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2012-05-10Run predial routine on local;2 channel where you would expect.Richard Mudgett
Before this patch, the predial routine executes on the ;1 channel of a local channel pair. Executing predial on the ;1 channel of a local channel pair is of limited utility. Any channel variables set by the predial routine executing on the ;1 channel will not be available when the local channel executes dialplan on the ;2 channel. * Create ast_pre_call() and an associated pre_call() technology callback to handle running the predial routine. If a channel technology does not provide the callback, the predial routine is simply run on the channel. Review: https://reviewboard.asterisk.org/r/1903/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-10Resolve FORWARD_NULL static analysis warningsKinsey Moore
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366168 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-10Coverity Report: Fix issues for error type CHECKED_RETURN for coreJonathan Rose
(issue ASTERISK-19658) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1905/ ........ Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366106 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-10Close the proper tcptls_session when session creation fails.Mark Michelson
(issue AST-998) Reported by: Thomas Arimont Tested by: Thomas Arimont ........ Merged revisions 366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366053 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-10Coverity Report: Fix issues for error type UNINIT in Core supported modulesJonathan Rose
(issue ASTERISK-19652) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1909/ ........ Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366049 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-09Block on frameout if the hardware has enough samples to complete a frame.Jonathan Rose
Fixes some problems with skipping audio in elaborate scenarios involving multiple codecs by making codec_dahdi operate in a more synchronous fashion similar to codec_g729. This change also fixes the use of file conversion tools from Asterisk's CLI. This change may cause the thread responsible for transcoding audio to block briefly (Shaun Ruffell describes this as 'several milliseconds') while waiting for the hardware transcoder. (closes issue ASTERISK-19643) reported by: Shaun Ruffell Patches: 0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch uploaded by Shaun Ruffell (license 5417) ........ Merged revisions 365989 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365990 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-09pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselectTzafrir Cohen
Allow menuselect to get its set of CFLAGS and LDFLAGS through the environment of Make: make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever" Review: https://reviewboard.asterisk.org/r/1907/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-09Improve FollowMe accept/decline DTMF string matching.Richard Mudgett
If you hit the wrong DTMF digit trying to accept/decline a FollowMe call, you had to wait for the prompt to repeat to try again. * Make FollowMe compare the last DTMF digits received to the accept/decline matching strings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-09Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt.Mark Michelson
chan_sip was coded under the assumption that a SIP dialog with an owner channel will always be destroyed after the owner channel has been hung up. However, there are situations where the SIP dialog can time out and auto destruct before the corresponding channel has hung up. A typical example of this would be if the 'h' extension in the dialplan takes a long time to complete. In such cases, __sip_autodestruct() would complain about the dialog being auto destroyed with an owner channel still in place. The problem is that even once the owner channel was hung up, the sip_pvt would still be linked in its ao2_container because nothing would ever unlink it. The fix for this is that if __sip_autodestruct() is called for a sip_pvt that still has an owner channel in place, the destruction is rescheduled for 10 seconds in the future. This will continue until the owner channel is finally hung up. (closes issue ASTERISK-19425) reported by David Cunningham Patches: ASTERISK-19425.patch uploaded by Mark Michelson (License #5049) (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by Dean Vesvuio ........ Merged revisions 365896 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365898 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-09Keep answered FollowMe calls until call accepted or last step times out.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-09Put winning FollowMe outgoing call on hold if the caller put it on hold.Richard Mudgett
The FollowMe caller call leg is usually answered and listening to MOH. The caller could put the call on hold while FollowMe is looking for a winner. The winning outgoing call is now immediately placed on hold if the caller has put the call on hold before the winning call was selected. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-09Restructure how the FollowMe outgoing channel list is handled.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-08Addendum to -r365766. Since it is no longer allocated.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-08Make FollowMe findmeexec() put the list head on the stack instead of ↵Richard Mudgett
mallocing it. Why this tiny struct was malloced instead of the 28k struct in the last change is beyond me. Just doing my part to help stamp out sillyness. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-08Add interrupt ('I') command to ExternalIVR.Sean Bright
Sending the 'I' command from an external process will cause the current playlist to be cleared, including stopping any audio file that is currently playing. This is useful when you want to interrupt audio playback only when specific DTMF is entered by the caller. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-08Make FollowMe app_exec() not declare a 28k struct on the stack.Richard Mudgett
Helping to stamp out stack abuse. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-08Simplify findmeexec() parameter passing.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-08* Fix FollowMe memory leak on error paths in app_exec().Richard Mudgett
* Fix FollowMe leaving recorded caller name file on error paths in app_exec(). * Use correct buffer dimension define in struct fm_args.namerecloc[]. This fixes unexpected namerecloc filename length restriction. ........ Merged revisions 365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365701 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-08* Fix accept/decline DTMF buffer overwrite in FollowMe.Richard Mudgett
* Made use MAX_YN_STRING define to make all accept/decline DTMF buffers the same size. Just using 20 isn't good enough when someone didn't get the memo. * Fix stupid use of a global variable in FollowMe. (ynlongest) * Fix bit field declarations in FollowMe. ........ Merged revisions 365631 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365632 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-08Send more accurate identification information in dialog-info SIP NOTIFYs.Mark Michelson
This uses the calling channel's caller ID and connected line information to populate the remote and local identities in the dialog-info NOTIFY when an extension is ringing. There is a bit of an oddity here, and that is that we seed the remote target with the To header of the outbound call rather than the from header. This is because it was reported that seeding with the from header caused hints to be broken with certain SNOM devices. A comment has been added to the code to explain this. (closes issue ASTERISK-16735) reported by Maciej Krajewski patches: local_remote_hint2.diff uploaded by Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark Michelson (license #5049) Tested by Niccolo Belli ........ Merged revisions 365574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365575 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07Change comment to use local channel name designators in features.cRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07Fix channel opaquification slip-up in r365477Matthew Jordan
Those channels are opaque now... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07Fix type punned compiler warning in test_config.cRichard Mudgett
........ Merged revisions 365476 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365478 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07Support VoiceMail d() option when extension does not exist in channel's contextMatthew Jordan
The VoiceMail d([c]) option is documented to accept digits for a new extension in context <c>, if played during the greeting. This option works fine if the extension being redirected to has an extension with the same initial digit in the channel's current context. If that digit did not happen to exist in some extension, a dialplan match would fail and the user would not be redirected. This patch fixes it such that if the <c> option is used, the extensions are matched in that context as opposed to the caller's original context. (closes issue ASTERISK-18243) Reported by: mjordan Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1892 ........ Merged revisions 365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365475 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-04Fix many issues from the NULL_RETURNS Coverity reportKinsey Moore
Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365399 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-04Fix local channel chains optimizing themselves out of a call.Richard Mudgett
* Made chan_local.c:check_bridge() check the return value of ast_channel_masquerade(). In long chains of local channels, the masquerade occasionally fails to get setup because there is another masquerade already setup on an adjacent local channel in the chain. * Made the outgoing local channel (the ;2 channel) flush one voice or video frame per optimization attempt. * Made sure that the outgoing local channel also does not have any frames in its queue before the masquerade. * Made do the masquerade immediately to minimize the chance that the outgoing channel queue does not get any new frames added and thus unconditionally flushed. * Made block indication -1 (Stop tones) event when the local channel is going to optimize itself out. When the call is answered, a chain of local channels pass down a -1 indication for each bridge. This blizzard of -1 events really slows down the optimization process. (closes issue ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec Davis Review: https://reviewboard.asterisk.org/r/1894/ ........ Merged revisions 365313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365320 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-04Fix core FINDING 2, FINDING 3, and FINDING 4 from Coverity's ↵Mark Michelson
CONSTANT_EXPRESSION_RESULT report. These three all are in RTP code that attempts to print the number of sequence number cycles in an RTCP RR report. The code was masking out the upper 16 bits and then shifting the number right by 16 bits. This led to an all zero result in all cases. The fix is to do the shift without the bit masking. (issue ASTERISK-19649) ........ Merged revisions 365298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365299 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-03Update security events unit testsMichael L. Young
The security events framework API was changed in Asterisk 10 but the unit tests were not updated at the same time. This patch does the following: * Adds two more security events that were added to the API * Add challenge, received_challenge and received_hash in the inval_password security event unit test (Closes issue ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael L. Young Patches: issue-asterisk-19760-trunk.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1897/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-03Update documentation references in CHANGES to reflect the correct pages on ↵Sean Bright
the wiki. The current CHANGES file refers to doc/ in many places and those files no longer exist. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-03Fix warning of Coverity Static analysis, change H225ProtocolIdentifierAlexandr Anikin
from value to pointer per functions that use this. (close issue ASTERISK-19670) Reported by: Matt Jordan Patches: ASTERISK-19670.patch (License #5415) ........ Merged revisions 365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365160 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-03Add IPv6 support to ExternalIVR.Sean Bright
Review: https://reviewboard.asterisk.org/r/1896/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-03Fix coverity static analysis warning, allocate full ie structureAlexandr Anikin
instead of without data buffer (close issue ASTERISK-19674) Reported by: Matt Jordan Patches: ASTERISK-19674.patch (License #5415) ........ Merged revisions 365143 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365155 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-02Multiple revisions 365006,365068Terry Wilson
........ r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race and local channel linkedids This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes the race condition by no longer scanning the channel list for "other" channels with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings and uses the refcount of the string as a counter of how many channels with the linkedid exist. Not only does this eliminate the race condition, but it also allows us to look up the linkedid by the hashed key instead of traversing the entire channel list. Review: https://reviewboard.asterisk.org/r/1895/ ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines Don't leak a ref if out of memory and can't link the linkedid If the ao2_link fails, we are most likely out of memory and bad things are going to happen. Before those bad things happen, make sure to clean up the linkedid references. This patch also adds a comment explaining why linkedid can't be passed to both local channel allocations and combines two ao2_ref calls into 1. Review: https://reviewboard.asterisk.org/r/1895/ ........ Merged revisions 365006,365068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365083 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-02Blocked revisions 365014Michael L. Young
........ Update security events unit tests The security events framework API was changed in Asterisk 10 but the unit tests were not updated at the same time. This patch does the following: * Adds two more security events that were added to the API * Add challenge, received_challenge and received_hash in the inval_password security event unit test (issue ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael L. Young Patches: issue-asterisk-19760-branch10.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1877/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-02Save the address on which a MESSAGE was received, so it can be used in MESSAGE()Jason Parker
This is useful in cases where chan_sip may be listening on multiple addresses. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-02Only log a failure to get read/write samples from factories if it didn't happenMatthew Jordan
In audiohook_read_frame_both, anytime samples are obtained from the read/write factories a debug statement is logged stating that samples were not obtained from the factories. This statement used to only occur if option_debug was turned on and no samples were obtained; in some refactoring when the option_debug statement was removed, the "else" clause was removed as well. This patch makes it so that those debug log statements only occur if the condition leading up to them actually happened. ........ Merged revisions 364965 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01Remove a function that has been marked unused since Asterisk 1.6.0.Mark Michelson
The reason I'm removing this is that Coverity reported a STRAY_SEMICOLON issue here. Since the function has been unused for so long, I just elected to remove it altogether. (closes issue ASTERISK-19660) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01Fixed __ao2_ref() validating user_data twice.Richard Mudgett
(closes issue ASTERISK-19755) Reported by: Gunther Kelleter Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter ........ Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364903 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01Fix Coverity-reported ARRAY_VS_SINGLETON error.Mark Michelson
As it turned out, this wasn't a huge deal. We were calling ast_app_parse_options() for a set of options of which none took arguments. The proper thing to do for this case is to pass NULL for the "args" parameter here. We were instead passing a seemingly-randomly chosen char * from the function. While this would never get written to, you can rest assured things would have gotten bad had new options (which took arguments) been added to func_volume. (closes issue ASTERISK-19656) ........ Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364900 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01* Fix error path resouce leak in local_request().Richard Mudgett
* Restructure local_request() to reduce indentation. ........ Merged revisions 364840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364845 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01Prevent a potential crash when using manager hooks.Jason Parker
Found by me while poking at DPMA-127. ........ Merged revisions 364841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364842 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01Play conf-placeintoconf message to the correct channelKinsey Moore
Correct the code in app_confbridge to play the conf-placeintoconf message to the marked user entering the bridge instead of to the conference while the marked user hears silence. (closes issue ASTERISK-19641) Reported-by: Mark A Walters ........ Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364787 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01Fix bad check in voicemail functions for ast_inboxcount2_funcJonathan Rose
Check looks for ast_inboxcount_func instead of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes issue ASTERISK-19718) Reported by: Corey Farrell Patches: ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364777 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-30Revert revision 360862.Mark Michelson
Revision 360862 was intended to improve identities sent in dialog-info NOTIFY requests. Some users reported that hint became broken once this was done. It's not clear exactly what part of the patch has caused this regression, but broken hints are bad. For now, this revision is being reverted so that the next releases of Asterisk do not have bad behavior in them. The original reported issue will have to be fixed differently in the next version of Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364707 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364708 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-30Merged revisions 364635 via svnmerge from Mark Murawki
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) | 10 lines Sanatize result from bfd_find_nearest_line (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file to null resulting in a crash when strrchr(file) runs (closes issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark Murawski ........ ........ Merged revisions 364650 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-30Fix use freed pointer in return value from call threadAlexandr Anikin
(issue ASTERISK-19663) Reported by: Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415) ........ Merged revisions 364649 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364651 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-29Fix error that caused truncate operations to failMatthew Jordan
Another very inappropriate placement of a ')' (again introduced in r362151) caused the various truncate operations to attempt to truncate the sound file at a position of '0'. (issue ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec ........ Merged revisions 364578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364579 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-29Fix configuring custom sound_leader_has_left in confbridge.confMichael L. Young
The configuration option to specify a custom sound_leader_has_left file for a conference bridge was not being parsed. This patch fixes it so that a custom sound file will now be used. (closes issue ASTERISK-19771) Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380) Review: https://reviewboard.asterisk.org/r/1884/ ........ Merged revisions 364536 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28Add support for lightweight NAT keepalive.Joshua Colp
If enabled using the keepalive option in sip.conf a small packet will be sent at a regular interval to keep the NAT mapping open. This is lightweight as the remote side does not need to parse and handle a SIP message. (closes issue AST-783) Review: https://reviewboard.asterisk.org/r/1756/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28md5: supress some compiler warnings.Russell Bryant
md5.c: In function ‘MD5Final’: md5.c:154:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] md5.c:155:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] There is an md5 unit test and it still passes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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