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Author	SHA1	Message	Date
Matthew Jordan	863493118b	Added clarification for the VERBOSITY setting to etc_default_asterisk ... Clarified that using the VERBOSITY setting in etc_default_asterisk is the same as using the -v command line switch, which causes Asterisk to launch in console mode. (closes issue ASTERISK-17030) Reported by: Jonas ........ Merged revisions 353550 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353551 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5861bab06d	Allow res_calendar to be unloaded ... The calendaring tech modules depend on res_calendar and initially res_calendar just bumped the use count so that it couldn't be unloaded. res_calendar can potentially create many threads and I've seen issues where the Asterisk shutdown has failed where it looked like these threads could be the culprit. This patch adds unload support for res_calendar. Unloading res_calendar will also unload the dependant tech modules as well. (closes issue ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/ ........ Merged revisions 353502 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353503 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353504 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
Alec L Davis	f92d6412ab	Merged revisions 353369 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r353369 | alecdavis | 2012-01-31 11:42:28 +1300 (Tue, 31 Jan 2012) | 9 lines Merged revisions 353368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan 2012) | 2 lines prevent debug messsges displaying -ve Cseq numbers. Missed in R353320 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	0ccc1f5274	Merged revisions 353321 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r353321 | alecdavis | 2012-01-31 11:16:22 +1300 (Tue, 31 Jan 2012) | 25 lines Merged revisions 353320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	c6489d7b32	Correct serious flaw in the top-level Makefile. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353319 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
Kevin P. Fleming	82f313b7b8	Clarify log WARNING message when port-zero SDP 'm' lines received. ... Previously, if an m-line in an SDP offer or answer had a port number of zero, that line was skipped, and resulted in an 'Unsupported SDP media type...' warning message. This was misleading, as the media type was not unsupported, but was ignored because the m-line indicated that the media stream had been rejected (in an answer) or was not going to be used (in an offer). ........ Merged revisions 353260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353261 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	843c7ef088	Allow softkey reject while device onhook. ... Fixes up softkey endcall. Previous code was a copy of onhook, now allows for endcall softkey to be used while device is still onhook. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353224 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	2e04182efc	Audit of ao2_iterator_init() usage for v10. Missed one. ... ........ Merged revisions 353039 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353040 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f2c2d83c92	Audit of ao2_iterator_init() usage for v10. ... Fix double format_cap iterator cleanup. ........ Merged revisions 352992 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	8a401484da	Make failed PauseMonitor and UnpauseMonitor with no valid channel not close AMI session. ... I also went ahead and took a little time to make sure that the manager value AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's how we handle this stuff these days. (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches: res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766) ........ Merged revisions 352959 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352965 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352981 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
Terry Wilson	5bfea5fdbf	Add aresult variable for CALENDAR_WRITE ... This patch adds a CALENDAR_SUCCESS=1/0 variable that is set to show whether or not CALENDAR_WRITE has passed. This patch also adds some debugging for caldav PUT responses and no longer treats responses with no body as an error (as a PUT gets a 201 Created with no body). (closes issue ASTERISK-16903) Reported by: Clod Patry Tested by: Terry Wilson Patches: calendarstatus.diff uploaded by Clod Patry (License #5138), slightly modified by Terry Wilson Review: https://reviewboard.asterisk.org/r/1692/ - This line, and those below, will be ignored-- M res/res_calendar.c M res/res_calendar_exchange.c M res/res_calendar_caldav.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	e0ca02fe21	Merged revisions 352863 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r352863 | alecdavis | 2012-01-27 13:08:03 +1300 (Fri, 27 Jan 2012) | 19 lines Merged revisions 352862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan 2012) | 12 lines rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer. If a BLF subscription exists for long enough, using %d may print negative version numbers. Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	075b8385a0	Fix outbound DTMF for inband mode (tell asterisk core to generate DTMF ... sounds). (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346) ........ Merged revisions 352807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352817 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	f4d98aeb28	Copy amaflags to sip_pvt from peer during create_addr_from_peer ... For whatever reason, we don't have a single function for copying data like this from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the sip_pvt, but it would probably be worth discussing this function along with the others that essentially just copy some amount of data from a peer to a private. (Closes issue ASTERISK-19029) Reported by: Matt Lehner ........ Merged revisions 352755 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352756 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	ed32b1c098	Merged revisions 352705 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r352705 | alecdavis | 2012-01-26 19:33:11 +1300 (Thu, 26 Jan 2012) | 27 lines Merged revisions 352704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan 2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make similar to other Notify messages. sample output: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="715" state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523"> <state>terminated</state> </dialog> </dialog-info> Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1693/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	5be89b07e2	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ... ........ Merged revisions 352643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352651 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352659 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
Kevin P. Fleming	261cde4675	Blocked revisions 352616 ... ........ 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. ........ Merged revisions 352612 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	080ea28515	Remove some extraneous debugging from registry memleak fix ... ........ Merged revisions 352551 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352556 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352565 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
Terry Wilson	4bf5e3716e	Clean up some SIP registry-related memory leaks ... 1) Be sure and free at unload the epa_backend we allocate at startup 2) Do the same sip_registry cleanup at unload we do at reload Review: https://reviewboard.asterisk.org/r/1689/ ........ Merged revisions 352514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352515 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352519 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
Jonathan Rose	973aeabf2d	Redocuments sip types peer, user, friend in sip.conf.sample ... There was faulty information in the sample config describing user as a synonym for friend so it has been changed to better elaborate on the differences between the three entity types. (closes issue ASTERISK-15537) Reported by: yarique ........ Merged revisions 352511 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352512 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	213f7a65b5	Fix channel opaquification of stringfields for chan_vpb ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	0fe9043233	Don't do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured. ... (closes issue ASTERISK-16550) reported by: Olle Johansson ........ Merged revisions 352424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352430 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	b68b7e55ac	Set core sounds version to 1.4.22. ... Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22! (closes issue ASTERISK-18978) Reported by: Cameron Twomey Patches: confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002 uploaded by Cameron Twomey ........ Merged revisions 352367 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352373 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352377 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
Richard Mudgett	2144ba5df2	Fix locking issues with channel datastores in func_odbc.c. ... * Fixed a potential memory leak when an existing datastore is manually destroyed by inline code instead of calling ast_datastore_free(). (closes issue ASTERISK-17948) Reported by: Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/ ........ Merged revisions 352291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352292 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	84aea92ba7	Blocked revisions 352288 ... ........ Blocked revisions 352287 ........ Move RTP timeout check to before bridged channel check so it is actually executed. (issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury (closes issue ASTERISK-14534) Reported by: kriborgen Patches: chan_sip.patch uploaded by kriborgen (license 6138) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352289 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
Matthew Jordan	59a42de303	Correctly apply FAXOPT settings (V17, V27, V29) before starting spandsp layer ... While the FAXOPT function could be used to set the modem capabilities, the input to that function was not being applied correctly to the spandsp layer. This patch applies the current model capabilities before starting the spandsp layer. (closes issue: ASTERISK-16409) Reported by: Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license 5081) spandsp-modems-10.diff uploaded by mnicholson (license 5081) ........ Merged revisions 352144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352149 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	de09749470	Add an announcement option to music-on-hold - plays sound when put on hold/between songs ... This is a feature patch which allows an 'announcement' option to be specified in musiconhold.conf which should be set to the name of a sound. If a valid sound is specified for this option, then it will be played on that music on hold class whenever a channel bound to that class is put on hold as well as when Asterisk is able to detect that a song has ended before starting the next song (excludes external players). (closes ASTERISK-18977) Reported by: Timo Teräs Patches: asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	a1bef6041d	Adds the ability to stop specific mixmonitors by using unique IDs set at monitor launch. ... MixMonitor receives a new option i(channel_variable) which stores the unique id at said variable. StopMixMonitor now accepts ID as an optional argument, which if included will make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI commands and AMI actions have been ammended to work with the IDs as well. In addition, monitors across a channel can now be listed be listed via CLI command "mixmonitor list <channel>" which will display all of the mixmonitors active on that channel along with the files they each have open. Created by Sergio González Martín. (closes issue ASTERISK-19096) Reported by: Sergio González Martín Review: https://reviewboard.asterisk.org/r/1643/ Review: https://reviewboard.asterisk.org/r/1682/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	74508e3bca	Fix sip_cfg.notifycid to be set with the defined enum values. ... The invalid value used when notifycid was enabled was benign. As far as the code was concerned -1 and 1 are equivalent. (closes issue ASTERISK-19232) Reported by: Eike Kuiper ........ Merged revisions 352090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352091 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352092 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	02408a2476	Remove XXX comment that is not necessary. ... ........ Merged revisions 352016 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352017 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	ab8ba431b6	Fix RTP reference leak. ... If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference for the RTP instance of the transferer. This fixes the issue by merging two similar but slightly conflicting sections of code into a single area. It also adds a stop_media_flows() call in the case that the transferer's UA never sends a BYE to us like it is supposed to. (issue ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/ ........ Merged revisions 352014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352015 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d0c765497d	Make CLI sip show channel list the complete route set. ... (closes issue ASTERISK-16877) Reported by: klaus3000 Patches: show-complete-routeset-patch.txt (license #5054) patch uploaded by klaus3000 (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c6fd4f5d74	SIP session timeout AMI event ... Add an AMI event in the Call category that is issued when a call is terminated due to either RTP stream inactivity or SIP session timer expiration. Event description: Event: SessionTimeout Source: source Channel: channel-name Uniqueid: channel-unique-id `source` can be either RTPTimeout or SIPSessionTimer (closes issue ASTERISK-16467) Patch-by: Kirill Katsnelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 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
Mark Michelson	b98a25ef93	Prevent potential buffer overflow on AMI MixMonitor command. ... Don't be alarmed. This only affected trunk, and it would have required manager access to your system. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	43621b05a9	More corrections for the ilbc code ... These changes are in a file that is not compiled by default, and so were missed on earlier checks. ........ Merged revisions 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351861 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago