mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
23778 Commits (79b2edea8f03c2c1e63f3472c128645d90c95363)
| Author | SHA1 | Message | Date |
|---|---|---|---|
David M. Lee
|
79b2edea8f |
Removing stray printf from r386540
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Mark Michelson
|
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 |
13 years ago |
|
David M. Lee
|
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 |
13 years ago |
|
David M. Lee
|
eddf4a1ae6 |
Ignore *.[oi] files in res/res_sip
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Joshua Colp
|
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 |
13 years ago |
|
Mark Michelson
|
8985bca752 |
REmove automerge properties.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
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 |
13 years ago |
Michael L. Young
|
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 |
13 years ago |
|
Michael L. Young
|
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 |
13 years ago |
|
David M. Lee
|
522a969741 |
Document JSON models in resource_*.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386462 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
David M. Lee
|
22aa2a6e9b |
Oops. Mustache doesn't like dictionaries
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386461 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Richard Mudgett
|
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 |
13 years ago |
Kinsey Moore
|
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 |
13 years ago |
|
Richard Mudgett
|
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
|
13 years ago |
Andrew Latham
|
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 |
13 years ago |
|
David M. Lee
|
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 |
13 years ago |
Olle Johansson
|
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 |
13 years ago |
Russell Bryant
|
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 |
13 years ago |
Matthew Jordan
|
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 |
13 years ago |
|
David M. Lee
|
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
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13 years ago |
|
David M. Lee
|
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 |
13 years ago |
|
David M. Lee
|
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 |
13 years ago |
Alec L Davis
|
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 |
13 years ago |
|
Kinsey Moore
|
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 |
13 years ago |
|
Kinsey Moore
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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 |
13 years ago |
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Kinsey Moore
|
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 |
13 years ago |
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David M. Lee
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c1ae5dc49b |
Fixed a typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385835 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Jason Parker
|
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 |
13 years ago |
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David M. Lee
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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 |
13 years ago |
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David M. Lee
|
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
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13 years ago |
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David M. Lee
|
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 |
13 years ago |
|
David M. Lee
|
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 |
13 years ago |
|
Matthew Jordan
|
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 |
13 years ago |
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Matthew Jordan
|
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 |
13 years ago |
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Michael L. Young
|
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
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13 years ago |
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Kinsey Moore
|
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
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13 years ago |
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Michael L. Young
|
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
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13 years ago |
|
Jason Parker
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ab6c0e74f1 |
Fix documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Kinsey Moore
|
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 |
13 years ago |
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Michael L. Young
|
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
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13 years ago |
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Alec L Davis
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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 |
13 years ago |
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Alec L Davis
|
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 |
13 years ago |
|
Jason Parker
|
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 |
13 years ago |
|
Richard Mudgett
|
13e2aae2ef |
Fix 'pri intense debug span' alias.
........ Merged revisions 385313 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
3afeac5e3b |
Eliminated dial_features_destroy() since it is equivalent to ast_free_ptr()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
eb2d144195 |
* Fix unlocked accesses to feature_list. The feature_list is now also
protected by the features_lock. * Made all calls to ast_find_call_feature() have the features_lock held. * Fixed set_config_flags() to actually use find_group() to look for feature groups in DYNAMIC_FEATURES. The code originally assumed all feature groups were listed in DYNAMIC_FEATURES. * Make everyone use ast_rdlock_call_features(), ast_unlock_call_features(), and new ast_wrlock_call_features() instead of directly calling the rwlock API on features_lock. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385277 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
David M. Lee
|
ff7ecd3dbf |
Fixed manager channelvars support.
For the events that have been ported to Stasis, this was broken in r384910, when a couple of lines of code was lost in a merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385236 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
1f3fff7b91 |
Use LDAP memory management functions instead of Asterisk's
When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk: invalid pointer errors) can occur as the memory is being allocated with Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the LDAP library's wrappers. This patch uses the LDAP library's wrappers where appropriate, so that compiling with MALLOC_DEBUG doesn't cause more problems than it solves. Note that the patch listed below was modified slightly for this commit to account for some additional memory allocation/deallocations. (closes issue ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham patches: issue18789-1.8-r316873.patch uploaded by seanbright (License 5060) ........ Merged revisions 385190 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385199 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
caf4a5f605 |
Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE
When a BYE request is processed in chan_sip, the current SIP dialog is detached from its associated Asterisk channel structure. The tech_pvt pointer in the channel object is set to NULL, and the dialog persists for an RFC mandated period of time to handle re-transmits. While this process occurs, the channel is locked (which is good). Unfortunately, operations that are initiated externally have no way of knowing that the channel they've just obtained (which is still valid) and that they are attempting to lock is about to have its tech_pvt pointer removed. By the time they obtain the channel lock and call the channel technology callback, the tech_pvt is NULL. This patch adds a few checks to some channel callbacks that make sure the tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit callbacks, which would crash if AMI initiated a DTMF on the channel at the same time as a BYE was received from the UA. This patch also adds checks on sip_transfer (as AMI can also cause a callback into this function), as well as sip_indicate (as lots of things can queue an indication onto a channel). Review: https://reviewboard.asterisk.org/r/2434/ (closes issue ASTERISK-20225) Reported by: Jeff Hoppe ........ Merged revisions 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385173 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
d09eeaa8eb |
Rename struct feature_ds to struct feature_datastore.
Because "struct feature_ds *feature_ds" is not a good thing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385142 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |