asterisk

Commit Graph

Author	SHA1	Message	Date
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
Tilghman Lesher	233b8364d3	On some platforms, O_RDONLY is not a flag to be checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX specification does not mandate how these 3 flags must be specified, only that one of the three must be specified in every call. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	7c6c99c317	AST-2012-004: Fix an error that allows AMI users to run shell commands sans authorization. As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI actions were able to run system commands by going through other AMI commands which did not require that authorization. Specifically, GetVar and Status allowed users to do this by setting their variable/s options to the SHELL or EVAL functions. Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that allowed users with originate permission to run MixMonitor and supply a shell command in the Data argument. That flaw is fixed in those versions of this patch. (closes issue ASTERISK-17465) Reported By: David Woolley Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) ........ Merged revisions 363117 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	88f80c1d54	AST-2012-006: Fix crash in UPDATE handling when no channel owner exists If Asterisk receives a SIP UPDATE request after a call has been terminated and the channel has been destroyed but before the SIP dialog has been destroyed, a condition exists where a connected line update would be attempted on a non-existing channel. This would cause Asterisk to crash. The patch resolves this by first ensuring that the SIP dialog has an owning channel before attempting a connected line update. If an UPDATE request is received and no channel is associated with the dialog, a 481 response is sent. (closes issue ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license 6283) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	9a3120c0c8	AST-2012-005: Fix remotely exploitable heap overflow in keypad button handling When handling a keypad button message event, the received digit is placed into a fixed length buffer that acts as a queue. When a new message event is received, the length of that buffer is not checked before placing the new digit on the end of the queue. The situation exists where sufficient keypad button message events would occur that would cause the buffer to be overrun. This patch explicitly checks that there is sufficient room in the buffer before appending a new digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant ........ Merged revisions 363100 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@363102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	93304431a3	Update app_dial M and U option GOTO return value documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	fe7f595e9b	OpenBSD doesn't have rawmemchr, use strchr (closes issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller (license 5434) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	e07ff6ed84	Document Speech* apps hangup on failure and suggest TryExec The Speech API apps return -1 on failure, which will hang up the channel. This may not be desirable behavior for some, but it isn't something that can be changed without breaking people's dialplans or writing an option to all of the Speech apps that does what TryExec already does. This patch documents the hangup behavior of the apps, and suggests TryExec as the solution. (closes issue AST-813) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362815 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	a59edad230	Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	e49fcbcb04	Add leading and trailing backslashes A couple of unit tests did not have have leading or trailing backslashes when setting their test category resulting in a warning message being displayed. Added the backslash where needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f1bb4eea3d	Update membermacro and membergosub documentation in queues.conf.sample. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	c2447e0cc8	Prevent a crash in ExternalIVR when the 'S' command is sent first. If the first command sent from an ExternalIVR client is an 'S' command, we were blindly removing the first element from the play list and deferencing it, even if it was NULL. This corrects that and also locks appropriately in one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	07a9b1744d	Handle multiple commands per connection via netconsole Asterisk would accept multiple NULL-delimited CLI commands via the netconsole socket, but would occasionally miss a command due to the command not being completely read into the buffer. This patch ensures that any partial commands get moved to the front of the read buffer, appended to, and properly sent. (closes issue ASTERISK-18308) Review: https://reviewboard.asterisk.org/r/1876/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	0e488d7cc4	Fix a variety of potential buffer overflows * chan_mobile: Fixed an overrun where the cind_state buffer (an integer array of size 16) would be overrun due to improper bounds checking. At worst, the buffer can be overrun by a total of 48 bytes (assuming 4-byte integers), which would still leave it within the allocated memory of struct hfp. This would corrupt other elements in that struct but not necessarily cause any further issues. * app_sms: The array imsg is of size 250, while the array (ud) that the data is copied into is of size 160. If the size of the inbound message is greater then 160, up to 90 bytes could be overrun in ud. This would corrupt the user data header (array udh) adjacent to ud. * chan_unistim: A number of invalid memmoves are corrected. These would move data (which may or may not be valid) into the ends of these buffers. * asterisk: ast_console_toggle_loglevel does not check that the console log level being set is less then or equal to the allowed log levels of 32. * frame: In ast_codec_pref_prepend, if any occurrence of the specified codec is not found, the value used to index into the array pref->order would be one greater then the maximum size of the array. * jitterbuf: If the element being placed into the jitter buffer lands in the last available slot in the jitter history buffer, the insertion sort attempts to move the last entry in the buffer into one slot past the maximum length of the buffer. Note that this occurred for both the min and max jitter history buffers. * tdd: If a read from fsk_serial returns a character that is greater then 32, an attempt to read past one of the statically defined arrays containing the values that character maps to would occur. * localtime: struct ast_time and tm are not the same size - ast_time is larger, although it contains the elements of tm within it in the same layout. Hence, when using memcpy to copy the contents of tm into ast_time, the size of tm should be used, as opposed to the size of ast_time. * extconf: this treats ast_timing's minmask array as if it had a length of 48, when it has defined the size of the array as 24. pbx.h defines minmask as having a size of 48. (issue ASTERISK-19668) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e9a0da476a	Add ability to ignore layer 1 alarms for BRI PTMP lines. Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7c0583212e	Handle case where an unknown format is used to get the preferred codec size In ast_codec_pref_getsize, if an unknown format is passed to the method, no preferred codec will be selected and a negative number will be used to index into the format list. The method now logs an unknown format as a warning, and returns an empty format list. (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@362368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	70bde6ffa7	Fix places in resources where a negative return value could impact execution This patch addresses a number of modules in resources that did not handle the negative return value from function calls adequately. This includes: * res_agi.c: if the result of the read function is a negative number, indicating some failure, the result would instead be treated as the number of bytes read. This patch now treats negative results in the same manner as an end of file condition, with the exception that it also logs the error code indicated by the return. * res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor to srcfd, and instead assigns a negative value, that file descriptor could later be passed to functions that require a valid file descriptor. If spawn_mp3 fails, we now immediately retry instead of continuing in the logic. * res_rtp_asterisk.c: if no codec can be matched between two RTP instances in a peer to peer bridge, we immediately return instead of attempting to use the codec payload type as an index to determine the appropriate negotiated codec. (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@362362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	0bdbd0d899	Fix places in main where a negative return value could impact execution This patch addresses a number of modules in main that did not handle the negative return value from function calls adequately, or were not sufficiently clear that the conditions leading to improper handling of the return values could not occur. This includes: * asterisk.c: A negative return value from the read function would be used directly as an index into a buffer. We now check for success of the read function prior to using its result as an index. * manager.c: Check for failures in mkstemp and lseek when handling the temporary file created for processing data returned from a CLI command in action_command. Also check that the result of an lseek is sanitized prior to using it as the size of a memory map to allocate. * translate.c: Note in the appropriate locations where powerof cannot return a negative value, due to proper checks placed on the inputs to that function. (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@362359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	2d7a927c81	Fix places where a negative return from ftello could be used as invalid input In a variety of locations in both reading and writing a file, the result from the C library function ftello is used as input to other functions. For the parameters and functions in question, a negative value is invalid input. This patch checks the return value from the ftello function to determine if we were able to determine the current position in the file stream and, if not, fail gracefully. (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@362355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
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

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