mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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1.6.0.8
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1.6.1-beta3
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10.10.0
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10.12.0
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10.12.2-digiumphones
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10.12.3-digiumphones
10.12.4
10.12.4-digiumphones
10.2.0
10.2.0-rc1
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10.3.0
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10.3.1
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10.4.0-digiumphones-rc1
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10.5.0
10.5.0-digiumphones
10.5.0-digiumphones-rc1
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10.5.0-rc1
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10.5.1-digiumphones
10.5.2
10.5.2-digiumphones
10.6.0
10.6.0-digiumphones
10.6.0-digiumphones-rc1
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10.6.1
10.6.1-digiumphones
10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
10.7.0-rc1
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10.7.1-digiumphones
10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
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10.9.0
10.9.0-digiumphones
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11.0.0
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11.1.1
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11.10.0
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11.11.0
11.11.0-rc1
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11.16.0
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11.2.1
11.2.2
11.20.0-rc1
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11.21.0
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12.0.0
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13.0.0
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13.0.1
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certified/1.8.11-cert1
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${ noResults }
5187 Commits (dde58df3182b7f99714e446a684d8ddb81f759bb)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
c2f68787a3 |
Merge "cel: Ensure only one dial status per channel exists." into 13
|
10 years ago |
|
|
ecc186a4cc |
Merge "stasis: Add setting subscription congestion levels." into 13
|
10 years ago |
|
|
e842a99e7c |
Merge "sorcery: Add setting object type congestion levels." into 13
|
10 years ago |
|
|
1ac47c5aae |
Merge "taskprocessors: Implement high/low water mark alerts." into 13
|
10 years ago |
|
|
caf6cccc5c |
cel: Ensure only one dial status per channel exists.
CEL wrongly assumed that a channel would only have a single dial event on it. This is incorrect. Particularly in a queue each call attempt to a member will result in a dial event, adding a new dial status in CEL without removing the old one. This would cause the container to grow with only one dial status being removed when the channel went away. The other dial status entries would remain leaking memory. This change fixes the memory leak by ensuring that only one dial status will only ever exist for each channel. The behavior during the scenario where multiple events are received has also been improved. For failure cases the first failure will be the dial status. If an answer dial status is received, though, it will take priority and the dial status for the channel will be answer. Memory usage has also been decreased by storing the minimal amount of information and the code has been cleaned up slightly. ASTERISK-25262 #close Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe |
10 years ago |
|
|
bd3b6e9f19 |
Merge "astfd: Not maximum size of a single file but maximum file descriptors." into 13
|
10 years ago |
|
|
c09af61c76 |
Merge "BuildSystem: Avoid 'ar cru' and use 'ar cr' instead." into 13
|
10 years ago |
|
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f834852137 |
Merge "Fix #include poll.h and sys/cdefs.h" into 13
|
10 years ago |
|
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0d84421f93 |
astfd: Not maximum size of a single file but maximum file descriptors.
With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", the maximum size of a single file was shown. Now, the maximum number of possible file descriptors is shown. ASTERISK-26097 Change-Id: Icf98d145774b38cac144ca76d19eaef42ce659a3 |
10 years ago |
|
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c9b873add8 |
Merge "ari/resource_channels: Add 'formats' to channel create/originate" into 13
|
10 years ago |
|
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9c5a0b814b |
Fix #include poll.h and sys/cdefs.h
POSIX defines poll.h, sys/poll.h should not be used at is c-library internal header which may or may not exist. Notable in musl it generates warning of being incorrect. And add explict include of sys/cdefs.h where needed. Change-Id: I142930df53fe7585a06b854b6faddc5301e024be |
10 years ago |
|
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557333ea4c |
stasis: Add setting subscription congestion levels.
Stasis subscriptions and message routers create taskprocessors to process the event messages. API calls are needed to be able to set the congestion levels of these taskprocessors for selected subscriptions and message routers. * Updated CDR, CEL, and manager's stasis subscription congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty call setup/teardown activity from triggering the taskprocessor overload alert. CDRs in particular need an extra high congestion level because they can take awhile to process the stasis messages. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: Id0a716394b4eee746dd158acc63d703902450244 |
10 years ago |
|
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110d772467 |
sorcery: Add setting object type congestion levels.
Sorcery creates taskprocessors for object types to process object observer callbacks. An API call is needed to be able to set the congestion levels of these taskprocessors for selected object types. * Updated PJSIP's contact and contact_status sorcery object type observer default congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty register/unregister and subscribe/unsubscribe activity from triggering the taskprocessor overload alert. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6 |
10 years ago |
|
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610eee2a36 |
taskprocessors: Implement high/low water mark alerts.
When taskprocessors get backed up, there is a good chance that we are being overloaded and need to defer adding new work to the system. * Implemented a high/low water alert mechanism for modules to check if the system is being overloaded and take appropriate action. When a taskprocessor is created it has default congestion levels set. A taskprocessor can later have those congestion levels altered for specific needs if stress testing shows that the taskprocessor is a symptom of overloading or needs to handle bursty activity without triggering an overload alert. * Add CLI "core show taskprocessor" low/high water columns. * Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was never a good thing to use when creating a taskprocessor because of the nature of how its references needed to be cleaned up on a partial creation. * Made res_pjsip's distributor check if the taskprocessor overload alert is active before placing a message representing brand new work onto a distributor serializer. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I182f1be603529cd665958661c4c05ff9901825fa |
10 years ago |
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d38b8e6399 |
BuildSystem: Avoid 'ar cru' and use 'ar cr' instead.
In several internal library projects, the files are archived with the help of 'ar cr'. Only the projects editline and the Objective Open H.323 stack implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier ignored since `D' is the default (see `U')". For consistency and to avoid this message all projects use 'ar cr' now. ASTERISK-26091 #close Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40 |
10 years ago |
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93237209eb |
Merge "core/dial: New channel variable FORWARDERNAME" into 13
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10 years ago |
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c27c232057 |
ari/resource_channels: Add 'formats' to channel create/originate
If you create a local channel and don't specify an originator channel to take capabilities from, we automatically add all audio formats to the new channel's capabilities. When we try to make the channel compatible with another, the "best format" functions pick the best format available, which in this case will be slin192. While this is great for preserving quality, it's the worst for performance and overkill for the vast majority of applications. In the absense of any other information, adding all formats is the correct thing to do and it's not always possible to supply an originator so a new parameter 'formats' has been added to the channel create/originate functions. It's just a comma separated list of formats to make availalble for the channel. Example: "ulaw,slin,slin16". 'formats' and 'originator' are mutually exclusive. To facilitate determination of format names, the format name has been added to "core show codecs". ASTERISK-26070 #close Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b |
10 years ago |
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e2132dd358 |
logging,cdr,cel: Fix stringfield memory leak.
The stringfields refactor to allow adding stringfields to the end of a
structure (
|
10 years ago |
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2de58c6d01 |
core/dial: New channel variable FORWARDERNAME
Added a new channel variable FORWARDERNAME which indicates which channel was responsible for a forwarding requests received on dial attempt. Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. ASTERISK-26059 #close Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2 |
10 years ago |
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e3c9ad6382 |
Merge "res_pjsip: add "via_addr", "via_port", "call_id" to contact" into 13
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10 years ago |
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230686f4ec |
res_pjsip: add "via_addr", "via_port", "call_id" to contact
As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contact. Added new fields ViaAddress, CallID to AMI event ContactStatus. ASTERISK-26011 Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576 |
10 years ago |
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a42bea3314 |
threadpool: Fix potential data race.
worker_start checked for ZOMBIE status without holding a lock. All other read/write of worker status are performed with a lock, so this check should do the same. ASTERISK-25777 #close Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781 |
10 years ago |
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935e0496c4 |
udptl: Don't eat sequence numbers until OK is received
Scenario: Local fax -> Asterisk w/ firewall -> Provider -> Remote fax * Local fax starts rtp call to remote fax * Remote fax starts t38 call back to local fax. * Local fax sends t38 no-signal to Asterisk before sending an OK. * udptl processes the frame and increments the expected sequence number. * chan_sip drops the frame because the call isn't up so nothing goes out the external interface to open the port for incoming packets. * Local fax sends OK and Asterisk sends OK to the remote fax. * Remote fax sends t38 packets which are dropped by the firewall. * Local fax re-sends t38 no-signal with the same sequence number. * udptl drops the frame because it thinks it's a dup. * Still no outgoing packets to open the firewall. * t38 negotiation fails. The patch drops frames t38 received before udptl sequence processing when the call hasn't been answered yet. The second no-signal frame is then seen as new and is relayed out the external interface which opens the port and allows negotiation to continue. ASTERISK-26034 #close Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9 |
10 years ago |
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12db14b9b1 |
Merge "logger: Add PID to syslog messages." into 13
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10 years ago |
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02d30e171e |
Use doubles instead of floats for conversions when comparing strings.
In 13.9.0, there was an issue where PJSIP contacts added to an AOR would be deleted at seemingly random times. One reason this was happening was because of an operation to retrieve the contacts whose expiration time was less than or equal to the current time. When retrieving existing contacts, the contact's expiration time and the current time were converted from a string to a float, and those two floats were compared. On some systems, including mine, this conversion was horribly off. For instance, I could regularly see the string "1463079214" get converted into 1463079168.000000. When switching from using a float to using a double, the conversion was as expected. Why was the conversion to float off? My best guess is that the conversion to float was attempting to store the entire value in the 23 bit significand of the IEEE-754 floating point number. In particular, if you take only the 23 most significant bits of 1463079214, you get the messed up 1463079168 that we were seeing in the conversion. It likely was possible to get a more precise value by composing the number using an exponent, but the conversion did not work that way. With a double, you have a 52 bit significand, allowing the entire value to fit there, and thereby allowing an accurate conversion. ASTERISK-26007 #close Reported by Greg Siemon Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070 |
10 years ago |
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36d66a23e0 |
logger: Add PID to syslog messages.
During refactoring of this support the addition of the PID to messages was removed. This change adds it back in. ASTERISK-25538 #close Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36 |
10 years ago |
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5261151286 |
Merge "file: Ensure nativeformats remains valid for lifetime of use." into 13
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10 years ago |
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60cc53ecb3 |
Merge "stasis_endpoints: Add new Status and Headers to ContactStatus" into 13
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10 years ago |
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dd00c71aae |
config_options.c: Expand #ifdef to contain whole if statement.
ASTERISK-25956 #close Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38 |
10 years ago |
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e6eb17efd9 |
stasis_endpoints: Add new Status and Headers to ContactStatus
ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail. ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail. These additions should be also in stasis_endpoints to include in command "manager show event ContactStatus" Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a |
10 years ago |
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fa11f4c920 |
file: Ensure nativeformats remains valid for lifetime of use.
It is possible for the nativeformats of a channel to change throughout its lifetime. As a result a user of it needs to either ensure the channel is locked when accessing the formats or keep a reference to the nativeformats themselves. This change fixes the file playback support so it keeps a reference to the nativeformats when accessing things. ASTERISK-25998 #close Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915 |
10 years ago |
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f39089f17c |
pjproject_bundled: Various fixes discovered during testing of OSes
For all OSes: * Disabled third-party codecs in pjproject and added '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the configure options since we don't use the pjsip codec capability. FreeBSD: * Added FreeBSD support to install_prereq. * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make". * Added __progname and environ to asterisk.exports.in. * Reverted the use of ldconfig to create shared library symlinks to ln. * Only enable epoll in pjproject if `uname -s` is Linux. * Added a patch to pjproject to take the name of the 'make' command from an environment variable if supplied. This is needed for the python bindings. (merged by Teluu into pjproject trunk 5/3/2016) FreeBSD support isn't complete. Still some general issues regarding make/gmake having nothing to do with pjproject. With some handholding it DOES build successfully. CentOS: Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH. CentOS 6/7 32/64 build and run the pjsip testsuite successfully. Ubuntu: No changes required. Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully. Debian: No changes required. Debian 6/7/8 32/64 build and run the pjsip testsuite successfully. There will utimately be a follow-up patch to create an install_prereq for the testsuite as I've discovered a few missing requirements. ASTERISK-25968 #close Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c |
10 years ago |
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a16aa467f5 |
Merge "config: Fix ast_config_text_file_save2 writability check for missing files" into 13
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10 years ago |
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9d8f59aaf2 |
Merge changes from topic 'system_stress_patches' into 13
* changes: test_message.c: Wait longer in case dialplan also processes the test message. Manager: Short circuit AMI message processing. manager.c: Eliminate most RAII_VAR usage. manager_channels.c: Fix allocation failure crash. |
10 years ago |
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4efc6b4315 |
Merge changes from topic 'system_stress_patches' into 13
* changes: Bridge system: Fix memory leaks and double frees on impart failure. bridge_softmix.c: Fix crash if channel fails to join mixing tech. |
10 years ago |
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b38f1146e5 |
config: Fix ast_config_text_file_save2 writability check for missing files
A patch I did back in 2014 modified ast_config_text_file_save2 to check the writability of the main file and include files before truncating and re-writing them. An unintended side-effect of this was that if a file doesn't exist, the check fails and the write is aborted. This patch causes ast_config_text_file_save2 to check the writability of the parent directory of missing files instead of checking the file itself. This allows missing files to be created again. A unit test was also added to test_config to test saving of config files. The regression was discovered when app_voicemail's passwordlocation=spooldir feature stopped working. ASTERISK-25917 #close Reported-by: Jonathan Rose Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80 |
10 years ago |
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09f8f8daa1 |
Merge "bridge: Hold off more than one imparting channel at a time." into 13
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10 years ago |
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ba63aa7c9e |
Manager: Short circuit AMI message processing.
Improve AMI message processing performance if there are no consumers listening for the messages. We now skip creating the AMI event message text strings. Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3 |
10 years ago |
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d5ee6acf28 |
manager.c: Eliminate most RAII_VAR usage.
* Made ast_manager_event_blob_create() not allocate the ao2 event object with a lock as it is not needed. Change-Id: I8e11bfedd22c21316012e0b9dd79f5918f644b7c |
10 years ago |
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7303e3dc96 |
manager_channels.c: Fix allocation failure crash.
An earlier allocation failure failed to create a channel snapshot for the AMI HangupRequest/SoftHangupRequest event which resulted in a crash in channel_hangup_request_cb(). Where the stasis message gets generated cannot tell if the NULL snapshot returned was because of an allocation failure or the channel was a dummy channel. * Made channel_hangup_request_cb() check if the channel blob has a snapshot and exit if it doesn't. * Eliminated the RAII_VAR usage in channel_hangup_request_cb(). Change-Id: I0b6a1c4e95cbb7d80b2a7054c6eadecc169dfd24 |
10 years ago |
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1e93f3d723 |
Bridge system: Fix memory leaks and double frees on impart failure.
You cannot reference the passed in features struct after calling ast_bridge_impart(). Even if the call fails. Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21 |
10 years ago |
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5e388d4188 |
bridge_softmix.c: Fix crash if channel fails to join mixing tech.
softmix_bridge_join() failed because of an allocation failure. To address this, the softmix bridge technology now checks if the channel failed to join softmix successfully. In addition, the bridge now begins the process of kicking the channel out of the bridge so we don't have channels partially in the bridge for very long. * Fix the test_channel_feature_hooks.c unit tests. The test channel must have a valid codec to join the simple_bridge technology. This patch makes joining a bridge more strict by not allowing partially joined channels to remain in the bridge. Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b |
10 years ago |
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e750ea9b5b |
lock.c: Check *lt before dereferencing it
*lt is NULL if t->tracking == 0 ASTERISK-25948 #close Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba |
10 years ago |
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9942d50aa5 |
bridge: Hold off more than one imparting channel at a time.
An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1 |
10 years ago |
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f4693d1897 |
bridge_channel.c: Ignore role setup failure in channel push.
We have to setup the channel roles after the bridge class push is called because the bridge class push callback may have set roles on the incoming channel. Since we have already partially pushed the channel into the bridge and reversing what we have already done could be problematic, the only thing we can do is press on to complete pushing the channel into the bridge. * Ignore any channel role setup errors after pushing the channel into a bridge. The channel may behave incorrectly in the bridge but we can no longer abort the push at this time. Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00 |
10 years ago |
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c7732a2600 |
Merge "Codecs: strip codec name while parsing allow/disallow options" into 13
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10 years ago |
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64ecd41c8f |
Codecs: strip codec name while parsing allow/disallow options
Failed registration using PJSIP/Realtime if one of the codec name in allow/disallow option is wrong or contains space. This patch strip codec name. ASTERISK-25914 Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d |
10 years ago |
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3f6c4667b8 |
core_unreal: Fix hangupcauses not getting set on Local channels
ASTERISK-25912 #close Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa |
10 years ago |
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736a2c303d |
Merge "lock: Add named lock capability" into 13
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10 years ago |
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772ff3048f |
lock: Add named lock capability
Locking some objects like sorcery objects can be tricky because the underlying ao2 object may not be the same for all callers. For instance, two threads that call ast_sorcery_retrieve_by_id on the same aor name might actually get 2 different ao2 objects if the underlying wizard had to rehydrate the aor from a database. Locking one ao2 object doesn't have any effect on the other even if those objects had locks in the first place. Named locks allow access control by keyspace and key strings. Now an "aor" named "1000" can be locked and any other thread attempting to lock "aor" "1000" will wait regardless of whether the underlying ao2 object is the same or not. Mutex and rwlocks are supported. This capability will initially be used to lock an aor when multiple threads may be attempting to prune expired contacts from it. Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45 |
10 years ago |