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2014-05-22ARI: Add ability to raise arbitrary User EventsScott Griepentrog
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
2014-05-22res_pjsip_refer: Fix bugs involving Parking/PJSIP/transfersJonathan Rose
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
2014-05-22UPGRADE: Add note for REF_DEBUG flagMatthew Jordan
........ Merged revisions 414345 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414346 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414347 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22res_corosync: Update module to work with Stasis (and compile)Matthew Jordan
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
2014-05-21core_unreal: Only block media frames when a generator is on both ends of an ↵Richard Mudgett
unreal channel. The fix for ASTERISK-12292 was a bit too aggressive. You could have generators pointed at each other on local channels but need to get other kinds of frames such as DTMF or CONNECTED_LINE frames accross. ........ Merged revisions 414269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414270 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414272 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-21pbx.c: prevent potential crash from recursive replace()Scott Griepentrog
Recurisve usage of replace() resulted in corruption of the temporary string storage and potential crash. By changing the string to be allocated separtely per instance, this is eliminated. ASTERISK-23650 #comment Reported by: Roel van Meer ASTERISK-23650 #close Review: https://reviewboard.asterisk.org/r/3539/ ........ Merged revisions 414214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414215 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414216 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-19Replace __ast_answer with ast_raw_answer in app_control_answerPaul Belanger
While load testing an ARI application, I noticed asterisk was returning HTTP 500 internal server errors on channels/:id/answer. After talking to #asterisk-dev, the issue appeared to be a lack of media flowing after __ast_answer() was called. So now, we call ast_raw_answer instead and no longer wait for media. ASTERISK-23758 #close Review: https://reviewboard.asterisk.org/r/3549/ ........ Merged revisions 414195 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-19Undo r414123Matthew Jordan
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
2014-05-18bridge_native_rtp/bridge_channel: Fix direct media issues due to frame hookMatthew Jordan
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
2014-05-16chan_dahdi: Fix analog dialtone detection.Richard Mudgett
* Check if waitingfordt (waitfordialtone) is enabled in dahdi_read() to allow the DSP to operate early enough to detect dialtone. * Made use the correct variable in my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve Davies Patches: dialtone_detect_fix (license #5012) patch uploaded by Steve Davies Review: https://reviewboard.asterisk.org/r/3534/ ........ Merged revisions 414067 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414068 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414069 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-16sig_pri.c: Pull the pri_dchannel() PRI_EVENT_RING case into its own function.Richard Mudgett
* Populate the CALLERID(ani2) value (and the special CALLINGANI2 channel variable) with the ANI2 value in addition to the PRI specific ANI2 channel variable. * Made complete snapshot staging with the channel lock held. All channel snapshots need to be done while the channel lock is held. ........ Merged revisions 414050 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414051 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-15app_meetme: Fix overwrite of DAHDI conference data structure.Richard Mudgett
Starting a conference recording using the admin menu overwrites the DAHDI conference data structure used to modify the admin user's conference mute mode. * Made no longer pass the user's DAHDI conference data structure into the menu functions. The menu now uses its own DAHDI conference data structure to start the recording channel. * Moved the unlock conf->playlock to before playing the conf-full message. No sense keeping the lock while that prompt is playing. The user is never going to get into the conference at that point. ........ Merged revisions 413991 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413992 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413993 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-14res_musiconhold: Minor cleanup.Walter Doekes
Fix a few free()'s that should be ast_free()'s. Reverted an old workaround that isn't necessary. Reorder a tiny bit of code. Remove a bit of commented-out code. Review: https://reviewboard.asterisk.org/r/3536/ ........ Merged revisions 413894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413895 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-13chan_sip: Add TLS and SRTP status to CLI command 'sip show channel'Jonathan Rose
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
2014-05-13h264: Fix H264 SDP payload format.Walter Doekes
https://tools.ietf.org/html/rfc3984#section-8.1 says profile-level-id takes 3 bytes in base16 (6 hex digits). This fixes video setup in certain cases. ASTERISK-23664 #close ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume Maudoux. Review: https://reviewboard.asterisk.org/r/3530/ ........ Merged revisions 413791 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413792 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-13rtp: Fix case typo in H263+ mime.Walter Doekes
http://tools.ietf.org/html/rfc3555#section-4.2.6 says the canonical mime subtype is "H263-1998", not "h263-1998". Original code was added in r183101 on 2009-03-19 02:26:50 +0100. This fixes issues with Polycom phones. ASTERISK-23665 #close ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume Maudoux, backported by me. Review: https://reviewboard.asterisk.org/r/3529/ ........ Merged revisions 413787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413788 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413789 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-13chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing ↵Richard Mudgett
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
2014-05-13Fix compiler warning from GCC 4.10 fixup.Richard Mudgett
........ Merged revisions 413766 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-12app_chanspy: Fix a test that was failing on account of r413551Jonathan Rose
ASTERISK-23381 #close ASTERISK-23381 #comment Reported by: Robert Moss Review: https://reviewboard.asterisk.org/r/3505/ ........ Merged revisions 413710 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413712 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-11framehooks: Add callback for determining if a hook is consuming frames of a ↵Joshua Colp
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
2014-05-11Undoing framehook support. Issues were uncovered by Bamboo.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-10framehooks: Add callback for determining if a hook is consuming frames of a ↵Joshua Colp
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
2014-05-09Fix 32bit build for func_envKinsey Moore
........ Merged revisions 413592 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413595 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413597 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09Allow Asterisk to compile under GCC 4.10Kinsey Moore
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
2014-05-09http.c: Remove dead code.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09app_chanspy: Fix a bug where Barge mode could failJonathan Rose
If the barge audiohook was attached prior to the spyee and its peer actually being bridged, the audiohook would not be applied and the connected peer would not be able to hear audio from the spy when the spy is in barge mode. (closes issue ASTERISK-23381) Reported by: Robert Moss Review: https://reviewboard.asterisk.org/r/3505/ ........ Merged revisions 413551 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413556 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-08app_queue: Extend documentation for various Manager actions and events.Joshua Colp
........ Merged revisions 413485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413486 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413487 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-07Ensure that presence state is decoded properly on Asterisk startup.Mark Michelson
The CustomPresence provider callback will automatically base64 decode stored data if the 'e' option was present when the state was set. However, since the provider callback was bypassed on Asterisk startup, encoded presence subtypes and messages were being sent instead. This fix makes it so the provider callback is always used when providing presence state updates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-07app_confbridge: Fixed "CBAnn" channels not going away.Richard Mudgett
Fixed a ref leak in conf_handle_talker_cb() everytime the conference bridge was found to report a channel's talker status change. The resulting leak caused the "CBAnn" channels and the conference bridge to never be destroyed. Thanks to Richard Kenner on the asterisk-user's list for locating the problem. Reported by: Richard Kenner ........ Merged revisions 413454 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-07app_confbridge: Fix ref leak in CLI "confbridge kick" command.Richard Mudgett
Fixed ref leak in the CLI "confbridge kick" command when the channel to be kicked was not in the conference. ........ Merged revisions 413451 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413452 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-07Fix encoding of custom prepare extra data.Mark Michelson
Patches: res_config_odbc-take2.patch by John Hardin (License #6512) ........ Merged revisions 413396 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413397 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-07Improve XML sanitization in NOTIFYs, especially for presence subtypes and ↵Mark Michelson
messages. Embedded carriage return line feed combinations may appear in presence subtypes and messages since they may be derived from user input in an instant messenger client. As such, they need to be properly escaped so that XML parsers do not vomit when the messages are received. ........ Merged revisions 413372 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-06Check for an act on failures to update contacts during registration.Mark Michelson
There was an underlying issue in a realtime backend where database updates would fail. Since we were not checking for failure, we would end up in a strange state where the old database entry was still present but Asterisk thought that it had been updated. Now when an entry fails to update, we print a warning and delete the old contact from sorcery so there is no mismatch between foreground and backend state. Patches: res_pjsip_registrar.patch by John Hardin (License #6512) ........ Merged revisions 413358 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-06Ensure that all parts of SQL UPDATEs and DELETEs are encoded.Mark Michelson
Patches: res_config_odbc.patch by John Hardin (License #6512) ........ Merged revisions 413304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413305 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413306 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-02Blocked revisions 413282Mark Michelson
........ Correct variable traversal logic in res_config_odbc's update_odbc function. Closes issue ASTERISK-23675 Reported by Leando Dardini Patches: asterisk-23675-odbc-linkedlist-traversal_12.diff uploaded by Michael L. Young (license #5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-02Prevent crashes in res_config_odbc due to uninitialized string fields.Mark Michelson
Patches: odbc-crash.patch by John Hardin (License #6512) ........ Merged revisions 413241 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413251 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413258 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-02Return the number of rows affected by a SQL insert, rather than an object ID.Mark Michelson
The realtime API specifies that the store callback is supposed to return the number of rows affected. res_config_pgsql was instead returning an Oid cast as an int, which during any nominal execution would be cast to 0. Returning 0 when more than 0 rows were inserted causes problems to the function's callers. To give an idea of how strange code can be, this is the necessary code change to fix a device state issue reported against chan_pjsip in Asterisk 12+. The issue was that the registrar would attempt to insert contacts into the database. Because of the 0 return from res_config_pgsql, the registrar would think that the contact was not successfully inserted, even though it actually was. As such, even though the contact was query-able and it was possible to call the endpoint, Asterisk would "think" the endpoint was unregistered, meaning it would report the device state as UNAVAILABLE instead of NOT_INUSE. The necessary fix applies to all versions of Asterisk, so even though the bug reported only applies to Asterisk 12+, the code correction is being inserted into 1.8+. Closes issue ASTERISK-23707 Reported by Mark Michelson ........ Merged revisions 413224 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413225 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413226 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-02res_pjsip_refer: Add Referred-By header on INVITE for blind transfers.Richard Mudgett
Per rfc3892, the Referred-By header in a REFER must be copied into the referenced request (IE. The outgoing INVITE to the transfer target). * Automatically put the Referred-By header in the outgoing INVITE message if the SIPREFERREDBYHDR channel variable is defined with a value. * Made chan_sip.c:get_refer_info() set SIPREFERREDBYHDR for inheritance so chan_pjsip has a better chance to interoperate. * Fixed refer_blind_callback() and refer_incoming_refer_request() to not modify the data in the pointer returned by pjsip_msg_find_hdr_by_name(). It seems wrong to modify that data since the calling routine doesn't own the buffer. ASTERISK-23501 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3514/ ........ Merged revisions 413210 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-02Parking: Add 'AnnounceChannel' argument to manager action 'Park'Jonathan Rose
(closes ASTERISK-23397) Reported by: Denis Review: https://reviewboard.asterisk.org/r/3446/ ........ Merged revisions 413196 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-01Make behavior of the PRESENCE_STATE 'e' option more consistent.Mark Michelson
When writing presence state, if 'e' is specified, then the presence state will be stored in the astdb encoded. However, consumers of presence state events or those that query for the presence state will be given decoded information. If base64 encoding is desired for consumers, then the information can be base64-encoded manually and the 'e' option can be omitted. closes issue ASTERISK-23671 Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3482 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-01Remove unnecessary repetition checks from res_pjsip_exten_stateMark Michelson
The PBX core already takes care of ensuring that repeated state changes are not communicated to exten state consumers. Because the check in res_pjsip_exten_state was incomplete, it was causing valid presence state changes not to be sent out. For instance, if the presence state did not change but the message or subtype did, then no presence-related NOTIFY request would be sent out. closes issue ASTERISK-23672 Reported by Mark Michelson ........ Merged revisions 413173 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-01res_pjsip: Add the ability to configure ciphers based on name.Joshua Colp
Previously this code would only accept the OpenSSL identifier instead of the documented name. ASTERISK-23498 #close ASTERISK-23498 #comment Reported by: Anthony Messina Review: https://reviewboard.asterisk.org/r/3491/ ........ Merged revisions 413159 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-30chan_sip.c: Fixed off-nominal message iterator ref count and alloc fail issues.Richard Mudgett
* 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
2014-04-30chan_pjsip: Fix deadlock when retrieving call-id of channel.Joshua Colp
If a task was in-flight which required the channel or bridge lock it was possible for the synchronous task retrieving the call-id to deadlock as it holds those locks. After discussing with Mark Michelson the synchronous task was removed and the call-id accessed directly. This should be safe as each object involved is guaranteed to exist and the call-id will never change. ........ Merged revisions 413140 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-30Websocket: Add session locking and delay closeKinsey Moore
This resolves a race condition where data could be written to a NULL FILE pointer causing a crash as a websocket connection was in the process of shutting down by adding locking to websocket session writes and by deferring session teardown until session destruction. (closes issue ASTERISK-23605) Review: https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan ........ Merged revisions 413123 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413124 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-30res_stasis: Add progress indications to operations which perform media.Joshua Colp
This change fixes operations which did not account for the fact that they may be executed on channels which have not been answered. These operations will now indicate progress when invoked. ASTERISK-23560 #close ASTERISk-23560 #comment Reported by: Jan Svoboda Review: https://reviewboard.asterisk.org/r/3495/ ........ Merged revisions 413121 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-30res_pjsip_sdp_rtp: Fix issue where sending a hold SDP twice could cause an ↵Joshua Colp
unhold. This change fixes a bug where if an SDP with media address and sendonly was received twice the underlying call would go off hold, instead of remaining on hold. This occured because the code did not properly take into account that the SDP may contain both a valid media address and the sendonly attribute. The code now examines the sendonly attribute and media address first, so if the SDP is received again no change will occur. ASTERISK-23558 #comment Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3472/ ........ Merged revisions 413119 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-30chan_pjsip: Add support for picking up calls in the configured pickup group.Joshua Colp
AST-1363 Review: https://reviewboard.asterisk.org/r/3478/ ........ Merged revisions 413117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-29Add "destroy" implementation for spinlock.George Joseph
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
2014-04-29chan_pjsip: Implement core ability to get Call-ID of a channel.Joshua Colp
This changes implement the "get_pvt_uniqueid" which is used to return the technology specific unique identifier. In the case of SIP this is the Call-ID of the dialog. Review: https://reviewboard.asterisk.org/r/3480/ ........ Merged revisions 413088 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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