when it gets a TEXT frame. On a normal
point-to-point call, the frames are forwarded between the two
correctly. bridge_softmix was not though so messages weren't
getting forwarded to conference bridge participants. Even if they
were, the bridging code had no way to tell the participants who
sent the message so it would look like it came from the bridge
itself.
* The TEXT frame type doesn't allow storage of any meta data, such
as sender, on the frame so a new TEXT_DATA frame type was added that
uses the new ast_msg_data structure as its payload. A channel
driver can queue a frame of that type when it receives a message
from outside. A channel driver can use it for sending messages
by implementing the new send_text_data channel tech callback and
setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech
properties. If set, the bridging/channel core will use it instead
of the original send_text callback and it will get the ast_msg_data
structure. Channel drivers aren't required to implement this. Even
if a TEXT_DATA enabled driver uses it for incoming messages, an
outgoing channel driver that doesn't will still have it's send_text
callback called with only the message text just as before.
* res_pjsip_messaging now creates a TEXT_DATA frame for incoming
in-dialog messages and sets the "from" to the display name in the
"From" header, or if that's empty, the caller id name from the
channel. This allows the chat client user to set a friendly name
for the chat.
* bridge_softmix now forwards TEXT and TEXT_DATA frames to all
participants (except the sender).
* A new function "ast_sendtext_data" was added to channel which
takes an ast_msg_data structure and calls a channel's
send_text_data callback, or if that's not defined, the original
send_text callback.
* bridge_channel now calls ast_sendtext_data for TEXT_DATA frame
types and ast_sendtext for TEXT frame types.
* chan_pjsip now uses the "from" name in the ast_msg_data structure
(if it exists) to set the "From" header display name on outgoing text
messages.
Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489
(cherry picked from commit
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8 years ago |
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8aa2222087 |
Merge "res_pjsip_endpoint_identifier_ip: Unregister the module for headers." into certified/13.21
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8 years ago |
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5de8b00f2c |
res_pjsip_endpoint_identifier_ip: Unregister the module for headers.
Asterisk uses Reference Counting to track whether a module can be unloaded. Every consumer who requires a module, increases the reference count. When the consumer goes, is unloaded itself, it has to decrease the reference count on all its used/required modules. That way core stop gracefully works on the command-line interface (CLI): One module after the other is unloaded. A recent change broke this for the module res_pjsip. ASTERISK-27861 Change-Id: I261abcb411d026bbb0691cc78f28300bfd3103a3 |
8 years ago |
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5aaf6d1605 |
res_pjsip: Register pjsip_transport_management not externally but internally.
The module (res_)pjsip_transport_management got moved into res_pjsip. It is no longer an independent/external module with (un)load_module and therefore has to register just internally with res_pjsip. ASTERISK-27860 Change-Id: Icd0413be7d2e98b92f51e6d6c353f2570bb4be95 |
8 years ago |
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13a85290fe |
Initialize 13.21-cert branch
A new branch was created for what will be Asterisk certified 13.21. A couple of things needed to be done to the branch in order to complete initialization: Modified the version file to reflect the certified version. Updated all extended modules to be disabled by default. Change-Id: Ie1dd3cb146391dea92c9e3ef906dde8d7241fee2 |
8 years ago |
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32f362c896 |
Update for 13.21.0-rc1
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8 years ago |
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b3cec9d445 |
Merge "res_pjsip: Fix deadlock on reliable transport shutdown." into 13
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8 years ago |
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ada0676e1d |
Merge "Build System: Enable python3 compatibility." into 13
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8 years ago |
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d474763a2c |
Merge "utils/pval: Add -lBlocksRuntime for compiler clang conditionally." into 13
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8 years ago |
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57aca68bbf |
Merge "utils: Add ast_assert_return" into 13
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8 years ago |
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e516c34c02 |
Merge "res_musiconhold: Don't restart MOH from beginning after announcement." into 13
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8 years ago |
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92f88f164a |
utils/pval: Add -lBlocksRuntime for compiler clang conditionally.
ASTERISK-27809 Change-Id: I930b364a33d54cc08dedfcd5bb45f7e83242f134 |
8 years ago |
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39c51394c8 |
utils: Add ast_assert_return
Similar to pjproject's PJ_ASSERT_RETURN macro, this one will do the following... If the assert passes... NoOp If the assert fails and AST_DEVMODE is defined, execute ast_assert() then, if DO_CRASH isn't set, return from the calling function with the supplied value. If the assert fails and AST_DEVMODE is not defined, return from the calling function with the supplied value. The macro will execute a return without a value if one isn't suppled. Change-Id: I0003844affeab550d5ff5bca7aa7cf8a559b873e |
8 years ago |
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ad0ba520b5 |
Merge "pjsip_scheduler.c: Add ability to trace scheduled tasks." into 13
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8 years ago |
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fb9cbd524b |
Merge "res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations." into 13
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8 years ago |
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f83f5c48b2 |
Merge "pjsip_scheduler.c: Fix some corner cases." into 13
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8 years ago |
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f2e140e8a7 |
res_musiconhold: Don't restart MOH from beginning after announcement.
This reverts a problem introduced by the fix for ASTERISK_24329. Now, when an announcement is played while waiting in a queue, music on hold will not restart from the beginning of the sound file and will instead pick up where it left off. However, the incorrect behavior in ASTERISK_24329 is now present again; if an announcement X seconds long is played when music on hold starts, music on hold will start X seconds into the file. ASTERISK-27774 #close Reported by: lvl Change-Id: I86b2885ee7063268f9b9747eddb788336ade989b |
8 years ago |
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b92ebdba5f |
pjsip_scheduler.c: Add ability to trace scheduled tasks.
When a scheduled task is created you can pass in the AST_SIP_SCHED_TASK_TRACK flag. This new flag causes scheduling events to be logged. Change-Id: I91967eb3d5a220915ce86881a28af772f9a7f56b |
8 years ago |
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12aa25b2e1 |
res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations.
ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3 |
8 years ago |
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dfdc9a2575 |
pjsip_scheduler.c: Fix some corner cases.
* Fix the periodic interval wander because it may take significant time between the sched thread queueing the task in the serializer and the serializer actually executing the task. The time it takes to actually execute the task was already taken into account. * Pass a schtd ref to the serializer when we queue a scheduled task on the serializer. We don't want it going away on us while it is in the serializer queue. * Skip the scheduled task if the task was canceled between queueing the task to the serializer and the serializer actually executing the task. * Reorder struct ast_sip_sched_task to avoid unnecessary padding. Removed task_id and added next_periodic. * Hold a ref to the passed in serializer so the serializer cannot go away on the scheduled task. ASTERISK_26806 Change-Id: I6c8046b75f6953792c8c30e55b836a4291143f24 |
8 years ago |
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af36823907 |
pjsip_scheduler.c: Sort "pjsip show scheduled_tasks" output.
* A side benefit is that the scheduled tasks are not completely blocked while the CLI command executes. * Adjusted the "Task Name" column width to have more room for longer names. Change-Id: Iec64aa463ee8b10eef90120e00c38b1fb444087e |
8 years ago |
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264af6bec0 |
Merge "res_pjsip_notify.c: enable in-dialog NOTIFY" into 13
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8 years ago |
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dfd0529abc |
Merge "pjsip_scheduler.c: Fix ao2 usage errors." into 13
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8 years ago |
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6aee468ba1 |
Merge "Build System: Strip '-std=c99' from CFLAGS provided by libraries." into 13
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8 years ago |
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7886354343 |
cdr_mysql: Compile error because MYSQL_PORT definition is missing
If it is not defined, it will add MYSQL_PORT definition. After some research on MySQL/MariaDB development tree, I couldn't find any reference to MYSQL_PORT definition in include files. ASTERISK-27782 #close Change-Id: Ieee56c836fc2e8bd021c456145bba04c6068bb77 |
8 years ago |
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724d926d46 |
res_pjsip_session: Rewrite o= with external_media_address.
It now appends the external IP address on the o= line of the SDP packet. The decision was made to write the numeric IP address as opposed to the RFC that states the FQDN should be used if and when available. We believe the usage of literal IP address will help avoid potential problems. ASTERISK-27614 #close Change-Id: I84f3360f3606b8c4e8d161edb228799ec0b8a302 |
8 years ago |
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6a4afe09ce |
res_pjsip_notify.c: enable in-dialog NOTIFY
This patch adds support to send in-dialog SIP NOTIFY commands on chan_pjsip channels, similar to the functionality recently added for chan_sip (ASTERISK_27461). This extends res_pjsip_notify to allow for in-dialog messages. ASTERISK-27697 Change-Id: If7f3151a6d633e414d5dc319d5efc1443c43dd29 |
8 years ago |
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0edc4ade93 |
Merge "res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge" into 13
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8 years ago |
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483d874836 |
Merge "chan_sip.c: Fix INVITE with replaces channel ref leak." into 13
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8 years ago |
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c4f02c975b |
pjsip_scheduler.c: Fix ao2 usage errors.
* Removed several invalid uses of OBJ_NOLOCK. These uses resulted in the 'tasks' container being accessed without a lock in a multi-threaded environment. A recipe for crashes. * Removed needlessly obtaining schtd object references. If the caller providing you a pointer to an object doesn't have a valid reference then you cannot safely get one from it. * Getting a ref to 'tasks' when you aren't copying the pointer into another location is useless. The 'tasks' container pointer is global. * Removed many unnecessary uses of RAII_VAR. * Make ast_sip_schedule_task() name parameter const. ASTERISK_26806 Change-Id: I5c62488e651314e2a1dbc01f5b078a15512d73db |
8 years ago |
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36f94cbcde |
Merge "Build System: Fixes for configure script." into 13
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8 years ago |
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7593359787 |
Build System: Enable python3 compatibility.
* Consistently use spaces in rest-api-templates/asterisk_processor.py. * Exclude third-party from docs/full-en_US.xml. * Add docs/full-en_US.xml to .gitignore. * Use list() to convert python3 view. * Use python3 print function. * Replace cmp() with equivalent equation. * Replace reference to out of scope subtype variable with name parameter. * Use unescaping triple bracket notation in mustache templates where needed. This causes behavior of Python2 to be maintained when using Python3. * Fix references to has_websocket / is_websocket in res_ari_resource.c.mustache. * Update calculation of has_websocket to use any(). * Use unicode mode for writing output file in transform.py. * Replace 'from swagger_model import *' with explicit import of required symbols. * Add missing 'import os' * Fix invalid reference to swagger_version from exception handler. I have not tested voicemailpwcheck.py, only the print syntax has been fixed. Change-Id: If5c5b556a2800d41a3e2cfef080ac2e151178c33 |
8 years ago |
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72b16ee400 |
res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge
There is a problem when an INVITE-with-Replaces transfer targets a channel in a ConfBridge. The transfer will unconditionally swap out the ConfBridge channel. Unfortunately, the ConfBridge state will not be aware of this change. Unexpected behavior will happen as a result since ConfBridge channels currently can only be replaced by a masquerade and not normal bridge channel moves. * We just need to pretend that the channel isn't in a bridge (like other transfer methods already do) so the transfer channel will masquerade into the ConfBridge channel. Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82 |
8 years ago |
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dcac292ae3 |
Merge "BuildSystem: Add support for building RADIUS with radcli." into 13
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8 years ago |
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9117bbb45d |
Merge "res_pjsip: Update authenticate_qualify documentation." into 13
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8 years ago |
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ea055386e0 |
chan_sip.c: Fix INVITE with replaces channel ref leak.
Given the below call scenario: A -> Ast1 -> B C <- Ast2 <- B 1) A calls B through Ast1 2) B calls C through Ast2 3) B transfers A to C When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to send an INVITE with replaces to Ast2. Ast2 then leaks a channel ref of the channel between Ast1 and Ast2. Channel ref leaks are easily seen in the CLI "core show channels" output. The leaked channels appear in the output but you can do nothing with them and they never go away unless you restart Asterisk. * Properly account for the channel refs when imparting a channel into a bridge when handling an INVITE with replaces in handle_invite_replaces(). The ast_bridge_impart() function steals a channel ref but the code didn't account for how many refs were held by the code at the time and which ref was stolen. * Eliminated RAII_VAR in handle_invite_replaces(). ASTERISK-27740 Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4 |
8 years ago |
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e94f8e4a24 |
res_pjsip: Update authenticate_qualify documentation.
Change-Id: I3811de0014b1ffe96d4a3b49cddd5d4ca02ee5d4 |
8 years ago |
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906db6a3ff |
app_agent_pool.c: Fix off nominal ref leak.
Change-Id: Ib427ffc2c802620eaafb08b1c2a17dddd8fb8eb6 |
8 years ago |
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07e408e588 |
Build System: Strip '-std=c99' from CFLAGS provided by libraries.
Asterisk requires GNU C extensions. On some systems certain libraries may incorrectly push -std=c99 into CFLAGS, thus breaking the build. This change causes that flag to be stripped so the Asterisk build is not broken by those libraries. This change is made for both pkgconfig and tool based libraries. ASTERISK-27629 #close Change-Id: I13389613b194abbac77becf90cd950dc168704db |
8 years ago |
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01b3c435d8 |
Build System: Fixes for configure script.
* Replace all 'else if' statements with 'elif'. * Use loop to detect versioned lua headers and libraries. The loop for detecting lua fixes a bug where LUA_INCLUDE would be appended with the directory of every lua version after the first one is found. Change-Id: I3276f9aee955014108345be6092f51c932b43a0f |
8 years ago |
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7888b6115a |
Merge "install_prereq: Add Slackware (somehow)." into 13
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8 years ago |
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cea1a22ef3 |
res_pjsip: Correct usages of pjproject's timer heap
Fix some timer heap initializations and cancels to try and prevent crashes and timer heap issues. Change-Id: I64885d190fa22097d1b55987091375541e57a7ee |
8 years ago |
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158c0efd4f |
pjroject_bundled: Add already-destroyed check to tsx_timer_callback
There have been cases that when the transaction timer callback is called the tsx is already destroyed. This causes a crash. We now check the tsx state and return if the tsx is already destroyed. Change-Id: If93acd5e48d9ca5bb553f2405d5afc836842fe1c |
8 years ago |