8406 Commits (e3de4bc46e0332050f3fd296202abe94d13c20a5)

Author	SHA1	Message	Date
Joshua Colp	7609249a22	Merge "crypto.h: Repair ./configure --with-ssl=PATH." into 13	7 years ago
Alexander Traud	2c3ad1e40d	crypto.h: Repair ./configure --with-ssl=PATH. ... ASTERISK-27908 Change-Id: Iac49d9f82faeb8a4611c6805906bd6d650b1b1d8	7 years ago
George Joseph	98da1971e3	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
Corey Farrell	d893e57c90	Fix GCC 8 build issues. ... This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	7 years ago
George Joseph	42abc9c430	Merge "BuildSystem: Add DragonFly BSD." into 13	7 years ago
George Joseph	c5d19565fe	Merge "bridge_softmix: Forward TEXT frames" into 13	7 years ago
Alexander Traud	7b219311eb	BuildSystem: Add DragonFly BSD. ... ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557	7 years ago
George Joseph	be7d4faed5	bridge_softmix: Forward TEXT frames ... Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	7 years ago
Alexander Traud	fb9634bcb3	chan_vpb: Avoid GNU old-style field designator extension. ... clang 6.0 warned about this. Beside that, this change removes the used variable 'desc'. ASTERISK-27808 Change-Id: Ia26bdcc0a562c058151814511cfcf70ecafa595b	7 years ago
Richard Mudgett	12aa25b2e1	res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. ... ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3	7 years ago
Jenkins2	0edc4ade93	Merge "res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge" into 13	7 years ago
Richard Mudgett	72b16ee400	res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge ... There is a problem when an INVITE-with-Replaces transfer targets a channel in a ConfBridge. The transfer will unconditionally swap out the ConfBridge channel. Unfortunately, the ConfBridge state will not be aware of this change. Unexpected behavior will happen as a result since ConfBridge channels currently can only be replaced by a masquerade and not normal bridge channel moves. * We just need to pretend that the channel isn't in a bridge (like other transfer methods already do) so the transfer channel will masquerade into the ConfBridge channel. Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82	7 years ago
Richard Mudgett	ea055386e0	chan_sip.c: Fix INVITE with replaces channel ref leak. ... Given the below call scenario: A -> Ast1 -> B C <- Ast2 <- B 1) A calls B through Ast1 2) B calls C through Ast2 3) B transfers A to C When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to send an INVITE with replaces to Ast2. Ast2 then leaks a channel ref of the channel between Ast1 and Ast2. Channel ref leaks are easily seen in the CLI "core show channels" output. The leaked channels appear in the output but you can do nothing with them and they never go away unless you restart Asterisk. * Properly account for the channel refs when imparting a channel into a bridge when handling an INVITE with replaces in handle_invite_replaces(). The ast_bridge_impart() function steals a channel ref but the code didn't account for how many refs were held by the code at the time and which ref was stolen. * Eliminated RAII_VAR in handle_invite_replaces(). ASTERISK-27740 Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4	7 years ago
Jenkins2	b9eb86b56e	Merge "chan_sip: Peers with distinct source ports don't match, regardless of transport." into 13	7 years ago
Alexander Traud	5b80e97fff	BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD. ... In the script ./configure, AST_EXT_LIB_CHECK checks for external libraries. Some libraries do not specify all their dependencies and require additional shared libraries. In AST_EXT_LIB_CHECK, this is the fifth parameter. However, if a library is specified there, it must exist on the platform, because ./configure tries to compile/link/execute a small app using those statements. For example, the library libdl.so is Linux specific and does not exist on BSD-like platforms. Furthermore, no supported platform/version was found, which still (ever?) requires those additional libraries. Therefore, they were simply removed. Finally, this change adds the error code ESTRPIPE to the channel driver chan_alsa for those platforms which lack it, again for example NetBSD. ASTERISK-27720 Change-Id: I3b21f2135f6cbfac7590ccdc2df753257f426e0b	7 years ago
Joshua Colp	f05ac26d4a	Merge "Replace direct checks of option_debug with DEBUG_ATLEAST macro." into 13	7 years ago
Corey Farrell	b81eadcefc	Replace direct checks of option_debug with DEBUG_ATLEAST macro. ... Checking option_debug directly is incorrect as it ignores file/module specific debug settings. This system-wide change replaces nearly all direct checks for option_debug with the DEBUG_ATLEAST macro. Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0	7 years ago
Jean Aunis	a35a654a52	chan_sip: Fix improper RTP framing on outgoing calls ... The "ptime" SDP parameter received in a SIP response was not honoured. Moreover, in the abscence of this "ptime" parameter, locally configured framing was lost during response processing. This patch systematically stores the framing information in the ast_rtp_codecs structure, taking it from the response or from the configuration as appropriate. ASTERISK-27674 Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c	7 years ago
Jenkins2	451fec3044	Merge "chan_unistim: NetBSD has an incompatible struct in_pktinfo." into 13	7 years ago
Alexander Traud	7b5e0960ef	chan_unistim: NetBSD has an incompatible struct in_pktinfo. ... ASTERISK-27714 Reported by: John Nemeth Change-Id: I1b84a89315a5f61222123d21bf35c59224da8990	7 years ago
Joshua Colp	4da0e19b89	chan_sip: Emit a second ringing event to ensure channel is found. ... When constructing a dialog-info+xml NOTIFY message a ringing channel is found if the state is ringing and further information is placed into the message. Due to the migration to the Stasis message bus this did not always work as expected. This change raises a second ringing event in such a way to guarantee that the event is received by chan_sip and another lookup is done to find the ringing channel. ASTERISK-24488 Change-Id: I547a458fc59721c918cb48be060cbfc3c88bcf9c	7 years ago
Richard Mudgett	ba63dad12d	chan_sip.c: Fix crash processing CANCEL. ... Check if initreq data string exists before using it when processing a CANCEL request. ASTERISK-27666 Change-Id: Id1d0f0fa4ec94e81b332b2973d93e5a14bb4cc97	7 years ago
Oron Peled	0fc3e831a7	chan_console: don't read and write at the same time ... It seems that the ALSA backend of PortAudio doesn't know how to both read and write at the same time by adding a per-device mutex. FIXME: currently only a draft version. Need to either auto-detect we work with the ALSA backend or add an extra configuration option to use this mutex. ASTERISK-27426 #close Change-Id: I635eacee45f5413faa18f5a3b606af03b926dacb	7 years ago
Jenkins2	5d74e07793	Merge "chan_unistim: Fix hold function ability to lock/crash asterisk" into 13	7 years ago
Igor Goncharovsky	1488efb3a8	chan_unistim: Fix hold function ability to lock/crash asterisk ... This patch fix chan_unistim hold functions to correctly support hold function in different states possible in case of multiple lines established on the phone ASTERISK-26596 #close Change-Id: Ib1e04e482e7c8939607a42d7fddacc07e26e14d4	7 years ago
Yasuhiko Kamata	c0a4a939cc	chan_sip: 3PCC patch for AMI "SIPnotify" ... A patch for sending in-dialog SIP NOTIFY message with "SIPnotify" AMI action. ASTERISK-27461 (created patch for 13 branch manually due to merge conflict) Change-Id: I255067f02e2ce22c4b244f12134b9a48d210c22a	7 years ago
Sean Bright	d2c836d24a	ice: Increase foundation buffer size ... Per RFC 5245, the foundation specified with an ICE candidate can be up to 32 characters but we are only allowing for 31. ASTERISK-27498 #close Reported by: Michele Prà Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf	7 years ago
Sean Bright	ce3d56920b	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Corey Farrell	82b6ba976f	Fix Common Typo's. ... Fix instances of: * Retreive * Recieve * other then * different then * Repeated words ("the the", "an an", "and and", etc). * othterwise, teh ASTERISK-24198 #close Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31	8 years ago
Jenkins2	eda4f59656	Merge "chan_sip: Fix memory leaks." into 13	8 years ago
Jenkins2	b539ea2a38	Merge "Remove constant conditionals (dead-code)." into 13	8 years ago
Corey Farrell	0e5d8ad09b	chan_sip: Fix memory leaks. ... In change_redirecting_information variables we use ast_strlen_zero to see if a value should be saved. In the case where the value is not NULL but is a zero length string we leaked. handle_response_subscribe leaked a reference to the ccss monitor instance. Change-Id: Ib11444de69c3d5b2360a88ba2feb54d2c2e9f05f	8 years ago
Oron Peled	fc86e58a5a	chan_console: Use correct parameter for 'set active' ... chan_console supports multiple devices but the CLI only works on a single device. 'console set active' selects this device. Sadly that CLI picks the wrong command-line parameter and will only work for a device called 'active'. ASTERISK-27490 #close Change-Id: I2f0e5fe63db19845bee862575b739360797dc73d	8 years ago
Corey Farrell	d6b2f457d9	Remove constant conditionals (dead-code). ... Some variables are set and never changed, making them constant. This means that code in the 'false' block of the conditional is unreachable. In chan_skinny and res_config_ldap I used preprocessor directive `#if 0` as I'm unsure if the unreachable code could be enabled in the future. Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059	8 years ago
Jenkins2	56a931f64c	Merge "aco: Minimize use of regex." into 13	8 years ago
Jenkins2	232a61006c	Merge "chan_pjsip.c: Improve ast_request() diagnostic msgs." into 13	8 years ago
Richard Mudgett	a368ad9229	chan_pjsip.c: Improve ast_request() diagnostic msgs. ... Attempting to dial PJSIP/endpoint when the endpoint doesn't exist and disable_multi_domain=no results in a misleading empty endpoint name message. The message should say the endpoint was not found. * Added missing endpoint not found message. * Added more information to the empty endpoint name msgs if available. * Eliminated RAII_VAR in request(). Change-Id: I21da85ebd62dcc32115b2ffcb5157416ebae51e4	8 years ago
Corey Farrell	e3bd95f55c	chan_sip: Add security event for calls to invalid extension. ... Log a message to security events when an INVITE is received to an invalid extension. ASTERISK-25869 #close Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f	8 years ago
Corey Farrell	501f4dcdd8	aco: Minimize use of regex. ... Remove nearly all use of regex from ACO users. Still remaining: * app_confbridge has a legitamate use of option name regex. * ast_sorcery_object_fields_register is implemented with regex, all callers use simple prefix based regex. I haven't decided the best way to fix this in both 13/15 and master. Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b	8 years ago
Jenkins2	992b7197b7	Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)" into 13	8 years ago
Jenkins2	9b94e440ae	Merge "chan_sip: Don't crash in Dial on invalid destination" into 13	8 years ago
Sean Bright	e1a358a6e4	chan_sip: Don't send trailing \0 on keep alive packets ... This is a partial fix for ASTERISK~25817 but does not address the comments regarding RFC 5626. Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420	8 years ago
Sean Bright	ce2c89ce68	chan_sip: Don't crash in Dial on invalid destination ... Stripping the DNID in a SIP dial string can result in attempting to call the argument parsing macros on an empty string, causing a crash. ASTERISK-26131 #close Reported by: Dwayne Hubbard Patches: dw-asterisk-master-dnid-crash.patch (license #6257) patch uploaded by Dwayne Hubbard Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e	8 years ago
Richard Mudgett	73b3390dbe	chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri) ... This patch does three things associated with the initial incoming INVITE request URI. 1) Add access to the full initial incoming INVITE request URI. 2) We were not setting DNID on incoming PJSIP channels. The DNID is the user portion of the initial incoming INVITE Request-URI. The value is accessed by reading CALLERID(dnid). 3) Fix CHANNEL(pjsip,target_uri) documentation. * The initial incoming INVITE request URI is now available using CHANNEL(pjsip,request_uri). * Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the initial incoming INVITE request URI user portion. * CHANNEL(pjsip,target_uri) now correctly documents that the target URI is the contact URI. * Refactored print_escaped_uri() out of channel_read_pjsip() to handle pjsip_uri_print() error condition when the buffer is too small. ASTERISK-27478 Change-Id: I512e60d1f162395c946451becb37af3333337b33	8 years ago
Sean Bright	f726f11974	utils: Add convenience function for setting fd flags ... There are many places in the code base where we ignore the return value of fcntl() when getting/setting file descriptior flags. This patch introduces a convenience function that allows setting or clearing file descriptor flags and will also log an error on failure for later analysis. Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7	8 years ago
Richard Mudgett	594faa192d	security-events: Fix SuccessfulAuth using_password declaration. ... The SuccessfulAuth using_password field was declared as a pointer to a uint32_t when the field was later read as a uint32_t value. This resulted in unnecessary casts and a non-portable field value reinterpret in main/security_events.c:add_json_object(). i.e., It would work on a 32 bit architecture but not on a 64 bit big endian architecture. Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935	8 years ago
Alexander Traud	64942276d1	chan_sip: Peers with distinct source ports don't match, regardless of transport. ... Previously, peers connected via TCP (or TLS) were matched by ignoring their source port. One cannot say anything when protocol:IP:port match, yes (see <http://stackoverflow.com/q/3329641>). However, when the ports do not match, the peers do not match as well. This change allows two peers connected to an Asterisk server via TCP (or TLS) behind a NAT (= same source IP address) to be differentiated via their port as well. ASTERISK-27457 Reported by: Stephane Chazelas Change-Id: Id190428bf1d931f2dbfd4b293f53ff8f20d98efa	8 years ago
George Joseph	0cdd31ee10	AST-2017-013: chan_skinny: Call pthread_detach when sess threads end ... chan_skinny creates a new thread for each new session. In trying to be a good cleanup citizen, the threads are joinable and the unload_module function does a pthread_cancel() and a pthread_join() on any sessions that are active at that time. This has an unintended side effect though. Since you can call pthread_join on a thread that's already terminated, pthreads keeps the thread's storage around until you explicitly call pthread_join (or pthread_detach()). Since only the module_unload function was calling pthread_join, and even then only on the ones active at the tme, the storage for every thread/session ever created sticks around until asterisk exits. * A thread can detach itself so the session_destroy() function now calls pthread_detach() just before it frees the session memory allocation. The module_unload function still takes care of the ones that are still active should the module be unloaded. ASTERISK-27452 Reported by: Juan Sacco Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd	8 years ago
Alexander Traud	41498dcb5d	chan_sip: ICE contained square brackets around IPv6 addresses. ... ASTERISK-27434 Change-Id: Iaeed89b4fa05d94c5f0ec2d3b7cd6e93d2d5a8f7	8 years ago
Richard Mudgett	8a7dd5cc44	chan_pjsip.c: Improve answer failure log messages. ... * Balanced the session->inv_session refs on answer failure. Change-Id: I33542d639d37e692cb46550b972a5fcfc3b804b8	8 years ago