22093 Commits (80efa3173315ba47b794e5c9f1b56f61a663a76c)

 

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Author	SHA1	Message	Date
Alexandr Anikin	1ab70391f8	check rtptimeouts in ooh323 channels as per config file ... (rtp voice, video, udptl except rtcp) (closes issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches: 19179-ooh323-2.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	5e5564b041	Fix request routing issue when outboundproxy is used. ... Asterisk was incorrectly setting the destination of CANCELs and ACKs for error responses to the URI of the initial INVITE. This resulted in further requests, such as INVITEs with authentication credentials, to be routed incorrectly. Instead, when these CANCEL or ACKs are to be sent, we should simply keep the destination the same as what it previously was. There is no need to alter it any. (closes issue ASTERISK-20008) Reported by Marcus Hunger Patches: ASTERISK-20008.patch uploaded by Mark Michelson (license #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	239470bc65	Fix monitoring calls put in a parking lot. ... * Fix a regression that was introduced by -r366167 which effectively disabled monitoring parked calls. (closes issue ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	977a9791be	Add a script to enable finding source files without support-levels defined. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	f83d1b98e8	Add support-level indications to many more source files. ... Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	6738e63081	Revert Makefile change to remove embedding res_adsi.so ... The change has resulted in a linking error for certain versions of GCC. This is much worse than the original issue, so for now, temporarily revert the change. A more thorough change will be sought out. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a64586e5c2	Fix a deadlock that occurs when func_volume is used on a local channel. ... This was discovered by trying to perform a call forward to an extension that makes use of func_volume. When the local channel is optimized away, the datastore on the local;2 channel would have its audiohook destroyed rather than detaching the audiohook from the channel and then destroying it. With this patch, func_volume's datastore destructor takes the proper route of detaching the audiohook and then destroying it. (closes issue ASTERISK-19611) reported by Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark Michelson (license #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	89ff2d1ed7	Mark res_smdi/res_adsi as 'core' supported modules ... Recently, various issues surrounding weak symbols have caused problems with modules that rely on that feature to be enabled in menuselect. This includes app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi, which, in certain circumstances, may not be enabled by default in menuselect. Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this patch marks both as 'core' supported modules. This will allow both app_voicemail and chan_dahdi to be enabled as well, regardless of whether or not that system supports weak symbols. (issue AST-900) Reported by: Thomas Arimont (issue AST-885) Reported by: Denis Alberto Martinez git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	b680746454	Remove forced linking of res_adsi.o ... In GCC 4.5+ the result is that Asterisk has a phantom module loaded at startup, claiming to be res_adsi. (closes issue ASTERISK-19920) reported by Leif Madsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	14b242451b	Do not install empty directories; add ASTLIBDIR ... r368830 modified the installation script to only create a directory if that directory does not exist. If some directory variable was empty, it would attempt to create the empty location. It also failed to create the ASTLIBDIR directory. This patch fixes it such that the correct directories are made and only created if a value specifying them actually exists. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	1a79e238a3	Do not perform install on existing directories ... If a directory already exists, performing a 'make install' will remove the permissions associated with the current directory and replace them with the permissions of the user executing the install. This patch changes this behavior to only perform an install on the directory if the directory does not exist. Thus, if a user later changes the permissions on that directory, those permissions will be preserved in subsequent installs. Review: https://reviewboard.asterisk.org/r/1986 Review: https://reviewboard.asterisk.org/r/1864 (closes issue ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger, Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded by mjordan) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	5919aa6cf4	Set the Caller ID "tag" on peers even if remote party information is present. ... On incoming calls, we were setting the cid_tag on the dialog only if there was no remote party information (Remote-Party-ID or P-Asserted-Identity) present. The Caller ID tag is an invented parameter, though, and should be set no matter the circumstance. (closes issue ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884) Reported by Trey Blancher git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	3863d34cdf	Fix deadlock potential with ast_set_hangupsource() calls. ... Calling ast_set_hangupsource() with the channel lock held can result in a deadlock because the function also locks the bridged channel. (issue ASTERISK-19537) (closes issue ASTERISK-19801) Reported by: Alec Davis (closes issue AST-891) Reported by: Guenther Kelleter Tested by: Guenther Kelleter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	0353a57671	Fix coverity UNUSED_VALUE findings in core support level files ... Most of these were just saving returned values without using them and in some cases the variable being saved to could be removed as well. (issue ASTERISK-19672) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	44e963a550	Fix compilation in dev-mode ... Backport a compilation fix in md5.c from trunk that only showed up in dev-mode under certain compiler versions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	aa8ed90f30	Fix POTS flash hook to orignate a second call deadlock. ... A deadlock can occur when a POTS phone tries to flash hook to originate a second call for 3-way or transfer. If another process is scanning the channels container when the POTS line flash hooks then a deadlock will occur. * Release the channel and private locks when creating a new channel as a result of a flash hook. (closes issue ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	e52f2967de	Fix a specific scenario where ACKs are not matched. ... If a dialog-starting INVITE contains a to-tag, then Asterisk will respond with a 481. In this case, the resulting incoming ACK would not be matched, so Asterisk would continue retransmitting the 481 until the transaction times out. There were two issues. Asterisk, upon creating a sip_pvt would generate a local tag. However, when the time came to transmit the 481, since there was a to-tag in the INVITE, Asterisk would place this original to-tag in the 481 response. When the ACK came in, Asterisk would attempt to match the to-tag in the ACK to the generated local tag. Unfortunately, Asterisk never actually transmitted a response with the generated local tag, so the to-tag in the ACK would not match. The other problem was that when the 481 was sent, nothing was set on the sip_pvt to indicate what CSeq is expected in the ACK. To fix the first problem, we zero out the to-tag seen in the incoming INVITE. This way, Asterisk, when time to send a response, will send its generated local tag instead. To fix the second problem, we set the sip_pvt's pendinginvite to the CSeq of the INVITE when we send a 481. (closes issue ASTERISK-19892) Reported by Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8a0f2ec66a	Add feature modifier to versions produced from branches ... Certain branches, such as Certified Asterisk, may have a modifier added to them that specifies the features available in that branch. For branches, this modifier is expected to be reflected in the location of the branch in subversion. For example, a subversion of URL of /certified/branches/1.8.11 would have a feature modifier of 'certified'. This is slightly different then how features are determined for tags, where the feature is part of the actual tag name, e.g., "10.5.0-digiumphones". In keeping with the nomenclature used for tags, the feature specifier for branches is translated and placed after the revision numbers. For the example given previously, this would result in a branch version of "Asterisk SVN-branch-1.8.11-cert-rXXXXXX". git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b563128877	Ensure overlapping hold flags do not conflict ... When changing between different modes of hold, the flags were not being cleared out properly causing a failure to change hold states. (closes issue ASTERISK-19919) Patch-by: Morten Tryfoss Reported-by: Morten Tryfoss git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	eae448bc55	Fix parked call performing a DTMF blind transfer after being retrieved. ... When a parked call was retrieved from the parking lot, it could not do a blind transfer because it caused the involved calls to be hung up unconditionally. * Made the ParkedCall application return the ast_bridge_call() return value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	ca256c003b	Resolve some build warnings ... My newly upgraded compiler caught these usages of uninitialized values. They weren't actually used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	6c63f45326	Ensure that pages and emails are sent using RFC822-compliant date format ... When localization was added to app_voicemail, these headers were altered when they should have remained in en_US format for RFC compliance. This reverts the changes to those two lines. (closes issue ASTERISK-19876) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	503fc9458a	Relay proper SIP responses on calling side. ... Revision 351130 broke corect HANGUPCAUSE setting for the 404 case in chan_sip. Other cases were also potentially broken. This patch fixes the relaying of causes to be what they used to be. (closes issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter Doekes (via a reviewboard test to be committed later) Patches: chan_sip.diff uploaded by Pavel Troller (license #6302) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c44df31f10	Document BLINDTRANSFER behavior change. ... (issue ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	0db9327800	Fix potential deadlock between masquerade and chan_local. ... * Restructure ast_do_masquerade() to not hold channel locks while it calls ast_indicate(). * Simplify many calls to ast_do_masquerade() since it will never return a failure now. If it does fail internally because a channel driver callback operation failed, the only thing ast_do_masquerade() can do is generate a warning message about strange things may happen and press on. * Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This change fixes half of the deadlock reported in ASTERISK-19801 between masquerades and chan_iax. (closes issue ASTERISK-19537) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1915/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	81e1b36378	Fix deadlock when Gosub used with alternate dialplan switches. ... Attempting to remove a channel from autoservice with the channel lock held will result in deadlock. * Restructured gosub_exec() to not call ast_parseable_goto() and ast_exists_extension() with the channel lock held. (closes issue ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	871cef109a	Improve SDP parsing warning messages ... * 'Unsupported media type' is only reported when that is in fact the case, not when a supported media type is included in an 'm' line that has an invalid format. * All warning messages related to parsing 'm' lines now include the 'm' line contents. * (minor bugfix) newline added to port-number-zero warning messages. * Warning messages improved to use RFC-specified terminology for various items. * Warnings for offers that include more than one port for a single media type now include the media type. Review: https://reviewboard.asterisk.org/r/1811/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	751a8bb6b3	Add documentation to function CHANNEL for options echocan_mode and buffers ... The ability to set "echocan_mode" and "buffers" through the dialplan was added to chan_dahdi some time ago. This patch adds some documentation to func_channel. (Closes issue ASTERISK-19911) Reported by: Dale Noll Tested by: Michael L. Young Patches: asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1949/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	bd85d458a2	Coverity Report: Fix issues for error type REVERSE_INULL (core modules) ... * Fixes findings: 0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f7d2d4601b	Use the DEADLOCK_AVOIDANCE() macro instead. ... (issue ASTERISK-19854) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	0bad4cbdde	Fix deadlock when executing CLI "pri show channels" and "ss7 show channels" commands. ... * Fix sig_pri_lock_owner() to avoid deadlock properly. * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid deadlock properly. * Code ss7_grab() better. (closes issue ASTERISK-19854) Reported by: Jaxon Patches: jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7) Tested by: Jaxon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ba26e7e249	Coverity Report: Fix issues for error type REVERSE_INULL (deprecated modules) ... * Fix only issue pointed out by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user(). * Change use of %i to %d in sscanf() in find_user(). The use of %i gives unexpected parsing because it can accept hex, octal, and decimal integer formats. * Changed other uses of %i in app_meetme() to use %d for consistency. (issue ASTERISK-19648) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	fd9f2351cd	AST-2012-008: Fix remote crash vulnerability in chan_skinny ... When a skinny session is unregistered, the corresponding device pointer is set to NULL in the channel private data. If the client was not in the on-hook state at the time the connection was closed, the device pointer can later be dereferenced if a message or channel event attempts to use a line's pointer to said device. The patches prevent this from occurring by checking the line's pointer in message handlers and channel callbacks that can fire after an unregistration attempt. (closes issue ASTERISK-19905) Reported by: Christoph Hebeisen Tested by: mjordan, Damien Wedhorn Patches: AST-2012-008-1.8.diff uploaded by mjordan (license 6283) AST-2012-008-10.diff uploaded by mjordan (license 6283) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	be10342c3f	AST-2012-007: Fix IAX receiving HOLD without suggested MOH class crash. ... * Made schedule_delivery() set the received frame f->data.ptr to NULL if the datalen is zero. * Fix queue_signalling() memcpy() size error. * Made queue_signalling() not use C++ keyword variable names. (closes issue ASTERISK-19597) Reported by: mgrobecker Patches: jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Michael L. Young git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	20b362fa4b	Fix pvt_sip for inbound call to use peer's allowtransfer setting ... The pvt_sip allowtransfer was not being set to that of the peer's setting. Therefore, the global allowtransfer setting was being used instead which would lead to calls not being transfered if the global setting was set to 'no' despite the setting on the peer being 'yes' and vice versa, calls would be allowed to transfer even if the peer's setting was 'no' but the global setting was 'yes'. (Closes issue ASTERISK-19856) Reported by: Jacek Tested by: Michael L. Young, Jacek Patches: issue-asterisk-19856-branch10-v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1923/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	4b03161fca	Fix Dial I option ignored if dial forked and one fork redirects. ... The Dial and Queue I option is intended to block connected line updates and redirecting updates. However, it is a feature that when a call is locally redirected, the I option is disabled if the redirected call runs as a local channel so the administrator can have an opportunity to setup new connected line information. Unfortunately, the Dial and Queue I option is disabled for *all* forked calls if one of those calls is redirected. * Make the Dial and Queue I option apply to each outgoing call leg independently. Now if one outgoing call leg is locally redirected, the other outgoing calls are not affected. * Made Dial not pass any redirecting updates when forking calls. Redirecting updates do not make sense for this scenario. * Made Queue not pass any redirecting updates when using the ringall strategy. Redirecting updates do not make sense for this scenario. * Fixed deadlock potential with chan_local when Dial and Queue send redirecting updates for a local redirect. * Converted the Queue stillgoing flag to a boolean bitfield. (closes issue ASTERISK-19511) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1920/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c88ad5e919	Fix WaitExten(x,m(musicclass)) string termination. ... The AST_CONTROL_HOLD MOH class from the WaitExten application can now be queued onto a channel, passed over local channels with the /m option, and passed over IAX channels. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	e456e4a8f1	Only call SSL_CTX_free if DO_SSL is defined. ... Thanks to Paul Belanger for pointing out this error. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f26d22b563	Update a peer's LastMsgsSent when the peer is notified of waiting messages ... Previously, MWI logic utilized a counter called 'lastmsgssent' to know whether or not MWI NOTIFY requests had been sent to a specific peer. When MWI notifications were changed to use the internal event framework, this value was no longer needed for its original purpose. Hence, it was no longer updated with the new/old message counts for a peer. However, the value was still presented when, either by AMI or CLI, a 'sip show peer [peer]' command was executed. The output of the command would always display the erroneous value of 32767/65535 for 'LastMsgsSent'. This patch makes it so that the value of lastmsgssent is updated appropriately. The value should now display the new/old message counts for a particular peer. (closes issue ASTERISK-17866) Reported by: Steve Davies patches by: ast-17866-rb1272.patch (License #5041 by irroot) Modified slightly for this commit Review: https://reviewboard.asterisk.org/r/1939 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	9753325d32	Fix race condition for CEL LINKEDID_END event ... This patch fixes to situations that could cause the CEL LINKEDID_END event to be missed. 1) During a core stop gracefully, modules are unloaded when ast_active_channels == 0. The LINKDEDID_END event fires during the channel destructor. This means that occasionally, the cel_* module will be unloaded before the channel is destroyed. It seemed generally useful to wait until the refcount of all channels == 0 before unloading, so I added a channel counter and used it in the shutdown code. 2) During a masquerade, ast_channel_change_linkedid is called. It calls ast_cel_check_retire_linkedid which unrefs the linkedid in the linkedids container in cel.c. It didn't ref the new linkedid. Now it does. Review: https://reviewboard.asterisk.org/r/1900/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	c191395381	Resolve crash in subscribing for MWI notifications ... ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the variable should definitely not be used after that. To solve this in the two cases that affect subscribing for MWI notifications, we instead save the ref locally, and unref them in the error conditions. (closes issue ASTERISK-19827) Reported by: B. R Review: https://reviewboard.asterisk.org/r/1940/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	3ddfb50fe4	Address MISSING_BREAK static analysis reports some more. ... This addresses core findings 4 and 6. Moises Silva helped me by stating that a break could be safely added to the case where it is added in chan_dahdi.c In say.c, I have added a comment indicating that static analysis complains but that it is currently unknown if this is correct. This fixes all core findings of this type. (closes issue ASTERISK-19662) reported by Matthew Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	eef4c09787	Fix memory leak of SSL_CTX structures in TLS core. ... SSL_CTX structures were allocated but never freed. This was a bigger issue for clients than servers since new SSL_CTX structures could be allocated for each connection. Servers, on the other hand, typically set up a single SSL_CTX for their lifetime. This is solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an ssl_ctx on it, it is freed so that a new one can take its place. 2. A companion to ast_ssl_setup() called ast_ssl_teardown() has been added so that servers can properly free their SSL_CTXs. (issue ASTERISK-19278) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	67268d9198	Fix more memory leaks ... This patch adds to what was fixed in r366880. Specifically, it addresses the following: * chan_sip: dispose of an allocated frame in off nominal code paths in sip_rtp_read * func_odbc: when disposing of an allocated resultset, ensure that any rows that were appended to that resultset are also disposed of * cli: free the created return string buffer in another off nominal code path (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	3bdb733c5d	Reorder and renumber tests appropriately ... It appears that a patch did not apply properly when adding tests 12 and 13 and test 11 was duplicated. These tests have been reordered and renumbered such that they make sense. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3edf245601	Fix a variety of memory leaks ... This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	a7733c579b	chan_sip: Fix missed locking of opposing pvt for directmedia acl from r366547 ... It also required deadlock avoidance since two sip_pvts structs needed to be locked simultaneously. Trunk handles it differently, so this is a 1.8 and 10 patch only. (issue AST-876) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7eb528fbf1	Fix checking bounds of array index after using it; improper sizeof ... This patch fixes two problems pointed out by a static analysis tool. * In chan_dahdi, when an event is handled the index of the sub channel is first obtained. In very off nominal cases, the method that determines the index can return a negative value. In the event handling code, whether or not the index returned is valid was being checked after that value was used to index into an array. This patch makes it so the value is checked before any indexing is done. * In res_calendar_ews, sizeof was being passed a pointer instead of the struct to determine the amount of memory to allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes issue ASTERISK-19671) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	82a5bbcece	Fix incorrect default port number for HTTP server. ... Thanks to Tzafrir Cohen for bringing this up on the Asterisk developers mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366650 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	5e7f34fa05	Correct misuse of ast_strip_quoted() when getting a Diversion header's reason parameter. ... The use here was assuming that the pointer would be updated, but the updated string is actually returned by ast_strip_quoted() instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago