asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	dbead14c3b	Put notice in CHANGES as well as UPGRADE.txt. ........ Merged revisions 404375 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	433c6f010f	res_pjsip: Ignore 401/407 responses for transactions and dialogs we don't know about. Under normal conditions it is unlikely we will ever receive a response for a transaction or dialog we don't know about but if any are received ignore them. ........ Merged revisions 404371 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	8402cd4cd9	res_pjsip_session: Fix SDP negotiation when resending an INVITE with authentication. The process for resending an INVITE with authentication involves restarting the UAC session. We were incorrectly passing in that a new offer is being sent, causing the SDP negotiation to get into a (technically speaking) funky state. ........ Merged revisions 404369 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	1b91ee6c4b	Fix a deadlock that occurred due to a conflict of masquerades. For the explanation, here is a copy-paste of the review board explanation: Initially, it was discovered that performing an attended transfer of a multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread started a masquerade and reached the point where it was calling the fixup() callback on the "original" channel. For chan_pjsip, this involves pushing a synchronous task to the session's serializer. The problem was that a task ahead of the fixup task was also attempting to perform a channel masquerade. However, since masquerades are designed in a way to only allow for one to occur at a time, the task ahead of the fixup could not continue until the masquerade already in progress had completed. And of course, the masquerade in progress could not complete until the task ahead of the fixup task had completed. Deadlock. The initial fix was to change the fixup task to be asynchronous. While this prevented the deadlock from occurring, it had the frightful side effect of potentially allowing for tasks in the session's serializer to operate on a zombie channel. Taking a step back from this particular deadlock, it became clear that the problem was not really this one particular issue but that masquerades themselves needed to be addressed. A PJSIP attended transfer operation calls ast_channel_move(), which attempts to both set up and execute a masquerade. The problem was that after it had set up the masquerade, the PBX thread had swooped in and tried to actually perform the masquerade. Looking at changes that had been made to Asterisk 12, it became clear that there never is any time now that anyone ever wants to set up a masquerade and allow for the channel thread to actually perform the masquerade. Everyone always is calling ast_channel_move(), performs the masquerade itself before returning. In this patch, I have removed all blocks of code from channel.c that will attempt to perform a masquerade if ast_channel_masq() returns true. Now, there is no distinction between setting up a masquerade and performing the masquerade. It is one operation. The only remaining checks for ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not want to interrupt a masquerade by hanging up the channel. Instead, now ast_hangup() will wait for a masquerade to complete before moving forward with its operation. The ast_channel_move() function has been modified to basically in-line the logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has been killed off for real. ast_channel_move() now has a lock associated with it that is used to prevent any simultaneous moves from occurring at once. This means there is no need to make sure that ast_channel_masq() or ast_channel_masqr() are already set on a channel when ast_channel_move() is called. It also means the channel container lock is not pulling double duty by both keeping the container locked and preventing multiple masquerades from occurring simultaneously. The ast_do_masquerade() function has been renamed to do_channel_masquerade() and is now internal to channel.c. The function now takes explicit arguments of which channels are involved in the masquerade instead of a single channel. While it probably is possible to do some further refactoring of this method, I feel that I would be treading dangerously. Instead, all I did was change some comments that no longer are true after this changeset. The other more minor change introduced in this patch is to res_pjsip.c to make ast_sip_push_task_synchronous() run the task in-place if we are already a SIP servant thread. This is related to this patch because even when we isolate the channel masquerade to only running in the SIP servant thread, we would still deadlock when the fixup() callback is reached since we would essentially be waiting forever for ourselves to finish before actually running the fixup. This makes it so the fixup is run without having to push a task into a serializer at all. (closes issue ASTERISK-22936) Reported by Jonathan Rose Review: https://reviewboard.asterisk.org/r/3069 ........ Merged revisions 404356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3ccd5dee18	udptl: Dead code elimination. ast_udptl_bridge was not used. Removing dead code starting with ast_udptl_bridge() eliminated the code in this change. Note: This code has actually been dead since Asterisk v1.4 when it was first put in. Review: https://reviewboard.asterisk.org/r/3079/ ........ Merged revisions 404354 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	c061f634da	res_fax.c: crash on framehook with no dsp in fax detect In fax_detect_framehook() a null pointer reference can occur where a voice frame is processed but no dsp is attached to the fax detection structure. The code block that rejects frames that detection cannot be processed on is checking for dsp but falls through when it should instead return, as this change implements. (closes issue ASTERISK-22942) Reported by: adomjan Review: https://reviewboard.asterisk.org/r/3076/ ........ Merged revisions 404351 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404352 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e4803bbd9e	Voicemail: Remove mailbox identifier format (box@context) assumptions in the system. This change is in preparation for external MWI support. Removed code from the system for normal mailbox handling that appends @default to the mailbox identifier if it does not have a context. The only exception is the legacy hasvoicemail users.conf option. The legacy option will only work for app_voicemail mailboxes. The system cannot make any assumptions about the format of the mailbox identifer used by app_voicemail. chan_sip and chan_dahdi/sig_pri had the most changes because they both tried to interpret the mailbox identifier. chan_sip just stored and compared the two components. chan_dahdi actually used the box information. The ISDN MWI support configuration options had to be reworked because chan_dahdi was parsing the box@context format to get the box number. As a result the mwi_vm_boxes chan_dahdi.conf option was added and is documented in the chan_dahdi.conf.sample file. Review: https://reviewboard.asterisk.org/r/3072/ ........ Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	2882c5f9f1	astdb: crash in sqlite3 during shutdown When Asterisk is shut down, the astdb_atexit() function releases (finalize) the previously initiated (prepared) SQL statements in sqlite3. Another thread making a subsequent request can cause a crash in sqlite3. This patch eliminates that issue by resetting the statement pointer after it is released/cleared. The sqlite3 code detects the null pointer, and aborts the operation cleanly. (closes issue AST-1265) Reported by: Alexander Hömig (closes issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged revisions 404344 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404345 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	eb235ad05f	channel: Add a missing ast_channel_unlock when allocating a Surrogate channel. ........ Merged revisions 404332 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alexandr Anikin	1370d40559	Handle temporary failures on gk registration Introduce new 'stopped' state for gk client and restart gk client on failures Remove ooh323 stack command lock as it is not need now. (closes issue ASTERISK-21960) Reported by: Dmitry Melekhov Patches: ASTERISK-21960.patch ASTERISK-21960-stacklockup-2.patch Tested by: Dmitry Melekhov ........ Merged revisions 404318 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404320 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	c2fd229fba	Fixup some skinny bugs causing Fracks and ao2 cleanup issues. Moved channel locking into setsubstate so that a process can complete working on a sub before another starts changing it. The existing code was causing some Fracks with schedule deletion. Removed multiple rtp cleanup. Now only cleansup up once, fixing ao2 object cleanup issues. ........ Merged revisions 404306 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7e9febbf86	app_cdr,app_forkcdr,func_cdr: Synchronize with engine when manipulating state When doing the rework of the CDR engine that pushed all of the logic into cdr.c and made it respond to changes in channel state over Stasis, we knew that accessing the CDR engine from the dialplan would be "slightly" non-deterministic. Dialplan threads would be accessing CDRs while Stasis threads would be updating the state of said CDRs - whereas in the past, everything happened on the dialplan threads. Tests have shown that "slightly" is in reality "very". This patch synchronizes things by making the dialplan applications/functions that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and CDR_PROP now all use Stasis to send their requests over to the CDR engine, and synchronize on the channel Stasis topic via a subscription so that they return their values/control to the dialplan at the appropriate time. While going through this, the following changes were also made: * DISA, which can reset the CDR when a user successfully authenticates, now just uses the ResetCDR app to do this. This prevents having to duplicate the same Stasis synchronization logic in that application. * Answer no longer disables CDRs. It actually didn't work anyway - calling DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer time - it just kills all CDRs on that channel, which isn't what the caller would intend. (closes issue ASTERISK-22884) (closes issue ASTERISK-22886) Review: https://reviewboard.asterisk.org/r/3057/ ........ Merged revisions 404294 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	af723c6572	Fixup skinny registration following network issues. On session registration, if device is already reporting that it is connected to a device, an innocuous packet (update time) is sent to the already connected device. If the tcp connection is down, the device will be unregistered and the new connection allowed. Without this patch, network issues can see a situation where a device can not reregister until after 3*timeout. ........ Merged revisions 404292 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	04dfe2d77f	Add AMI event for presence state. Review: https://reviewboard.asterisk.org/r/3039/ ........ Merged revisions 404275 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404279 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	47c0b78690	ooh323c: Fix gcc 4.6.3 compiler warnings. ........ Merged revisions 404212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404219 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404263 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	ed0c493222	chan_oss.c: channel being locked twice and unlocked once Removed channel lock as it is now being down in ast_channel_alloc ........ Merged revisions 404261 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	28c0cb28d0	channel locking: Add locking for channel snapshot creation Original commit message by mmichelson (asterisk 12 r403311): "This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such." The above was initially committed and then reverted at r403398. The problem was found to be in core_local.c in the publish_local_bridge_message function. The ast_unreal_lock_all function locks and adds a reference to the returned channels and while they were being unlocked they were not being unreffed when no longer needed. Fixed by unreffing the channels. Also in bridge.c a lock was obtained on "other->chan", but then an attempt was made to unlock "other" and not the previously locked channel. Fixed by unlocking "other->chan" (closes issue ASTERISK-22709) Reported by: John Bigelow ........ Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alexandr Anikin	86b5e11607	Introduce new config option 'aniasdni'. If yes then asterisk set dialed number as own id back to the caller on incoming h.323 calls. Option can be set globally or per user section. (closes issue ASTERISK-22020) Reported by: Ross Beer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e2630fcd51	channels: Return allocated channels locked. This change makes ast_channel_alloc return allocated channels locked. By doing so no other thread can acquire, lock, and manipulate the channel before it is completely set up. (closes issue AST-1256) Review: https://reviewboard.asterisk.org/r/3067/ ........ Merged revisions 404204 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alexandr Anikin	e23b8d19a4	Implement module reload command for chan_ooh323 (close issue ASTERISK-22817) Patches: ooh323_module_reload.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ec79aabdb9	ari: Bump the version of ARI to 1.0.0 (closes issue ASTERISK-23007) ........ Merged revisions 404184 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	513756b956	res_calendar: Protect channel when adding datastore. This change adds a missing channel lock when adding a datastore to a channel. ........ Merged revisions 404135 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404136 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404137 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	2a6df87fcd	func_strings: Documentation fix for QUOTE() Example output was inaccurate. (issue ASTERISK-22970) (closes issue ASTERISK-22970) Reported by: Gareth Palmer Patches: func_strings.patch uploaded by Gareth Palmer (license 5169) ........ Merged revisions 404081 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404087 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404099 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c07fa05aa7	LICENSE: Update language to include ARI ........ Merged revisions 404050 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	655ce29aac	tests: fix ast_bridge_base_new calls not using the additional arguments r404042 gave ast_bridge_base_new two new arguments for setting a bridge creator and name. Unfortunately since a couple test modules aren't compiled by default, I missed the fact that this change impacted those tests and caused compilation failures against them. ........ Merged revisions 404048 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	f7c60b8fb6	Several components: fixing Typos in comments and code, "avaliable" instead of "available" (issue ASTERISK-23021) (closes issue ASTERISK-23021) Reported by: Jeremy Lainé Tested by: Rusty Newton Patches: available.patch uploaded by Jeremy Lainé (license 6561) ........ Merged revisions 404046 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b0bb03e916	bridging: Give bridges a name and a known creator Bridges have two new optional properties, a creator and a name. Certain consumers of bridges will automatically provide bridges that they create with these properties. Examples include app_bridgewait, res_parking, app_confbridge, and app_agent_pool. In addition, a name may now be provided as an argument to the POST function for creating new bridges via ARI. (closes issue AFS-47) Review: https://reviewboard.asterisk.org/r/3070/ ........ Merged revisions 404042 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	91d0f30506	res_sorcery_config: Output an error message when an object can't be created. If object creation fails an error message will now be output with the id, type, and configuration file. ........ Merged revisions 404029 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	9fc2cc178a	framehooks: Re-iterate if framehook provides different frame. Framehooks can be used in a reactive manner to execute specific logic when a frame is received with a certain type and payload. Since it is possible for framehooks to provide frames it was possible for this reactive framehook to be unaware of frames it is looking for. This change makes it so that when framehooks return a modified frame the code will now re-iterate (from the beginning) and call any previous framehooks that have not provided a modified frame themselves. Review: https://reviewboard.asterisk.org/r/3046/ ........ Merged revisions 404027 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	27f37f6e3d	Changed the default for live_dangerously to no ........ Merged revisions 404006 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	f257293f04	Setting svn:ignore ........ Merged revisions 403748 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4d43d22a4b	Blocked revisions 403779-403780 ........ Multiple revisions 403779-403780 ........ r403779 \| rmudgett \| 2013-12-13 13:48:05 -0600 (Fri, 13 Dec 2013) \| 12 lines app_voicemail: Voicemail callback registration/unregistration function improvements. * The voicemail registration/unregistration functions now take a struct of callbacks instead of a lengthy parameter list of callbacks. * The voicemail registration/unregistration functions now prevent a competing module from interfering with an already registered callback supplying module. ........ Merged revisions 403643 from http://svn.asterisk.org/svn/asterisk/trunk ........ r403780 \| rmudgett \| 2013-12-13 13:55:31 -0600 (Fri, 13 Dec 2013) \| 8 lines test_voicemail_api: Add check for a registered voicemail provider before tests. It is much nicer diagnosing a test failure if app_voicemail is actually loaded. ........ Merged revisions 403726 from http://svn.asterisk.org/svn/asterisk/trunk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	76042b1eb4	ari/resource_channels: When creating a channel, specify a default format (SLIN) When creating channels via ARI, the current code fails to provide any default format capabilities. For non-virtual channels this isn't really a problem - the channels typically receive their capabilities as a result of the underlying channel driver configuration. For virtual channels (such as Local channels), the lack of any format capabilities causes the Asterisk core to make some 'odd' choices with respect to the translation paths. The issue reporter had some paths that had 3 hops on each channel leg, causing multiple transcodings and some really crappy audio/performance. By specifying a baseline of SLIN, we prevent that from occurring. Note that this is what AMI does when it performs an Originate, as does res_clioriginate. Review: https://reviewboard.asterisk.org/r/3068/ (issue ASTERISK-22962) Reported by: Matt DiMeo ........ Merged revisions 403993 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	744556c01d	security: Inhibit execution of privilege escalating functions This patch allows individual dialplan functions to be marked as 'dangerous', to inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. Also, the ABI was changed to something more reasonable, since Asterisk 12 does not yet have a public release. (closes issue ASTERISK-22905) Review: http://reviewboard.digium.internal/r/432/ ........ Merged revisions 403913 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403917 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403959 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	00dcee2a64	transfers: Fix bug setting both BLINDTRANSFER and ATTENDEDTRANSFER The ast_bridge_set_transfer_variables function is supposed to wipe whichever variable isn't being set. Instead it was setting both to the new value. Oops. (issue AFS-24) ........ Merged revisions 403957 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	102d448486	pbx.c: put copy of ast_exten.data on stack to prevent memory corruption During dialplan execution in pbx_extension_helper(), the contexts global read lock prevents link list corruption, but was released with a pointer to the ast_exten and data later used in variable substitution. Instead, this patch removes pbx_substitute_variables() and locates a copy of the ast_exten data on the stack before releasing the lock, where ast_exten could get free'd by another thread performing a module reload. (issue AST-1179) Reported by: Thomas Arimont (issue AST-1246) Reported by: Alexander Hömig Review: https://reviewboard.asterisk.org/r/3055/ ........ Merged revisions 403862 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403863 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403864 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	3322180d4b	app_sms: BufferOverflow when receiving odd length 16 bit message This patch prevents an infinite loop overwriting memory when a message is received into the unpacksms16() function, where the length of the message is an odd number of bytes. (closes issue ASTERISK-22590) Reported by: Jan Juergens Tested by: Jan Juergens ........ Merged revisions 403856 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4ddf45fd24	pjsip/dialplan_functions: Use the right buffer length when printing URIs While entertaining, sizeof(buflen) is not the same as buflen. Doh. ........ Merged revisions 403823 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d7b84b0368	res_pjsip: Apply outbound proxy to all SIP requests. Objects which are involved in SIP request creation and sending now allow an outbound proxy to be specified. For cases where an endpoint is used the outbound proxy specified there will be applied. (closes issue ASTERISK-22673) Reported by: Antti Yrjola Review: https://reviewboard.asterisk.org/r/3022/ ........ Merged revisions 403811 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3a5cc054ed	res_stasis: Expose event for call forwarding and follow forwarded channel. This change adds an event for when an originated call is redirected to another target. This event contains the original channel and the newly created channel. If a stasis subscription exists on the original originated channel for a stasis application then a new subscription will also be created on the stasis application to the redirected channel. This allows the application to follow the call path completely. (closes issue ASTERISK-22719) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3054/ ........ Merged revisions 403808 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	661ac14911	documentation: Add PJSIP technology to messaging documentation ........ Merged revisions 403796 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	de4a18d1aa	test.c: Fix too sticky unit test failed status. Rerunning a failed unit test after loading any required modules should allow the test to report a pass status if it now passes. ........ Merged revisions 403782 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	63b2c28c4b	Transfers: Make Asterisk set ATTENDEDTRANSFER/BLINDTRANSFER more reliably There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER wouldn't be set on channels involved with blind and attended transfers. This would happen with features that were initialized by channel driver specific mechanisms in multiparty calls. This patch resolves those cases while attempted to keep the behavior for setting those variables as consistent as possible. (closes issue AFS-24) Review: https://reviewboard.asterisk.org/r/3040/ ........ Merged revisions 403781 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	84e1790beb	bridge_native_rtp: Deadlock during 4-way conference creation The change contains a slightly adjusted patch that was on the issue (submitted by kmoore). A fix was made by adding in a bridge lock while calling bridge_start/stop from the framehook callback. Since the framehook callback is not called from the bridging core the bridge is not locked, but needs to be before calling bridge_start. (closes issue ASTERISK-22749) Reported by: Kinsey Moore Review: https://reviewboard.asterisk.org/r/3066/ Patches: lock_inversion.diff uploaded by kmoore (license 6273) ........ Merged revisions 403767 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	f425c4a086	ARI: Allow specifying channel variables during a POST /channels Added the ability to specify channel variables when creating/originating a channel in ARI. The variables are sent in the body of the request and should be formatted as a single level JSON object. No nested objects allowed. For example: {"variable1": "foo", "variable2": "bar"}. (closes issue ASTERISK-22872) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3052/ ........ Merged revisions 403752 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	ce18946de4	ARI: Adding a channel to a bridge while a live recording is active blocks Added the ability to have rules that are checked when adding and/or removing channels to/from a bridge. In this case, if a channel is currently recording and someone attempts to add it to a bridge an "is recording" rule is checked, fails, and a 409 conflict is returned. Also command functions now return an integer value that can be descriptive of what kind of problems, if any, occurred before or during execution. (closes issue ASTERISK-22624) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2947/ ........ Merged revisions 403749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	fc8c0ef28f	channels/Makefile: clean pjsip directory ........ Merged revisions 403736 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403737 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3a5e4317f5	test_voicemail_api: Add check for a registered voicemail provider before tests. It is much nicer diagnosing a test failure if app_voicemail is actually loaded. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	d2eb007bf0	realtime: Create extensions in alembic ast-db-manage contribution When the alembic scripts were written for creating Asterisk realtime databases the extensions table for dialplan wasn't included. This update creates the extensions table. (closes issue ASTERISK-22815) Reported by: Zone Conkle Review: https://reviewboard.asterisk.org/r/3064/ ........ Merged revisions 403713 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	5583d2c629	chan_pjsip: Revert r403587 This patch was intended to eliminate a deadlock that occurs when masquerades occur in pjsip channels, but has some potential side effects. Mark Michelson is currently working on addressing this problem from another angle. (issue ASTERISK-22936) Reported by: Jonathan Rose ........ Merged revisions 403705 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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