8492 Commits (c98c1b3f744fba671ff23a212bb970cc2d38b203)

Author	SHA1	Message	Date
Joshua Colp	25764691b0	Merge "netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API"	8 years ago
Matthew Fredrickson	9f9dce05b2	netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API ... This function originally was used in chan_sip to enable some simplifying assumptions and eventually was copy and pasted into res_pjsip_logger and res_hep. Since it's replicated in three places, it's probably best to move it into the public netsock2 API for these modules to use. Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04	8 years ago
Corey Farrell	b5914d90ac	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	8 years ago
Tzafrir Cohen	6301531416	chan_dahdi: Configurable dialed digit timeouts ... Analog phones dial overlap dialing and it is chan_dahdi's job to read the numbers. It has three timeout constants that this commit converts to channel-level configuration options: * firstdigit_timeout: Default time (ms) to detect first digit * interdigit_timeout: Default time (ms) to detect following digits * matchdigit_timeout: Default time (ms) to wait in case of ambiguous match. This happens when the dialed digits match a number in the current context but are also the prefix of another number. Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213	8 years ago
George Joseph	3bad41257b	Merge "BuildSystem: Add DragonFly BSD."	8 years ago
Jenkins2	9c430569d4	Merge "bridge_softmix: Forward TEXT frames"	8 years ago
Alexander Traud	efe40ff671	BuildSystem: Add DragonFly BSD. ... ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557	8 years ago
George Joseph	4fb7967c73	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	8 years ago
Alexander Traud	3d9345e3ae	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	8 years ago
Richard Mudgett	237d341bbd	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	8 years ago
Jenkins2	fabfe701bb	Merge "res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge"	8 years ago
George Joseph	8af759c088	Merge "chan_sip.c: Fix INVITE with replaces channel ref leak."	8 years ago
Richard Mudgett	0c03eab962	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	8 years ago
Joshua Colp	c7bd554094	pjsip / res_rtp_asterisk: Add support for sending REMB ... This change allows chan_pjsip to be given an AST_FRAME_RTCP containing REMB feedback and pass it to res_rtp_asterisk. Once res_rtp_asterisk receives the frame a REMB RTCP feedback packet is constructed with the appropriate contents and sent to the remote endpoint. ASTERISK-27776 Change-Id: Ic53f821c1560d8924907ad82c4d9c0bc322b38cd	8 years ago
Richard Mudgett	d72a2966da	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	8 years ago
Joshua Colp	e14b0e960d	res_rtp_asterisk: Add support for raising additional RTCP messages. ... This change extends the existing AST_FRAME_RTCP frame type to be able to contain additional RTCP message types, such as feedback messages. The payload type is contained in the subclass which allows knowing what is in the frame itself. The RTCP feedback message type is now handled and REMB[1] messages are raised with their containing information. This also fixes a bug where all feedback messages were triggering video updates instead of just FIR and FUR. Finally RTCP frames are now passed up through the Asterisk core to what is handling the channel, mapped appropriately in the case of bridging, and written to an outgoing stream. Since RTCP frames are on a per-stream basis this is only done on multistream capable channels. [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 ASTERISK-27758 ASTERISK-26366 Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e	8 years ago
Jenkins2	e45e8466f1	Merge "chan_sip: Peers with distinct source ports don't match, regardless of transport."	8 years ago
Jenkins2	beab57a576	Merge "BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD."	8 years ago
Alexander Traud	00789174f6	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	8 years ago
Corey Farrell	572a508ef2	loader: Convert reload_classes to built-in modules. ... * acl (named_acl.c) * cdr * cel * ccss * dnsmgr * dsp * enum * extconfig (config.c) * features * http * indications * logger * manager * plc * sounds * udptl These modules are now loaded at appropriate time by the module loader. Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so the module loader will abort startup on failure of these modules. Some of these modules are still initialized or shutdown from outside the module loader. logger.c is initialized very early and shutdown very late, manager.c is initialized by the module loader but is shutdown by the Asterisk core (too much uses it without holding references). Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f	8 years ago
Jenkins2	1485719531	Merge "Replace direct checks of option_debug with DEBUG_ATLEAST macro."	8 years ago
Corey Farrell	c8a521b6c8	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	8 years ago
Jean Aunis	75a35ee5e8	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	8 years ago
Jenkins2	24c72ea286	Merge "chan_unistim: NetBSD has an incompatible struct in_pktinfo."	8 years ago
Alexander Traud	aabbb49e33	chan_unistim: NetBSD has an incompatible struct in_pktinfo. ... ASTERISK-27714 Reported by: John Nemeth Change-Id: I1b84a89315a5f61222123d21bf35c59224da8990	8 years ago
Joshua Colp	259c80675e	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	8 years ago
Richard Mudgett	8372138cce	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	8 years ago
Oron Peled	5b8fea93d1	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	8 years ago
Corey Farrell	527cf5a570	Remove redundant module checks and references. ... This removes references that are no longer needed due to automatic references created by module dependencies. In addition this removes most calls to ast_module_check as they were checking modules which are listed as dependencies. Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e	8 years ago
Igor Goncharovsky	33d5ab3e69	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	8 years ago
Corey Farrell	9cfdb81e91	loader: Add dependency fields to module structures. ... * Declare 'requires' and 'enhances' text fields on module info structure. * Rename 'nonoptreq' to 'optional_modules'. * Update doxygen comments. Still need to investigate dependencies among modules I cannot compile. Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf	8 years ago
Joshua Colp	6d5f4768a4	chan_sip: Check that an iostream exists before accessing. ... Before getting the file descriptor for an iostream check that it is present. ASTERISK-27534 Change-Id: Ie0aa1394007a37c30e337ea1176a6fb3a63bc99c	8 years ago
Sean Bright	15f8b9b8bf	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	8 years ago
Corey Farrell	c2529a352c	astobj.h: Remove from Asterisk core. ... This is the old ASTOBJ macro's which are no longer used except by the deprecated netsock.c. Move it to the chan_iax2 include folder so it does not get used elsewhere. Change-Id: I7e4ae96678b36b9f41d3cae14b167f110eb5d349	8 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Corey Farrell	1b80ffa495	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	968993a297	Merge "chan_sip: Fix memory leaks."	8 years ago
Jenkins2	262eda7801	Merge "Remove constant conditionals (dead-code)."	8 years ago
Corey Farrell	3c037ef972	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
Corey Farrell	b3e839debd	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
Oron Peled	c02e256407	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
Jenkins2	11a1e07ad2	Merge "netsock: Remove from Asterisk core."	8 years ago
Joshua Colp	5224fd3ab4	Merge changes from topic 'faster-aco' ... * changes: aco: Minimize use of regex. aco: Create ways to minimize use of regex.	8 years ago
Corey Farrell	064c74e4af	netsock: Remove from Asterisk core. ... This moves netsock.c / netsock.h to the chan_iax2 module. netsock.h has been marked deprecated since 13.0.0, chan_iax2 is the only remaining user. Change-Id: I28c6578043bac18de5ea608e136acec4f83d5dd3	8 years ago
Jenkins2	e79f0c0709	Merge "chan_pjsip.c: Improve ast_request() diagnostic msgs."	8 years ago
Richard Mudgett	4a461bcde4	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	6f8b34f9c1	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	bf2d35931d	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
Joshua Colp	e411b7d112	Merge "chan_sip: 3PCC patch for AMI "SIPnotify""	8 years ago
Joshua Colp	c50905756b	Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)"	8 years ago