3296 Commits (4666efb84920d416d6b291d61389b0eb6941c7c4)

Author	SHA1	Message	Date
Matthew Jordan	4666efb849	Fix remotely exploitable stack overflow in HTTP manager ... There exists a remotely exploitable stack buffer overflow in HTTP digest authentication handling in Asterisk. The particular method in question is only utilized by HTTP AMI. When parsing the digest information, the length of the string is not checked when it is copied into temporary buffers allocated on the stack. This patch fixes this behavior by parsing out pre-defined key/value pairs and avoiding unnecessary copies to the stack. (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	d9c2ea6d15	udptl: Ensure fec[] in udptl_build_packet() is initialized. ... Scan results indicated that this array could be used uninitialized. At a quick look, it looks correct. In any case, initializing it is a Good Thing (tm). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5b61050209	Fix deadlock potential with some ast_indicate/ast_indicate_data calls. ... Calling ast_indicate()/ast_indicate_data() with the channel lock held can result in a deadlock with a local channel because of how local channels need to avoid deadlock. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	6625f0a50c	Fix incorrect jitter buffer overflow due to missed resynchronizations ... When a change in time occurs, such that the timestamps associated with frames being placed into an adaptive jitter buffer (implemented in jitterbuf.c) are significantly different then the previously inserted frames, the jitter buffer checks to see if it needs to be resynched to the new time frame. If three consecutive packets break the threshold, the jitter buffer resynchs itself to the new timestamps. This currently only occurs when history is calculated, and hence only on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other hand, are never passed to the history calculations. Because of this, if the jump in time is greater then the maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are dropped and no resynchronization occurs. Alterntively, if the overfill logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the buffer, but with a time reference that is not applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads from the jitter buffer reach the errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch threshold. Note that this only impacts chan_iax2, as other consumers of the adaptive jitter buffer use the abstract jitter buffer API, which does not use JB_TYPE_CONTROL frames. Review: https://reviewboard.asterisk.org/r/1814/ (closes issue ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7141a29e1b	Fix Dial m and r options and forked calls generating warnings for voice frames. ... When connected line support was added, the wait_for_answer() variable single changed its meaning slightly. Unfortunately, the places where single was used did not necessarily get updated to reflect that change. Also audio/video frames were sent to all forked calls when the endpoints were never made compatible. * Don't pass audio/video media frames when the channels have not been made compatible. * Added handling of AST_CONTROL_SRCCHANGE to app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also pass a requested MOH class. (closes issue ASTERISK-16901) Reported by: Chris Gentle (closes issue ASTERISK-17541) Reported by: clint Review: https://reviewboard.asterisk.org/r/1805/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	5a87fa5de9	Fix bogus reads/writes of console log levels in asterisk.c ... This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact that logger.c implements 32 log levels (because of the custom log level stuff). asterisk.c uses this define to size an array of levels per remote console. This array is modified in ast_console_toggle_loglevel(), which is called by the "logger set level" CLI command. While the documentation for the CLI command doesn't make it terribly obvious, you can use this CLI command to toggle a custom log level on a remote console, as well. However, doing so led to an invalid array index in asterisk.c. This array is read from any time a log message is written to a console. So, all custom log level messages resulted in a bogus read if a remote console was connected. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	a66d14f2ae	Fix inaccurate sizeof() in sched.c. ... This code just needed sizeof(int), not sizeof(int *). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	48b1c5e5a6	Fix incorrect sizeof() usage in features.c. ... This didn't actually result in a bug anywhere, luckily. The only place where the result of these memcpys was used is in app_dial, and the only field that it read out of ast_call_feature was the first one, which is an int, so these memcpys always copied just enough to avoid a problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	c8f8fcc518	Fix incorrect sizeof() on a pointer in MD5Final(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	f5067300be	Don't use a buffer after it goes out of scope. ... 's' is set to 'workspace'. Make sure 'workspace' doesn't go out of scope while the reference to it via 's' is still used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	c546ff5110	Fix setting CDR variables in the hangup extension ... A previous CDR fix for setting CDR variables during a bridge via custom dialplan features broke setting CDR variables in the hangup extension. This patch fixes the issue. Review: https://reviewboard.asterisk.org/r/1794/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	7622a34c89	Make hints for invalid SIP devices return Unavail, not idle ... This patch drastically simplifies the device state aggegation code. The old method was not only overly complex, but also made it impossible to return AST_DEVICE_INVALID from the aggregation code. The unit test update is as a result of fixing that bug. The SIP change stems from a bug introduced by removing a DNS lookup for hostname-based SIP channels. (closes issue ASTERISK-16702) Review: https://reviewboard.asterisk.org/r/1808/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	47244a11d6	Eliminate double close of file descriptor in manager.c ... The process_output function in manager.c attempted to call fclose and close immediately afterwards. Since fclose implies close, this resulted in a potential double free on file descriptors. This patch changes that behavior and also adds error checking to fclose and close depending on which was deemed necessary. Also error messages. Thanks to Rosen Iliev for pointing out the location of the problem. (closes issue ASTERISK-18453) Reported By: Jaco Kroon Review: https://reviewboard.asterisk.org/r/1793/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	f1277cc0e0	Fix case-sensitivity for device-specific event subscriptions and CCSS ... This change fixes case-sensitivity for device-specific subscriptions such that the technology identifier is case-insensitive while the remainder of the device string is still case-sensitive. This should also preserve the original case of the device string as passed in to the event system. CCSS is the only feature affected as it is the only consumer of device-specific event subscriptions. The second part of this patch addresses similar case-sensitivity issues within CCSS itself that prevented it from functioning correctly after the fix to the events system. This adds a unit test to verify that the event system works as expected. (closes issue ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a36c4234a4	Remove ISDN hold restriction for non-bridged calls. ... The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	ed7570328a	Fix race condition that can cause important control frames (such as a hangup) to be missed. ... This takes two actions. 1. Move the reading of the alertpipe in __ast_read() to immediately before the removal of frames from the readq. This means we won't do something silly like read from the alertpipe, then ignore the fact that there's a frame to get from the readq since channel's fdno is the AST_TIMING_FD. 2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because if the rate is 0 and the timingfunc is NULL, it means that the channel's timing fd is being invalidated, so any pending reads should not occur. This may actually solve more issues than the referenced one below, but it's not known at this time for sure. (closes issue ASTERISK-19223) reported by Frank-Michael Wittig Review: https://reviewboard.asterisk.org/r/1779 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	496d1009dd	Second attempt to get optimal translation paths when codec_resample is used. ... This borrows code heavily from changes made in translation code in Asterisk 10. This uses the quality and sample rate change of translation in order to pick paths rather than the computational cost of translations. Computational cost is used solely in determining if a single translation step from a specific translator is better than the same translation step provided by a different translator. (closes issue ASTERISK-16821) reported by Andrew Lindh Review: https://reviewboard.asterisk.org/r/1772 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	8417ceb751	Prevent outbound SIP NOTIFY packets from displaying a port of 0 ... In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. (closes issue ASTERISK-19430) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	100721d217	Fix callerid of Originated calls. ... Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	c026fb96b0	Copy CDR variables when set during a bridge ... This patch makes sure amaflags, accountcode, and userfield get copied to the bridge CDR when set during a bridge (like via a custom feature). (closes issue ASTERISK-16990) Review: https://reviewboard.asterisk.org/r/1721/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	032962f1a2	Allow SRTP policies to be reloaded ... Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f49ff3ff9c	Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension. ... Custom parking extensions may not be coded such that the first and only extension priority is the Park application. These custom parking extensions will not be recognized as parking extensions. When a call is blind transferred to an extension that is not recognized as a parking extension, the normal blind transfer code causes the transferred channel to start executing dialplan. Calls that get parked in this manner do not know the original channel name that parked the call so the original parker could never be called back if the parked call is not retrieved before the timeout time. The parking space is also announced to the call being parked as a side effect of not knowing the original parking channel. * Fix handling of BLINDTRANSFER channel variable for call parking. * Fixed SIP blind transfer using the wrong dialplan context variable to check for the parking extension. (closes issue ASTERISK-19322) Reported by: aragon Tested by: rmudgett, jparker Review: https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	9a3c569772	Track module use count for res_calendar ... If the res_calendar module was followed immediately by one of the calendar tech modules and "core stop gracefully" was run, Asterisk would crash. This patch adds use count tracking for res_calendar so that it is unloaded after the tech modules when shutting down gracefully. It is now not possible to unload all the of the calendar modules via "module unload res_calednar.so", but it is still possible to unload them all via "module unload -h res_calendar.so". Review: https://reviewboard.asterisk.org/r/1752/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	1ba154c100	Revert change to translate.c as it has caused an infinite loop to occur in circumstances. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	706d34fade	Revert a change to audio_audiohook_write_list that had no affect. ... When I made this change initially, I was under the false impression that the audiohooks structure remained on the channel after all of the hooks had been detached. This is not the case, ast ast_read takes care of removing the audiohooks structure if the lists are empty. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9ce7947aa6	Fix AMI Redirect ExtraChannel not redirecting to the same exten and context. ... The astman_get_header() never returns NULL so the check by the code for NULL would never fail. (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches: 0018325.patch (license #6116) patch uploaded by Nuno Borges (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	ae3a465d8f	Make the config parser remove escaping backslashes ... The config parser in Asterisk does not currently remove a backslash that is used to escape a semicolon which would otherwise be interpreted as the start of a comment. The change here causes that backslash to be removed, but does not create a real escape system in the config parser. The biggest complication with a real escape system would be breaking existing configs everywhere (parsing \\ as \ and breaking on escaped non-semicolon characters) even though it would be the "right" way to do things. (closes issue ASTERISK-17121) Review: https://reviewboard.asterisk.org/r/1724/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	5aa7a6986e	Remove outdated comment. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	c4334ee50a	Fix translation path choices. ... This change makes it so computational cost is not taken into account when deciding if a multistep path is better than a single-step path. This means that the only time a multistep path will be chosen is if no single-step path exists. This ensures a better quality translation even if it turns out to be slightly slower. (closes issue ASTERISK-16821) reported by Andrew Lindh Review: https://reviewboard.asterisk.org/r/1715 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	fdd8444090	Remove some unnecessary locking from ast_hangup(). ... This patch removes some unnecessary locking of the channels container in ast_hangup(). The reason this came up is that this lock can very quickly block the entire system. If any of the channel cleanup code decides to block, it causes a problem for the whole system. For example, when audiohooks get destroyed, if that blocks for a while waiting on the mixmonitor thread to exit because it's busy blocking on some I/O, it causes a problem for many other threads in the meantime. Review: https://reviewboard.asterisk.org/r/1712/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	294793a822	Add missing headers to AMI UnParkedCall event to uniquely identify the call. ... The AMI UnParkedCall event was missing the Parkinglot and Uniqueid headers that the AMI ParkedCall event contains. (closes issue ASTERISK-19240) Reported by: Michael Yara git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	47b3aa6362	Fix TLS port binding behavior as well as reload behavior: ... * Removes references to tlsbindport from http.conf.sample and manager.conf.sample * Properly bind to port specified in tlsbindaddr, using the default port if specified. * On a reload, properly close socket if the service has been disabled. A note has been added to UPGRADE.txt to indicate how ports must be set for TLS. (closes issue ASTERISK-16959) reported by Olaf Holthausen (closes issue ASTERISK-19201) reported by Chris Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas Review: https://reviewboard.asterisk.org/r/1709 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c7c7d4dab4	Fix memory leak in error paths for action_originate(). ... * Fix memory leak of vars in error paths for action_originate(). * Moved struct fast_originate_helper tech and data members to stringfields. * Simplified ActionID header handling for fast_originate(). * Added doxygen note to ast_request() and ast_call() and the associated channel callbacks that the data/addr parameters should be treated as const char *. Review: https://reviewboard.asterisk.org/r/1690/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353454 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	d699845a55	Re-link peers by IP when dnsmgr changes the IP ... Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it anytime an address resolves to something different. There are a couple of issues with this. First, the ast_sockaddr is usually the address of an ast_sockaddr inside a refcounted struct and we never bump the refcount of those structs when using dnsmgr. This makes it possible that a refresh could happen after the destructor for that object is called (despite ast_dnsmgr_release being called in that destructor). Second, the module using dnsmgr cannot be aware of an address changing without polling for it in the code. If an action needs to be taken on address update (like re-linking a SIP peer in the peers_by_ip table), then polling for this change negates many of the benefits of having dnsmgr in the first place. This patch adds a function to the dnsmgr API that calls an update callback instead of blindly updating the address itself. It also moves calls to ast_dnsmgr_release outside of the destructor functions and into cleanup functions that are called when we no longer need the objects and increments the refcount of the objects using dnsmgr since those objects are stored on the ast_dnsmgr_entry struct. A helper function for returning the proper default SIP port (non-tls vs tls) is also added and used. This patch also incorporates changes from a patch posted by Timo Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	d42eaedb15	Find even more network interfaces. ... The previous change made the code look for emN and pciN in addition to what it did originally, which was search for ethN. However, it needed to be looking for pciN#N, so that's what it does now. This also moves the memset() to be before every ioctl(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	1efffc8d95	Add 'L16-256' MIME subtype alias for slin16. ... Asterisk has supported the 'L16' MIME subtype for 16kHz signed linear (PCM) audio for quite some time, but some endpoints refer to it as 'L16-256'. This commit adds this as an alias for the existing format. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	45bf0ea5b0	Update ast_set_default_eid() to find more network interfaces. ... As of Fedora 15, ethN is not the name of ethernet interfaces. The names are emN or pciN. Update some code that searched for interfaces named ethN to look for the new names, as well. For more information about why this change was made, see this page: http://domsch.com/blog/?p=455 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a55030f4fa	Audit of ao2_iterator_init() usage for v1.8. ... Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	d4dc9894f2	Avoid unnecessary rebuilds of main/test.c. ... main/test.c includes "asterisk/version.h", when it should include "asterisk/ast_version.h" instead (and it should use the ast_get_version() and ast_get_version_num() functions). This commit modifies it to extract the Asterisk version information using the proper APIs, and as a result means that main/test.c no longer needs to be rebuilt when a Subversion checkout is updated or modified. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	e41e647429	Fix grammar of comment. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	e2a7c10b97	Fix blind transfers from failing if an 'h' extension is present. ... This prevents the 'h' extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) Review: https://reviewboard.asterisk.org/r/1685 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8a536e73cb	Fix ast_app_dtget() time unit inconsistency. ... Note: Noone calls ast_app_dtget() with the timeout parameter of zero so the bad code normally will never get executed. * Fix unnecessary floating point division in func_timeout.c timeout_write() when all other values are integers. (closes issue ASTERISK-16817) Reported by: Dmitry Andrianov git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	91e4333579	Add missing code to set direct RTP setup information during dialing. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	2263f9d10e	Allow only one thread at a time to do asterisk cleanup/shutdown. ... Add locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1662/ Review: https://reviewboard.asterisk.org/r/1658/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	2f0051982a	Make sure asterisk builds on OpenBSD ... OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not 'struct ucred', which causes compilation of main/asterisk.c to fail in read_credentials(). This allows configure to check for sockpeercred and asterisk to deal with it properly. (closes issue ASTERISK-18929) Reported-by: Barry Miller Patch-by: Barry Miller git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	773074f3b7	Add missing CEL logging fields to various CEL backends. ... * Add missing eventextra to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and EventExtra to cel_manager.c. * Add missing userdeftype support for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample. (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	f22392dbdc	Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_END ... (closes issue ASTERISK-17940) Reporter: Nic Colledge Patches: features_18.patch uploaded by Nic Colledge (license 6245) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	929c8c3cb7	Remove extraneous BRIDGEPEER AMI VarSet event on a CEL dummy channel. ... (closes issue ASTERISK-19180) Reported by: Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license #5909) patch uploaded by Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c1f9b1277e	Make Asterisk -x command line parameter imply -r parameter presence. ... The Asterisk -x command line parameter is documented inconsistently. * Made the -x documentation and behavior consistent. * Since this is also a new year, updated the copyright notices while here. (closes issue ASTERISK-19094) Reported by: Eugene Patches: issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified) Tested by: Eugene git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	49a980a427	Allow playback of formats that don't support seeking ... ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. This regression was introduced in r158062. (closes issue ASTERISK-18994) Patch-by: Timo Teras git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago