asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	163c734ae7	MoH: Don't restart stream on repeated start calls Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. This includes an extra check to prevent the errors previously experienced in the testsuite and has 100+ test runs behind it. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416440 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416441 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0c896d8b9b	chan_dahdi: Adds support for major update to libss7. * SS7 support now requires libss7 v2.0 or later. The new libss7 is not backwards compatible. * Added SS7 support for connected line and redirecting. * Most SS7 CLI commands are reworked as well as new SS7 commands added. See online CLI help. * Added several SS7 config option parameters described in chan_dahdi.conf.sample. * ISUP timer support reworked and now requires explicit configuration. See ss7.timers.sample. Special thanks to Kaloyan Kovachev for his support and persistence in getting the original patch by adomjan updated and ready for release. SS7-27 #close Reported by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Igor Goncharovskiy	a1e0a5e4b0	We have faced situation when using CDR and CEL by sqlite3 modules. With system having high load (~100 concurrent calls created by sipp) we found many cdr and cel records missed. There is special finction in sqlite3, that make able to fix this situation - sqlite3_wait_timeout, that also can replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this function can be used for aastdb and res_config_sqlite3 to avoid missed writes to sqlite db.	11 years ago
Matthew Jordan	8313964d72	channels/chan_sip: Forbid remote bridging if T.38 is negotiated When a framehook is removed - such as the fax gateway framehook - the bridge framework will re-evaluate the bridge mixing technologies to see if it can improve the bridging. When this occurs, get_rtp_info will be called to determine if local or remote bridging can be used. Using remote bridging will cause a fax to fail, as direct media negotiation will cause some small number of packets to not arrive at the remote endpoint. This patch forces local native bridging if T.38 negotiation is in progress or has been established. ........ Merged revisions 416318 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	072b61bbed	channel_internal_api: Publish a snapshot change when linkedids change Snapshots are now not published quite as much as they used to. One instance where they are not published any longer is during bridge enter and exit - the state of the channel doesn't change, the bridge does. However, channels are changed when a linkedid is propagated; previously, the channel's state would be updated and published during the bridge enter event. Now this must be explicitly done. ........ Merged revisions 416300 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	705d0c3e81	test_stasis_endpoints: Remove expected channel snapshot We no longer publish a channel snapshot when it is associated with an endpoint; after all, the channel itself hasn't changed - the endpoint state has changed. This updates the channel_messages unit test accordingly. ........ Merged revisions 416298 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b52c6d0903	MoH: Undo commit r416150 (1.8) This patch reverts r416150. When the comparison between mohclass->name and state->class->name is made, you are not guaranteed that (a) state->class is non-NULL or that state or state->class are in a safe state. Crashes caught by the bridges/transfer_capabilities test. ........ Merged revisions 416251 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416252 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416255 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	eade1d490c	res_manager_devicestate and res_manager_presencestate missing support level Add MODULEINFO comment block to define support level core for these new modules. Review: https://reviewboard.asterisk.org/r/3620/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Kinsey Moore	0acc626530	MoH: Don't restart stream on repeated start calls Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416150 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416151 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416152 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b2012ccb0a	CEL: Expose parking retreiver in extra field This exposes the retreiver of a parked call under the "retreiver" key of the extra field when this information is available. Review: https://reviewboard.asterisk.org/r/3608/ ........ Merged revisions 416148 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Rusty Newton	9ec5064383	main/pbx - documentation - enhance 'core show hints' and 'core show hint' help text Adds descriptive help text to 'core show hints' and 'core show hint'. The text describes the various columns for the sake of clarity. It takes into account recent changes to the content displayed by the commands https://reviewboard.asterisk.org/r/3604/ and https://reviewboard.asterisk.org/r/3611/. ASTERISK-23764 Review: https://reviewboard.asterisk.org/r/3610/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	27430374db	Fix build in devmode for GCC 4.10 ........ Merged revisions 415980 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Jonathan Rose	435343cfea	Blocked revisions 415838 ........ Correct UPGRADE.txt notes in r415825 The change was marked against the wrong version of Asterisk. My apologies. ........ Merged revisions 415837 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	fa8c58fefb	app_queue: delayed state can cause early leavewhenempty ringing In app_queue, device state changes arrive in event messages and update the queue member status value. That value is checked in get_member_status() to decide that the caller should leave when there are no available members. Although event messages can be delayed by other activity, there is no adverse affect by lagged status except in one specific case: there is only one available member, it was just rung, and leavewhenempty is enabled set for ringing members. This change adds a direct check of the device state only under this condition where the caller may be dropped incorrectly, resolving this issue without affecting performance of app_queue normally. AST-1248 #close Review: https://reviewboard.asterisk.org/r/3595/ Reported by: Thomas Arimont ........ Merged revisions 415833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415835 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415836 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	70b976f084	MixMontior: Add class authorization requirements to MixMonitor AMI commands MixMonitor AMI commands StartMixMonitor and StopMixMonitor lacked class authorization. StopMixMonitor now requires that the manager user either have the call or system class authorization. StartMixMonitor is a slightly larger issue since it can execute shell commands if the right arguments are passed into it, and we consider this a permission escalation. A security release will be issued for problem this shortly. ASTERISK-23609 #close Reported by: Corey Farrell ........ Merged revisions 415825 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415832 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	870394c051	res_pjsip_pubsub: unauthenticated remote crash in PJSIP pub/sub framework A remotely exploitable crash vulnerability exists in the PJSIP channel driver's pub/sub framework. If an attempt is made to unsubscribe when not currently subscribed and the endpoint's "sub_min_expiry" is set to zero, Asterisk tries to create an expiration timer with zero seconds, which is not allowed, so an assertion raised. The fix was to reject a subscription that is attempting to unsubscribe when not being already subscribed. Asterisk now checks for this situation appropriately and responds with a 400 instead of crashing. AST-2014-005 ASTERISK-23489 #close ........ Merged revisions 415812 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	e6cb6974fe	Fix potential deadlock situation in res_pjsip. SIP transaction timeouts are handled in the PJSIP monitor thread. When this happens on a subscription, and the subscription is destroyed, the subscription destruction is dispatched synchronously to the threadpool. The issue is that the PJSIP dialog is locked by the monitor thread, and then the dispatched task attempts to lock the dialog. This leads to a deadlock that causes SIP traffic to no longer be accepted on the Asterisk server. The fix here is to treat the monitor thread as if it were a threadpool thread when it attempts to dispatch synchronous tasks. This way, the dispatched task turns into a simple function call within the same thread, and the locking issue is averted. AST-2014-008 ASTERISK-23802 #close ........ Merged revisions 415794 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Walter Doekes	3b0ad74e17	safe_asterisk: Overwrite old safe_asterisk on make install. From now on, make install will overwrite safe_asterisk with the latest version. You need to move any local modifications to files inside /etc/asterisk/startup.d, if you have any. See also commits r394939 and r397938. ASTERISK-21965 #close Patches: safe_asterisk.patch uploaded by jkister (License 6232, modified by me) ........ Merged revisions 415748 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	71b3c9a749	format.c: Fix misuse of hash container function. The supplied hash function to a container must be idempotent given the object's key value to figure out which container bucket the object belongs in. Returning a random number or the current container count is not idempotent. The "computed hash" value doesn't help find the object later in those cases. * Fixed the format_list container to actually be a list since that is how the container is used. Conceptually, if more than 283 formats were added to the format_list then odd things may have happened before the fix. ........ Merged revisions 415728 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415729 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	d5298f2a1b	CLI: correct presence information on core show hints Adds presence to core show hint and changes presence string conversion to use the correct function. ASTERISK-23858 #close Review: https://reviewboard.asterisk.org/r/3611/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	d7ed0a1ece	CLI: add presence information to core show hints Adds presence state value to output of core show hints. Also reformats the output slightly so it doesn't use as much space as it would otherwise. Was: 1000@demo : SIP/1000 State:Unavailable Watchers 0 Now: 1000@demo : SIP/1000 State:Unavailable Presence:Idle Watchers 0 AFS-53 #close Review: https://reviewboard.asterisk.org/r/3604/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	2bd6a010a6	Fix build in dev mode due to signed/unsigned mismatch ........ Merged revisions 415678 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	a0adb8a26b	PJSIP: PJSIPNotify - Strip content-length headers and add documentation Documentation for how to add custom headers/content to notifies created with the PJSIPNotify manager action was a little sparse and it also wasn't vetting application of Content-length headers like its chan_sip equivalent was (so two Content-length headers could be applied... and PJSIP determines the content length anyway, so it just opens people up for error). This patch also flips the variable order so that the variables are interpreted in the same order as they are put in the AMI action. Review: https://reviewboard.asterisk.org/r/3587/ ........ Merged revisions 415658 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Alexandr Anikin	c63bf80c38	chan_ooh323: fix loading module failure if there no accessible h323_log or ooh323 config file change return 1 to return AST_MODULE_LOAD_FAILURE on module load routine few cosmetic changes ASTERISK-23814 #close (closes issue ASTERISK-23814) Reported by: Igor Goncharovsky Patches: ASTERISK-23814-ast11.patch ........ Merged revisions 415599 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415602 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	45db91dd98	chan_pjsip: Fix bug where custom SIP headers could be duplicated on outgoing INVITEs. When using PJSIP_HEADER() to add custom headers to outgoing INVITE requests, certain situations could result in the headers being duplicated. For instance, if the request were retransmitted, or if the INVITE were re-sent with authentication credentials, the custom headers would be re-added to the request. The fix here is to, after adding the custom headers to the outbound INVITE, remove the datastore that holds the custom headers to add. This way, there is no risk in accidentally adding them if the session supplement is called into a second or third time. ........ Merged revisions 415579 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	ce733a02b4	safe_asterisk: Cleanup additions to r415132. * Replaced a stray echo that should've been a message call in safe_asterisk. This replaces a conditional log message by a slightly different message. Please update your log parsing scripts. * Made the $NOTIFY mail Subject more verbose by adding the machine name and exitstatus. (Note that a 'make install' still won't overwrite your old safe_asterisk if it exists. See ASTERISK-21965.) ASTERISK-23492 #close ........ Merged revisions 415521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415522 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415523 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8da7f0248f	autoservice: stop thread on graceful shutdown This change adds thread shutdown to autoservice for graceful shutdowns only. ast_register_cleanup is backported to 1.8 to allow this. The logger callid is also released on shutdown in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3594/ ........ Merged revisions 415463 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415464 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415465 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 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	11 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	11 years ago
Kevin Harwell	4308aa5648	core uri: Custom uri parsing error when no query parameters If using the custom URI parsing code (not external uriparser lib) and there was no query parameters the resulting pointer would be NULL and then an attempt was made to subtract from it. The pointer is now set to a valid value if there is no query parameter(s). Also, in the 'ast_uri_make_host_with_port' function when setting the terminator on the resulting string it was writing it one past the end of allocated memory. It now writes the string terminator appropriately. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	5510e3c699	PJSIP: Remove premature write of raw formats Currently, there are situations that can occur when using chan_pjsip and certain dialplan applications (notably ChanSpy()) that can cause the channel to get no audio with scrolling warnings about format mismatches. This is caused by a failure to update translation paths on a mid-call native format update since the raw formats have already been updated by res_pjsip_sdp_rtp.c in set_caps(). Removing the premature raw format updates allows the translation paths to be setup correctly and the raw read and write formats with them. AFS-63 #close ........ Merged revisions 415342 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Rusty Newton	4e292ea3af	configs/cli_aliases.conf: Two new aliases, plus enhancements for context names. Changed naming of included alias templates to avoid confusion between version names. For example, asterisk12 was for asterisk 1.2, so I changed it to asterisk_1dot2, so that later we can use asterisk_12 for Asterisk 12. Added alias for "features reload" to the template for Asterisk 11 style syntax template, as features reload was removed in 12, but you can still do "module reload features" Added alias for "pjsip reload" to the friendly template. It is shorter than "module reload res_pjsip.so" and if some are like me; I constantly forget that reloading chan_pjsip doesn't parse config. Remembering "pjsip reload" is just easier. ASTERISK-23654 #close ASTERISK-23654 #comment Fixed by adding two new aliases and enhancements for context names. Review: https://reviewboard.asterisk.org/r/3572/ ........ Merged revisions 415301 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b0abea6028	config: Fix indentation and missing curlies in config_text_file_load(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	61b0be0600	config: Fix config files not reloading when only an included file changes. The twisted logic determining if a config file should be reloaded was mostly broken and disabled. The incorrect test that ASTERISK-23383 fixed actually reenabled the broken logic. The incorrect test was causing the timestamp to always be cleared which caused config files with includes to always be reloaded. * Made wildcard includes always cause a reload. Determining if a file was deleted cannot be determined without restructuring the cache to determine if any files are missing from the last files actually loaded. Also without refactoring config_text_file_load(), the glob loop couldn't check more than one file for changes anyway. * Made remove the cache entry if the file no longer exists when trying to get its timestamp or it is no longer a regular file. This fixes the corner case where the file was loaded, then deleted, then the config reloaded, then the file restored with the same timestamp, and then the config reloaded again. * Made remove the cache entry include list when actually loading the file. This gets rid of any stale includes the file had from the last time the file was loaded. ASTERISK-23683 #close Reported by: tootai Review: https://reviewboard.asterisk.org/r/3575/ ........ Merged revisions 415225 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415229 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415230 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 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	11 years ago
Matthew Jordan	fd45b82247	app_confbridge: Allow muting of users waiting to enter a ConfBridge Prior to this patch, users waiting to enter a ConfBridge were not considered when muted via the CLI or via AMI. Instead, a confusing message would be emitted stating that the channel did not exist. This patch allows a user to be muted when waiting to enter a ConfBridge conference. This is equivalent to start when muted, only toggled via the CLI or AMI. Review: https://reviewboard.asterisk.org/r/3582 #ASTERISK-23824 #close patches: rb3582.patch uploaded by tm1000 (License 6524) ........ Merged revisions 415206 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415207 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	6fd8940d4c	PJSIP: Send initial connected line information This makes chan_pjsip send connected line information when it is called so that connected line information is available on the connected channel. (closes issue DPMA-442) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3584/ ........ Merged revisions 415191 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	cfb9c8bff1	safe_asterisk: Cleanup and debian compatibility. Cleans up the safe_asterisk script and adds the ASTSAFE_FOREGROUND option that allows the debian asterisk init script to capture the right pid. * Drop the vim #modeline which wasn't used. Use test consistently without the odd configure xno syntax. Double quote all paths. General cleanup. * Don't output message()s to the console but only to TTY if set. * Allow TTY to be "no" as well as empty (debian compatibility with debian/patches/safe_asterisk-config). * Add option to export ASTSAFE_FOREGROUND=1 from the init script that calls this to disable backgrounding. Debian uses a similar method in debian/patches/safe_asterisk-nobg). ASTERISK-23492 #close Review: https://reviewboard.asterisk.org/r/3574/ ........ Merged revisions 415132 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415171 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415172 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6b423e311b	chan_pjsip: Add debug in RTP Engine glue callback This patch adds some debug statements that aid with determining why a direct media request may or may not be initiated. ........ Merged revisions 415117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4f603c5da3	res_pjsip_session: Add debug statement for session refreshes This small patch adds a debug level 3 statement indicating how a session refresh is being sent - either as a re-INVITE or as an UPDATE - and where the session refresh is going. ........ Merged revisions 415115 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	db2ee74883	app_confbridge: Correct verification of conference name length Conference names were not checked for maximum length, allowing unexpected behaviour. This change adds checking to ensure the maximum length is not exceeded. The maximum length is also changed from 32 to AST_MAX_EXTENSION. ASTERISK-23035 #close Reported by: Iñaki Cívico Tested by: Iñaki Cívico Patches: confbridge-enforce_max-1.8.patch uploaded by coreyfarrell (license 5909) confbridge-enforce_max-11up.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 415060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415066 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415078 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	795af210a3	func_odbc: Fix fixed size buffers fix (r414968). The change that removed the fixed size buffers in odbc-related code -- removing arbitrary column width limits -- was incomplete. This change adds: no segfault on writesql without insertsql and return value checks after strdup. While I was in the vicinity I cleaned up the linefeeds in the odbc function descriptions, moved some code for clarity, removed some blobs and noted (but didn't fix) that the 'odbc write ... exec' CLI command doesn't behave as the dialplan equivalent when insertsql= is used. ASTERISK-23582 #close Review: https://reviewboard.asterisk.org/r/3579/ ........ Merged revisions 414997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414998 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414999 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	962b78bca1	bridge_native_rtp: Take the bridge type choice of both channels into account. The bridge_native_rtp module currently uses the bridge result of the first channel that joins a bridge as the ultimate result. This means that if the first channel has direct media enabled but the second does not a direct media bridge will still occur. This change makes it so that both sides are taken into account. If either side forbids the bridge or responds with a local bridge result then either a generic or local bridge occurs. ASTERISK-23541 #close Reported by: Justin E Review: https://reviewboard.asterisk.org/r/3577/ ........ Merged revisions 414975 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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