22093 Commits (80efa3173315ba47b794e5c9f1b56f61a663a76c)

 

All Branches   

Search

Author	SHA1	Message	Date
Jonathan Rose	923a66d764	chan_sip: Check the right channel's host address for directmediapermit/deny ... Prior to this patch, when checking the addresses for directmediapermit and directmediadeny, Asterisk would check the host address of the channel permit/deny was specified, which differs from the expectations of both our users and the development team. Instead, directmediapermit/deny now checks against the address of the channel that the peer with the ACL is connected to. (issue AST-876) Review: https://reviewboard.asterisk.org/r/1899/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	2deb898bcd	Fix two more coverity constant expression result findings. ... These correspond to findings 0 and 1 in the core findings of ASTERISK-19649. After contacting Mark Spencer, he was unsure of what the intent behind these lines of code were, so they are being axed. For Asterisk 1.8 and 10, the output of debugging DUNDi frames will not be changed, but for trunk the "Retry" portion will be omitted since it does not properly distinguish retransmissions from initial frames. (closes issue ASTERISK-19649) Reported by Matthew Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	fd520e0d19	Fix broken reinvite glare scenario. ... To make a long story short, reinvite glares were broken because Asterisk would invert the To and From headers when ACKing a 491 response. The reason was because the initreq of the dialog was being changed to the incoming glared reinvite instead of being set to the outgoing glared reinvite. This change has three parts * In handle_incoming, we never will reject an ACK because it has a to-tag present, even if we think the request may be out of dialog. * In handle_request_invite, we do not change the initreq when receiving a reinvite to which we will respond with a 491. * In handle_request_invite, several superflous settings up pendinginvite have been removed since this is dones automatically by transmit_response_reliable Review: https://reviewboard.asterisk.org/r/1911 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	dac30f1523	format_mp3: Fix a possible crash in mp3_read(). ... This patch fixes a potential crash in mp3_read() by not assuming that dbuf has enough data to finish filling up the output buffer. The patch also makes sure that the dbuf state gets reset after we know we read everything out of it already. In passing, this patch includes some other cleanups of this module, including stripping trailing whitespace, formatting fixes based on coding guidelines, and removing a number of unused members from the private state struct. (closes issue ASTERISK-19761) Reported by: Chris Maciejewsk Tested by: Chris Maciejewsk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	91154705df	* Made ast_change_name() hold the channels container lock while changing the channel name. ... * Eliminate redundant list not empty check in clone_variables(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	a94fcae21b	Resolve FORWARD_NULL static analysis warnings ... 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	ae528efea3	Coverity Report: Fix issues for error type CHECKED_RETURN for core ... (issue ASTERISK-19658) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1905/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	965dd3a7d8	Close the proper tcptls_session when session creation fails. ... (issue AST-998) Reported by: Thomas Arimont Tested by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	a7650f8002	Coverity Report: Fix issues for error type UNINIT in Core supported modules ... (issue ASTERISK-19652) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1909/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	50b43947f1	Block on frameout if the hardware has enough samples to complete a frame. ... 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	3a9a0b9cea	Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt. ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	55980149c7	* Fix FollowMe memory leak on error paths in app_exec(). ... * Fix FollowMe leaving recorded caller name file on error paths in app_exec(). * Use correct buffer dimension define in struct call_followme.moh[] and struct fm_args.namerecloc[]. This fixes unexpected namerecloc filename length restriction. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e163e37e24	* Fix accept/decline DTMF buffer overwrite in FollowMe. ... * 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. * Fix FollowMe n option documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	2cb787371c	Send more accurate identification information in dialog-info SIP NOTIFYs. ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	4547358e75	Fix type punned compiler warning in test_config.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	2c8f87dcb4	Support VoiceMail d() option when extension does not exist in channel's context ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365474 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	eafb430562	Fix findings 0-3, 5, and 8 for Coverity MISSING_BREAK errors. ... (Issue ASTERISK-19662) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	83d3444284	Fix many issues from the NULL_RETURNS Coverity report ... 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	186a89e99c	Fix local channel chains optimizing themselves out of a call. ... * 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	c6d9524482	Fix core FINDING 2, FINDING 3, and FINDING 4 from Coverity's 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	4bddfcf335	Fix warning of Coverity Static analysis, change H225ProtocolIdentifier ... from value to pointer per functions that use this. (close issue ASTERISK-19670) Reported by: Matt Jordan Patches: ASTERISK-19670.patch (License #5415) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	0b9bca22bd	Fix coverity static analysis warning, allocate full ie structure ... instead of without data buffer (close issue ASTERISK-19674) Reported by: Matt Jordan Patches: ASTERISK-19674.patch (License #5415) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	c068460b12	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365068 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	7da21c68c4	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	a11540c95b	Fixed __ao2_ref() validating user_data twice. ... (closes issue ASTERISK-19755) Reported by: Gunther Kelleter Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	095054e4a1	Fix Coverity-reported ARRAY_VS_SINGLETON error. ... 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	be16f2da64	Prevent a potential crash when using manager hooks. ... Found by me while poking at DPMA-127. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fde9505a95	* Fix error path resouce leak in local_request(). ... * Restructure local_request() to reduce indentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	dcd64542bc	Play conf-placeintoconf message to the correct channel ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	5b45764bb7	Fix bad check in voicemail functions for ast_inboxcount2_func ... 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	dba70c1340	Revert improved identities sent in dialog-info NOTIFY requests in r360862 ... 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	97787d8c5e	Fix use freed pointer in return value from call thread ... (issue ASTERISK-19663) Reported by: Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Murawki	2a6bdce6b4	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f248f6eb71	Fix error that caused truncate operations to fail ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	139a7459cd	Don't attempt to make use of the dynamic_exclude_static ACL if DNS lookup fails. ... (closes issue ASTERISK-18321) Reported by Dan Lukes Patches: ASTERISK-18321.patch by Mark Michelson (license #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	f0eb58d0d5	Fix ast_parse_arg numeric type range checking and add tests ... ast_parse_arg wasn't checking for strto* parse errors or limiting the results by the actual range of the numeric types. This patch fixes that and adds unit tests as well. Review: https://reviewboard.asterisk.org/r/1879/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c527c4d197	Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines ... The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds, between two timeval structs, and return the difference in a 64-bit integer. Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval struct are large enough to hold the calculated values before it returns. On 64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit machines, however, a long may be less (32-bits), in which case, the calculation can overflow. This overflow caused significant problems in MixMonitor, which uses the method to determine if an audio factory, which has not presented audio to an audiohook, is merely late in providing said audio or will never provide audio. In an overflow situation, the audiohook would incorrectly determine that an audio factory that will never provide audio is merely late instead. This led to situations where a MixMonitor never recorded any audio. Note that this happened most frequently when that MixMonitor was started by the ConfBridge application itself, or when the MixMonitor was attached to a Local channel. (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski Tested by: Michael L. Young Patches: 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan) (closes issue ASTERISK-19471) Reported by: feyfre Tested by: feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1889/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	0536634ff1	Allow SIP pvts involved in Replaces transfers to fall out of reference sooner ... Unref the SIP pvt stored in the refer structure as soon as it is no longer needed so that the pvt and associated file descriptors can be freed sooner. This change makes a reference decrement unnecessary in code that handles SIP BYE/Also transfers which should not touch the reference anyway. (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski Tested by: Maciej Karjewski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3ef885b576	Allow for reloading SRTP crypto keys within the same SIP dialog ... As a continuation of the patch in r356604, which allowed for the reloading of SRTP keys in re-INVITE transfer scenarios, this patch addresses the more common case where a new key is requested within the context of a current SIP dialog. This can occur, for example, when certain phones request a SIP hold. Previously, once a dialog was associated with an SRTP object, any subsequent attempt to process crypto keys in any SDP offer - either the current one or a new offer in a new SIP request - were ignored. This patch changes this behavior to only ignore subsequent crypto keys within the current SDP offer, but allows future SDP offers to change the keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Review: https://reviewboard.asteriskorg/r/1885/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	faec22add3	Update Pickup application documentation. (With feeling this time.) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	28fef5789b	Fix DTMF atxfer running h exten after the wrong bridge ends. ... When party B does an attended transfer of party A to party C, the attending bridge between party B and C should not be running an h exten when the bridge ends. Running an h exten now sets a softhangup flag to ensure that an AGI will run in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the attending bridge between party B and C. (closes issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	bbd95e031d	Add more constness to the end_buf pointer in the netconsole ... issue ASTERISK-18308 Review: https://reviewboard.asterisk.org/r/1876/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	c4ed0550e8	Fix reference leaks involving SIP Replaces transfers ... The reference held for SIP blind transfers using the Replaces header in an INVITE was never freed on success and also failed to be freed in some error conditions. This caused a file descriptor leak since the RTP structures in use at the time of the transfer were never freed. This reference leak and another relating to subscriptions in the same code path have now been corrected. (closes issue ASTERISK-19579) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	2ecce90e93	chan_sip: [general] maxforwards, not checked for a value greater than 255 ... The peer maxforwards is checked for both '< 1' and '> 255', but the default 'maxforwards' in the [general] section is only checked for '< 1' alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1888/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	d6ab0313c6	Update Pickup application documentation. (Even better) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	78b487007c	Update Pickup application documentation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1302d291c7	Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call. ... Some switches may not handle the call-deflection/call-rerouting message if the call is disconnected too soon after being sent. Asteisk was not waiting for any reply before disconnecting the call. * Added a 5 second delay before disconnecting the call to wait for a potential response if the peer does not disconnect first. (closes issue ASTERISK-19708) Reported by: mehdi Shirazi Patches: jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	74a9fcd6c4	Clear ISDN channel resetting state if the peer continues to use it. ... Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in response to a RESTART request. * Made the second SETUP received after sending a RESTART request clear the channel resetting state as if the peer had sent the expected RESTART ACKNOWLEDGE before continuing to process the SETUP. The peer may not be sending the expected RESTART ACKNOWLEDGE. (issue ASTERISK-19608) (issue AST-844) (issue AST-815) Patches: jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	262ee9fd02	Fix recalled party B feature flags for a failed DTMF atxfer. ... 1) B calls A with Dial option T 2) B DTMF atxfer to C 3) B hangs up 4) C does not answer 5) B is called back 6) B answers 7) B cannot initiate transfers anymore * Add dial features datastore to recalled party B channel that is a copy of the original party B channel's dial features datastore. * Extracted add_features_datastore() from add_features_datastores(). * Renamed struct ast_dial_features features_caller and features_callee members to my_features and peer_features respectively. These better names eliminate the need for some explanatory comments. * Simplified code accessing the struct ast_dial_features datastore. (closes issue ASTERISK-19383) Reported by: lgfsantos git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	0dbc13d013	Hangup affected channel in error paths of bridge_call_thread(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago