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
1.6.1.1
1.6.1.10
1.6.1.10-rc1
1.6.1.10-rc2
1.6.1.10-rc3
1.6.1.11
1.6.1.12
1.6.1.12-rc1
1.6.1.13
1.6.1.13-rc1
1.6.1.14
1.6.1.15-rc1
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1.6.1.16
1.6.1.17
1.6.1.18
1.6.1.18-rc1
1.6.1.18-rc2
1.6.1.19
1.6.1.19-rc1
1.6.1.19-rc2
1.6.1.19-rc3
1.6.1.2
1.6.1.20
1.6.1.20-rc1
1.6.1.20-rc2
1.6.1.21
1.6.1.22
1.6.1.23
1.6.1.24
1.6.1.25
1.6.1.3-rc1
1.6.1.4
1.6.1.5
1.6.1.5-rc1
1.6.1.6
1.6.1.7-rc1
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1.6.1.8
1.6.1.9
1.6.2.0
1.6.2.0-beta1
1.6.2.0-beta2
1.6.2.0-beta3
1.6.2.0-beta4
1.6.2.0-rc1
1.6.2.0-rc2
1.6.2.0-rc3
1.6.2.0-rc4
1.6.2.0-rc5
1.6.2.0-rc6
1.6.2.0-rc7
1.6.2.0-rc8
1.6.2.1
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1.6.2.14
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1.6.2.15-rc1
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1.6.2.16
1.6.2.16-rc1
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1.6.2.16.2
1.6.2.17
1.6.2.17-rc1
1.6.2.17-rc2
1.6.2.17-rc3
1.6.2.17.1
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1.6.2.17.3
1.6.2.18
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1.6.2.20
1.6.2.21
1.6.2.22
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1.6.2.24
1.6.2.3-rc1
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1.6.2.4
1.6.2.5
1.6.2.6
1.6.2.6-rc1
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1.6.2.7
1.6.2.7-rc1
1.6.2.7-rc2
1.6.2.7-rc3
1.6.2.8
1.6.2.8-rc1
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1.8.0
1.8.0-beta1
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1.8.0-beta3
1.8.0-beta4
1.8.0-beta5
1.8.0-rc1
1.8.0-rc2
1.8.0-rc3
1.8.0-rc4
1.8.0-rc5
1.8.1
1.8.1-rc1
1.8.1.1
1.8.1.2
1.8.10.0
1.8.10.0-rc1
1.8.10.0-rc2
1.8.10.0-rc3
1.8.10.0-rc4
1.8.10.1
1.8.11.0
1.8.11.0-rc1
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1.8.11.1
1.8.12.0
1.8.12.0-rc1
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1.8.12.0-rc3
1.8.12.1
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1.8.13.0-rc1
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1.8.14.0-rc1
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1.8.15-cert4
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1.8.16.0-rc1
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1.8.19.0-tc1
1.8.19.1
1.8.2
1.8.2-rc1
1.8.2.1
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1.8.2.3
1.8.2.4
1.8.20.0
1.8.20.0-rc1
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1.8.21.0-rc1
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1.8.22.0-rc1
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1.8.23.0
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1.8.23.1
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1.8.4-rc1
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1.8.4.1
1.8.4.2
1.8.4.3
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1.8.7.1
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1.8.8.0
1.8.8.0-rc1
1.8.8.0-rc2
1.8.8.0-rc3
1.8.8.0-rc4
1.8.8.0-rc5
1.8.8.1
1.8.8.2
1.8.9.0
1.8.9.0-rc1
1.8.9.0-rc2
1.8.9.0-rc3
1.8.9.1
1.8.9.2
1.8.9.3
10.0.0
10.0.0-beta1
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10.0.1
10.1.0
10.1.0-rc1
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10.1.1
10.1.2
10.1.3
10.10.0
10.10.0-digiumphones
10.10.0-digiumphones-rc1
10.10.0-digiumphones-rc2
10.10.0-rc1
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10.10.1
10.10.1-digiumphones
10.11.0
10.11.0-digiumphones
10.11.0-digiumphones-rc1
10.11.0-digiumphones-rc2
10.11.0-digiumphones-rc3
10.11.0-rc1
10.11.0-rc2
10.11.0-rc3
10.11.1
10.11.1-digiumphones
10.12.0
10.12.0-digiumphones
10.12.0-digiumphones-rc1
10.12.0-digiumphones-rc2
10.12.0-rc1
10.12.0-rc2
10.12.1
10.12.1-digiumphones
10.12.2
10.12.2-digiumphones
10.12.3
10.12.3-digiumphones
10.12.4
10.12.4-digiumphones
10.2.0
10.2.0-rc1
10.2.0-rc2
10.2.0-rc3
10.2.0-rc4
10.2.1
10.3.0
10.3.0-rc1
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10.3.0-rc3
10.3.1
10.4.0
10.4.0-digiumphones-rc1
10.4.0-digiumphones-rc2
10.4.0-rc1
10.4.0-rc2
10.4.0-rc3
10.4.1
10.4.2
10.5.0
10.5.0-digiumphones
10.5.0-digiumphones-rc1
10.5.0-digiumphones-rc2
10.5.0-rc1
10.5.0-rc2
10.5.1
10.5.1-digiumphones
10.5.2
10.5.2-digiumphones
10.6.0
10.6.0-digiumphones
10.6.0-digiumphones-rc1
10.6.0-digiumphones-rc2
10.6.0-rc1
10.6.0-rc2
10.6.1
10.6.1-digiumphones
10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
10.7.0-rc1
10.7.1
10.7.1-digiumphones
10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
10.8.0-digiumphones-rc2
10.8.0-rc1
10.8.0-rc2
10.9.0
10.9.0-digiumphones
10.9.0-digiumphones-rc1
10.9.0-digiumphones-rc2
10.9.0-digiumphones-rc3
10.9.0-rc1
10.9.0-rc2
10.9.0-rc3
11.0.0
11.0.0-beta1
11.0.0-beta2
11.0.0-rc1
11.0.0-rc2
11.0.1
11.0.2
11.1.0
11.1.0-rc1
11.1.0-rc2
11.1.0-rc3
11.1.1
11.1.2
11.10.0
11.10.0-rc1
11.10.1
11.10.2
11.11.0
11.11.0-rc1
11.12.0
11.12.0-rc1
11.12.1
11.13.0
11.13.0-rc1
11.13.1
11.14.0
11.14.0-rc1
11.14.0-rc2
11.14.1
11.14.2
11.15.0
11.15.0-rc1
11.15.0-rc2
11.15.1
11.16.0
11.16.0-rc1
11.17.0
11.17.0-rc1
11.17.1
11.18.0-rc1
11.19.0-rc1
11.2.0
11.2.0-rc1
11.2.0-rc2
11.2.1
11.2.2
11.20.0-rc1
11.20.0-rc2
11.20.0-rc3
11.21.0
11.21.0-rc1
11.21.0-rc2
11.21.0-rc3
11.21.1
11.22.0-rc1
11.3.0
11.3.0-rc1
11.3.0-rc2
11.4.0
11.4.0-rc1
11.4.0-rc2
11.4.0-rc3
11.5.0
11.5.0-rc1
11.5.0-rc2
11.5.1
11.6-cert11
11.6.0
11.6.0-rc1
11.6.0-rc2
11.6.1
11.7.0
11.7.0-rc1
11.7.0-rc2
11.8.0
11.8.0-rc1
11.8.0-rc2
11.8.0-rc3
11.8.1
11.9.0
11.9.0-rc1
11.9.0-rc2
11.9.0-rc3
12.0.0
12.0.0-alpha1
12.0.0-alpha2
12.0.0-beta1
12.0.0-beta2
12.1.0
12.1.0-rc1
12.1.0-rc2
12.1.0-rc3
12.1.1
12.2.0
12.2.0-rc1
12.2.0-rc2
12.2.0-rc3
12.3.0
12.3.0-rc1
12.3.0-rc2
12.3.1
12.3.2
12.4.0
12.4.0-rc1
12.5.0
12.5.0-rc1
12.5.1
12.6.0
12.6.0-rc1
12.6.1
12.7.0
12.7.0-rc1
12.7.0-rc2
12.7.1
12.7.2
12.8.0
12.8.0-rc1
12.8.0-rc2
12.8.1
12.8.2
13.0.0
13.0.0-beta1
13.0.0-beta2
13.0.0-beta3
13.0.1
13.0.2
13.1-cert2
13.1.0
13.1.0-rc1
13.1.0-rc2
13.1.1
13.2.0
13.2.0-rc1
13.2.1
13.3.0
13.3.0-rc1
13.3.1
13.3.2
13.4.0-rc1
13.5.0-rc1
13.6.0-rc1
13.6.0-rc2
13.6.0-rc3
13.7.0
13.7.0-rc1
13.7.0-rc2
13.7.0-rc3
13.7.1
13.8.0-rc1
certified/1.8.11-cert1
certified/1.8.11-cert10
certified/1.8.11-cert2
certified/1.8.11-cert3-rc1
certified/1.8.11-cert3-rc2
certified/1.8.11-cert4
certified/1.8.11-cert5
certified/1.8.11-cert5-rc1
certified/1.8.11-cert5-rc2
certified/1.8.11-cert6
certified/1.8.11-cert7
certified/1.8.11-cert8
certified/1.8.11-cert9
certified/1.8.11-cert9-rc1
certified/1.8.15-cert1
certified/1.8.15-cert1-rc1
certified/1.8.15-cert1-rc2
certified/1.8.15-cert1-rc3
certified/1.8.15-cert2
certified/1.8.15-cert3
certified/1.8.15-cert4
certified/1.8.15-cert5
certified/1.8.15-cert6
certified/1.8.15-cert7
certified/1.8.28-cert1
certified/1.8.28-cert1-rc1
certified/1.8.28-cert2
certified/1.8.28-cert3
certified/1.8.28-cert4
certified/1.8.28-cert5
certified/1.8.6-cert1
certified/11.2-cert1
certified/11.2-cert1-rc1
certified/11.2-cert1-rc2
certified/11.2-cert2
certified/11.2-cert3
certified/11.6-cert1
certified/11.6-cert1-rc1
certified/11.6-cert1-rc2
certified/11.6-cert10
certified/11.6-cert11
certified/11.6-cert12
certified/11.6-cert2
certified/11.6-cert3
certified/11.6-cert4
certified/11.6-cert5
certified/11.6-cert6
certified/11.6-cert7
certified/11.6-cert8
certified/11.6-cert9
certified/13.1-cert1
certified/13.1-cert1-rc1
certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
23985 Commits (dbdb2b1b3a618a79b3e42c77c05cb2688b4da08d)
Author | SHA1 | Message | Date |
---|---|---|---|
|
3aac3d2ee5 |
Tweak res_sip priority so it gets loaded first before all other SIP stuff.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386760 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
e5eb77002b |
Update res_config_sqlite to use the ast_variable lists.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386746 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
02be50b1ac |
Add support for a realtime sorcery module.
This change does the following: 1. Adds the sorcery realtime module 2. Adds unit tests for the sorcery realtime module 3. Changes the realtime core to use an ast_variable list instead of variadic arguments 4. Changes all realtime drivers to accept an ast_variable list Review: https://reviewboard.asterisk.org/r/2424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
c5f1eecb15 |
Add missing module dependencies to various res_sip* modules
This patch updates the various res_sip modules with their proper menuselect options and proper dependencies, such that Asterisk still has a snowball's chance in hell of compiling without pjproject. Much thanks to snuffy(-home|-work) for making everyone's life easier with this patch. Review: https://reviewboard.asterisk.org/r/2472/ (closes issue ASTERISK-21669) Reported by: snuffy patches: xml-depends.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386686 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
2d1cbb4311 |
Clean up memory leak in config file on off nominal paths when glob is allowed
If a system allows for its usage, Asterisk will use glob to help parse Asterisk .conf files. The config file loading routine was leaking the memory allocated by the glob() routine when the config file was in an unmodified or invalid state. This patch properly calls globfree in those off nominal paths. (closes issue ASTERISK-21412) Reported by: Corey Farrell patches: config_glob_leak.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 386672 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386677 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386685 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
562d0b4d18 |
By popular demand, putting the about-to-load-module printf back.
But now it only prints during the initial startup, and prints at verbose 1 level. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
b3e3dfa51c |
Clean up resources in features on exit
This patch cleans up two things features: * It properly unregisters the CLI commands that features registered * It cancels and performs a pthread_join on the created parking thread. This not only properly joins a non-detached thread, but also prevents disposing of the parking lots prior to the parking thread completely exiting. (closes issue ASTERISK-21407) Reported by: Corey Farrell patches: features_shutdown-r2.patch uploaded by Corey Farrell (License 5909) ........ Merged revisions 386641 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386642 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386676 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
79b2edea8f |
Removing stray printf from r386540
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
09af343789 |
Add an \extref doxygen pointer for libuuid.
Thanks to Olle Johansson for suggesting this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
946eb5ede0 |
Example of how to use the Stasis message bus
In order to get people familiar with the Stasis message bus, it would be useful to have something of a tutorial. Since I'm not clever enough to think of some cool integration we could do with Twitter, I settled for something that might actually be useful. This patch adds a res_statsd.so module, which implements a basic statsd[1] client. Statsd is a very simple statistics gathering server, which can publish its results to a backend graphing engine, like Graphite[2]. There are several different Statsd server implementations[3], so you can pick what works best for your environment. The actual example of how to use the Stasis message bus is in res_chan_stats.so. This module demonstrates how to use subscriptions and the message router by monitoring messages and posting channels stats to the statsd server. A wiki page walking through res_chan_stats.so is forthcoming. [1]: https://github.com/etsy/statsd/ [2]: http://graphite.readthedocs.org/en/latest/ [3]: http://joemiller.me/2011/09/21/list-of-statsd-server-implementations/ Review: https://reviewboard.asterisk.org/r/2460/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
eddf4a1ae6 |
Ignore *.[oi] files in res/res_sip
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
8c1f423cf7 |
Don't bind to anything in the sample configuration so we don't clash with chan_sip on a "make samples" right now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386577 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
8985bca752 |
REmove automerge properties.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
74f2318051 |
Merge the pimp_my_sip branch into trunk.
The pimp_my_sip branch is being merged at this point because it offers basic functionality, and from an API standpoint, things are complete. SIP work is *not* feature-complete; however, with the completion of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have been created, and thus it is possible for developers to attempt to create new SIP work. API documentation can be found in the doxygen in the code, but usability documentation is still lacking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
b4c881c86e |
Fix Displaying Symmetric RTP Global Setting
* Use comedia_string() to display correctly the symmetric rtp setting when running "sip show settings" ........ Merged revisions 386486 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386487 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
735026ccf6 |
Change Case On Forcerport For Consistency
* Change "ForcerPort" to "Forcerport" to match everywhere else it is displayed ........ Merged revisions 386483 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386484 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386485 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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522a969741 |
Document JSON models in resource_*.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386462 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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22aa2a6e9b |
Oops. Mustache doesn't like dictionaries
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386461 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
72828808c8 |
confbridge: Make search the conference bridges container using OBJ_KEY.
* Make confbridge config parsing user profile, bridge profile, and menu container hash/cmp functions correctly check the OBJ_POINTER, OBJ_KEY, and OBJ_PARTIAL_KEY flags. * Made confbridge load_module()/unload_module() free all resources on failure conditions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386375 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
e6d77dcdb5 |
Fix some bad whitespace
This crept in with the RESTful HTTP interface merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386352 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
c137d12111 |
Fix crash when AMI redirect action redirects two channels out of a bridge.
The two party bridging loops were changing the bridge peer pointers without the channel locks held. Thus when ast_channel_massquerade() tested and used the pointer there is a small window of opportunity for the pointers to become NULL even though the masquerade code has the channels locked. (closes issue ASTERISK-21356) Reported by: William luke Patches: jira_asterisk_21356_v11.patch (license #5621) patch uploaded by rmudgett Tested by: William luke ........ Merged revisions 386256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386286 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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e05870ccae |
Doxygen - Markup Guidelines
Expand on a commit by OEJ to use the Coding-Guidelines (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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1c21b8575b |
This patch adds a RESTful HTTP interface to Asterisk.
The API itself is documented using Swagger, a lightweight mechanism for documenting RESTful API's using JSON. This allows us to use swagger-ui to provide executable documentation for the API, generate client bindings in different languages, and generate a lot of the boilerplate code for implementing the RESTful bindings. The API docs live in the rest-api/ directory. The RESTful bindings are generated from the Swagger API docs using a set of Mustache templates. The code generator is written in Python, and uses Pystache. Pystache has no dependencies, and be installed easily using pip. Code generation code lives in rest-api-templates/. The generated code reduces a lot of boilerplate when it comes to handling HTTP requests. It also helps us have greater consistency in the REST API. (closes issue ASTERISK-20891) Review: https://reviewboard.asterisk.org/r/2376/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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1871017cc6 |
Fix mistake in Doxygen.
Doxygen is only *ONE* comment that applies to the NEXT piece of code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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1cb52c6026 |
sla: remove redundant locking.
sla.lock was already locked in the only place that sla_check_reload() was called. Remove the redundant locking of sla.lock done in this function. Less recursive locking is A Good Thing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386190 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
511cf82367 |
Prevent res_timing_pthread from blocking callers
There were several reports of deadlock when using res_timing_pthread. Backtraces indicated that one thread was blocked waiting for the write to the pipe to complete and this thread held the container lock for the timers. Therefore any thread that wanted to create a new timer or read an existing timer would block waiting for either the timer lock or the container lock and deadlock ensued. This patch changes the way the pipe is used to eliminate this source of deadlocks: 1) The pipe is placed in non-blocking mode so that it would never block even if the following changes someone fail... 2) Instead of writing bytes into the pipe for each "tick" that's fired the pipe now has two states--signaled and unsignaled. If signaled, the pipe is hot and any pollers of the read side filedescriptor will be woken up. If unsigned the pipe is idle. This eliminates even the chance of filling up the pipe and reduces the potential overhead of calling unnecessary writes. 3) Since we're tracking the signaled / unsignaled state, we can eliminate the exta poll system call for every firing because we know that there is data to be read. (closes issue ASTERISK-21389) Reported by: Matt Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches: 0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch uploaded by sruffell (License 5417) (closes issue ASTERISK-19754) Reported by: Nikola Ciprich (closes issue ASTERISK-20577) Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported by: Henry Fernandes (closes issue ASTERISK-17467) Reported by: isrl (closes issue ASTERISK-17458) Reported by: isrl Review: https://reviewboard.asterisk.org/r/2441/ ........ Merged revisions 386109 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386159 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386160 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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e61cc22404 |
cli.c: Properly initialize debug_modules and verbose_modules.
This avoids some lock errors on the core set {debug,verbose} commands. ........ Merged revisions 386049 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386051 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386054 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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9c696e665f |
Allow WebSocket connections on more URL's
This patch adds the concept of ast_websocket_server to res_http_websocket, allowing WebSocket connections on URL's more more than /ws. The existing funcitons for managing the WebSocket subprotocols on /ws still work, so this patch should be completely backward compatible. (closes issue ASTERISK-21279) Review: https://reviewboard.asterisk.org/r/2453/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386020 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
aff127a737 |
Fix lock errors on startup.
In messages.c, there are several places in the code where we create a tmp_tech_holder and pass that into an ao2_find call. Unfortunately, we weren't initializing the rwlock on the tmp_tech_holder, which the hash function was locking. It's apparently harmless, but still not the best code. This patch extracts all that copy/pasted code into two functions, msg_find_by_tech and msg_find_by_tech_name, which properly initialize and destroy the rwlock on the tmp_tech_holder. Review: https://reviewboard.asterisk.org/r/2454/ ........ Merged revisions 386006 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386019 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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fed7249ad1 |
res_xmpp and res_jabber need to search 'cachable' in the attrib section of the received IE, not data.
(issue ASTERISK-20175) (closes issue ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2452/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385939 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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1f71abdfae |
Allow res_corosync to build
ast_enable_distributed_devstate is no longer applicable to how the distributed device state system works and is no longer necessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385886 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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71a01725b8 |
Move presence state distribution to Stasis-core
Convert presence state events to Stasis-core messages and remove redundant serializers where possible. Review: https://reviewboard.asterisk.org/r/2410/ (closes issue ASTERISK-21102) Patch-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385862 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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191cf99ae1 |
Move device state distribution to Stasis-core
In the move from Asterisk's event system to Stasis, this makes distributed device state aggregation always-on, removes unnecessary task processors where possible, and collapses aggregate and non-aggregate states into a single cache for ease of retrieval. This also removes an intermediary step in device state aggregation. Review: https://reviewboard.asterisk.org/r/2389/ (closes issue ASTERISK-21101) Patch-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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c1ae5dc49b |
Fixed a typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385835 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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f4870aa71c |
Don't unnecessarily rebuild things on every run of 'make'.
Review: https://reviewboard.asterisk.org/r/2449/ ........ Merged revisions 385745 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385768 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385782 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
4b9e9fd2c0 |
Avoid unused variable warning when not in devmode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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c599aca553 |
Moved core logic from app_stasis to res_stasis
After some discussion on asterisk-dev, it was decided that the bulk of the logic in app_stasis actually belongs in a resource module instead of the application module. This patch does that, leaves the app specific stuff in app_stasis, and fixes up everything else to be consistent with that change. * Renamed test_app_stasis to test_res_stasis * Renamed app_stasis.h to stasis_app.h * This is still stasis application support, even though it's no longer in an app_ module. The name should never have been tied to the type of module, anyways. * Now that json isn't a resource module anymore, moved the ast_channel_snapshot_to_json function to main/stasis_channels.c, where it makes more sense. Review: https://reviewboard.asterisk.org/r/2430/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385742 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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2450722f52 |
DTMF events are now published on a channel's stasis_topic. AMI was
refactored to use these events rather than producing the events directly in channel.c. Finally, the code was added to app_stasis to produce DTMF events on the WebSocket. The AMI events are completely backward compatible, including sending events on transmitted DTMF, and sending DTMF start events. The Stasis-HTTP events are somewhat simplified. Since DTMF start and DTMF send events are generally less useful, Stasis-HTTP will only send events on received DTMF end. (closes issue ASTERISK-21282) (closes issue ASTERISK-21359) Review: https://reviewboard.asterisk.org/r/2439 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385734 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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c6cf12408e |
Fix the svn:keywords property on several files.
Normally I think keyword expansion is silly, but the one time it would have been good, it didn't work because the property had quotes in it. This patch fixes obviously busted svn:keywords properties. ........ Merged revisions 385683 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385689 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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2dd40e9707 |
Calculate the timestamp for outbound RTP if we don't have timing information
This patch calculates the timestamp for outbound RTP when we don't have timing information. This uses the same approach in res_rtp_asterisk. Thanks to both Pietro and Tzafrir for providing patches. (closes issue ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035) rtp-timestamp.patch uploaded by pbertera (License 5943) ........ Merged revisions 385636 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385637 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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5111744214 |
Don't attempt to create a voice frame on a read error
Prior to this patch, a read error in snd_pcm_readi would still be treated as a nominal result when constructing a voice frame from the expected data. Since the value returned is negative, as opposed to the number of samples read, this could result in a crash. With this patch, we now return a null frame when a read error is detected. Note that the patch on ASTERISK-21329 was modified slightly for this commit, in that we bail immediately on detecting the read error, rather than bypassing the construction of the voice frame. (closes issue ASTERISK-21329) Reported by: Keiichiro Kawasaki patches: chan_alsa.diff uploaded by kawasaki (License 6489) ........ Merged revisions 385633 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385634 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385635 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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a7c5183d67 |
Fix Manager Segfault When app_queue Is Unloaded
When app_queue is unloaded, some manager commands are not being unregistered which result in a segfault. This patch corrects this. (closes issue ASTERISK-21397) Reported by: Peter Katzmann, Corey Farrell Tested by: Corey Farrell Patches: asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L. Young (license 5026) asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2444/ ........ Merged revisions 385593 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385594 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385595 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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510b3b3594 |
Allow codec_resample to be unloaded
Ensure that trans_size is correct to prevent uninitialized entries from preventing reload. (closes issue ASTERISK-21401) Reported by: Corey Farrell Tested by: Corey Farrell Patches: codec_resample-unload.patch uploaded by Corey Farrell ........ Merged revisions 385582 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385585 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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1a09839e6b |
Fix app_voicemail Segfault And A Few Memory Leaks
The original report was that app_voicemail would crash. This was caused by ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being performed for that return status. After adding the initial patch to fix this issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered. During review, Walter Doekes (wdoekes) suggested adding a helper function in order to determine if we had a valid configuration or not. This patch does the following: * Creates a helper function to check if the configuration is valid * Adds calls to the new helper function where appropiate * Fixes memory leaks where the code returned without running ast_config_destroy() on the configuration that was loaded (closes issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco Kroon, Michael L. Young Patches: asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon (license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2443/ ........ Merged revisions 385551 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385557 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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ab6c0e74f1 |
Fix documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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7f885dc31d |
Expose channel snapshot manager blob generation
These functions are already used in one branch (jrose's parking branch) and will soon be used in other branches as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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fcbb9f0c8d |
Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
When we reload Asterisk or chan_sip, the flags force_rport and comedia that are turned on and off when using the auto_force_rport and auto_comedia nat settings go back to the default setting off. These flags are turned on when needed or off when not needed at the time that a peer registers, re-registers or initiates a call. This would apply even when only the default global setting "nat=auto_force_rport" is being used, which in this case would only affect the force_rport flag. Everything is good except for the following: The nat setting is set to auto_force_rport and auto_comedia. We reload Asterisk and the peer's registration has not expired. We load in the settings for the peer which turns force_rport and comedia back to off. Since the peer has not re-registered or placed a call yet, those flags remain off. We then initiate a call to the peer from the PBX. The force_rport and comedia flags stay off. If NAT is involved, we end up with one-way audio since we never checked to see if the peer is behind NAT or not. This patch does the following: * Moves the checking of whether a peer is behind NAT into its own function * Create a function to set the peer's NAT flags if they are using the auto_* NAT settings * Adds calls in sip_request_call() to these new functions in order to setup the dialog according to the peer's settings (closes issue ASTERISK-21374) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2421/ ........ Merged revisions 385473 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385474 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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e5b0de5535 |
IAX2 defer_full_frames fail to get sent
Ensure iax2_process_thread is signalled when a deferred frame is queued to it. (closes issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2426/ ........ Merged revisions 385429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385430 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385431 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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3959535615 |
IAX2, prevent network thread starting before all helper threads are ready
On startup, it's possible for a frame to arrive before the processing threads were ready. In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive before we are at ast_cond_wait, the signal will be ignored. The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued. Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled after each thread is started. (issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2427/ ........ Merged revisions 385402 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385403 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385406 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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67cc3b53c9 |
Blocked revisions 385356
........ Add dependency on libuuid, for res_rtp_asterisk pjproject is what actually requires libuuid. (closes issue ASTERISK-21125) reported by Private Name (Ed. note: Really? Private Name? I am rolling my eyes so hard right now.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |