3235 Commits (a1435aa3fad5bda73a66dbccf3982787eff55ea2)

Author	SHA1	Message	Date
Matt Jordan	4c9f613309	ARI: Add the ability to subscribe to all events ... This patch adds the ability to subscribe to all events. There are two possible ways to accomplish this: (1) On initial WebSocket connection. This patch adds a new query parameter, 'subscribeAll'. If present and True, Asterisk will subscribe the applications to all ARI events. (2) Via the applications resource. When subscribing in this manner, an ARI client should merely specify a blank resource name, i.e., 'channels:' instead of 'channels:12354'. This will subscribe the application to all resources of the 'channels' type. ASTERISK-24870 #close Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6	10 years ago
Mark Michelson	9d1f176e29	res_pjsip: Copy default_from_user to avoid crash. ... The default_from_user retrieval function was pulling the default_from_user from the global configuration struct in an unsafe way. If using a database as a backend configuration store, the global configuration struct is short-lived, so grabbing a pointer from it results in referencing freed memory. The fix here is to copy the default_from_user value out of the global configuration struct. Thanks go to John Hardin for discovering this problem and proposing the patch on which this fix is based. ASTERISK-25390 #close Reported by Mark Michelson Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c	10 years ago
Mark Michelson	ac62928d6b	res_pjsip: Change default from user value. ... When Asterisk sends an outbound SIP request, if there is no direct reason to place a specific value for the username in the From header, Asterisk would generate a UUID. For example, this would happen when sending outbound OPTIONS requests when qualifying or when sending outbound INVITE requests when originating (if no explicit caller ID were provided). The issue is that some SIP providers reject these sorts of requests with a "Name too long" error response. This patch aims to fix this by changing the default outbound username in From headers to "asterisk". This value can be overridden by changing the default_from_user option in the global options if desired. ASTERISK-25377 #close Reported by Mark Michelson Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190	10 years ago
Joshua Colp	1b1561f4c8	res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items. ... The keepalive support in res_pjsip_sdp_rtp currently assumes that a stream will only be negotiated once. This is false. If the stream is replaced and later added back it can be negotiated again causing multiple keepalive scheduled items to exist. This change explicitly deletes the existing keepalive scheduled item before adding the new one. The res_pjsip_sdp_rtp module also does not stop RTP keepalives or timeout timer if the stream has been replaced. This change adds a callback to the session media interface to allow a media stream to be stopped without the resources being destroyed. This allows the scheduled items and RTP to be stopped when the stream no longer exists. ASTERISK-25356 #close Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de	10 years ago
Joshua Colp	2a4eee0cd9	res_pjsip: Add common ast_sip_get_host_ip API. ... Modules commonly used the pj_gethostip function for retrieving the IP address of the host. This function does not cache the result and may result in a DNS lookup occurring, or additional work. If the DNS server is unreachable or network issues arise this can cause the pj_gethostip function to block for a period of time. This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string function which does the same thing but caches the host IP address at module load time. This results in no additional work being done each time the local host IP address is needed. ASTERISK-25342 #close Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e	10 years ago
Matt Jordan	a0f451c35e	main/format: Add an API call for retrieving format attributes ... Some codecs that may be a third party library to Asterisk need to have knowledge of the format attributes that were negotiated. Unfortunately, when the great format migration of Asterisk 13 occurred, that ability was lost. This patch adds an API call, ast_format_attribute_get, to the core format API, along with updates to the unit test to check the new API call. A new callback is also now available for format attribute modules, such that they can provide the format attribute values they manage. Note that the API returns a void *. This is done as the format attribute modules themselves may store format attributes in any particular manner they like. Care should be taken by consumers of the API to check the return value before casting and dereferencing. Consumers will obviously need to have a priori knowledge of the type of the format attribute as well. Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3	10 years ago
Scott Emidy	df9ce36366	ARI: Retrieve existing log channels ... An http request can be sent to get the existing Asterisk logs. The command "curl -v -u user:pass -X GET 'http://localhost:8088 /ari/asterisk/logging'" can be run in the terminal to access the newly implemented functionality. * Retrieve all existing log channels ASTERISK-25252 Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808	10 years ago
Scott Emidy	e9f1bc08cb	ARI: Creating log channels ... An http request can be sent to create a log channel in Asterisk. The command "curl -v -u user:pass -X POST 'http://localhost:088/ari/asterisk/logging/mylog? configuration=notice,warning'" can be run in the terminal to access the newly implemented functionality for ARI. * Ability to create log channels using ARI ASTERISK-25252 Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782	10 years ago
Scott Emidy	78364132ce	ARI: Deleting log channels ... An http request can be sent to delete a log channel in Asterisk. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/logging/mylog'" can be run in the terminal to access the newly implemented functionally for ARI. * Able to delete log channels using ARI ASTERISK-25252 Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6	10 years ago
Mark Duncan	aed068844c	res/res_rtp_asterisk: Add ECDH support ... This will add ECDH support to Asterisk. It will detect auto ECDH support in OpenSSL (1.0.2b and above) during ./configure. If this is available, it will use it, otherwise it will fall back to prime256v1 (this behavior is consistent with other projects such as Apache and nginx). This fixes WebRTC being broken in Firefox 38+ due to Firefox now only supporting ciphers with perfect forward secrecy. ASTERISK-25265 #close Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b	10 years ago
Joshua Colp	20ee33e22e	Merge topic 'misc_rtp_tweaks' into 13 ... * changes: rtp_engine.h: No sense allowing payload types larger than RFC allows. rtp_engine.c: Minor tweaks. rtp_engine.h: Misc comment fixes. chan_sip.c: Tweak glue->update_peer() parameter nil value.	10 years ago
Benjamin Ford	1ae762634c	ARI: Rotate log channels. ... An http request can be sent to rotate a specified log channel. If the channel does not exist, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/logging/logChannelName/rotate'" can be run in the terminal to access this new functionality. * Added the ability to rotate log files through ARI ASTERISK-25252 Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01	10 years ago
Richard Mudgett	89b21fd9a3	rtp_engine.h: No sense allowing payload types larger than RFC allows. ... * Tweaked add_static_payload() to not use magic numbers. Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b	10 years ago
Richard Mudgett	e20f435b60	rtp_engine.h: Misc comment fixes. ... Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43	10 years ago
Joshua Colp	2749721791	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	10 years ago
Mark Michelson	d9094ddd73	res_pjsip: Add rtp_keepalive endpoint option. ... This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b	10 years ago
Richard Mudgett	e31cb6b248	strings.h: Fix issues with escape string functions. ... Fixes for issues with the ASTERISK-24934 patch. * Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is an empty string. If it were an empty string the functions returned NULL as if there were a memory allocation failure. This failure caused the AMI VarSet event to not get posted if the new value was an empty string. * Fixed dest buffer overwrite potential in ast_escape() and ast_escape_c(). If the dest buffer size is smaller than the space needed by the escaped s parameter string then the dest buffer would be written beyond the end by the nul string terminator. The num parameter was really the dest buffer size parameter so I renamed it to size. * Made nul terminate the dest buffer if the source string parameter s was an empty string in ast_escape() and ast_escape_c(). * Updated ast_escape() and ast_escape_c() doxygen function description comments to reflect reality. * Added some more unit test cases to /main/strings/escape to cover the empty source string issues. ASTERISK-25255 #close Reported by: Richard Mudgett Change-Id: Id77fc704600ebcce81615c1200296f74de254104	10 years ago
Benjamin Ford	1aafadf814	ARI: Added new functionality to reload a single module. ... An http request can be sent to reload an Asterisk module. If the module can not be reloaded or is not already loaded, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/modules/{moduleName}'" (or something similar, based on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be reloaded through http requests ASTERISK-25173 Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1	10 years ago
Mark Michelson	585d98fbb6	Merge "ARI: Added new functionality to get information on a single module." into 13	10 years ago
Mark Michelson	ca65ddcd19	Merge "bridge.c: Fixed race condition during attended transfer" into 13	10 years ago
Benjamin Ford	73e35d20de	ARI: Added new functionality to get information on a single module. ... An http request can be sent to retrieve information on a single module, including the resource name, description, use count, status, and support level. The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari /asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Information on a single module can now be retrieved ASTERISK-25173 Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463	10 years ago
Kevin Harwell	97ee0ee6c6	bridge.c: Fixed race condition during attended transfer ... During an attended transfer a thread is started that handles imparting the bridge channel. From the start of the thread to when the bridge channel is ready exists a gap that can potentially cause problems (for instance, the channel being swapped is hung up before the replacement channel enters the bridge thus stopping the transfer). This patch adds a condition that waits for the impart thread to get to a point of acceptable readiness before allowing the initiating thread to continue. ASTERISK-24782 Reported by: John Bigelow Change-Id: I08fe33a2560da924e676df55b181e46fca604577	10 years ago
Matt Jordan	e51d86682a	Merge "ARI: Added new functionality to get all module information." into 13	10 years ago
Benjamin Ford	47ea312b24	ARI: Added new functionality to get all module information. ... An http request can be sent to retrieve a list of all existing modules, including the resource name, description, use count, status, and support level. The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/ asterisk/modules" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Information on modules can now be retrieved Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0	10 years ago
Joshua Colp	b74b071369	res_sorcery_memory_cache: Backport to 13 ... Gerrit is complaining of conflicts when trying to create a patch series of all of the cherry-picked master commits, so I have instead squashed it all into one commit. ASTERISK-25067 #close Reported by: Matt Jordan Change-Id: I6dda90343fae24a75dc5beec84980024e8d61eb9	10 years ago
Joshua Colp	7386a761c1	Merge "PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error." into 13	10 years ago
Richard Mudgett	0422433f47	PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error. ... When res_pjsip body generator modules were generating XML or XPIDF response bodies, there was a chance that the generated body would be the exact size of the supplied buffer. Adding the nul string terminator would then write beyond the end of the buffer and potentially corrupt memory. * Fix MALLOC_DEBUG high fence violations caused by adding a nul string terminator on the end of a buffer for XML or XPIDF response bodies. * Made calls to pj_xml_print() safer if the XML prolog is requested. Due to a bug in pjproject, the return value could be -1 _or_ AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough. * Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the return value of pj_xml_print() when the supplied buffer is not large enough. ASTERISK-25168 Reported by: Carl Fortin Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de	10 years ago
Richard Mudgett	ada7346792	res_pjsip: Need to use the same serializer for a pjproject SIP transaction. ... All send/receive processing for a SIP transaction needs to be done under the same threadpool serializer to prevent reentrancy problems inside pjproject and res_pjsip. * Add threadpool API call to get the current serializer associated with the worker thread. * Pick a serializer from a pool of default serializers if the caller of res_pjsip.c:ast_sip_push_task() does not provide one. This is a simple way to ensure that all outgoing SIP request messages are processed under a serializer. Otherwise, any place where a pushed task is done that would result in an outgoing out-of-dialog request would need to be modified to supply a serializer. Serializers from the default serializer pool are picked in a round robin sequence for simplicity. A side effect is that the default serializer pool will limit the growth of the thread pool from random tasks. This is not necessarily a bad thing. * Made pjsip_distributor.c save the thread's serializer name on the outgoing request tdata struct so the response can be processed under the same serializer. This is a cherry-pick from master. **** ASTERISK-25115 Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a NOTE: session_inv_on_state_changed() is disassociating the dialog from the session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED. Unfortunately this is a tad too soon because our BYE request transaction has not completed yet. ASTERISK-25183 #close Reported by: Matt Jordan Change-Id: I8bad0ae1daf18d75b8c9e55874244b7962df2d0a	10 years ago
Matt Jordan	0a1700d286	Merge "threadpool, res_pjsip: Add serializer group shutdown API calls." into 13	10 years ago
Matt Jordan	e5499c3233	Merge "sorcery: Add ast_sorcery_object_unregister() API call." into 13	10 years ago
Richard Mudgett	84c12f9e0c	threadpool, res_pjsip: Add serializer group shutdown API calls. ... A module trying to unload needs to wait for all serializers it creates and uses to complete processing before unloading. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059	10 years ago
Richard Mudgett	20f3d77ab9	sorcery: Add ast_sorcery_object_unregister() API call. ... Find and unlink the specified sorcery object type to complement ast_sorcery_object_register(). Without this function you cannot completely unload individual modules that use sorcery for configuration. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88	10 years ago
Richard Mudgett	890c923786	AMI: Add Linkedid to the standard channel snapshot AMI event headers. ... * The AMI version is bumped to 2.8.0. ASTERISK-25189 #close Reported by: John Hardin Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac	10 years ago
Mark Michelson	db0521f905	Merge "res_pjsip_mwi: Set up unsolicited MWI upon registration." into 13	10 years ago
Richard Mudgett	2602a7484b	test.c: Add unit test registration checks for summary and description. ... Added checks when a unit test is registered to see that the summary and description strings do not end with a new-line '\n' for consistency. The check generates a warning message and will cause the /main/test/registrations unit test to fail. * Updated struct ast_test_info member doxygen comments. Change-Id: I295909b6bc013ed9b6882e85c05287082497534d	10 years ago
Joshua Colp	80e82dc97f	res_pjsip_mwi: Set up unsolicited MWI upon registration. ... The res_pjsip_mwi previously required a reload to set up the proper subscriptions to allow unsolicited MWI to work. This change makes it so the act of registering will also cause this to occur. This is particularly useful if realtime is involved as no reload needs to occur within Asterisk to cause the MWI information to get sent. ASTERISK-25180 #close Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252	10 years ago
Kevin Harwell	31c77b157b	res_pjsip: Add option to force G.726 to be treated as AAL2 packed. ... Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615	10 years ago
Richard Mudgett	a2b718f4f6	res_pjsip.h: Fix some doxygen comments. ... Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976	11 years ago
Matt Jordan	8ea1c0aa81	Merge "Fix unsafe uses of ast_context pointers." into 13	11 years ago
Corey Farrell	55c8daf88b	Fix unsafe uses of ast_context pointers. ... Although ast_context_find, ast_context_find_or_create and ast_context_destroy perform locking of the contexts table, any context pointer can become invalid at any time that the contexts table is unlocked. This change adds locking around all complete operations involving these functions. Places where ast_context_find was followed by ast_context_destroy have been replaced with calls ast_context_destroy_by_name. ASTERISK-25094 #close Reported by: Corey Farrell Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa	11 years ago
Kevin Harwell	f5d5aa67dc	AMI: Escape string values. ... So this issue is a bit complicated. Since it is possible to pass values to AMI that contain a '\r\n' (or other similar sequences) these values need to be escaped. One way to solve this is to escape the values and then pass the escaped values to the AMI variable parameter string building function. However, this puts the onus on the pre-build function to escape all string values. This potentially requires a fair amount of changes along with a lot of string allocations/freeing for all values. Surely there is a way to push this complexity down a level into the string building function itself? This of course is possible, but ends up requiring a way to distinguish between strings that need to be escaped and those that don't. The best way to handle this is by introducing a new format specifier in the format string. For instance a %s (no escape) and %S (escape). However, that is a bit weird and unexpected. So faced with those possibilities this patch implements a limited version of the first option. Instead of attempting to escape all string values this patch only escapes those values that make sense. This approach limits the number of changes and doesn't suffer from the odd format specifier problem. ASTERISK-24934 #close Reported by: warren smith Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0	11 years ago
Ivan Poddubny	888bb49618	Fix buffer overflow in slin sample frames generation. ... The length of frames retured by sample functions was twice as large as real, what caused global buffer overflow caught by AddressSanitizer. ASTERISK-24717 #close Reported by: Badalian Vyacheslav Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6	11 years ago
George Joseph	1558a89129	Revert "endpoint/stasis: Eliminate duplicate events on endpoint status change" ... This reverts commit 35c699086a. Change-Id: Ia98c2b4820cf579a5b9bb75e9e05d7a233205fb7	11 years ago
George Joseph	35c699086a	endpoint/stasis: Eliminate duplicate events on endpoint status change ... When an endpoint was created, it's messages were being forwarded to both the tech endpoint topic and the all endpoints topic. Since the tech topic was also forwarded to all, this was resulting in duplicate messages whenever an endpoint published. This patch causes the endpoint to only forward to the tech topic and lets the tech topic forward to all. To accomplish this, the existing stasis_cp_single_create function (which both creates and forwards) was cloned and split into 2 functions, one that creates the topic and one that sets up the forwarding. This allows endpoint_internal_create to create the topic from the endpoint_all cache without forwarding it there, then allows it to do the forward to the tech's topic. ASTERISK-25137 #close Reported-by: Vitezslav Novy ASTERISK-25116 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c	11 years ago
George Joseph	262d590819	res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes ... Add a new ContactStatus AMI event. Publish the following status/state changes: Created Removed Reachable Unreachable Unknown Contact URI, new status/state, aor and endpoint names, and the last qualify rtt result are included in the event. ASTERISK-25114 #close Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com>	11 years ago
Corey Farrell	0d266cbe02	Stasis: Fix unsafe use of stasis_unsubscribe in modules. ... Many uses of stasis_unsubscribe in modules can be reached through unload. These have been switched to stasis_unsubscribe_and_join. Some subscription callbacks do nothing, for these I've created a noop callback function in stasis.c. This is used by some modules that monitor MWI topics in order to enable cache, since the callback does not become invalid after dlclose it is safe to use stasis_unsubscribe on these, even during module unload. ASTERISK-25121 #close Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c	11 years ago
George Joseph	60e2fbfe62	res_pjsip: Refactor endpt_send_transaction (qualify_timeout) ... This patch refactors the transaction timeout processing to eliminate calling the lower level public pjsip functions and reverts to calling pjsip_endpt_send_request again. This is the result of me noticing a possible incompatibility with pjproject-2.4 which was causing contact status flapping. The original version of this feature used the lower level calls to get access to the tsx structure in order to cancel the transaction when our own timer expires. Since we no longer have that access, if our own timer expires before the pjsip timer, we call the callbacks and just let the pjsip transaction take it's own course. When the transaction ends, it discovers the callbacks have already been run and just cleans itself up. A few messages in pjsip_configuration were also added/cleaned up. ASTERISK-25105 #close Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com>	11 years ago
Matt Jordan	620054c527	Merge "audiohook.c: Difference in read/write rates caused continuous buffer resets" into 13	11 years ago
Kevin Harwell	b1e8c0b9eb	audiohook.c: Difference in read/write rates caused continuous buffer resets ... Currently, everytime a sample rate change occurs (on read or write) the associated factory buffers are reset. If the requested sample rate on a read differed from that of a write then the buffers are continually reset on every read and write. This has the side effect of emptying the buffer, thus there being no data to read and then write to a file in the case of call recording. This patch fixes it so that an audiohook_list's rate always maintains the maximum sample rate among hooks and formats. Audiohook sample rates are only overwritten by this value when slin native compatibility is turned on. Also, the audiohook sample rate can only overwrite the list's sample rate when its rate is greater than that of the list or if compatibility is turned off. This keeps the rate from constantly switching/resetting. ASTERISK-24944 #close Reported by: Ronald Raikes Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f	11 years ago
Matt Jordan	31cc24aad6	res/res_http_websocket: Add a pre-session established callback ... This patch updates http_websocket and its corresponding implementation with a pre-session established callback. This callback allows for WebSocket server consumers to be notified when a WebSocket connection is attempted, but before we accept it. Consumers can choose to reject the connection, if their application specific logic allows for it. As a result, this patch pulls out the previously private websocket_protocol struct and makes it public, as ast_websocket_protocol. In order to preserve backwards compatibility with existing modules, the existing APIs were left as-is, and new APIs were added for the creation of the ast_websocket_protocol as well as for adding a sub-protocol to a WebSocket server. In particular, the following new API calls were added: * ast_websocket_add_protocol2 - add a protocol to the core WebSocket server * ast_websocket_server_add_protocol2 - add a protocol to a specific WebSocket server * ast_websocket_sub_protocol_alloc - allocate a sub-protocol object. Consumers can populate this with whatever callbacks they wish to support, then add it to the core server or a specified server. ASTERISK-24988 Reported by: Joshua Colp Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2	11 years ago