3376 Commits (055a19e1286fbb7a3559c9928fad4e7d08af5329)

Author	SHA1	Message	Date
Mark Michelson	0f4489dc0f	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 ........ Merged revisions 353770 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353820 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	23bc964e1c	Constify some more channel driver technology callback parameters. ... Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	797d633139	Remove inconsistency in CEL eventtype for user defined events. ... The CEL eventtype field for ODBC and PGSQL backends should be USER_DEFINED instead of the user defined event name supplied by the CELGenUserEvent application. If the field is output as a number, the user defined name does not have a value and is always output as 21 for USER_DEFINED and the userdeftype field would be required to supply the user defined name. The following CEL backends (cel_odbc, cel_pgsql, cel_custom, cel_manager, and cel_sqlite3_custom) can be independently configured to remove this inconsistency. * Allows cel_manager, cel_custom, and cel_sqlite3_custom to behave the same way. (closes issue ASTERISK-17189) Reported by: Bryant Zimmerman Review: https://reviewboard.asterisk.org/r/1669/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a99b3c817b	Fix ExtenSpy and simplify the channel search functions. ... When ast_channel name was opaquified, the channel search functions did not get converted correctly. As a result ExtenSpy which uses a channel iterator search by exten@context could never find anything. * Updated the doxygen documentation for the search functions in channel.h. Review: https://reviewboard.asterisk.org/r/1702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2d7a40de58	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/ ........ Merged revisions 353454 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353463 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	de57235ac6	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/ ........ Merged revisions 353371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353397 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	92ef8a6fe1	Address OpenSSL initialization issues when using third-party libraries. ... When Asterisk is used with various third-party libraries (CURL, PostgresSQL, many others) that have the ability themselves to use OpenSSL, it is possible for conflicts to arise in how the OpenSSL libraries are initialized and shutdown. This patch addresses these conflicts by 'wrapping' the important functions from the OpenSSL libraries in a new shared library that is part of Asterisk itself, and is loaded in such a way as to ensure that *all* calls to these functions will be dispatched through the Asterisk wrapper functions, not the native functions. This new library is optional, but enabled by default. See the CHANGES file for documentation on how to disable it. Along the way, this patch also makes a few other minor changes: * Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to more closely match what is used during run-time configuration. * Corrects some errors in the configure script where AC_CHECK_TOOLS was used instead of AC_PATH_PROG. * Adds a new variable for linker flags in the build system (DYLINK), used for producing true shared libraries (as opposed to the dynamically loadable modules that the build system produces for 'regular' Asterisk modules). * Moves the Makefile bits that handle installation and uninstallation of the main Asterisk binary into main/Makefile from the top-level Makefile. * Moves a couple of useful preprocessor macros from optional_api.h to asterisk.h. Review: https://reviewboard.asterisk.org/r/1006/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	dd35aa1555	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(). ........ Merged revisions 353175 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353176 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	7023350098	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. ........ Merged revisions 353126 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353127 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353128 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	3b785264b0	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 ........ Merged revisions 353077 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353078 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	27b69e7d29	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/ ........ Merged revisions 352955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352956 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	9ee8a74461	Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". ... A long time ago, in a land far far away, we added "asterisk/ast_version.h", which provides the ast_get_version() and ast_get_version_num() functions. These were added so that modules that needed the version information for the Asterisk instance they were loaded in could actually get it (as opposed the version that they were compiled against). We changed everything in the tree to use the new mechanism (although later main/test.c was added using the old method). However, the old mechanism was never removed, and as a result, new code is still trying to use it. This commit removes asterisk/version.h and replaces it with a header that will generate a compile-time error if you try to use it (the error message tells you which header you should use instead). It also removes the Makefile and build_tools bits that generated the file, and it updates main/test.c to use the 'proper' method of getting the Asterisk version information. This is an API change and thus is being committed for trunk only, but it's a fairly minor one and definitely improves the situation for out-of-tree modules. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cbe57b11cb	Fixes for sending SIP MESSAGE outside of calls. ... * Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA function in the authorization attempt. * Pass up better From header contents for SIP to use. Now is in the "display-name" <URI> format expected by MessageSend. (Note that this is a behavior change that could concievably affect some people.) * Block user from adding standard headers that are added automatically. (To, From,...) * Allow the user to override the Content-Type header contents sent by MessageSend. * Decrement Max-Forwards header if the user transferred it from an incoming message. * Expand SIP short header names so the dialplan and other code only has to deal with the full names. * Documents what SIP expects in the MessageSend(from) parameter. (closes issue ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/1683/ ........ Merged revisions 352520 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	50de9578aa	Eliminate unnecessary rebuilds of main/format*.c. ... These files have no need to include "asterisk/version.h", and doing so forces them to be rebuilt each time a Subversion checkout moves between 'modified' and 'unmodified' states. ........ Merged revisions 352516 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields ... Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	c3c6b5a0ba	Fix grammar of comment. ... ........ Merged revisions 352230 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352231 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	0920c50341	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 ........ Merged revisions 352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352228 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	20c6ff71b6	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 ........ Merged revisions 352029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352035 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	778fa4abaf	Various parking improvements. ... * Adds per-parking lot options comebackcontext and comebackdialtime * Makes comebacktoorigin settable per parking lot * Sets a PARKER channel variable when comebacktoorigin is disabled (closes issue ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches: asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231 with updates by me. Review: https://reviewboard.asterisk.org/r/1674 Review: https://reviewboard.asterisk.org/r/963 Reviewed by Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	c60d15222c	Add ABS() absolute value function to the expression parser. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	409751e2dc	Sort the output of 'database showkey' as well. ... You can pass wildcards (%) to the database CLI commands, so this will sort the returned list of matches. ........ Merged revisions 350978 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	35fef9a7dc	Add missing code to set direct RTP setup information during dialing. ... ........ Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350976 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	382d14a214	Sort the output of 'database show' by key. ... This more closely mimics the behavior of 'database show' before the conversion to sqlite3. ........ Merged revisions 350938 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	ef0de1358d	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/ ........ Merged revisions 350888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350889 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	76888b5990	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 ........ Merged revisions 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350731 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ec2b28d913	Remove some dead code in ast_bridge_call(). ... None of the parameters to ast_bridge_call() can be NULL for the bridge to work so no need to check for it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	523c95e146	Add missing CEL logging fields to various CEL backends. ... Multiple revisions 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51 -0600 (Fri, 13 Jan 2012) | 12 lines 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 ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13 Jan 2012) | 8 lines Use compatible names for event extra data for various CEL backends. * Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190) ........ Merged revisions 350555,350571 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350585 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a8276fe8ef	Fix crash from bridge channel hangup race condition in ConfBridge ... This patch addresses two issues in ConfBridge and the channel bridge layer: 1. It fixes a race condition wherein the bridge channel could be hung up 2. It removes the deadlock avoidance from the bridging layer and makes the bridge_pvt an ao2 ref counted object Patch by David Vossel (mjordan was merely the commit monkey) (issue ASTERISK-18988) (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628) (closes issue ASTERISK-19100) Reported by: Matt Jordan Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1654/ ........ Merged revisions 350550 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	19a9761084	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) ........ Merged revisions 350501 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350502 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9988918829	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 ........ Merged revisions 350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350453 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350454 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	9748f19e96	Always treat arguments to get_by_name_cb as strings ... Initially, support was left in for the old style of searching, even though it wasn't actually used. In the case of name_len != 0, the OBJ_KEY flag isn't passed because we aren't matching on a full key and therefor can't use the hash function to optimize. The code left in to support the old way of searching unfortunately treated a prefix search like this as though an ast_channel struct was passed as an arg and caused a crash. This patch also adds needed parentheses around some matching conditions. (closes issue ASTERISK-19182) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b7e814aea5	Fix compiler warnings reported by gcc v4.2.4. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions ... There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	a2a3b3ee4b	Fix shutdown handling of sqlite3 astdb. ... If a db_sync was scheduled just before shutdown, the atexit code calling db_sync would have no effect, causing the astdb commit thread to stay alive. This caused the SIP/realtime_sipregs test to fail. (The fallback kill would run the atexit code again and that would wreak havoc.) This fixes that the atexit kill condition is picked up properly. (closes issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1659 ........ Merged revisions 350180 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	70b246f338	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 ........ Merged revisions 350075 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350076 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	6fa808447b	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 ........ Merged revisions 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349732 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	fd04da5114	Fix an issue where dsp.c would interpret multiple dtmf events from a single key press. ... When receiving calls from a mobile phone into a DISA system on a connection with significant interference, the reporter's Asterisk system would interpret DTMF incorrectly and replicate digits received. This patch resolves that by increasing the number of frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and adjusts dtmf_detect function to reset hits and misses only when an edge is detected. (closes issue ASTERISK-17493) Reported by: Alec Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546) Review: https://reviewboard.asterisk.org/r/1130/ ........ Merged revisions 349728 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349729 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ebf40f1129	Ensures Asterisk closes when receiving terminal signals in 'no fork' mode. ... When catching a signal, in no fork mode the console thread is identical to the thread responsible for catching the signal and closing Asterisk, which requires it to first dispense with the console thread. Prior to this patch, if these threads were identical, upon receiving a killing signal, the thread will send an URG signal to itself, which we also catch and then promptly do nothing with. Obviously this isn't useful behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 349672 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349673 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	573e1e5dc0	Fix documentation for SayNumber to reflect the fact that language is changed in CHANNEL() ... (closes issue ASTERISK-18962) reported by: Nir Simionovich ........ Merged revisions 349450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349451 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	2b2d34b3c9	Constify tag argument in REF_DEBUG related code. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	24a6c9b815	Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop ... Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary. (issue ASTERISK-19040) (issue ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/ ........ Merged revisions 349339 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349340 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	9e48f6799d	Use ast_audiohook_write_list_empty to determine if our lists are empty instead ... of duplicating that logic. ........ Merged revisions 349289 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349290 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	fdda494776	Improve T.38 gateway V.21 preamble detection. ... This commit removes the V.21 preamble detection code previously added to the generic DSP implementation in Asterisk, and instead enhances the res_fax module to be able to utilize V.21 preamble detection functionality made available by FAX technology modules. This commit also adds such support to res_fax_spandsp, which uses the Spandsp modem tone detection code to do the V.21 preamble detection. There should be no functional change here, other than much more reliable V.21 preamble detection (and thus T.38 gateway initiation). ........ Merged revisions 349248 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	8017be6fa9	Once an audiohook is attached to a channel, we continue to transcode all of the ... frames, even after all of the hooks are detached. This patch short-cicuits us out before we transcode unnecessarily. ........ Merged revisions 349144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349145 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	19a4928fee	INFO/Record request configurable to use dynamic features ... Adds two new options to SIP peers allowing them to specify features (dynamic or builtin) to use when sending INFO/record requests. Recordonfeature activates whatever feature is specified when recieving a record: on request while recordofffeature activates whatever feature is specified when receiving a record: off request. Both of these features can be disabled by setting the feature to an empty string. (closes issue ASTERISK-16507) Reported by: Jon Bright Review: https://reviewboard.asterisk.org/r/1634/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	32e35e5fcd	Fix extension state callback references in chan_sip. ... Chan_sip gives a dialog reference to the extension state callback and assumes that when ast_extension_state_del() returns, the callback cannot happen anymore. Chan_sip then reduces the dialog reference count associated with the callback. Recent changes (ASTERISK-17760) have resulted in the potential for the callback to happen after ast_extension_state_del() has returned. For chan_sip, this could be very bad because the dialog pointer could have already been destroyed. * Added ast_extension_state_add_destroy() so chan_sip can account for the sip_pvt reference given to the extension state callback when the extension state callback is deleted. * Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy() and handle_statechange() now that the struct ast_state_cb has a destructor to call. * Ensure that ast_extension_state_add_destroy() will never return -1 or 0 for a successful registration. * Fixed pbx.c statecbs_cmp() to compare the correct information. The passed in value to compare is a change_cb function pointer not an object pointer. * Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for deadlocking when those locks are held during the callback. * Removed unused lock declaration for the pbx.c store_hints list. (closes issue ASTERISK-18844) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/1635/ ........ Merged revisions 348940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348952 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	cf0c9830bf	Add Asterisk TestSuite event hooks to support ConfBridge testing ... This patch adds initial testsuite event hooks so that ConfBridge tests can be executed in the Asterisk TestSuite. (issue ASTERISK-19059) ........ Merged revisions 348846 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	eb37d38b7d	Update documentation for MESSAGE_SEND_STATUS variable. ... (Closes issue ASTERISK-19056) Reported by: Yuri Patches: 348360.diff uploaded by Yuri (license #5242) ........ Merged revisions 348605 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	be74e6f16e	Clean-up on isle five for __ast_request_and_dial() and ast_call_forward(). ... * Add locking when a channel inherits variables and datastores in __ast_request_and_dial() and ast_call_forward(). Note: The involved channels are not active so there was minimal potential for problems. * Remove calls to ast_set_callerid() in __ast_request_and_dial() and ast_call_forward() because the set information is for the wrong direction. * Don't use C++ keywords for variable names in ast_call_forward(). * Run the redirecting interception macro if defined when forwarding a call in ast_call_forward(). Note: Currently will never execute because the only callers that supply a calling channel supply a hungup or zombie channel. * Make feature_request_and_dial() put the transferee into autoservice when it calls ast_call_forward() in case a redirection interception macro is run. Note: Currently will never happen because the caller channel (Party B) is always hungup at this time. * Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame to silence a log message. ........ Merged revisions 348464 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348465 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e71bad4958	Fix cut and past error in ast_call_forward(). ... (issue ASTERISK-18836) ........ Merged revisions 348401 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348405 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago