asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	86e8ab5ed4	voicemail API callbacks: Extract the sayname API call to its own registerd callback. * Extract the sayname API call to its own registerd callback. This allows the app_directory and app_chanspy applications to say a mailbox owner's name using an alternate provider when app_voicemail is not available because you are using res_mwi_external. app_directory still uses the voicemail.conf file. AFS-64 #close Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	d87f8c429e	pjsip cli: Change Identify to show CIDR notation instead of netmasks. * Added ast_sockaddr_cidr_bits() to count the 1 bits in an ast_sockaddr. * Added ast_ha_join_cidr() which uses ast_sockaddr_cidr_bits() for the netmask instead of ast_sockaddr_stringify_addr. * Changed res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr() instead of ast_ha_join() for the CLI output. This is a CLI change only. AMI was not affected. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3652/ ........ Merged revisions 416737 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	bd0aa4fb04	res_http_websocket: read/write string fixup There was a problem when reading a string from the websocket. It assumed the received data had a null terminator and tried to write the data to an ast_str. This of course could/would read past the end of the given buffer while writing the data to the internal buffer of ast_str. Modified the the code to correctly place a null terminator on the result string. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416394 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9cc1a8e893	stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	13e697f8c0	AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages. ASTERISK-23673 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/ ........ Merged revisions 416066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416067 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416070 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4ca5745dbe	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	58f4c18ab6	res_pjsip_pubsub: Persist subscriptions in sorcery so they are recreated on startup. This change makes res_pjsip_pubsub persist inbound subscriptions in sorcery. By default this uses the local astdb but it can also be configured to store within an outside database. When Asterisk is started these subscriptions are recreated if they have not expired. Notifications are sent to the devices which have subscribed and they are none the wiser that the system has restarted. Review: https://reviewboard.asterisk.org/r/3598/ ........ Merged revisions 415766 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	20a14e568f	bridges/bridge_native_rtp: Reconfigure bridge on removal of framehook This patch is a re-do of r414122. When r414122 was merged, a major problem with it was uncovered. UNBRIDGE soft hangup flags have a catastrophic effect on the pbx core if they leak out from the bridge layer: the channel gets hung up. With the number of threads involved in a blind transfer, and with the initial patch, it was likely that this would occur. This caused a large number of test failures This patch is nearly identical with the one proposed in r414122, save for the following changes: - We explicitly clear the UNBRIDGE flag when setting an after goto on a channel in a bridge - Defensively, if we encounter an UNBRIDGE flag in the pbx core, we handle it https://reviewboard.asterisk.org/r/3585/ ........ Merged revisions 415443 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	30b7ba05e7	bridge.h: Remove redundant struct ast_bridge_channel forward declaration. ........ Merged revisions 415427 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	5ca495ed2f	chan_sip: Fix order of variables specified in SIPNotify action Prior to this patch, sequential variables would be ordered in reverse from the order specified in the manager action. Review: https://reviewboard.asterisk.org/r/3588/ ........ Merged revisions 415359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415390 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415410 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	077c4187d9	Split astobj2.c into more maintainable components. Split astobj2.c into the following files to improve maintainability. astobj2.c - object primitives, object primitive misc and initialization code. astobj2_private.h - internal object declarations needed by the containers. astobj2_container.c - generic conainer and container misc code. astobj2_container_hash.c - hash container specific code. astobj2_container_rbtree.c - rbtree container specific code. astobj2_container_private.h - generic container definitions and rtti prototypes. https://reviewboard.asterisk.org/r/3576/ ........ Merged revisions 415317 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	e763d70470	res_http_websocket: Create a websocket client Added a websocket server client in Asterisk. Asterisk has a websocket server, but not a client. The ability to have Asterisk be able to connect to a websocket server can potentially be useful for future work (for instance this could allow ARI to connect back to some external system, although more work would be needed in order to incorporate that). Also a couple of things to note - proxy connection support has not been implemented and there is limited http response code handling (basically, it is connect or not). Also added an initial new URI handling mechanism to core. Internet type URI's are parsed into a data structure that contains pointers to the various parts of the URI. (closes issue ASTERISK-23742) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/3541/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	53968c00b3	TALK_DETECT: A channel function that raises events when talking is detected This patch adds a new channel function TALK_DETECT that, when set on a channel, causes events indicating the start/stop of talking on a channel to be emitted to both AMI and ARI clients. The function allows setting both the silence threshold (the length of silence after which we decide no one is talking) as well as the talking threshold (the amount of energy that counts as talking). Parameters can be updated on a channel after talk detection has been enabled, and talk detection can be removed at any time. The events raised by the function use a nomenclature similar to existing AMI/ARI events. For AMI: ChannelTalkingStart/ChannelTalkingStop For ARI: ChannelTalkingStarted/ChannelTalkingFinished Review: https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close Reported by: Matt Jordan ........ Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	fb5690ce4b	Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6107712857	AMI/ARI: Update version numbers Update the semantic versioning of ARI to 1.3.0 and AMI to 2.3.0 to account for backwards compatible changes going from 12.2.0 to 12.3.0. ........ Merged revisions 414765 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	69125a7ae2	res_pjsip_session: Fix leaked video RTP ports. Simply enabling PJSIP to negotiage a video codec (e.g., h264) would leak video RTP ports if the codec were not negotiated by an incoming call. * Made add_sdp_streams() associate the handler with the media stream if the handler handled the media stream. Otherwise, when the ast_sip_session_media object was destroyed it didn't know how to clean up the RTP resources. * Fixed sdp_requires_deferral() associating the handler with the media stream when deciding if the SDP processing needs to be deferred for T.38. Like the leaked video RTP ports, the T.38 handler needs to clean up allocated resources from deciding if SDP processing needs to be deffered. * Cleaned up some dead code in handle_incoming_sdp() and sdp_requires_deferral(). ASTERISK-23721 #close Reported by: cervajs Review: https://reviewboard.asterisk.org/r/3571/ ........ Merged revisions 414749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	cf21644d6a	ARI: Add ability to raise arbitrary User Events User events can now be generated from ARI. Events can be signalled with arbitrary json variables, and include one or more of channel, bridge, or endpoint snapshots. An application must be specified which will receive the event message (other applications can subscribe to it). The message will also be delivered via AMI provided a channel is attached. Dialplan generated user event messages are still transmitted via the channel, and will only be received by a stasis application they are attached to or if the channel is subscribed to. This change also introduces the multi object blob mechanism used to send multiple snapshot types in a single message. The dialplan app UserEvent was also changed to use multi object blob, and a new stasis message type created to handle them. ASTERISK-22697 #close Review: https://reviewboard.asterisk.org/r/3494/ ........ Merged revisions 414405 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d00882108f	res_pjsip_refer: Fix bugs involving Parking/PJSIP/transfers PJSIP would never send the final 200 Notify for a blind transfer when transferring to parking. This patch fixes that. In addition, it fixes a reference leak when performing blind transfers to non-bridging extensions. Review: https://reviewboard.asterisk.org/r/3485/ ........ Merged revisions 414400 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9cee08f502	res_corosync: Update module to work with Stasis (and compile) This patch fixes res_corosync such that it works with Asterisk 12. This restores the functionality that was present in previous versions of Asterisk, and ensures compatibility with those versions by restoring the binary message format needed to pass information from/to them. The following changes were made in the core to support this: * The event system has been partially restored. All event definition and event types in this patch were pulled from Asterisk 11. Previously, we had hoped that this information would live in res_corosync; however, the approach in this patch seems to be better for a few reasons: (1) Theoretically, ast_events can be used by any module as a binary representation of a Stasis message. Given the structure of an ast_event object, that information has to live in the core to be used universally. For example, defining the payload of a device state ast_event in res_corosync could result in an incompatible device state representation in another module. (2) Much of this representation already lived in the core, and was not easily extensible. (3) The code already existed. :-) * Stasis message types now have a message formatter that converts their payload to an ast_event object. * Stasis message forwarders now handle forwarding to themselves. Previously this would result in an infinite recursive call. Now, this simply creates a new forwarding object with no forwards set up (as it is the thing it is forwarding to). This is advantageous for res_corosync, as returning NULL would also imply an unrecoverable error. Returning a subscription in this case allows for easier handling of message types that are published directly to an aggregate topic that has forwarders. Review: https://reviewboard.asterisk.org/r/3486/ ASTERISK-22912 #close ASTERISK-22372 #close ........ Merged revisions 414330 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	42a1dee02d	Undo r414123 The Test Suite caught a few problems, undoing until those are resolved git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	17ff4d9282	bridge_native_rtp/bridge_channel: Fix direct media issues due to frame hook This patch fixes issues with direct media bridges that occur after a blind transfer. These issues were caught by the (currently failing) pjsip/transfers/blind_transfer/caller_direct_media test. The test currently fails primarily for two reasons: (1) When Bob and Charlie (the transfer target and the transfer destination) enter a bridge together, the framehook remains on the transfer target channel until both channels are in the bridge. As it consumes voice frames, the initial bridge type is a simple bridge. The framehook is removed when both channels are in the bridge; however, this does not currently cause the bridging framework to re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a framehook is removed so the bridge can re-evaluate itself. (2) When a channel leaves a native RTP bridge, it may be leaving due to being hung up. Sending a re-INVITE to a channel that is about to be hung up is not nice - in fact, there's a good chance we'll send the BYE request before the channel has had a chance to send back a 200 OK. To be somewhat nicer, this patch adds a function to channel.h that allows the bridging framework to query for exactly why a channel is leaving a bridge via the channel's soft hangup flags. This allows it to only send the re-INVITE if there's a chance the channel will survive the native bridging experience. Review: https://reviewboard.asterisk.org/r/3535/ ........ Merged revisions 414122 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	e81b873fa2	chan_sip: Add TLS and SRTP status to CLI command 'sip show channel' ASTERISK-23564 #close Reported by: Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ ........ Merged revisions 413876 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413877 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	8b6ab4782a	chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing is enabled. When overlap dialing is enabled, the lack of inband audio available information in the SETUP_ACKNOWLEDGE events causes an interoperability problem with SIP. sig_pri doesn't know if there is dialtone present when a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an AST_CONTROL_PROGRESS frame. The SIP channel driver then sends out a 183 Session Progress and blocks the desired 180 Ringing message when the ALERTING message comes in. * Made the configure script detect if the installed version of libpri supports the SETUP_ACKNOWLEDGE enhancements. * Using the new API, made generate an AST_CONTROL_PROGRESS frame on an incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio is present instead of assuming that dialtone is present. * Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio available indication only if dialtone is expected. The change also makes the fallback behaviour of sending the PROGRESS message better by sending it only if dialtone is expected. * Changed receiving a PROCEEDING message to not generate an AST_CONTROL_PROGRESS frame if the progress indication ie indicates non-end-to-end-ISDN. This helps interoperability with SIP. * Changed sending a PROCEEDING message in response to an AST_CONTROL_PROCEEDING frame to not indicate inband audio available. It was silly to do so anyway because the channel driver doesn't know if inband audio is even available. This helps interoperability with SIP. This patch and a corresponding change in libpri work together to allow Asterisk to control the inband audio available progress indication ie on the SETUP_ACKNOWLEDGE message when dialtone is present. AST-1338 #close Reported by: Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/ ........ Merged revisions 413714 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413765 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413771 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d134150be2	framehooks: Add callback for determining if a hook is consuming frames of a specific type. In the past framehooks have had no capability to determine what frame types a hook is actually interested in consuming. This has meant that code has had to assume they want all frames, thus preventing native bridging. This change adds a callback which allows a framehook to be queried for whether it is consuming a frame of a specific type. The native RTP bridging module has also been updated to take advantange of this, allowing native bridging to occur when previously it would not. ASTERISK-23497 #comment Reported by: Etienne Lessard ASTERISK-23497 #close Review: https://reviewboard.asterisk.org/r/3522/ ........ Merged revisions 413681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e2ed86e4ca	Undoing framehook support. Issues were uncovered by Bamboo. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3b3e4b9b95	framehooks: Add callback for determining if a hook is consuming frames of a specific type. In the past framehooks have had no capability to determine what frame types a hook is actually interested in consuming. This has meant that code has had to assume they want all frames, thus preventing native bridging. This change adds a callback which allows a framehook to be queried for whether it is consuming a frame of a specific type. The native RTP bridging module has also been updated to take advantange of this, allowing native bridging to occur when previously it would not. ASTERISK-23497 #comment Reported by: Etienne Lessard ASTERISK-23497 #close Review: https://reviewboard.asterisk.org/r/3522/ ........ Merged revisions 413650 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	abd3e4040b	Allow Asterisk to compile under GCC 4.10 This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	20750e261b	chan_sip.c: Fixed off-nominal message iterator ref count and alloc fail issues. * Fixed early exit in sip_msg_send() not destroying the message iterator. * Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy() tolerant of a NULL iter parameter in case ast_msg_var_iterator_init() fails. * Made ast_msg_var_iterator_destroy() clean up any current message data ref. * Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(), ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy() use iter instead of i. * Eliminated RAII_VAR usage in res_pjsip_messaging.c:vars_to_headers(). ........ Merged revisions 413139 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413142 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	c6ed85748c	Add "destroy" implementation for spinlock. The original commit for spinlock was missing "destroy" implementations. Most of them are no-ops but phtread_spin and pthread_mutex do need their locks destroyed. ........ Merged revisions 413102 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	64045f0b57	This patch adds support for spinlocks in Asterisk. There are cases in Asterisk where it might be desirable to lock a short critical code section but not incur the context switch and yield penalty of a mutex or rwlock. The primary spinlock implementations execute exclusively in userspace and therefore don't incur those penalties. Spinlocks are NOT meant to be a general replacement for mutexes. They should be used only for protecting short blocks of critical code such as simple compares and assignments. Operations that may block, hold a lock, or cause the thread to give up it's timeslice should NEVER be attempted in a spinlock. The first use case for spinlocks is in astobj2 - internal_ao2_ref. Currently the manipulation of the reference counter is done with an ast_atomic_fetchadd_int which works fine. When weak reference containers are introduced however, there's an additional comparison and assignment that'll need to be done while the lock is held. A mutex would be way too expensive here, hence the spinlock. Given that lock contention in this situation would be infrequent, the overhead of the spinlock is only a few more machine instructions than the current ast_atomic_fetchadd_int call. ASTERISK-23553 #close Review: https://reviewboard.asterisk.org/r/3405/ ........ Merged revisions 412976 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b9d7dfcc62	ARI: Make bridges/{bridgeID}/play queue sound files Previously multiple play actions against a bridge at one time would cause the sounds to play simultaneously on the bridge. Now if a sound is already playing, the play action will queue playback to occur after the completion of other sounds currently on the queue. (closes issue ASTERISK-22677) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3379/ ........ Merged revisions 412639 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	51b6c49681	Originated calls: Fix several originate call problems. * Restore the reason value set by pbx_outgoing_attempt() to use AST_CONTROL_xxx values as all the consumers were expecting rather than cause codes. * Fixed the dial routines to set cause codes for more than just ast_request() so pbx_outgoing_attempt() reason codes will function. * Fix inconsistent locked_channel return status in pbx_outgoing_attempt(). The chanel may not have been locked or the channel may have been a stale pointer. * Fixed the OutgoingSpoolFailed channel to run dialplan whenever the dialing fails for an originate exten and 1 < synchronous. * Fix incorrect ast_cond_wait() usage in pbx_outgoing_attempt(). Indroduced by issue ASTERISK-22212 patch. * Made struct pbx_outgoing use the ao2 lock instead of its own lock for the cond wait mutex. No sense in having two locks associated with the same struct when only one is needed. Review: https://reviewboard.asterisk.org/r/3421/ ........ Merged revisions 412581 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	a8742e327f	ARI: Add tones playback resource Adds a tones URI type to the playback resource. The tone can be specified by name (from indications.conf) or by a tone pattern. In addition, tonezone can be specified in the URI (by appending ;tonezone=<zone>). Tones must be stopped manually in order for a stasis control to move on from playback of the tone. Tones may be paused, resumed, restarted, and stopped. They may not be rewound or fast forwarded (tones can't be controlled in a way that lets you skip around from note to note and pausing and resuming will also restart the tone from the beginning). Tests are currently in development for this feature (https://reviewboard.asterisk.org/r/3428/). (closes issue ASTERISK-23433) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3427/ ........ Merged revisions 412535 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	b4210a0081	Stasis: Add a usage note on stasis_app_get_bridge This function returns an ast_bridge without a refcount bump and the caller must increment the count if it intends to hold the pointer. (closes issue ASTERISK-23588) Review: https://reviewboard.asterisk.org/r/3450/ Reported by: Matt Jordan ........ Merged revisions 412439 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	5b7a769fd8	(mix)monitor: Add options to enable a periodic beep Add an option to enable a periodic beep to be played into a call if it is being recorded. If enabled, it uses the PERIODIC_HOOK() function internally to play the 'beep' prompt into the call at a specified interval. This option is provided for both Monitor() and MixMonitor(). Review: https://reviewboard.asterisk.org/r/3424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d28af99e65	chan_sip.c: Fix channel staging assertion failure. The failing assertion ensures that the final snapshot gets generated so CDR records can get finalized. The only place where a channel staging snapshot flag could be left set is in chan_sip.c:handle_request_bye(). The function could return before clearing the flag because the channel could dissappear while the function had to have the channel unlocked. * Fixed handle_request_bye() channel snapshot staging coverage area to not have a return in the middle of it and be unable to clear the staging flag. * Pushed the channel snapshot staging coverage area into ast_rtp_instance_set_stats_vars() to ensure that the staging is not interrutped. * Made callers of ast_rtp_instance_set_stats_vars() not call it with any channels or channel driver private locks held to eliminate the deadlock potential. The callers must hold references to the passed in channel and rtp objects. * Eliminated sip_hangup() trying to get the bridge peer. It is futile at this point because the channel could never be in a bridge. Review: https://reviewboard.asterisk.org/r/3431/ ........ Merged revisions 412385 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d6e2c50058	bridging: Ensure locking during snapshot creation While the vast majority of bridge snapshot creation is locked properly, there are currently some instances that are not. This adds the missing locking to ensure bridge state is not malleable during snapshot creation. (closes issue ASTERISK-22904) Review: https://reviewboard.asterisk.org/r/3415/ Reported by: Matt Jordan ........ Merged revisions 412193 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4f30c7e91f	main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output This patch does the following: (1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables REF_DEBUG globally throughout Asterisk. (2) The ref debug log file is now created in the AST_LOG_DIR directory. Every run will now blow away the previous run (as large ref files sometimes caused issues). We now also no longer open/close the file on each write, instead relying on fflush to make sure data gets written to the file (in case the ao2 call being performed is about to cause a crash) (3) It goes with a comma delineated format for the ref debug file. This makes parsing much easier. This also now includes the thread ID of the thread that caused ref change. (4) A new python script instead for refcounting has been added in the contrib/scripts folder. (5) The old refcounter implementation in utils/ has been removed. Review: https://reviewboard.asterisk.org/r/3377/ ........ Merged revisions 412114 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412115 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412153 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	03beadb6e9	internal_timing: Remove the option and always make it enabled if a timing module is loaded. The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). * Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411717 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	eefcb79bfb	Prevent duplicate sorcery wizards from being applied to sorcery object types. This commit contains several changes to sorcery: 1) Application of sorcery configuration based on module name is automatically performed when sorcery is opened for a module. 2) Sorcery will not attempt to apply the same wizard to an object type more than once. 3) Sorcery gives more exact results when attempting to apply a wizard, whether as the default or based on configuration. Sorcery unit tests still pass for me after making these changes. Review: https://reviewboard.asterisk.org/r/3326 ........ Merged revisions 411159 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ef0c9fe4d8	res_hep/res_hep_pjsip: Add a HEPv3 capture agent module and a logger for PJSIP This patch adds the following: (1) A new module, res_hep, which implements a generic packet capture agent for the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based on a patch provided by Alexandr Dubovikov; I basically just wrapped it up, added configuration via the configuration framework, and threw in a taskprocessor. (2) A new module, res_hep_pjsip, which forwards all SIP message traffic that passes through the res_pjsip stack over to res_hep for encapsulation and transmission to a HEPv3 capture server. Much thanks to Alexandr for his Asterisk patch for this code and for a lot of patience waiting for me to port it to 12/trunk. Due to some dithering on my part, this has taken the better part of a year to port forward (I still blame CDRs for the delay). ASTERISK-23557 #close Review: https://reviewboard.asterisk.org/r/3207/ ........ Merged revisions 411534 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	597f25db69	Update API versions and UPGRADE/CHANGES for 12.2.0 This patch does the following: * It updates the AMI version to 2.2.0 to indicate backwards compatible changes have been made since the last release * It updates the ARI version to 1.2.0 to indicate backwards compatible changes have been made since the last release * It updates the UPGRADE/CHANGES files with changes that were not mentioned ........ Merged revisions 411529 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	44409401ec	main/formats: Fix crash in ast_format_cmp during non-clean shutdown. * Update asterisk.h to reflect availability of ast_register_cleanup in 11.9. * Use ast_register_cleanup for format_attr_shutdown. (closes issue ASTERISK-23103) Reported by: JoshE ........ Merged revisions 411310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	2bf37a417d	Add a "message_context" option for PJSIP endpoints. ........ Merged revisions 411157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c1c8300e27	res_pjsip: Fix contact authenticate_qualify endpoint lookup when qualifing a contact. * Fixed bad use of ao2_find() in on_endpoint(). * Replaced use of find_endpoints() with find_an_endpoint() since only the first found endpoint is ever needed. * Fixed qualify_contact_cb() to update the contact with the aor authenticate_qualify setting. Otherwise, permanent contacts in the aor type sections would have a config line order dependancy. * Fixed off nominal path contact ref leak in qualify_contact(). The comment saying the unref is not needed was wrong. * Fixed off nominal path use of the endpoint parameter if it is NULL in send_out_of_dialog_request(). * Added missing off nominal path unref of pjsip tdata in send_out_of_dialog_request(). * Fixed off nominal path failing to call the callback in send_request_cb() when the request is challenged for authentication. * Eliminated silly RAII_VAR() use in qualify_contact_cb(). * Updated ast_sip_send_request() doxygen to better reflect reality. (closes issue ASTERISK-23254) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/3381/ ........ Merged revisions 411141 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	1ba13718fc	assigned-uniqueids: Miscellaneous cleanup and fixes. * Fix memory leak in ast_unreal_new_channels(). Made it generate the ;2 uniqueid on a stack variable instead of mallocing it. * Made send error response to ARI and AMI requests instead of just logging excessive uniqueid length and allowing truncation. action_originate() and ari_channels_handle_originate_with_id(). * Fixed minor truncating uniqueid hole when generating the ;2 uniqueid string length. Created public and internal lengths of uniqueid. The internal length can handle a max public uniqueid plus an appended ;2. * free() and ast_free() are NULL tolerant so they don't need a NULL test before calling. * Made use better struct initialization format instead of the position dependent initialization format. Also anything not explicitly initialized in the struct is initialized to zero by the compiler. * Made ast_channel_internal_set_fake_ids() use the safer ast_copy_string() instead of strncpy(). Review: https://reviewboard.asterisk.org/r/3371/ ........ Merged revisions 410949 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	cc40bf5317	res_pjsip: Enable PJSIP DNS client support. This change enables DNS client support within PJSIP. System nameservers are automatically discovered using res_init or res_ninit. If this fails then PJSIP will resort to using gethostbyname for resolution. By enabling this support we gain SRV support, failover, and weight support. (closes issue ASTERISK-23435) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3343/ ........ Merged revisions 410795 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	eba91d2a98	Revert changes to sorcery that accidentally got committed. These changes were still up for review and have not been approved yet. I must have had the changes in my working copy when making a different change. ........ Merged revisions 410696 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	d44aefeef4	Fix stuck channel in ARI through the introduction of synchronous bridge actions. Playing back a file to a channel in an ARI bridge would attempt to wait until the playback concluded before returning. The method used involved signaling the waiting thread in the ARI custom playback function. The problem with this is that there were some corner cases that were not accounted for: * If a bridge channel could not be found, then we never would attempt the playback but would still attempt to wait for the playback to complete. * If the bridge playfile action failed to queue, we would still attempt to wait for the playback to complete. * If the bridge playfile action were queued but some circumstance caused the playback not to occur (the bridge dies, the channel is removed from the bridge), then we would never be notified. The solution to this is to move the waiting logic into the bridge code. A new bridge API function is added to queue a synchronous action on a bridge. The waiting thread is notified when the queued frame has been freed, either due to an error occurring or due to successful playback. As a failsafe, the waiting thread has a 10 minute timeout just in case there is a frame leak somewhere. Review: https://reviewboard.asterisk.org/r/3338 ........ Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	a0a51a65e0	framehook.h: Fix some doc typos. There were a number of instances in this header file where "function all" was intended to be "function call". This patch fixes that up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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3014 Commits (86e8ab5ed47bccf0ee80acfd711eb81b0991522c)