asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	eb8bbe660e	channel.c: Make CHECK_BLOCKING() save thread LWP id for messages. * Removed an unnecessary call to ast_channel_blocker_set() in __ast_read(). ASTERISK-27625 Change-Id: I342168b999984666fb869cd519fe779583a73834	7 years ago
Richard Mudgett	080508d2eb	channel.c: Fix usage of CHECK_BLOCKING() The CHECK_BLOCKING() macro is used to indicate if a channel's handling thread is about to do a blocking operation (poll, read, or write) of media. A few operations such as ast_queue_frame(), soft hangup, and masquerades use the indication to wake up the blocked thread to reevaluate what is going on. ASTERISK-27625 Change-Id: I4dfc33e01e60627d962efa29d0a4244cf151a84d	7 years ago
Richard Mudgett	0989b63047	autoservice: Don't start channel autoservice if the thread is a user interface. Executing dialplan functions from either AMI or ARI by getting a variable could place the channel into autoservice. However, these user interface threads do not handle the channel's media so we wind up with two threads attempting to handle the media. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: If2dc94ce15ddabf923ed1e2a65ea0ef56e013e49	7 years ago
Richard Mudgett	da54605b8a	ARI POST DTMF: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e	7 years ago
Richard Mudgett	7d874c1af7	AMI PlayDTMF Action: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905	7 years ago
Jenkins2	986cffa18e	Merge "app_confbridge: Enable sending events to participants"	7 years ago
Jenkins2	c59142e273	Merge "app_mp3: remove 10 seconds of silence after mp3 playback"	7 years ago
Sam Wierema	4c7ab73468	app_mp3: remove 10 seconds of silence after mp3 playback This patch changes the way asterisk polls output from mpg123, instead of waiting for 10 seconds(when playing an http url) it now uses a timeout of one second and iterates 10 times using this same timeout. The main difference is that for every timeout asterisk receives it now checks if mpg123 is still running before poll again. ASTERISK-27752 Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620	7 years ago
Jenkins2	c857c71342	Merge "tests/test_utils: Repair ./configure --with-ssl=PATH."	7 years ago
Joshua Colp	f851baed76	Merge "res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP."	7 years ago
Alexander Traud	9d7958672b	tests/test_utils: Repair ./configure --with-ssl=PATH. ASTERISK-27914 Change-Id: Ibcab8f556ee77776f203cff8b06d776a673b7bc4	7 years ago
ktyerman	e1908ea484	chan_iax2: better handling for timeout and EINTR The iax2 module is not handling timeout and EINTR case properly. Mainly when there is an interupt to the kernel thread. In case of ast_io_wait recieves a signal, or timeout it can be an error or return 0 which eventually escapes the thread loop, so that it cant recieve any data. This then causes the modules receive queue to build up on the kernel and stop any communications via iax in asterisk. The proposed patch is for the iax module, so that timeout and EINTR does not exit the thread. ASTERISK-27705 Reported-by: Kirsty Tyerman Change-Id: Ib4c32562f69335869adc1783608e940c3535fbfb	7 years ago
George Joseph	e7a7506f9c	app_confbridge: Enable sending events to participants ConfBridge can now send events to participants via in-dialog MESSAGEs. All current Confbridge events are supported, such as ConfbridgeJoin, ConfbridgeLeave, etc. In addition to those events, a new event ConfbridgeWelcome has been added that will send a list of all current participants to a new participant. For all but the ConfbridgeWelcome event, the JSON message contains information about the bridge, such as its id and name, and information about the channel that triggered the event such as channel name, callerid info, mute status, and the MSID labels for their audio and video tracks. You can use the labels to correlate callerid and mute status to specific video elements in a webrtc client. To control this behavior, the following options have been added to confbridge.conf: bridge_profile/enable_events: This must be enabled on any bridge where events are desired. user_profile/send_events: This must be set for a user profile to send events. Different user profiles connected to the same bridge can have different settings. This allows admins to get events but not normal users for instance. user_profile/echo_events: In some cases, you might not want the user triggering the event to get the event sent back to them. To prevent it, set this to false. A change was also made to res_pjsip_sdp_rtp to save the generated msid to the stream so it can be re-used. This allows participant A's video stream to appear as the same label to all other participants. Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e	7 years ago
Alexander Traud	b01fc2ef3d	res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP. Previously, Asterisk used its script ./configure, to test whether OpenSSL was built with no-srtp (or was simply too old). However, the header file <openssl/opensslconf.h> is the preferred way to detect the local configuration of OpenSSL. As a positive side-effect the script ./configure does not interleave the detection of the Open Settlement Protocol Toolkit (OSPTK) with the detection of individual features of OpenSSL anymore. Change-Id: I3c77c7b00b2ffa2e935632097fa057b9fdf480c0	7 years ago
Jenkins2	a6a22debf1	Merge "rtp: Don't negotiate dynamic codecs using payload."	7 years ago
Jenkins2	8b4c72caea	Merge "res_rtp_asterisk: Allow OpenSSL configured with no-deprecated."	7 years ago
Jenkins2	076ff479ac	Merge "crypto.h: Repair ./configure --with-ssl=PATH."	7 years ago
Joshua Colp	782064852a	Merge "res_crypto: Allow OpenSSL configured with no-deprecated."	7 years ago
Jenkins2	a46e658bc0	Merge "res_srtp: Repair ./configure --with-ssl=PATH."	7 years ago
Jenkins2	56ddc0bd80	Merge "func_odbc: NODATA if SQLNumResultCols returned 0 columns on readsql"	7 years ago
Joshua Colp	41175caee0	rtp: Don't negotiate dynamic codecs using payload. In Asterisk there are some dynamic codecs that have a fixed payload number. This number was being improperly used to negotiate the codec, instead of using the name and sample rate. This could result in the wrong payload number being negotiated for a codec. This change makes it so that only static payloads will be negotiated using their payload number. ASTERISK-27848 Change-Id: Ia865830170fd3f808cdb33104f3d4c4ffdc77570	7 years ago
Jenkins2	200956dcc7	Merge "chan_pjsip: Register for "BEFORE_MEDIA" responses"	7 years ago
Kevin Harwell	7d01ac13a1	Merge "AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses."	7 years ago
Sean Bright	b649682caa	AST-2018-007: iostreams potential DoS when client connection closed prematurely Before Asterisk sends an HTTP response (at least in the case of errors), it attempts to read & discard the content of the request. If the client lies about the Content-Length, or the connection is closed from the client side before "Content-Length" bytes are sent, the request handling thread will busy loop. ASTERISK-27807 Change-Id: I945c5fc888ed92be625b8c35039fc6d2aa89c762	7 years ago
Richard Mudgett	81ac32a85f	AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses. When endpoint specific ACL rules block a SIP request they respond with a 403 forbidden. However, if an endpoint is not identified then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints. * Made endpoint specific ACL rules now respond with a 401 unauthorized which is the same as if an endpoint were not identified. The fix is accomplished by replacing the found endpoint with the artificial endpoint which always fails authentication. ASTERISK-27818 Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32	7 years ago
Alexander Traud	0743ad6422	res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. Furthermore, allow OpenSSL configured with no-dh. Additionally, this change allows auto-negotiation of the elliptic curve/group for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect. ASTERISK-27910 Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537	7 years ago
Alexander Traud	99aed78078	crypto.h: Repair ./configure --with-ssl=PATH. ASTERISK-27908 Change-Id: Iac49d9f82faeb8a4611c6805906bd6d650b1b1d8	7 years ago
Alexander Traud	ca682f0030	res_crypto: Allow OpenSSL configured with no-deprecated. The header <openssl/rsa.h> had to be included explicitly. ASTERISK-27906 Change-Id: I41743801eed998c039d73db7a0762d104a4f75b2	7 years ago
Alexander Traud	234bf4b7ff	res_srtp: Repair ./configure --with-ssl=PATH. ASTERISK-27905 Change-Id: Ibb7dc148a0048f4f9c3b12937ba4240dff0d15e2	7 years ago
Alexei Gradinari	65ff2f057a	func_odbc: NODATA if SQLNumResultCols returned 0 columns on readsql The functions acf_odbc_read/cli_odbc_read ignore a number of columns returned by the SQLNumResultCols. If the number of columns is zero it means no data. In this case, a SQLFetch function has to be not called, because it will cause an error. ASTERISK-27888 #close Change-Id: Ie0f7bdac6c405aa5bbd38932c7b831f90729ee19	7 years ago
George Joseph	1725eaf8fb	chan_pjsip: Register for "BEFORE_MEDIA" responses chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant it was not updating HANGUPCAUSE for 4XX responses. If the remote end sent a "180 Ringing", then a "486 Busy", the hangup cause was left at "180 Normal Clearing". * Removed chan_pjsip_incoming_response from the original session supplement (which was handling only "AFTER MEDIA") and added it to a new session supplement which accepts both "BEFORE_MEDIA" and "AFTER_MEDIA". * Also cleaned up some cleanup code in load module. ASTERISK-27902 Change-Id: If9b860541887aca8ac2c9f2ed51ceb0550fb007a	7 years ago
Alexander Traud	9f2eb17005	ooh323c: GCC 8.1 warned about output truncated before terminating nul. ASTERISK-27901 Change-Id: I5a8e894f4924ef52e3094f6870656a559d67f3d7	7 years ago
George Joseph	7b5fc5d20f	Merge "pjsip_options: handle modification of qualify options in realtime"	7 years ago
George Joseph	fdc931ffc6	Merge "pjsip_options: show/reload AOR qualify options using CLI"	7 years ago
George Joseph	49196c3a7b	Merge "app_confbridge: Add talking indicator for ConfBridgeList AMI response"	7 years ago
George Joseph	13ac76a230	Merge "cdr_mysql: my_connect_db(): reduce indentation"	7 years ago
George Joseph	5978883d13	Merge "cdr_mysql: split mysql init out of my_load_module"	7 years ago
Joshua Colp	c98c1b3f74	Merge "bridge_channel.c: Fix Deadlock when using Local channels and fax gateway"	7 years ago
Joshua Colp	2151903a16	Merge "tcptls: Allow OpenSSL configured with no-dh."	7 years ago
George Joseph	76339b1962	Merge "tcptls.h: Repair ./configure --with-ssl=PATH."	7 years ago
Alexei Gradinari	7af5e86821	pjsip_options: show/reload AOR qualify options using CLI Currentrly pjsip_options code does not handle the situation when the AOR qualify options were changed. Also there is no way to find out what qualify options are using. This patch add CLI commands to show and synchronize Aor qualify options: pjsip show qualify endpoint <id> Show the current qualify options for all Aors on the PJSIP endpoint. pjsip show qualify aor <id> Show the PJSIP Aor current qualify options. pjsip reload qualify endpoint <id> Synchronize the qualify options for all Aors on the PJSIP endpoint. pjsip reload qualify aor <id> Synchronize the PJSIP Aor qualify options. ASTERISK-27872 Change-Id: I1746d10ef2b7954f2293f2e606cdd7428068c38c	7 years ago
Alexei Gradinari	e46b442e38	pjsip_options: handle modification of qualify options in realtime Currentrly pjsip_options code does not handle the situation when the qualify options were changed in realtime database. Only 'module reload res_pjsip' helps. This patch add a check on contact add/update observers if the contact qualify options are different than local aor qualify options. If the qualify options were modified then synchronize the pjsip_options AOR local state. ASTERISK-27872 Change-Id: Id55210a18e62ed5d35a88e408d5fe84a3c513c62	7 years ago
George Joseph	99aad2f0af	Merge "tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated."	7 years ago
Joshua Colp	8b08a8437e	Merge "app_meetme: Fix manager event documentation for several events."	7 years ago
Pirmin Walthert	e078558038	bridge_channel.c: Fix Deadlock when using Local channels and fax gateway ast_indicate is invoked with the bridge locked. As ast_indicate locks the other end of the bridge as well this can lead to a deadlock in some situations. (Especially when a different thread does the same in the reverse order). This patch calls ast_indicate after unlocking the bridge which fixes the deadlock. Calling ast_indicate with these parameters without locking the bridge should be safe as this is done at different places without a bridge lock. ASTERISK-27094 #close Reported-by: David Brillert Change-Id: I5f86c1e2ce75b9929a36ab589b18c450e62ea35f	7 years ago
George Joseph	437ab41881	app_sendtext: Allow content types other than text/plain There was no real reason to limit the conteny type to text/plain other than that's what it was limited to before. Now any text/* content type will be allowed for channel drivers that don't support enhanced messaging and any type will be allowed for channel drivers that do support enhanced messaging. Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9	7 years ago
William McCall	a7f4121238	app_confbridge: Add talking indicator for ConfBridgeList AMI response When an AMI client connects, it cannot determine if a user was talking prior to a transition in the user speaking state (which would generate a ConfbridgeTalking event). This patch causes app_confbridge to track the talking state and make this state available via ConfBridgeList. ASTERISK-27877 #close Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6	7 years ago
Joshua Colp	227f59fdac	Merge "ast_coredumper: Fix output directory and variable precedence"	7 years ago
Richard Mudgett	6bbede84fb	app_meetme: Fix manager event documentation for several events. The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and MeetmeTalkRequest AMI events were documented with sending out a Usernum header when the User header was actually output. * Change the online documentation to match reality. ASTERISK-27873 ASTERISK-25261 Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39	7 years ago
Joshua Colp	c63cd006ba	Merge "libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated."	7 years ago

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