21925 Commits (bcd63be3cdc21f6733411863685c08c53ca9bf0c)

 

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Author	SHA1	Message	Date
Jonathan Rose	bcd63be3cd	Make use of va_args more appropriate to form in various res_config modules plus utils. ... A number of va_copy operations weren't matched with a corresponding va_end in res_config_odbc. Also, there was a potential for va_end to be invoked twice on the same va_arg in utils, which would mean invoking va_end on an undefined variable... which is bad. va_end is removed from various functions in config_pgsql and config_curl since they aren't making their own copy. The invokers of those functions are responsible for calling va_end on them. (issue ASTERISK-19451) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1848/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8cbe9f9fa7	Fix error that caused seek format operations to set max file size to '1' or '0' ... A very inappropriate placement of a ')' (introduced in r362151) caused the maximum size of a file to be set as the result of a comparison operation, as opposed to the result of the ftello operation. This resulted in seeking being restricted to the beginning of the file, or 1 byte into the file. Thanks to the Asterisk Test Suite for properly freaking out about this on at least one test. (issue ASTERISK-19655) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	d84e70a95c	Turn off warning message when bind address is set to any. ... When a bind address is set to an ANY address (udpbindport=::), a warning message is displayed stating that "Address remapping activated in sip.conf but we're using IPv6, which doesn't need it. Please remove 'localnet' and/or 'externaddr' settings." But if one is running dual stack, we shouldn't be told to turn those settings off. This patch checks if the bind address is an ANY address or not. The warning message will now only be displayed if the bind address is NOT an ANY address and IPv6 is being used. Also, updated the copyright year. (closes issue ASTERISK-19456) Reported by: Michael L. Young Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7a72dc706a	Fix negative return handling in channel drivers ... In chan_agent, while handling a channel indicate, the agent channel driver must obtain a lock on both the agent channel, as well as the channel the agent channel is using. To do so, it attempts to lock the other channel first, then unlock the agent channel which is locked prior to entry into the indicate handler. If this unlock fails with a negative return value, which can occur if the object passed to agent_indicate is an invalid ao2 object or is NULL, the return value is passed directly to strerror, which can only accept positive integer values. In chan_dahdi, the return value of dahdi_get_index is used to directly index into the sub-channel array. If dahd_get_index returns a negative value, it would use that value to index into the array, which could cause an invalid memory access. If dahdi_get_index returns a negative number, we now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	0da3b6c793	Fix handling of negative return code when storing voicemails in ODBC storage ... When storing a voicemail message using an ODBC connection to a database, the voicemail message is first stored on disk. The sound file associated with the message is read into memory before being transmitted to the database. When this occurs, a failure in the C library's lseek function would cause a negative value to be passed to the mmap as the size of the memory map to create. This would almost certainly cause the creation of the memory map to fail, resulting in the message being lost. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b3a38a51c7	Check for IO stream failures in various format's truncate/seek operations ... For the formats that support seek and/or truncate operations, many of the C library calls used to determine or set the current position indicator in the file stream were not being checked. In some situations, if an error occurred, a negative value would be returned from the library call. This could then be interpreted inappropriately as positional data. This patch checks the return values from these library calls before using them in subsequent operations. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	6afefc4eb1	Make ForkCDR e option not set end time of the newly forked CDR log ... Prior to this patch, ForkCDR's e option would immediately set the end time of the forked CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time being roughly the same as it's beginning time (which is in turn roughly the same as the original's end time). (closes issue ASTERISK-19164) Reported by: Steve Davies Patches: cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	680b627906	Send relative path named recordings to the meetme directory instead of sounds ... Prior to this patch, no effort was made to parse the path name to determine a proper destination for recordings of MeetMe's r option. This fixes that. Review: https://reviewboard.asterisk.org/r/1846/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	7a33e9ba9d	Make trunkfreq take effect when set ... Previously, setting trunkfreq had no effect on initial load or on reload and only ever used the default value. This causes trunkfreq to be used appropriately on initial load and reload. (closes issue ASTERISK-19521) Patch-by: Jaco Kroon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	3bb065e39a	Simplify build system architecture optimization ... This change to the build system rips out any usage of PROC along with architecture-specific optimizations in favor of using -march=native where it is supported. This fixes broken builds on 64bit Intel systems and results in better optimized code on systems running GCC 4.2+. Review: https://reviewboard.asterisk.org/r/1852/ (closes issue ASTERISK-19462) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b856533029	Prevent invalid access of free'd memory if DAHDI channel during an MWI event ... In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. * Rework the -r361705 patch to better manage the cs and mtd allocated resources. * Fixed use of mwimonitoractive flag to be correct if the mwi_thread() fails to start. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	40beb62845	Fix crash caused by unloading or reloading of res_http_post ... When unlinking itself from the registered HTTP URIs, res_http_post could inadvertently free all URIs registered with the HTTP server. This patch modifies the unregister method to only free the URI that is actually being unregistered, as opposed to all of them. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c359eeb9c6	Allow func_curl to exit gracefully if list allocation fails during write ... If the global_curl_info data structure could not be allocated, the datastore associated with the operation would be free'd, but the function would not return. This would later dereference the datastore, almost certainly causing Asterisk to crash. With this patch, if the data structure is not allocated the method will return an error code, and not attempt any further operation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361753 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	246ad9bf0d	Prevent invalid access of free'd memory if DAHDI channel during an MWI event ... In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. This patch makes it so that we only free the caller ID structure if a DAHDI channel is successfully created, and we bump the gains back up if we fail to make a DAHDI channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	41f3d27d20	Change SHARED function to use a safe traversal when modifying a variable ... When the SHARED function modifies a variable, it removes it from its list of variables and reinserts the new value at the head of the list of variables. Doing this inside a standard list traversal can be dangerous, as the standard list traversal does not account for the list being changed. While the code in question should not cause a use after free violation due to its breaking out of the loop after freeing the variable, it could lead to a maintenance issue if the loop was modified. This also fixes a violation reported by a static analysis tool, which also makes this code easier to maintain in the future. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	1d3ba1f2a7	Fix memory leak in res_calendar_ews when event email address node is empty ... If the XML calendar data returned by a Microsoft Exchange Web Service specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address is provided, a condition existed where an ast_calendar_attendee struct would be allocated but not appended to the list of attendees. Because of that, the memory associated with the attendee would never be freed. This patch frees the memory if no e-mail address is provided. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	304af5d7cc	Fix memory leak when using MeetMeAdmin 'e' option with user specified ... A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command (eject last user that joined) is used in conjunction with a specified user. Regardless of the command being executed, if a user is specified for the command, MeetMeAdmin will look up that user. Because the 'e' option kicks the last user that joined, as opposed to the one specified, the reference to the user specified by the command would be leaked when the user variable was assigned to the last user that joined. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	4148e51555	Add missing newlines to CLI logging ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	be62cac9ee	Fix typo in svn:keywords ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	d521fa4af3	Fix typo in svn:keywords ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	5fcb78840f	Remove a few more files related to chan_usbradio and app_rpt. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5c318b19c2	Fix a typo in the warning messages for an ignored media stream ... Added a '\n' to the warning messages when we ignore a media stream due to the port number being '0'. (closes issue ASTERISK-19646) Reported by: Badalian Vyacheslav git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	ab87db4419	Remove unnecessary error message in app_dial.c ... The error message for failure to stop autoservice after a gosub or macro call during a dial was removed for macro while Asterisk 1.4 was still being actively developed. The corresponding gosub error message was never removed. (closes issue ASTERISK-19551) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	3560b4a9b5	Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined ... There were a few instances of restarting music on hold in meetme that would cause Asterisk to revert to the default class of music on hold for no adequate reason. Review: https://reviewboard.asterisk.org/r/1844/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	78e9a8e5f0	Fix some stuff involving calls to memcpy and memset ... The important parts of the patch were already applied through other updates. (closes issue ASTERISK-19445) Reported by: Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto Dei (license 5027) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	04756101e0	Make 'help devstate change' display properly (get rid of excess comma) ... (closes issue ASTERISK-19444) Reported by: Makoto Dei Patches: devstate-change-usage-truncate.patch uploaded by Makoto Dei (license 5027) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	ed76cdda72	Replace GNU old-style field designator extensions to fix clang warnings ... (issue ASTERISK-19540) Reported by: Makoto Dei Patches: clang-gnu-designator.patch uploaded by Makoto Dei (license 5027) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	abd7fa4476	Make the MeetMeAdmin N command (mute all nonadmins) not mute admins ... (Closes Issue ASTERISK-19335) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b1152d145f	Fix the display of documentation for Transfer ... This came up while fixing documentation generation for many other cases where the argument separator was not being displayed properly. Now that it is displayed properly, it shows up in the wrong place for Transfer since the '/' is only required if Tech is present. (related to issue ASTERISK-18168) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361040 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	063aa93c46	Stop sending out RTCP if RTP is inactive ... This change prevents Asterisk from sending RTCP receiver reports during a remote bridge since it is no longer receiving media and should not be reporting anything. (related to ASTERISK-19366) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1df90a2cf1	Fix logger deadlock on Asterisk shutdown. ... The logger_thread() had an exit path that failed to release the logmsgs list lock. * Make logger_thread() exit path unlock the logmsgs list lock. * Made ast_log() not queue any messages to the logmsgs list if the close_logger_thread flag is set. (issue ASTERISK-19463) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	82fbd4a401	Fix potential race condition during call pickup. ... Prior to this patch, a connected line update was queued during call pickup and then an answer frame was queued. The original caller would presumably then have his connected line updated and then the call would be answered. In actuality, the answer frame was not how the call ended up being answered. Rather, an odd section in app_dial that checks if the called channel's state is up. The result is that the order of the connected line update and the answer were variable. In most cases, this wasn't actually a bad thing. However, if the 'I' option was passed to dial, the connected line update would be inhibited. The fix is to queued the connected line after the answer frame is queued. This way the race in app_dial is between two conditions resulting in an answer. This way the connected line update occurs after the answer every time. (closes issue ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by Mark Michelson (license 5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	58565827e6	Improve accuracy of identifying information sent in dialog-info SIP NOTIFY requests. ... This change makes use of connected party information in addition to caller ID in order to populate local and remote XML elements in the dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by: Maciej Krajewski Tested by: Maciej Krajewski Patches: local_remote_hint2.diff uploaded by Mark Michelson (license 5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	d58fe85724	Destroy configs when they are no longer used ... https://reviewboard.asterisk.org/r/1834/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	390e8c1f73	Make a debug message regarding subscription changes more accurate. ... I was getting confused during some testing why Asterisk was saying that a subscription was being added when it was clearly being removed. This fixes that confusion. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	fe267f5925	Updates config with bootstrap where I changed configure.ac in r360488 ... (issue ASTERISK-17842) Reported by: Bryon Clark git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	aa74e83c19	Fix BETTER_BACKTRACES library detection for Fedora/RedHat/CentOS ... (closes ASTERISK-17842) Reported by: Bryon Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch uploaded by Bryon Clark (license 6157) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	165789485e	Update CHANGES for r360471 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360474 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	bcf902419e	Fix Asterisk version typo ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	df02086396	Increase verbosity level for ast_verb messages ... While this does not fix the issue of the CLI being flooded by 'doing dnsmgr_lookup' messages, increasing the verbosity level above 5 should help minimize it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	e068e29190	func_curl: Fix leak of an ast_str in error handling code path. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	dea36a06bb	app_page: Fix a memory leak on every Page(). ... dial_list is a dynamically allocated array that is allocated at the beginning of Page() based on how many devices will be dialed. This was never being freed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	093121050f	app_jack: fix datastore memory leak in error handling path. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	2b2c5f27fc	Rebuild parsers. ... This is needed to include the last fix to main/ast_expr2.y. The changes look much bigger as this regeneration of the code was done with newer versions of flex and bison. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	17cd5abb22	expression parser: Fix (theoretical) memory leak. ... Fix a memory leak that is very unlikely to actually happen. If a malloc() succeeded, but the following strdup() failed, the memory from the original malloc() would be leaked. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c04afdbc70	Make number not available presentation also set screening to network provided. ... Q.951 indicates that when the presentation indicator is "Number not available due to interworking" for a number then the screening indicator field should be "Network provided". * Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE when the presentation is "Number not available due to interworking". This fix makes Asterisk consistent and it also makes it consistent with earlier branches as far as this presentation value is concerned. * Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions handle the "Number not available due to interworking" case better in sig_pri.c. This change is possible because the minimum required libpri version (v1.4.11) has the necessary defines in libpri.h. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f9246f83c9	Add missing initialization of update_redirecting in chan_sip.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	94b6251ea2	Update install_prereq script to include missing GSM library for debian amd move SQLite3. ... (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches: debian_install_prereq.diff uploaded by Andrew Latham (license 5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tzafrir Cohen	00dc354511	Also detect gmime 2.6 ... Also detect gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	cfdc12387b	Ensure Asterisk sends a BYE when pending on the final response to a re-INVITE ... When Asterisk detects a hangup and cannot send a BYE due to a pending INVITE, it sets the pendingbye flag and waits for the final response to that INVITE. When the response is received, it transmits the BYE. If, however, that INVITE request is a pending re-INVITE, it needs to first send a CANCEL request to terminate the pending re-INVITE. In that circumstance, Asterisk was, in some scenarios, clearing the pendingbye flag after processing the CANCEL request and not checking for a pending BYE when receiving the final 487 response to the INVITE. This patch ensures that if the pendingbye flag is set, it is honored regardless of the nature of the INVITE request currently in flight. (closes issue ASTERISK-19365) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283) Review: https://reviewboard.asterisk.org/r/1807 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago