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2013-04-22This patch adds a RESTful HTTP interface to Asterisk.David M. Lee
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
2013-04-19Prevent res_timing_pthread from blocking callersMatthew Jordan
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
2013-04-18Allow WebSocket connections on more URL'sDavid M. Lee
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
2013-04-16res_xmpp and res_jabber need to search 'cachable' in the attrib section of ↵Alec L Davis
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
2013-04-16Allow res_corosync to buildKinsey Moore
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
2013-04-16Move device state distribution to Stasis-coreKinsey Moore
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
2013-04-15Avoid unused variable warning when not in devmodeDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-15Moved core logic from app_stasis to res_stasisDavid M. Lee
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
2013-04-15Fix the svn:keywords property on several files.David M. Lee
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
2013-04-14Calculate the timestamp for outbound RTP if we don't have timing informationMatthew Jordan
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
2013-04-10Use LDAP memory management functions instead of Asterisk'sMatthew Jordan
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
2013-04-08Don't attempt a websocket protocol removal if res_http_websocket isn't thereMatthew Jordan
This patch sets the protocols container provided by res_http_websocket to NULL when the module gets unloaded and adds the necessary checks when adding/ removing a websocket protocol. This prevents some FRACKing on an invalid pointer to the disposed container if a module that uses res_http_websocket is unloaded after it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-08Stasis application WebSocket supportDavid M. Lee
This is the API that binds the Stasis dialplan application to external Stasis applications. It also adds the beginnings of WebSocket application support. This module registers a dialplan function named Stasis, which is used to put a channel into the named Stasis app. As a channel enters and leaves the Stasis diaplan application, the Stasis app receives a 'stasis-start' and 'stasis-end' events. Stasis apps register themselves using the stasis_app_register and stasis_app_unregister functions. Messages are sent to an application using stasis_app_send. Finally, Stasis apps control channels through the use of the stasis_app_control object, and the family of stasis_app_control_* functions. Other changes along for the ride are: * An ast_frame_dtor function that's RAII_VAR safe * Some common JSON encoders for name/number, timeval, and context/extension/priority Review: https://reviewboard.asterisk.org/r/2361/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-06Add a res_sorcery_astdb module which uses the astdb to persist objects.Joshua Colp
Review: https://reviewboard.asterisk.org/r/2420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-28Add uuid wrapper API call ast_uuid_generate_str().Richard Mudgett
* Updated test_uuid.c to test the new API call. * Made system use the new API call to eliminate "10's of lines" where used. * Fixed untested ast_strdup() return in stasis_subscribe() by eliminating the need for it. struct stasis_subscription now contains the uniqueid[] string. * Fixed some issues in exchangecal_write_event(): Create uid with enough space for a UUID string to avoid a realloc. Fix off by one error if the calendar event provided a UUID string. There is no need to check for NULL before calling ast_free(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-27Convert MWI state message type to the new stasis naming conventionKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-27Fix white noise on SRTP decryptionKinsey Moore
When res_rtp_asterisk.c was altered to avoid attempting to apply unprotect algorithms to non-audio RTP packets, the test used was incorrect. This caused the audio packets to not be decrypted and resulted in loud white noise on the other endpoint (or both endpoints depending on the call legs involved). The test now properly checks the version field in the RTP header to ensure that RTP and RTCP are decrypted while other types of packets are not. (closes issue ASTERISK-21323) Reported by: andrea Tested by: Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff uploaded by Kinsey Moore ........ Merged revisions 384048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384049 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-27AST-2013-001: Prevent buffer overflow through H.264 format negotiationMatthew Jordan
The format attribute resource for H.264 video performs an unsafe read against a media attribute when parsing the SDP. The value passed in with the format attribute is not checked for its length when parsed into a fixed length buffer. This patch resolves the vulnerability by only reading as many characters from the SDP value as will fit into the buffer. (closes issue ASTERISK-20901) Reported by: Ulf Harnhammar patches: h264_overflow_security_patch.diff uploaded by jrose (License 6182) ........ Merged revisions 383973 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-25Properly delimit post data in res_config_curl.Sean Bright
........ Merged revisions 383667 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383668 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-22Move more channel events to Stasis; move res_json.c to main/json.c.David M. Lee
This patch started out simply as fixing the bouncing tests introduced in r382685, but required some other changes to give it a decent implementation. To fix the bouncing tests, the UserEvent and Newexten AMI events needed to be refactored to dispatch via Stasis. Dispatching directly to AMI resulted in those events sometimes getting ahead of the associated Newchannel events, which would understandably confuse anyone. I found that instead of creating a zillion different message types and structures associated with them, it would be preferable to define a message type that has a channel snapshot and a blob of structured data with a small bit of additional information. The JSON object model provides a very nice way of representing structured data, so I went with that. * Move JSON support from res_json.c to main/json.c * Made libjansson-dev a required dependency * Added an ast_channel_blob message type, which has a channel snapshot and JSON blob of data. * Changed UserEvent and Newexten events so that they are dispatched via ast_channel_blob messages on the channel's topic. * Got rid of the ast_channel_varset message; used ast_channel_blob instead. * Extracted the manager functions converting Stasis channel events to AMI events into manager_channel.c. (issue ASTERISK-21096) Review: https://reviewboard.asterisk.org/r/2381/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-20Pass the sorcery instance to wizards for CUD operations as well as retrieve.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16Make sure things compile...Kinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16Transition MWI to Stasis-coreKinsey Moore
Remove MWI's dependency on the event system by moving it to Stasis-core. This also introduces forwarding topic pools in Stasis-core which aggregate many dynamically allocated topics into a single primary topic. Review: https://reviewboard.asterisk.org/r/2368/ (closes issue ASTERISK-21097) Patch-by: Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16Add support for using XMPP buddy state via device state.Joshua Colp
This change allows you to use XMPP buddy state in places where device state can be used be used, such as dialplan hints. If at least one resource is available the buddy is considered available. Now your phone can reflect their IM status too! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16Fix a bug where resources were not found due to hashing on the priority itself.Joshua Colp
........ Merged revisions 383266 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-13Always set the RTP instance data in the RTP engineMatthew Jordan
Not informing the RTP engine of the instance data creates shrapnel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Update DoxygenAndrew Latham
Push some cleanups upstream before testing another ticket. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Fix a crash when res_xmpp is configured using a username without a domain.Joshua Colp
(closes issue ASTERISK-21156) Reported by: amsoft2001 ........ Merged revisions 382923 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12Switch to using external pjproject libraries.Jason Parker
ICE/STUN/TURN support in res_rtp_asterisk is also now optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Load sorcery modules earlier, so they can actually be used.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-07Add a 'secret' probation strictrtp mode to handle delayed changes in RTP sourceMatthew Jordan
Often, Asterisk may realize that a change in the source of an RTP stream is about to occur and ask that the RTP engine reset it's lock on the current RTP source. In certain scenarios, it may take awhile for the new remote system to send RTP packets, while the old remote system may continue providing RTP during that time period. This causes Asterisk to re-lock onto the old source, thereby rejecting the new source when the old source stops sending RTP and the new source begins. This patch prevents that by having a constant secondary, 'secret' probation mode enabled when an RTP source has been chosen. RTP packets from other sources are always considered, but never chosen unless the current RTP source stops sending RTP. Review: https://reviewboard.asterisk.org/r/2364 (closes issue AST-1124) Reported by: John Bigelow Tested by: John Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested by: John Bigelow ........ Merged revisions 382573 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28While the ICE negotiation is occurring leave strictrtp in an open state, ↵Joshua Colp
media can and will come from different places. ........ Merged revisions 382298 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28Fix a bug with ICE and strictrtp where media could get dropped.Joshua Colp
If the end result of the ICE negotiation resulted in the path for media changing it was possible for the strictrtp code to discard the RTP packets. This change causes strictrtp to enter learning mode once again when the ICE negotiation has completed successfully. ........ Merged revisions 382296 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-28Don't undefine bzero()/bcopy().Jason Parker
This was causing build failures against external libraries that happened to use them, unless silly hacks were added to the modules that used those headers. Review: https://reviewboard.asterisk.org/r/2359/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-22Fix FastAGI To Properly Check For A ConnectionMichael L. Young
When IPv6 support was added to FastAGI, the intent was to have the ability to check all addresses resolved for a host since we might receive an IPv4 address and an IPv6 address. The problem with the current code, is that, since we are doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring this instead of handling it. We break out of the loop and continue on. When we later call ast_poll(), it succeeds but we never check if we have a connection or not on the socket level. We then attempt to send data to the host address that we think is setup and it fails. We then check the errno and see that we have "connection refused" and then return with agi failed. This patch does the following: * Handles EINPROGRESS by creating the function handle_connection() - ast_poll() was moved into this function - This function checks the results of the connection on the socket level after calling ast_poll() * Continues to the next address if the above fails to create a connection * Once all addresses resolved are tried and we still are unable to establish a connection, then we return that the FastAGI call failed (closes issue ASTERISK-21065) Reported by: Jeremy Kister Tested by: Jeremy Kister, Michael L. Young Patches: asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2330/ ........ Merged revisions 381893 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-19Add The Status Of A Module To The Output Of "CLI> module show"Michael L. Young
When a module's configuration is not loadable, we still load the module but it is not in a running state. When trying to troubleshoot, let's say, why chan_motif is ignoring inbound XMPP traffic, there is no way to indicate that a loaded module is not currently running. (closes issue ASTERISK-21108) Reported by: Rusty Newton Tested by: Michael L. Young Patches: asterisk-21108_add_status-v2.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2331/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-16Add support for retrieving multiple objects from sorcery using a regex on ↵Joshua Colp
their id. Review: https://reviewboard.asterisk.org/r/2329/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-15Add CLI configuration documentationMatthew Jordan
This patch allows a module to define its configuration in XML in source, such that it can be parsed by the XML documentation engine. Documentation is generated in a two-pass approach: 1. The documentation is first generated from the XML pulled from the source 2. The documentation is then enhanced by the registration of configuration options that use the configuration framework This patch include configuration documentation for the following modules: * chan_motif * res_xmpp * app_confbridge * app_skel * udptl Two new CLI commands have been added: * config show help - show configuration help by module, category, and item * xmldoc dump - dump the in-memory representation of the XML documentation to a new XML file. Review: https://reviewboard.asterisk.org/r/2278 Review: https://reviewboard.asterisk.org/r/2058 patches: on review 2058 uploaded by twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-11Minor fixes to res_json and test_json.David M. Lee
* Made input checking more consistent with other Asterisk code * Added validation to ast_json_dump_new_file * Fixed tests for ownereship semantics (issue ASTERISK-20887) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-11Fix crash in res_xmpp when deleting pubsub node from CLIMatthew Jordan
An error existed in res_xmpp where it would attempt to delete attributes from a node that itself was also deleted. Per the iksemel documentation, attributes added using iks_insert are copied to the parent node's stack, and will be reclaimed when that node is itself destroyed. (closes issue ASTERISK-20982) Reported by: marcelloceschia patches: delete-node-fix.diff uploaded by marcelloceschia (License 6036) ........ Merged revisions 381159 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-07Fix a bug where a changed configuration file might not be available to all ↵Joshua Colp
sorcery object types. Since res_sorcery_config used a static name of "res_sorcery_config" to inform the configuration file API that it asked for the configuration file it was possible during a reload for some sorcery object types not to receive the new configuration file. This change introduces a UUID on a per-sorcery config instance basis so that the unchanged state is kept on an instance basis and not for the res_sorcery_config module as a whole. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-04Fix how we build pjproject.Jason Parker
Allow parallel builds, better tolerate failures, build faster. This also stops running dependencies before top-level configure has been run. (closes issue ASTERISK-20815) Review: https://reviewboard.asterisk.org/r/2292/ ........ Merged revisions 380816 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-31Multiple revisions 380735-380736Jason Parker
........ r380735 | qwell | 2013-01-31 15:40:09 -0600 (Thu, 31 Jan 2013) | 1 line Fix a few compiler warnings. ........ r380736 | qwell | 2013-01-31 15:42:34 -0600 (Thu, 31 Jan 2013) | 1 line Ignore warnings caused by PJ_TODO()s in pjproject. ........ Merged revisions 380735-380736 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-31Multiple revisions 380671-380673Jason Parker
........ r380671 | qwell | 2013-01-31 12:59:28 -0600 (Thu, 31 Jan 2013) | 4 lines Remove a cross-compile workaround. ar and ranlib can be easily detected with autoconf. ........ r380672 | qwell | 2013-01-31 13:00:38 -0600 (Thu, 31 Jan 2013) | 2 lines Always check for libm, regardless of configure options. ........ r380673 | qwell | 2013-01-31 13:03:03 -0600 (Thu, 31 Jan 2013) | 7 lines Add support for parallel builds of pjproject. Also adds proper dependency checking, and direct .a file targets. We don't take advantage of this currently, but we will soon. (issue ASTERISK-20815) ........ Merged revisions 380671-380673 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-30Fix memory leak in res_calendar_icalendarMatthew Jordan
The ICalendar module had a systemic memory leak on each fetch of data from the ICalendar source. The previous fetched data was not being properly disposed. This patch makes it so that before each fetch of data, we dispose of the previously fetched data. (closes issue ASTERISK-21012) Reported by: Joel Vandal Tested by: Joel Vandal ........ Merged revisions 380451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 380452 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-25Make sorcery modules global, since they are required by other modules that ↵Jason Parker
are global. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-25Add a missing '\' to a log message.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-25Merge the sorcery data access layer API.Joshua Colp
Sorcery is a unifying data access layer which provides a pluggable mechanism to allow object creation, retrieval, updating, and deletion using different backends (or wizards). This is a fancy way of saying "one interface to rule them all" where them is configuration, realtime, and anything else that comes along. Review: https://reviewboard.asterisk.org/r/2259/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-22res_fax_spandsp: fix t38 transmission bug caused by not returning successJonathan Rose
This patch fixes the problem, but the issue includes a test which is still being considered for the automated test suite. (issue ASTERISK-20919) Reported by: NITESH BANSAL Patches: patch_ast_fax_spandsp.patch uploaded by NITESH BANSAL (license 6418) ........ Merged revisions 379949 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-22Add ControlPlayback manager actionMatthew Jordan
This patch adds the capability for asynchronous manipulation of audio being played back to a channel though a new AMI action "ControlPlayback". The ControlPlayback action supports a number of operations, the availability of which depend on the application being used to send audio to the channel. When the audio playback was initiated using the ControlPlayback application or CONTROL STREAM FILE AGI command, the audio can be paused, stopped, restarted, reversed, or skipped forward. When initiated by other mechanisms (such as the Playback application), the audio can be stopped, reversed, or skipped forward. Review: https://reviewboard.asterisk.org/r/2265/ (closes issue ASTERISK-20882) Reported by: mjordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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