22116 Commits (1393623e9b50681b41fad99ecf87f19005ed8ed4)

 

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Author	SHA1	Message	Date
Jonathan Rose	1393623e9b	chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK ... Under certain conditions, a SIP transaction involving directmedia wouldn't trigger a re-invite because the SDP answer was included in an ACK instead of in a message that we would have triggered the invite with. This patch just queues a source change control frame if the dialog is using directmedia when we find sdp for an ACK. (closes issue AST-913) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	b0d337d2cb	Fix bug where final queue member would not be removed from memory. ... If a static queue had realtime members, then there could be a potential for those realtime members not to be properly deleted from memory. If the queue's members were loaded from realtime and then all the members were deleted from the backend, then the queue would still think these members existed. The reason was that there was a short- circuit in code such that if there were no members found in the backend, then the queue would not be updated to reflect this. Note that this only affected static queues with realtime members. Realtime queues with realtime members were unaffected by this issue. (closes issue ASTERISK-19793) reported by Marcus Haas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	01780bfda2	Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog destruction ... The other instance of this bug was fixed by jcolp/file in r121496. If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed. (closes issue ASTERISK-20119) Patch-by: Misha Vodsedalek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	af74988006	Add test instrumentation ... This adds test instrumentation for loading and unloading of modules and for certain actions in MeetMe to be used in the testsuite or any other consumer of AMI events. These will only be generated when Asterisk is built with TEST_FRAMEWORK enabled. (issue PQ-1131) (issue PQ-1133) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	4e0e1535c9	Fix problem where incorrect pointer was checked for nullity. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	53f102ed7c	Fix a couple of documentation problems in app_queue.c ... * The RemoveQueueMember app made mention of options that could be passed in, but no options are supported. I have removed the listing of options from the documentation. * The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value that could be set. (closes issue AST-949) reported by Steve Pitts (closes issue AST-954) reported by Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	eefc33dd34	remove ALREADYGONE flag on ooh323 call data by ooh323_indicate ... (CONGESTION/BUSY) due to call hasn't gone there really. This indication arrive from asterisk core not h.323 stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov Patches: ASTERISK-19308.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	c6d75bc2f5	Send re-register packets by GRQ (gatekeeper request) interval ... (close issue ASTERISK-20094) Patches: ASTERISK-20094-2.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	af738476ca	Use better libss7 detection test and move libpri compile test. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	74d2044626	Fix to resend GRQ/RRQ if RRJ (registration reject) is received ... (close issue ASTERISK-20094) Patches: ASTERISK-20094.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	ec6801aa0d	change opening h323 logfile with append mode instead of overwrite ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	483223b4d3	Correct documentation for the MeetMe x flag ... The documentation for the x flag for MeetMe incorrectly described its function as closing down the conference when the last marked user left. It actually causes the users with that flag to leave the conference when the last marked user exits. The functionality of this flag is not changing. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	38bbed178e	Fix Not Unreferencing A Spied Channel ... When a channel hangs up while being spied upon and the option to exit the ChanSpy application when the spied on channel hangs up is set, ast_autochan_destroy is not being called and therefore a reference to the spied upon channel is not removed. The symptom being reported was that when using func_group in the dialplan and calling "group show channels" at the cli, the spied upon channel was still being shown while "core show channels" showed that the channel was not up. This patch calls ast_autochan_destroy when a spied upon channel hangs up and the option to exit the ChanSpy application is set, removing the reference to the channel allowing the count for the group that the spied channel was part of to be decremented. (closes issue ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	13c82a2b30	Do not define a cause that doesn't actually exist ... AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause information. As such, it should not be defined and translatable as a cause. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b887a43667	Fix the analog dial *0 flash-hook of bridged peer feature. ... The flash-hook the bridged peer feature now correctly determines if the bridged peer is another chan_dahdi channel, that it is an analog channel, and that it has the correct signaling for an FXO port. It now also flash-hooks the correct channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	f051750307	Add missing AST_CAUSE_* -> text translations ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	7f3839ae03	Improve debug message for temporary outbound proxies. ... Thanks to Paul Belanger for pointing this out. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	9d402e02a5	Seriously? Another compilation error fixed. ... Somebody beat me. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	dca8a1b14e	Remove unused variable. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	1119a574ed	Fix error in the "IPorHost" section of a SIP dialstring. ... This is based on the review request posted by Walter Doekes (referenced lower in the commit message) The main fix here is to treat the IPorHost portion of the dial string as a temporary outbound proxy. This ensures requests get sent to the proper location. Due to the age of the request, some parts were no longer relevant. For instance, the request moved outbound proxy parsing code into a single method. This is done in a previous commit, so it was not necessary to do again. Also, the review request fixed some errors with regards to request routing for CANCEL and ACK requests. This has also been fixed in more recent commits. (closes issue ASTERISK-19677) reported by Walter Doekes Review https://reviewboard.asterisk.org/r/1859 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	6a198f0045	Revert alloca changes for utils ... These changes were a tad overzealous in the utils directory. Unfortunately, these don't compile with a "make". git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	f1cd5b6e93	Schedule pokes of registered SIP peers within a given timespan after SIP reload ... With a large number of SIP peers registered, performing a SIP reload causes a flood of SIP OPTIONS request packets. These are immediately sent out, and, as responses come back, can cause peers to be flagged as 'lagged' due to handling of the many response messages. This fix prevents this "packet storm" and schedules the pokes for a random time. That time varies between 1 ms and the peer's qualify time, or, if the qualify time is unknown, the global qualifyfreq setting. The committed patch has some very small modifications to the patch schmidts wrote for the review. (closes issue ASTERISK-19154) Reported by: Nicolo Mazzon patches: issue19154.patch license #6034 uploaded by schmidts Review: https://reviewboard.asterisk.org/r/1652 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	377caa7fb1	Clean up and ensure proper usage of alloca() ... This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	80efa31733	Help mitigate potential reinvite glare scenarios. ... When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. For those who are having some deja vu, that's because this patch was originally committed to trunk since there is a new configuration option added. After seeing a bug report about audio being slow to set up on SIP calls, it became apparent that this patch would be the best solution for resolving the issue. The patch is unintrusive and will have no effect unless the option is explicitly enabled. (closes issue AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857) reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	395b4b4898	Release B channel allocation on error path in chan_misdn. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	e0c0f9934d	res_agi: Add message indicating need for \n character in verbose message ... The while loop responsible for reading AGI messages from a fastAGI service can end up looping indefinitely when an AGI script fails to indicate the end of a message with a \n character. This patch adds an indication that we are expecting a \n character to end the message to make it more clear to users that this is necessary if they are receiving this warning over and over. (issue ASTERISK-20061) Reported by: Eike Kuiper git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	cb4bdd6531	Rewrite a comment that didn't adequately explain the code it was documenting. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	4f5d328eca	chan_oss: fix "sample rate" error message ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	3e75c69c3b	Improve documentation for the SHELL() dialplan function. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	aac6e33ba4	Free any datastores attached to dummy channels. ... Revision 370205 added the use of a datastore attached to a dummy channel to resolve a memory leak, but ast_dummy_channel_destructor() in this branch did not free datastores, resulting in a continued (but slightly smaller) memory leak. This patch backports the change to free said datastores from the Asterisk trunk. (related to issue AST-916) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ad8a86bef0	Fix compiler warnings. ... gcc (GCC) 4.2.4 has problems casting away constness. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	2660ebd91a	Fix compilation error when MALLOC_DEBUG is enabled ... To fix a memory leak in CEL, a channel datastore was introduced whose destruction function pointer was pointed to the ast_free macro. Without MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free. With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a different place then utils.h, and became undefined. This patch resolves this by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be ast_free, which is defined to be free. (issue AST-916) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	c545c76c87	Handle extremely out of order RFC 2833 DTMF ... The current implementation of RFC 2833 DTMF handling in res_rtp_asterisk will, if a packet arrives out of order, drop the packet. This is to prevent duplicate ton generation in the Asterisk core. Since the RTP layer does not buffer data itself, this is the only option the RTP layer currently has for handling packets that arrive out of order. For the most part, this doesn't matter. For a particular digit, so long as a BEGIN packet arrives before the first END packet, the digit will be produced. If subsequent BEGIN packets arrive interleaved with the ENDs, they will be dropped; likewise, if the BEGIN or END packets themselves are out of order, those packets are dropped but sufficient information is conveyed to the Asterisk core to produce the appropriate digit. For certain sequences of DTMF packets - most notably when, for a particular digit, an END packet arrives before any BEGIN packet for that digit - this is a real problem. When an END arrives before any BEGINs, the END packet is dropped - but at the same time, it causes subsequent BEGIN packets for that digit to be ignored. When the next in order END packet arrives, it too is dropped - Asterisk believes that there was no initial BEGIN. The solution this patch provides is to trust the END packet to convey the information needed for the Asterisk core to produce the DTMF digit. If we receive an END packet, and it: * Has a timestamp greater then the last timestamp received from an END packet * Does not have the same sequence number as the last received sequence number (and is thus not an END packet retransmission) Then we send the END frame up to the Asterisk core. It contains enough DTMF information for Asterisk to produce the digit. On the other hand, if we receive a BEGIN or continuation packet that occurs with a timestamp equal to or less then the last END timestamp, then we've received something out of order - but we already have received enough information to produce the digit. These packets are dropped. Much thanks goes to Olle Johansson (oej) for providing the idea for this solution. Review: https://reviewboard.asterisk.org/r/2033/ (closes issue ASTERISK-18404) Reported by: Stephane Chazelas Tested by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	c81a9be09f	Resolve severe memory leak in CEL logging modules. ... A customer reported a significant memory leak using Asterisk 1.8. They have tracked it down to ast_cel_fabricate_channel_from_event() in main/cel.c, which is called by both in-tree CEL logging modules (cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event that they log. The cause was an incorrect assumption about how data attached to an ast_channel would be handled when the channel is destroyed; the data is now stored in a datastore attached to the channel, which is destroyed along with the channel at the proper time. (closes issue AST-916) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2053/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	ecbaf1ee3f	Ensure that all ast_datastore_info structures are 'const'. ... While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	3d331187af	Code cleanup and bugfix in chan_sip outboundproxy parsing. ... The bug was clearing the global outboundproxy when a peer-specific outboundproxy was bad. The cleanup reduces duplicate code. Review: https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	7ae8833eeb	Add comments about the BUILD_NATIVE change ... This is a significant change and mention of it should have gone into UPGRADE.txt and CHANGES. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d8ff69cb1b	Add missing ast_hangup() calls on some analog exception paths. ... Make starting analog_ss_thread() or __analog_ss_thread() failure paths hangup the channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c77948d604	Include Expires header for SIP PUBLISH requests ... RFC3903 requres SIP PUBLISH requests to have Expires headers, so add them. Review: https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	bc4a0338e5	Prevent double uri_escaping in chan_sip when pedantic is enabled ... If pedantic mode is enabled, outbound invites will have double-escaped contacts. This avoids setting an already-escaped string into a field where it is expected to be unescaped. (closes issue ASTERISK-20023) Reported by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	84081cf8e5	Correct Documentation For DEC Function ... The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	be2a4c0d0d	Allow the REALTIME() function to report errors back to the caller. ... Also, do more error checking on the arguments specified to the REALTIME() function and clarify the documentation. While I was editing the file, a few coding guidelines fixups, as well. Review: https://reviewboard.asterisk.org/r/2031/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	913225a79a	Improve Goto and GotoIf related documentation ... Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	66f69a53c3	Add Digium phones context to sip_notify sample config. ... This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	6b24019080	chan_sip: Fix small behavioral change accidentally introduced in r369750 ... When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed the return value, which would likely make the indication not be sent in audio. This fixes that while still removing the warning message. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	a611188a8d	chan_sip: Add case for FLASH control frames so that we don't display a warning. ... chan_sip channels can receive flash control frames when connected to analog phones and possibly for other reasons. There really isn't a reason to warn when these frames are received, we can safely ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	a35a2ec1b5	Remove a superfluous and dangerous freeing of an SSL_CTX. ... The problem here is that multiple server sessions share a SSL_CTX. When one session ended, the SSL_CTX would be freed and set NULL, leaving the other sessions unable to function. The code being removed is superfluous because the SSL_CTX structures for servers will be properly freed when ast_ssl_teardown is called. (closes issue ASTERISK-20074) Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license #5049) Testers: Trevor Helmsley git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	9a4ee83043	Fix bridging thread leak. ... The bridge thread was exiting but was never being reaped using pthread_join(). This has been fixed now by calling pthread_join() in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	8339cc9684	AST-2012-011: Resolve heap corruption issue with voicemail ... The heard and deleted arrays in the voicemail state structure were not handled properly following the memory leak fix in r354890 and a fix for an invalid free in r356797. This could result in accessing and writing into freed memory. The allocation for these arrays has been reworked to avoid the possibility of invalid frees, access of freed memory, and crashes that were occurring as a result of this. Locking around accesses and modifications of the voicemail state structure members dh_arraysize, heard, and deleted has been added to prevent simultaneous modification and access when IMAP storage is in use. If IMAP storage is not in use, this locking is not compiled in. Review: https://reviewboard.asterisk.org/r/1994/ (closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license 6273) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	48667d72f1	Do not send a BYE when a provisional response arrives during a re-INVITE ... Commits r369557 and r369579 were done to improve handling of re-INVITEs when the UA that was supposed to receive the re-INVITE fails to respond. A limitation of those patches occurred when a UA sent a provisional response to the re-INVITE. This triggered a sending of a BYE in check_pending. This patch tweaks the handling of the re-INVITE such that a BYE is not sent in response to those messages. (issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies patches: (reinvite_tweak.diff license #5012 by Steve Davies) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago