1951 Commits (e2cd14876fcf0ca1af389112362ee120c60412b2)

Author	SHA1	Message	Date
David M. Lee	e2cd14876f	Just a couple of Stasis-HTTP nitpick fixes. ... * Fixed crash when res_stasis_http is unloaded before the implementation modules. * Cleaned up test initialization for test_stasis_http.so. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	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	13 years ago
Joshua Colp	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	13 years ago
Joshua Colp	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	13 years ago
Matthew Jordan	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	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
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
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
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
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
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	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
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	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
David M. Lee	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
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	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
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	a5df2542c3	Don't attempt a websocket protocol removal if res_http_websocket isn't there ... 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	13 years ago
David M. Lee	a2a53cc306	Stasis application WebSocket support ... 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	13 years ago
Joshua Colp	426095bc55	Add a res_sorcery_astdb module which uses the astdb to persist objects. ... Review: https://reviewboard.asterisk.org/r/2420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	a1c94fece8	Add uuid wrapper API call ast_uuid_generate_str(). ... * 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	13 years ago
Kinsey Moore	1a2a4578d2	Convert MWI state message type to the new stasis naming convention ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	15bbfb941f	Fix white noise on SRTP decryption ... 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	13 years ago
Matthew Jordan	4b5a0e1932	AST-2013-001: Prevent buffer overflow through H.264 format negotiation ... 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	13 years ago
Sean Bright	d484f366f5	Properly delimit post data in res_config_curl. ... ........ 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	13 years ago
David M. Lee	cf9324b25e	Move more channel events to Stasis; move res_json.c to main/json.c. ... 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	13 years ago
Joshua Colp	07d01e1c41	Pass the sorcery instance to wizards for CUD operations as well as retrieve. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	6300aa6ae4	Make sure things compile... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	99aa02d17f	Transition MWI to Stasis-core ... 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	13 years ago
Joshua Colp	5d45596f62	Add support for using XMPP buddy state via device state. ... 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	13 years ago
Joshua Colp	2f89b7a6eb	Fix a bug where resources were not found due to hashing on the priority itself. ... ........ 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	13 years ago
Matthew Jordan	95849b1a83	Always set the RTP instance data in the RTP engine ... 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	13 years ago
Andrew Latham	e29737179a	Update Doxygen ... 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	13 years ago
Joshua Colp	9a992c6cba	Fix a crash when res_xmpp is configured using a username without a domain. ... (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	13 years ago
Jason Parker	1cb917096b	Switch to using external pjproject libraries. ... 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	13 years ago
Jason Parker	c55592a324	Load sorcery modules earlier, so they can actually be used. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	80b8c2349c	Add a 'secret' probation strictrtp mode to handle delayed changes in RTP source ... 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	13 years ago
Joshua Colp	3a8caa351e	While the ICE negotiation is occurring leave strictrtp in an open state, 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	13 years ago
Joshua Colp	50a74cbd2a	Fix a bug with ICE and strictrtp where media could get dropped. ... 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	13 years ago
Jason Parker	6acc9ceb76	Don't undefine bzero()/bcopy(). ... 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	13 years ago
Michael L. Young	e9bcf9826a	Fix FastAGI To Properly Check For A Connection ... 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	13 years ago
Michael L. Young	d1f8e338b0	Add The Status Of A Module To The Output Of "CLI> module show" ... 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	13 years ago
Joshua Colp	cce1c9547f	Add support for retrieving multiple objects from sorcery using a regex on 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	13 years ago
Matthew Jordan	d04ab3c645	Add CLI configuration documentation ... 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	13 years ago
David M. Lee	f2b9c12704	Minor fixes to res_json and test_json. ... * 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	13 years ago
Matthew Jordan	4682d32d34	Fix crash in res_xmpp when deleting pubsub node from CLI ... 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	13 years ago
Joshua Colp	abd17dc849	Fix a bug where a changed configuration file might not be available to all 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	13 years ago