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Author	SHA1	Message	Date
Joshua Colp	ecd4cde521	Merge "ARI: Deleting log channels"	11 years ago
Joshua Colp	1b89cbb3b0	Merge "res_pjsip: Ensure sanitized XML is NULL terminated."	11 years ago
Joshua Colp	12d7c8a740	Merge "res_pjsip_pubsub: More accurately persist packet."	11 years ago
Joshua Colp	58effbc3f6	Merge "res_rtp_asterisk.c: Fix off-nominal crash potential."	11 years ago
Scott Emidy	f19c4930c2	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	11 years ago
Mark Michelson	382334cc06	res_pjsip_pubsub: More accurately persist packet. ... The pjsip_rx_data structure has a pkt_info.packet field on it that is the packet that was read from the transport. For datagram transports, the packet read from the transport will correspond to the SIP message that arrived. For streamed transports, however, it is possible to read multiple SIP messages in one packet. In a recent case, Asterisk crashed on a system where TCP was being used. This is because at some point, a read from the TCP socket resulted in a 200 OK response as well as an incoming SUBSCRIBE request being stored in rdata->pkt_info.packet. When the SUBSCRIBE was processed, the combination 200 OK and SUBSCRIBE was saved in persistent storage. Later, a restart of Asterisk resulted in the crash because the persistent subscription recreation code ended up building the 200 OK response instead of a SUBSCRIBE request, and we attempted to access request-specific data. The fix here is to use the pjsip_msg_print() function in order to persist SUBSCRIBE requests. This way, rather than using the raw socket data, we use the parsed SIP message that PJSIP has given us. If we receive multiple SIP messages from a single read, we will be sure only to save off the relevant SIP message. There also is a safeguard put in place to make sure that if we do end up reconstructing a SIP response, it will not cause a crash. ASTERISK-25306 #close Reported by Mark Michelson Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2	11 years ago
Joshua Colp	ca84a4b235	Merge "rtp_engine.c: Must protect mime_types_len with mime_types_lock."	11 years ago
Joshua Colp	7ebca6795b	Merge "res_pjsip_sdp_rtp.c: Fixup some whitespace."	11 years ago
Joshua Colp	4b6c657a82	res_pjsip: Ensure sanitized XML is NULL terminated. ... The ast_sip_sanitize_xml function is used to sanitize a string for placement into XML. This is done by examining an input string and then appending values to an output buffer. The function used by its implementation, strncat, has specific behavior that was not taken into account. If the size of the input string exceeded the available output buffer size it was possible for the sanitization function to write past the output buffer itself causing a crash. The crash would either occur because it was writing into memory it shouldn't be or because the resulting string was not NULL terminated. This change keeps count of how much remaining space is available in the output buffer for text and only allows strncat to use that amount. Since this was exposed by the res_pjsip_pidf_digium_body_supplement module attempting to send a large message the maximum allowed message size has also been increased in it. A unit test has also been added which confirms that the ast_sip_sanitize_xml function is providing NULL terminated output even when the input length exceeds the output buffer size. ASTERISK-25304 #close Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302	11 years ago
Joshua Colp	3d4db97253	Merge "res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list."	11 years ago
Joshua Colp	8a48c9e6cb	Merge "res_http_websocket: Debug write lengths."	11 years ago
Joshua Colp	7351d33a1f	res_rtp_asterisk: Don't leak temporary key when enabling PFS. ... A change recently went in which enabled perfect forward secrecy for DTLS in res_rtp_asterisk. This was accomplished two different ways depending on the availability of a feature in OpenSSL. The fallback method created a temporary instance of a key but did not free it. This change fixes that. ASTERISK-25265 Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396	11 years ago
Mark Michelson	c63316eec1	res_http_websocket: Debug write lengths. ... Commit 39cc28f6ea attempted to fix a test failure observed on 32 bit test agents by ensuring that a cast from a 32 bit unsigned integer to a 64 bit unsigned integer was happening in a predictable place. As it turns out, this did not cause test runs to succeed. This commit adds several redundant debug messages that print the payload lengths of websocket frames. The idea here is that this commit will not cause tests to succeed for the faulty test agent, but we might deduce where the fault lies more easily this way by observing at what point the expected value (537) changes to some ungangly huge number. If you are wondering why something like this is being committed to the branch, keep in mind that in commit 39cc28f6ea I noted that the observed test failures only happen when automated tests are run. Attempts to run the tests by hand manually on the test agent result in the tests passing. Change-Id: I14a65c19d8af40dadcdbd52348de3b0016e1ae8d	11 years ago
Matt Jordan	3ba6099a9e	Merge "res_http_websocket: Avoid passing strlen() to ast_websocket_write()."	11 years ago
Matt Jordan	8672f0bbbd	Merge "res/res_rtp_asterisk: Add ECDH support"	11 years ago
Mark Michelson	35a98161df	res_http_websocket: Avoid passing strlen() to ast_websocket_write(). ... We have seen a rash of test failures on a 32-bit build agent. Commit 48698a5e21 solved an obvious problem where we were not encoding a 64-bit value correctly over the wire. This commit, however, did not solve the test failures. In the failing tests, ARI is attempting to send a 537 byte text frame over a websocket. When sending a frame this small, 16 bits are all that is required in order to encode the payload length on the websocket frame. However, ast_websocket_write() thinks that the payload length is greater than 65535 and therefore writes out a 64 bit payload length. Inspecting this payload length, the lower 32 bits are exactly what we would expect it to be, 537 in hex. The upper 32 bits, are junk values that are not expected to be there. In the failure, we are passing the result of strlen() to a function that expects a uint64_t parameter to be passed in. strlen() returns a size_t, which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit unsigned value to somewhere where a 64-bit unsigned value is expected would cause no problems. In fact, in manual runs of failing tests, this works just fine. However, ast_websocket_write() uses the Asterisk optional API, which means that rather than a simple function call, there are a series of macros that are used for its declaration and implementation. These macros may be causing some sort of error to occur when converting from a 32 bit quantity to a 64 bit quantity. This commit changes the logic by making existing ast_websocket_write() calls use ast_websocket_write_string() instead. Within ast_websocket_write_string(), the 64-bit converted strlen is saved in a local variable, and that variable is passed to ast_websocket_write() instead. Note that this commit message is full of speculation rather than certainty. This is because the observed test failures, while always present in automated test runs, never occur when tests are manually attempted on the same test agent. The idea behind this commit is to fix a theoretical issue by performing changes that should, at the least, cause no harm. If it turns out that this change does not fix the failing tests, then this commit should be reverted. Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67	11 years ago
Joshua Colp	8eef7392c4	Merge topic 'misc_rtp_tweaks' ... * changes: rtp_engine.h: No sense allowing payload types larger than RFC allows. rtp_engine.c: Minor tweaks. rtp_engine.h: Misc comment fixes.	11 years ago
Joshua Colp	4cf21fed27	Merge "chan_sip.c: Tweak glue->update_peer() parameter nil value."	11 years ago
Mark Michelson	b002e09214	Merge "ARI: Channels added to Stasis application during WebSocket creation ..."	11 years ago
Mark Michelson	92ddda68aa	Merge "ARI: Rotate log channels."	11 years ago
Joshua Colp	94f7427b17	Merge "res_pjsip_session.c: Fix crashes seen when call cancelled."	11 years ago
Benjamin Ford	1f02d20da4	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	11 years ago
Joshua Colp	d61ea4bcc2	Merge "dns_core: Allow zero-length DNS responses."	11 years ago
Ashley Sanders	fe804b09b3	ARI: Channels added to Stasis application during WebSocket creation ... ... Prior to ASTERISK-24988, the WebSocket handshake was resolved before Stasis applications were registered. This was done such that the WebSocket would be ready when an application is registered. However, by creating the WebSocket first, the client had the ability to make requests for the Stasis application it thought had been created with the initial handshake request. The inevitable conclusion of this scenario was the cart being put before the horse. ASTERISK-24988 resolved half of the problem by ensuring that the applications were created and registered with Stasis prior to completing the handshake with the client. While this meant that Stasis was ready when the client received the green-light from Asterisk, it also meant that the WebSocket was not yet ready for Stasis to dispatch messages. This patch introduces a message queuing mechanism for delaying messages from Stasis applications while the WebSocket is being constructed. When the ARI event processor receives the message from the WebSocket that it is being created, the event processor instantiates an event session which contains a message queue. It then tries to create and register the requested applications with Stasis. Messages that are dispatched from Stasis between this point and the point at which the event processor is notified the WebSocket is ready, are stashed in the queue. Once the WebSocket has been built, the queue's messages are dispatched in the order in which they were originally received and the queue is concurrently cleared. ASTERISK-25181 #close Reported By: Matt Jordan Change-Id: Iafef7b85a2e0bf78c114db4c87ffc3d16d671a17	11 years ago
Mark Michelson	86034227ca	dns_core: Allow zero-length DNS responses. ... A testsuite test recently failed due to a crash that occurred in the DNS core. The problem was that the test could not resolve an address, did not set a result on the DNS query, and then indicated the query was completed. The DNS core does not handle the case of a query with no result gracefully, and so there is a crash. This changeset makes the DNS system resolver set a result with a zero-length answer in the case that a DNS resolution failure occurs early. The DNS core now also will accept such a response without treating it as invalid input. A unit test was updated to no longer treat setting a zero-length response as off-nominal. Change-Id: Ie56641e22debdaa61459e1c9a042e23b78affbf6	11 years ago
Richard Mudgett	f49bef08a2	rtp_engine.c: Fix performance issue with several channel drivers that use RTP. ... ast_rtp_codecs_get_payload() gets called once or twice for every received RTP frame so it would be nice to not allocate an ao2 object to then have it destroyed shortly thereafter. The ao2 object gets allocated only if the payload type is not set by the channel driver as a negotiated value. The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323. * Made static_RTP_PT[] an array of ao2 objects that ast_rtp_codecs_get_payload() can return instead of an array of structs that must be copied into a created ao2 object. ASTERISK-25296 #close Reported by: Richard Mudgett Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0	11 years ago
Richard Mudgett	33a465249b	res_rtp_asterisk.c: Fix off-nominal crash potential. ... ASTERISK-25296 Reported by: Richard Mudgett Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b	11 years ago
Richard Mudgett	5f925d48b7	rtp_engine.c: Must protect mime_types_len with mime_types_lock. ... Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e	11 years ago
Richard Mudgett	ba7dd38470	res_pjsip_sdp_rtp.c: Fixup some whitespace. ... Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973	11 years ago
Richard Mudgett	3751bf0971	res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list. ... Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2	11 years ago
Richard Mudgett	e2d5d4db35	rtp_engine.h: No sense allowing payload types larger than RFC allows. ... * Tweaked add_static_payload() to not use magic numbers. Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b	11 years ago
Richard Mudgett	bc1eae55cb	rtp_engine.c: Minor tweaks. ... * Fix off nominial ref leak of new_type in ast_rtp_codecs_payloads_set_m_type(). * No need to lock static_RTP_PT_lock in ast_rtp_codecs_payloads_set_m_type() and ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type parameter sanity check. * No need to create ast_rtp_payload_type ao2 objects with a lock since the lock is not used. Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4	11 years ago
Richard Mudgett	d122c1e50b	chan_sip.c: Tweak glue->update_peer() parameter nil value. ... Change glue->update_peer() parameter from 0 to NULL to better indicate it is a pointer. Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd	11 years ago
Richard Mudgett	d12dc97fc9	rtp_engine.h: Misc comment fixes. ... Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43	11 years ago
Richard Mudgett	077c58cd5c	res_pjsip_session.c: Fix crashes seen when call cancelled. ... Two testsuite tests crashed in the same place as a result of an INVITE being CANCELed. tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp The session pointer is no longer in the inv->mod_data[session_module.id] location because the INVITE transaction has reached the terminated state. ASTERISK-25297 #close Reported by: Richard Mudgett Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427	11 years ago
Joshua Colp	82989bc14f	Merge "Add a test event for inband ringing."	11 years ago
Mark Michelson	5fcd1bc556	res_http_websocket: Properly encode 64 bit payload ... A test agent was continuously failing all ARI tests when run against Asterisk 13. As it turns out, the reason for this is that on those test runs, for some reason we decided to use the super extended 64 bit payload length for websocket text frames instead of the extended 16 bit payload length. For 64-bit payloads, the expected byte order over the network is 7, 6, 5, 4, 3, 2, 1, 0 However, we were sending the payload as 3, 2, 1, 0, 7, 6, 5, 4 This meant that we were saying to expect an absolutely MASSIVE payload to arrive. Since we did not follow through on this expected payload size, the client would sit patiently waiting for the rest of the payload to arrive until the test would time out. With this change, we use the htobe64() function instead of htonl() so that a 64-bit byte-swap is performed instead of a 32 bit byte-swap. Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a	11 years ago
Mark Michelson	8fb8988fd4	Add a test event for inband ringing. ... This event is necessary for the bridge_wait_e_options test to be able to confirm that ringing is being played on the local channel that runs the BridgeWait() application with the e(r) option. ASTERISK-25292 #close Reported by Kevin Harwell Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e	11 years ago
Mark Duncan	1d081ec970	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	11 years ago
Jonathan Rose	687597ca8c	holding_bridge: ensure moh participants get frames ... Currently, if a blank musiconhold.conf is used, musiconhold will fail to start for a channel going into a holding bridge with an anticipation of getting music on hold. That being the case, no frames will be written to the channel and that can pose a problem for blind transfers in PJSIP which may rely on frames being written to get past the REFER framehook. This patch makes holding bridges start a silence generator if starting music on hold fails and makes it so that if no music on hold functions are installed that the ast_moh_start function will report a failure so that consumers of that function will be able to respond appropriately. ASTERISK-25271 #close Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99 (cherry picked from commit 8458b8d441)	11 years ago
Joshua Colp	309dd2a409	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	11 years ago
Joshua Colp	a105461f9e	Merge "res_pjsip: Add rtp_keepalive to sample config file."	11 years ago
Mark Michelson	a0c31c7a05	res_pjsip: Add rtp_keepalive to sample config file. ... Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19	11 years ago
Mark Michelson	d97bed46b7	Local channels: Alternate solution to ringback problem. ... Commit 54b25c80c8 solved an issue where a specific scenario involving local channels and a native local RTP bridge could result in ringback still being heard on a calling channel even after the call is bridged. That commit caused many tests in the testsuite to fail with alarming consequences, such as not sending DialBegin and DialEnd events, and giving incorrect hangup causes during calls. This commit reverts the previous commit and implements and alternate solution. This new solution involves only passing AST_CONTROL_RINGING frames across local channels if the local channel is in AST_STATE_RING. Otherwise, the frame does not traverse the local channels. By doing this, we can ensure that a playtones generator does not get started on the calling channel but rather is started on the local channel on which the ringing frame was initially indicated. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39	11 years ago
Matt Jordan	ffb679f7b5	Merge "audiohook: Use manipulated frame instead of dropping it."	11 years ago
Mark Michelson	43173f073c	Merge "audiohook: Read the correct number of samples based on audiohook format."	11 years ago
Joshua Colp	1cc99ba8b6	audiohook: Use manipulated frame instead of dropping it. ... Previous changes to sample rate support in audiohooks accidentally removed code responsible for allowing the manipulate audiohooks to work. Without this code the manipulated frame would be dropped and not used. This change restores it. ASTERISK-25253 #close Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13	11 years ago
Mark Michelson	0b7148e262	Local channels: Do not block control -1 payloads. ... Control frames with a -1 payload are used as a special signal to stop playtones generators on channels. This indication is sent both by app_dial as well as by ast_answer() when a call is answered in case any tones were being generated on a calling channel. This control frame type was made to stop traversing local channel pairs as an optimization, because it was thought that it was unnecessary to send these indications, and allowing such unnecessary control frames to traverse the local channels would cause the local channels to optimize away less quickly. As it turns out, through some special magic dialplan code, it is possible to have a tones being played on a non-local channel, and it is important for the local channel to convey that the tones should be stopped. The result of having tones continue to be played on the non-local channel is that the tones play even once the channel has been bridged. By not blocking the -1 control frame type, we can ensure that this situation does not happen. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815	11 years ago
Joshua Colp	e5fe8d40c8	audiohook: Read the correct number of samples based on audiohook format. ... Due to changes in audiohooks to support different sample rates the underlying storage of samples is in the format of the audiohook itself and not of the format being requested. This means that if a channel is using G722 the samples stored will be at 16kHz. If something subsequently reads from the audiohook at a format which is not the same sample rate as the audiohook the number of samples needs to be adjusted. Given the following example: 1. Channel writing into audiohook at 16kHz (as it is using G722). 2. Chanspy reading from audiohook at 8kHz. The original code would read 160 samples from the audiohook for each 20ms of audio. This is incorrect. Since the audio in the audiohook is at 16kHz the actual number needing to be read is 320. Failure to read this much would cause the audiohook to reset itself constantly as the buffer became full. This change adjusts the requested number of samples by determining the duration of audio requested and then calculating how many samples that would be in the audiohook format. ASTERISK-25247 #close Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d	11 years ago
Joshua Colp	3143bb2e88	Merge "Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c"	11 years ago