asterisk

Commit Graph

Author	SHA1	Message	Date
Leif Madsen	53c1ecd8ea	Add note about how Authenticate() application with option 'd' works. (closes issue ASTERISK-17422) Reported by: Leif Madsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	8daf83a53b	Modify comments in MeetMe application documentation about DAHDI. The MeetMe application documentation has some comments about usage of DAHDI, and they were a bit outdated relative to modern DAHDI releases. This patch changes the comment to just tell the user that a functional DAHDI timing source is required, and no longer mention 'dahdi_dummy', since that module does not exist in current DAHDI releases. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e9dc0ae56d	Use int for storing ao2_container_count instad of size_t AST-676 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	33f73e2ae2	Simplify queue membercount code Despite an ominous sounding comment stating that membercount was for "logged in" members only and thus we couldn't use ao2_container_count(), I could not find a single place in the code where that seemed to be accurate. The only time we decremented membercount was when we were marking something dead or actually removing it. The only places we incremented it were either after ao2_link(), or trying to correct for having set it to 0 during a reload. In every case where we were correcting the value, it seemed that we were trying to make the count actually match what ao2_container_count() would return. The only place I could find where we made a determination about something being "logged in" or not, we didn't trust the membercount, but instead looked at devicestate, paused, etc. This patch removes membercount, replaces its use with ao2_container_count, and manually adds the results of ao2_container_count to a "membercount" field for ast_data queue query results. This patch also would fix AST-676, but as it is slightly riskier than the previously committed fix, the two commits have been made separately. Reivew: https://reviewboard.asterisk.org/r/1541/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f927ef5571	Properly update membercount for reloaded members Since q->membercount is set to 0 before reloading, it is important to increment it again for reloaded members as well as added. (closes issue AST-676) Review: https://reviewboard.asterisk.org/r/1541/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	4037f1366d	Revert Janitor patch 341906 For now git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	4f690341cf	Whitespace Fixups / Add Braces This janitorial patch is related to work on RB1538 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	fb6e8a5575	Fix previous commit git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	902b38d21d	Voicemail compiler flags are 'core' support git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	88bf8d3316	Fixes some support level info so that it can be read by menuselect. (issue ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4051609b5c	Update MeetMe p and X option documentation when interacting with the s option. ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	de9e8e501e	Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2fb42fc3da	Initialize option flags for SendURL application. (closes issue ASTERISK-18574) Reported by: marcelloceschia git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@339776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1a4ba9305a	Fix Dial F option notes formatting. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@339511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	e83a93313c	Make documentation for Dial() options 'F' and 'F()' more clear. (Closes issue ASTERISK-18646) Reported by: Physis Heckman Tested by: Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@339144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
TransNexus OSP Development	7d656e1330	Remove r338137 and r338138. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
TransNexus OSP Development	9e2e3778af	Updated for OSP Toolkit 4.0.0. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	32fc932cf5	Upgrade app_macro to core git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f2e1640435	Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	3b2f5e7d4c	Make sure a CDR is on the stack for call in the Queue. Only let update_cdr act on the last CDR in the stack. In some circumstances [Attended transfer to queue] a CDR record is not inserted for this call where it should. (closes issue ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	c4cd620d7a	More silly spacing changes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	6e94c27f6c	Dumb little spacing fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	f13c3b3fd2	Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	32c717b97c	Document applications that play audio and do not answer unanswered calls. This patch is part of an effort to document early media and its usage. If you are interested in contributing to this documentation effort, there are probably other applications worth documenting as well as an Asterisk wiki article at https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	07a3a611a9	Made Dial d and H options no longer immediately auto-answer the calling leg. The Dial d and H options break DTMF attended transfer atxferdropcall option. 1) Party A calls party B. 2) Party B does a DTMF attended transfer to Party C. If the dialplan uses the Dial d or H options to call Party C then the Dial application answers the call immediately before initiating the call leg to Party C. The premature answer causes the transfer code to not invoke the atxferdropcall=no behavior for a blonde transfer since Party C has "answered". The transfer code thinks that Party B has "consulted" with Party C when Party B hangs up and completes the transfer to Party A. Party A now hears ringback until Party C actually answers. ASTERISK-13294 Dial d option. ASTERISK-11067 Dial H option to disconnect before answer. The referenced issues made Dial answer with the d and H options because many SIP and ISDN phones cannot send DTMF before the call is connected. * Made require the dialplan to control when or if the call needs to be answered to use the Dial application d and H options. (The call is no longer surprise answered when using the Dial d or H options.) Review: https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA AST-666 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	f94fa3dba3	Locking order in app_queue.c causes deadlocks.	14 years ago
Richard Mudgett	5c5122d104	Remove obsolete todo comment about PICKUPRESULT. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	28952b7ea5	Meetme should have 'core' support level (closes issue ASTERISK-18542) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	263a410438	Ensure frames are not written to dialed channel if ringback is requested When a single channel was dialed and there was media to be forwarded to the calling channel, the media was written without regard for ringback causing silence to be heard in some circumstances. This regression was introduced when the meaning of "single" changed to mean only the number of channels dialed. (closes issue ASTERISK-18083) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7dc49195d8	Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	74f9e66b41	peroid typo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	4b1398a82d	Make SQL query in app_voicemail.c portable LIMIT is not portable.	14 years ago
Matthew Jordan	92ad64998c	Fixed improperly formatted TestEvent AMI message in app_voicemail git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	3a29ee54db	Fixed incorrect pointer copy to structure copy in revision 333339 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	792c3a2d56	Bug fixes for voicemail user emailsubject / emailbody. This code change fixes a few issues with the voicemail user override of emailbody and emailsubject, including escaping the strings, potential memory leaks, and not overriding the voicemail defaults. Revision 325877 fixed this for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent check-in prevented 325877 from being applied to 10. This check-in resolves both issues, and applies the changes to 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by: Sebastien Couture Tested by: mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1374 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	185f890c89	Memory Leak in app_queue The patch that was committed in the 1.6.x versions of Asterisk for ASTERISK-15862 actually fixed two issues. One was not applicable to 1.8 but the other is. queue_leak.patch fixes the portion applicable to 1.8. (closes issue ASTERISK-18265) Reported by: Fred Schroeder Patches: queue_leak.patch (license #5049) patch uploaded by mmichelson Tested by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b1f11e0df4	Revert previous commit. Not ready yet. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c0ce03d77f	Signed git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	def9e8fe45	Reference leaks in app_queue. * Fixed load_realtime_queue() leaking a queue reference when it overwrites q when processing a realtime queue. (issue ASTERISK-18265) * Make join_queue() unreference the queue returned by load_realtime_queue() when it is done with the pointer. The load_realtime_queue() returns a reference to the just loaded realtime queue. * Fixed queues container reference leak in queues_data_provider_get(). * queue_unref() should not return q that was just unreferenced. * Made logic in __queues_show() and queues_data_provider_get() when calling load_realtime_queue() easier to understand. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	56549c96ab	Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	b92d5952e1	Unlock the channel before calling update_queue. Holding the channel lock when calling update_queue which attempts to lock the queue lock can cause a deadlock. This deadlock involves the following chain: 1. hold chan lock -> wait queue lock 2. hold queue lock -> wait agent list lock 3. hold agent list lock -> wait chan list lock 4. hold chan list lock -> wait chan lock git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	427d1167cd	Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	0688f632dc	Use proper values for 64-bit option flags. Also, reusing bits es no bueno, so change the value of a duplicate. (issue ASTERISK-18239) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	42b5040b71	Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	ae2e5eea83	Change support for ConfBridge() in 1.8 to Extended. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	3b50c5a387	Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more appropriate anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	31a1b94622	Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5c06d1dbb0	Update PickupChan documentation. The PickupChan uses the ampersand as the argument separator. Was documented as: PickupChan(channel[,channel2[,...][,options]]) Fixed documentation to: PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) This is a continuation of ASTERISK-17494 for v1.8 and later. (closes issue ASTERISK-18144) Reported by: Erik Smith Patches: pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith Tested by: Erik Smith git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	9769bb7d34	MeetMe requests a PIN twice in some circumstances If a call to MeetMe includes both the dynamic(D) and always request PIN(P) options, MeetMe will ask for the PIN two times: once for creating the conference and once for entering the conference. This behavior was introduced in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN entry for joining a conference. (closes AST-601) Review: https://reviewboard.asterisk.org/r/1305/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	f5ec4864bd	app_dial may double free a channel datastore When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it. (closes issue ASTERISK-17917) Reported by: Mark Murawski Tested by: Mark Murawski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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3963 Commits (d63e58203a796bb59e20b51cb705254100621c9a)