2504 Commits (66e993de951dcb3fe536f51b6f4b77a2c5e52180)

Author	SHA1	Message	Date
Russell Bryant	dd50b9e8b5	Merged revisions 222878 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	db7b4ec65e	fixes an ast_netsock_list memory leak. ... ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9456ab2724	Deadlock in channel masquerade handling ... Channels are stored in an ao2_container. When accessing an item within an ao2_container the proper locking order is to first lock the container, and then the items within it. In ast_do_masquerade both the clone and original channel must be locked for the entire duration of the function. The problem with this is that it attemptes to unlink and link these channels back into the ao2_container when one of the channel's name changes. This is invalid locking order as the process of unlinking and linking will lock the ao2_container while the channels are locked!!! Now, both the channels in do_masquerade are unlinked from the ao2_container and then locked for the entire function. At the end of the function both channels are unlocked and linked back into the container with their new names as hash values. This new method of requiring all channels and tech pvts to be unlocked before ast_do_masquerade() or ast_change_name() required several changes throughout the code base. (closes issue #15911) Reported by: russell Patches: masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, atis (closes issue #15618) Reported by: lmsteffan Patches: deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671) Tested by: lmsteffan, dvossel Review: https://reviewboard.asterisk.org/r/387/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	1c9fe00920	Recorded merge of revisions 222152 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	20743ec07d	Allow non-compliant T.38 endpoints to be supportable via configuration option. ... Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c1c25181af	Merged revisions 221970 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines Ensure the result of the hash function is positive. Negative array offsets suck. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ba10edfcac	Initialize a variable that we check immediately upon startup. ... (closes issue #15973) Reported by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	bd7ca4b764	One more off-by-one in trunk ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8c7b3cf738	Merged revisions 221776 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines Fix a bunch of off-by-one errors ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	19ba91cd22	Remove ability to control T.38 FAX error correction from udptl.conf. ... chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	10ce6cd757	Use rtp properties instead of adding a callback ... Thanks, Josh. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	865daf4858	Merged revisions 221086 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a4ece92018	Merged revisions 221200 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines Avoid a potential NULL dereference. (closes issue #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt uploaded by tilghman (license 14) Tested by: kobaz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	01181a27a0	Fix channel reference leak. ... ast_cel_report_event would geet a reference to the bridged channel. However, certain return paths, such as if CEL was not enabled, would result in a reference leak. All return paths now properly unref the channel. (closes issue #15991) Reported by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	cee8c6cd47	Get rid of annoying and cryptic debug messages. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	d04158f5b1	Eliminate unnecessary include of version.h in manager.c. ... Including version.h here causes this file to get recompiled after every commit or update, which is not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	8c30540269	Correct sense of logic test committed in revision 220494. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	aabdc575a5	Don't use hash-based lookups for ast_channel_get_by_name_prefix(). ... ast_channel_get_full() tries to use OBJ_POINTER to optimize name-based channel lookups, but this will not work properly when the channel's full name was not supplied; for name-prefix searches, there is no value in doing a hash-based lookup, and in fact doing so could result in many channels being skipped. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	17180120bf	Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by default (starting in 1.6.3). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	90746d26f3	fixes tcptls_session memory leak caused by ref count error ... (closes issue #15939) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/375/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	f150b48bc0	Add bridge related dial flags to the bridge app ... Most of the functionality here is gained simply by setting the feature flag on the bridge config. However, the dial limit functionality has been moved from app_dial to the features code and has been made public so both app_dial and the bridge app can use it. (closes issue #13165) Reported by: tim_ringenbach Patches: app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	1cf5422dc8	Merged revisions 220288 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines Implicitly sending a progress signal breaks some applications. Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	07f9778f5b	Merged revisions 219653 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines Really stop the stream, when ast_closestream() is called. (closes issue #15129) Reported by: bmh Patches: 20090918__issue15129.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/372/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	b27a54b8de	Merged revisions 219136 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3093ccb619	Merged revisions 219023 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	077b44c43f	Merged revisions 218867 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines Fixes CID pattern matching behavior to mirror that of extension pattern matching. Pattern matching for extensions uses a type of scoring system, giving values for specificity to each character in the pattern. Unfortunately, this is done character by character, in order. This does lead to some less specific patterns being first in line for matching, but it will usually get the job done. This patch merely brings CID matching to the same level as extension matching. This patch does not attempt to tackle the problem shared by extension matching. (closes issue #14708) Reported by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	3031ca468d	Do not attempt to add a parking extension if an error occurred while reading the configuration. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	1ca9bc4e1e	Check the origination priority for more matches, not the current priority. ... Found by Pavel Troller on the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	2ee2947a55	Properly terminate the response to the manager Ping action. ... In passing, correct the formatting of the Timestamp attribute so that there is a space after the colon and before the value. (closes issue #15861) Reported by: Ivan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ad69df830d	Enable turning off the application delimiter warning with the 'dontwarn' option. ... Suggested on the -dev list, and implemented in an alternate way by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	7348bacf05	Merged revisions 216435 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216506 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	98f18d56b8	Merged revisions 216430 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	6f32731568	make sure 'start' is always initialized. ... Makes asterisk compile with --enable-dev-mode git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	7f745ecd73	Document language prompt submission process. ... This patch adds a document describing the language prompt submission process, licensing terms and other issues related to that process. In addition, it modifies the sound file searching process to support language codes with any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts can be named with gender, customer/company, etc. suffices as well. (closes issue #15771) Reported by: jtodd Patches: language-criteria.txt uploaded by jtodd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d09f9fd00a	Merge code associated with AST-2009-006 ... (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	f914f65634	- lock channel before looking for a channel variable ... - Init the parkings list member of struct parkinglot. Thanks Sean for the explanation why this should be here. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	57a9927143	Close up to the soft open file limit (same on Linux, but varies drastically on OS X). ... Also, a Makefile fix for Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches: 20090901__issue14542.diff.txt uploaded by tilghman (license 14) Tested by: jtodd, tilghman Change-type: bugfix git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	cf0076c5f3	Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS frames are properly ... decoded. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0f6b01f914	Fix a trunk compilation warning. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	006b0b480b	Properly initialize the session to prevent a crash. ... (closes issue #15774) Reported by: lasko Patches: 20090831__issue15774.diff.txt uploaded by tilghman (license 14) Tested by: lasko git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	9f7a3466ef	Various patches, to enable Asterisk to once again compile on Mac OS X. ... One note on defining _POSIX_C_SOURCE: while this feature test macro works to require certain behaviors on Linux, it works differently on *BSD platforms to REMOVE certain API calls that are not in the POSIX specification, such as vasprintf(3). Thus, defining it while depending upon vasprintf (and other extensions to the POSIX standard) to be defined is a recipe to ensure that Asterisk is only buildable on Linux. Hence, this define which was meant to INCREASE portability, effectively ensures the opposite. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	40c13bd1b0	Merged revisions 214701 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) | 8 lines Modify comment to be a bit more accurate. We have kept this comment around long enough, that it's pretty clear that we're keeping the code, because changing the code would require a pretty fundamental architectural shift. We've also taken criticism in some quarters, because it was believed that it was referring to the code being nasty. No, the code isn't nasty, just the operation itself is rather odd. Fixed for eternity (probably not). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b8c75efa7e	Ensure that we check for the special value CONFIG_STATUS_FILEINVALID. ... (closes issue #15786) Reported by: a_villacis Patches: asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch uploaded by a villacis (license 660) (Plus a few of my own, to catch the remaining places within manager.c where it could have been a problem) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	29e1e05e13	Add two new dialplan variables when using features ... Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature. Added DYNAMIC_PEERNAME which holds the unique channel name on the other side and is set when a dynamic feature is triggered. (closes issue #14663) Reported by: tamiel Patches: 20090313_features.diff uploaded by tamiel (license 712) Tested by: tamiel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	2794b198ce	Merged revisions 214194 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) | 19 lines ast_write() ignores ast_audiohook_write() results In ast_write(), if a channel has a list of audiohooks, those lists are written to and the resulting frame is what ast_write() should continue with. The problem was the returned audiohook frame was not being handled at all, and the original frame passed into it did not contain the mixed audio, so essentially audio was being lost. One result of this was chan_spy's whisper mode no longer worked. To complicate the issue, frames passed into ast_write may either be a single frame, or a list of frames. So, as the list of frames is processed in the audiohook_write, the returned frames had to be added to a new list. (closes issue #15660) Reported by: corruptor Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c18df321f0	Merged revisions 214068-214069 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009) | 6 lines Fix pronunciation of German dates. (closes issue #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded by Benjamin Kluck (license 803) ........ r214069 | tilghman | 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should always compile before committing... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c1b4f0c4c9	Merged revisions 213970 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009) | 7 lines Improve error message by informing user exactly which function is missing a parethesis. (closes issue #15242) Reported by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by loloski (license 68) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	ad37760473	Make LOAD_ORDER actually work ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	53fd27c005	Fix a crash by checking the proper pointer for validity before deferencing it. ... (closes issue #15751) Reported by: atis Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license 242) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	8b707e913b	Fix compile when certain G711 menuselect options are enabled. ... (closes issue #15697) Reported by: slavon git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago