asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	b719f56c72	res_pjsip_sdp_rtp: Restore removed NULL check. When sending an RTP keepalive, we need to be sure we're not dealing with a NULL RTP instance. There had been a NULL check, but the commit that added the rtp_timeout and rtp_hold_timeout options removed the NULL check. Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64	10 years ago
Joshua Colp	e18c300550	res_http_websocket: When shutting down a session don't close closed socket Due to the use of ast_websocket_close in session termination it is possible for the underlying socket to already be closed when the session is terminated. This occurs when the close frame is attempted to be written out but fails. Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b	10 years ago
Joshua Colp	0bc6fe6f7d	Merge "res_http_websocket: Forcefully terminate on write errors." into 13	10 years ago
Joshua Colp	b4e9416138	res_http_websocket: Forcefully terminate on write errors. The res_http_websocket module will currently attempt to close the WebSocket connection if fatal cases occur, such as when attempting to write out data and being unable to. When the fatal cases occur the code attempts to write a WebSocket close frame out to have the remote side close the connection. If writing this fails then the connection is not terminated. This change forcefully terminates the connection if the WebSocket is to be closed but is unable to send the close frame. ASTERISK-25312 #close Change-Id: I10973086671cc192a76424060d9ec8e688602845	10 years ago
Richard Mudgett	c126afe18f	res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message. If the saved SUBSCRIBE message is not parseable for whatever reason then Asterisk could crash when libpjsip tries to parse the message and adds an error message to the parse error list. * Made ast_sip_create_rdata() initialize the parse error rdata list. The list is checked after parsing to see that it remains empty for the function to return successful. ASTERISK-25306 Reported by Mark Michelson Change-Id: Ie0677f69f707503b1a37df18723bd59418085256	10 years ago
Matt Jordan	47d9ff1741	Merge "res/res_format_attr_silk: Expose format attributes to other modules" into 13	10 years ago
Joshua Colp	c57b78d4c9	Merge "Replace htobe64 with htonll" into 13	10 years ago
Joshua Colp	f733bc10b1	Merge "res_pjsip_pubsub: More accurately persist packet." into 13	10 years ago
Matt Jordan	8e194047ac	res/res_format_attr_silk: Expose format attributes to other modules This patch adds the .get callback to the format attribute module, such that the Asterisk core or other third party modules can query for the negotiated format attributes. Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c	10 years ago
David M. Lee	26f0559a94	Replace htobe64 with htonll We don't have a compatability function to fill in a missing htobe64; but we already have one for the identical htonll. Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac	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
Joshua Colp	cf27200391	Merge "ARI: Deleting log channels" into 13	10 years ago
Joshua Colp	4b1bd40d7e	Merge "res_pjsip: Ensure sanitized XML is NULL terminated." into 13	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 Michelson	e25569ef95	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	10 years ago
Joshua Colp	9182c9e4e6	Merge "res_rtp_asterisk.c: Fix off-nominal crash potential." into 13	10 years ago
Joshua Colp	ba40b07ddc	Merge topic 'misc_rtp_tweaks' into 13 * changes: rtp_engine.c: Must protect mime_types_len with mime_types_lock. res_pjsip_sdp_rtp.c: Fixup some whitespace.	10 years ago
Joshua Colp	8521a86367	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	10 years ago
Joshua Colp	c07fa843ec	Merge "res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list." into 13	10 years ago
Mark Michelson	56d11d4198	Merge "res_rtp_asterisk: Don't leak temporary key when enabling PFS." into 13	10 years ago
Joshua Colp	9a12804e59	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	10 years ago
Mark Michelson	27dc2094e9	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	10 years ago
Matt Jordan	1aa23a5d1b	Merge "res_http_websocket: Avoid passing strlen() to ast_websocket_write()." into 13	10 years ago
Mark Michelson	39cc28f6ea	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	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
Mark Michelson	e28fbebc57	Merge "ARI: Rotate log channels." into 13	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	84262749d2	res_rtp_asterisk.c: Fix off-nominal crash potential. ASTERISK-25296 Reported by: Richard Mudgett Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b	10 years ago
Richard Mudgett	a93b7a927c	res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list. Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2	10 years ago
Richard Mudgett	741fa0d26d	res_pjsip_sdp_rtp.c: Fixup some whitespace. Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973	10 years ago
Richard Mudgett	13eb491e35	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	10 years ago
Mark Michelson	48698a5e21	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	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
Michael Cargile	a23adcca3d	res/res_musiconhold: Add a warning when MOH does not exist Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b	10 years ago
Matt Jordan	03064daeb2	res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf Misconfiguring sorcery.conf with a 'config' wizard with no extra data will currently crash Asterisk on startup, as the wizard requires a comma delineated list to parse. This patch updates res_sorcery_config to check for the presence of the data before it starts manipulating it. Change-Id: I4c97512e8258bc82abe190627a9206c28f5d3847	10 years ago
Matt Jordan	9d7f689b4b	Merge "ARI: Add support for push configuration of dynamic object" into 13	10 years ago
Matt Jordan	8bcf6d2801	ARI: Add support for push configuration of dynamic object This patch adds support for push configuration of dynamic, i.e., sorcery, objects in Asterisk. It adds three new REST API calls to the 'asterisk' resource: * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current object given its ID. This returns back a list of ConfigTuples, which define the fields and their present values that make up the object. * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an object. A body may be passed with the request that contains fields to populate in the object. The same format as what is retrieved using the GET operation is used for the body, save that we specify that the list of fields to update are contained in the "fields" attribute. * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic object from its backing storage. Note that the success/failure of these operations is somewhat configuration dependent, i.e., you must be using a sorcery wizard that supports the operation in question. If a sorcery wizard does not support the create or delete mechanisms, then the REST API call will fail with a 403 forbidden. ASTERISK-25238 #close Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c	10 years ago
Matt Jordan	80eaf0b025	Merge "res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer()." into 13	10 years ago
Matt Jordan	174f0e9d4d	Merge "res_pjsip_session.c: Add some helpful comments and minor tweaks." into 13	10 years ago
Matt Jordan	bba2c44ac4	Merge "res_pjsip_session.c: Fix off nominal crash potential in debug message." into 13	10 years ago
Richard Mudgett	243c0d1609	parking_applications.c: Fix ast_verb() line terminator. Change-Id: I8797238c71563e243c48c6145b4f1ae58f91f775	10 years ago
Richard Mudgett	c782320c68	res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park. setup_park_common_datastore() was assuming that a non-NULL string returned for the ATTENDEDTRANSFER and BLINDTRANSFER channel variables are not empty strings. Things got crashy as a result. * Made setup_park_common_datastore() treat the channel variable values the same whether they are NULL or empty for ATTENDEDTRANSFER and BLINDTRANSFER. ASTERISK-25254 #close Reported by: Richard Mudgett Change-Id: I9a9c174b33f354f35f82cc6b7cea8303adbaf9c2	10 years ago
Richard Mudgett	2735dd5b2d	res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer(). Change-Id: I9e115dee74bd72e06081d0ee73ecdeb886caa5fb	10 years ago
Richard Mudgett	3d0ca343ca	res_pjsip_session.c: Add some helpful comments and minor tweaks. Change-Id: I742aeeaf5f760593f323a00fb691affe22e35743	10 years ago
Richard Mudgett	8d08bb179c	res_pjsip_session.c: Fix off nominal crash potential in debug message. Change-Id: I09928297927ee85f7655289acee3a586816466bc	10 years ago
Mark Michelson	ca41785774	Merge "ARI: Fixed unload mode for unload module." into 13	10 years ago
Benjamin Ford	3384e64ef6	ARI: Fixed unload mode for unload module. Changed the unload mode to AST_FORCE_SOFT from AST_FORCE_FIRM, which would unload a module even if it was in use. * Changed unload mode to proper mode ASTERISK-25173 Change-Id: If2402487b5bce05d9770f25f65f5c8e292ad5533	10 years ago
Matt Jordan	00d858da87	Merge "res_pjsip_session.c: Fix crash on call disconnect." into 13	10 years ago

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3087 Commits (bd867cd0787b124984caf2604478212651ea4c03)