20998 Commits (51ce432d0743ed839af28bc620fa2bc94eed8245)

 

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Author	SHA1	Message	Date
Brett Bryant	51ce432d07	This patch fixes a bug with MeetMe behavior where the 'P' option for always ... prompting for a pin is ignored for the first caller. (closes issue #18070) Reported by: mav3rick Review: https://reviewboard.asterisk.org/r/1132/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Brett Bryant	a54ab29087	Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null ... value. (closes issue #18821) Reported by: cmaj Patches: patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx uploaded by cmaj (license 830) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	2f95620a2f	Don't use static declared buf in parse_name_andor_addr ... This function isn't used anywhere yet, but we definitely don't want to keep the same value for buf between calls to the function. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	a00e99ec56	Merged revisions 311496 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011) | 2 lines Fixes memory leak in MeetMe AMI action ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	7cf95da39a	Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL. ... This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings. (closes issue #18759) Reported by: bklang Patches: null-strings.patch uploaded by bklang (license 919) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	87b246e421	Properly populate the LOCALSTATIONID channel variable. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4f3cf039f4	Race condition when ISDN CallRerouting/CallDeflection invoked. ... The queued AST_CONTROL_BUSY could sometimes be processed before the call_forward dial string is recognized. * Moved setting the call_forwarding dial string after sending a response to the initiator and just queue an empty frame to wake up the media thread instead of an AST_CONTROL_BUSY. * Added check for empty rerouting/deflection number and respond with an error. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	93601856b6	Merged revision 310986 from ... https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines Dial() o option broke when connected line feature added. The patch restores the o option behavior and adds the ability to specify the CallerID. The Dial o and f options are complementary to each other. The o option stores the CallerID on the outgoing channel as the channel's CallerID. The f option forces the CallerID sent by the outgoing channel. o(x) - The argument 'x' is optional. If not present, then specify that the CallerID that was present on the *calling* channel be stored as the CallerID on the *called* channel. This was the behavior of Asterisk 1.0 and earlier. If present, then specify the CallerID stored on the *called* channel. Note that o(${CALLERID(all)}) is similar to option o without parameters. f(x) - The argument 'x' is optional and its presence changes the behavior of this option. If not present, then force the outgoing CallerID on a call-forward or deflection to the dialplan extension for this Dial() using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If present, then force the outgoing CallerID to 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA SWP-3096 .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ef01ba5ff2	This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. ... In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy. (closes issue #18742) Reported by: jkister Tested by: jkister, jcovert, jrose Review: http://reviewboard.digium.internal/r/106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	0e86babe2f	Merged revisions 311140 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar 2011) | 4 lines Don't write items to the manager socket twice. AST-2011-003 (closes issue 0018987) Reported by: ks-steven ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	d0829474b2	Merged revisions 311049 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300 (Thu, 17 Mar 2011) | 17 lines Merged revisions 311048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar 2011) | 12 lines Remove extra quote in indications.conf Picking low hanging fruit. (closes issue #18971) Reported by: IgorG Patches: based on indications.conf.sample.diff uploaded by IgorG (license 20) Tested by: IgorG ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	3cc8ae22cd	Merged revisions 310998 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) | 11 lines Fix crash on fdopen failure See security advisory AST-2011-004 (closes issue #18845) Reported by: cmaj Patches: patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt uploaded by cmaj (license 830) patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt uploaded by cmaj (license 830) Tested by: cmaj, twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	d37bdd02dc	Merged revisions 310992 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) | 4 lines Don't keep trying to write to a closed connection See security advisory AST-2011-003. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	bec22e5c1f	Merged revisions 310889 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines Don't delay DTMF in core bridge while listening for DTMF features This patch is mostly the work of Olle Johansson. I did some cleanup and added the silence generating code if transmit_silence is set. When a channel listens for DTMF in the core bridge, the outbound DTMF is not sent until we have received DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds. Some products see this delay and the time skew on RTP packets that results and start ignoring the audio that is sent afterward. With this change, the DTMF_BEGIN frame is inspected and checked. If it matches a feature code, we wait for DTMF_END and activate the feature as before. If transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a multi-digit feature. If it doesn't match a feature, the frame is forwarded along with the DTMF_END without delay. By doing it this way, DTMF is not delayed. (closes issue #15642) Reported by: jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396) Tested by: globalnetinc, jde (closes issue #16625) Reported by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ Review: https://reviewboard.asterisk.org/r/1125/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	feb0df1202	Fix branch compile. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	7ac0138d31	core show locks: display ThreadID in hexadecimal ... Allow easier cross referencing of thread ID's with GDB backtraces (closes issue #18968) Reported by: alecdavis Patches: bug18968.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	5082c5d44b	Introduce t.38 parameters control functionality not full but enough for ... Send/RcvFax support Introduce t.38 controls between asterisk core and channel/proto layers. Not all parameters are transferred from proto layers but *Fax apps tested and work ok. (issue #18693) Reported by: benngard2 Patches: issue-18693.patch uploaded by may213 (license 454) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1a7d232a5a	Undoes 310726 for further analysis ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	89eaad8ab8	Moves data store destruction from channel destruction to hangup in channel.c ... This moves the data store destruction and app signaling events for a call to ast_hangup so that threads which wait for data store destruction don't become stuck forever when attached to an application/function/etc that keeps the channel open. (closes issue #18742) Reported by: jkister Patches: patch.diff uploaded by jrose (license 1225) Tested by: jkister, jcovert, jrose Review: https://reviewboard.asterisk.org/r/1136/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a6bb331e18	Merged revisions 310635 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines Merged revisions 310633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and TDM410 The last character in the caller id message is getting a framing error. The checksum is the last character in the message. A framing error in the checksum could be because: 1) The sender did not send a full stop bit. 2) The sender cut off the FSK carrier too soon. 3) The sender opted to send zero of the specified zero to 10 trailing mark bits and round-off errors in the code resulted in the code not being where it thought it was in the demodulated bit stream. Bit 8 of 'b' is set when parity error. Bit 9 of 'b' is set when framing error. Made ignore the framing and parity error bits if the errored character is the checksum. We can tolerate a framing/parity error there. The checksum character validates the message. (closes issue #18474) Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek (license 636) (with modifications) Tested by: nivek ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	7ea558865a	Merged revisions 310585 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | 8 lines Adds 'p' as an option to func_volume. When it is on, the old behavior with DTMF controlling volume adjustment will be enforced. When it is off, DTMF will not be processed by the function. Programmed by Jonathan Rose Reviewed by David Vossel, Leif Madsen, and Russell Bryant http://reviewboard.digium.internal/r/93/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	49fa80b8d3	Merged revisions 310448 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310448 | tilghman | 2011-03-12 14:24:54 -0600 (Sat, 12 Mar 2011) | 38 lines Recorded merge of revisions 310435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011) | 31 lines Add AELSub, which provides a stable entry point into AEL subroutines. This commit needs some explanation, given that we're adding a new application into an existing release branch. This is generally a violation of our release policy, except in very limited circumstances, and I believe this is one of those circumstances. The problem that this solves is one of the sanity of using multiple dialplan languages to define a dialplan. In the case of the reporter, he or she is using AEL is define subroutines, while using Realtime extensions to invoke those subroutines. While you can do this, it's based upon the reality of AEL using actual dialplan extensions; however, there is no guarantee that the details of _how_ AEL is compiled into extensions will remain stable. In fact, at the time of this commit, it has already changed twice, once in a fundamental way. Now normally, a new application would only be added to trunk. However, this application is explicitly to create a stable user-level API between versions, and adding it to trunk only will not solve the user's problem of switching between 1.6.2 and 1.8, nor will it help anybody switching from 1.8 to 1.10. Therefore, it needs to go into existing release branches. For the sake of consistency, and also because one of the changes was between 1.4 and 1.6.x, I am also electing to commit this to 1.4. (closes issue #18910) Reported by: alexandrekeller Patches: 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman (license 14) 20110304__issue18919__1.4.diff.txt uploaded by tilghman (license 14) Tested by: alexandrekeller ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	782d757faf	Merged revisions 310414 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011) | 7 lines Transactional handles should be used for the insertbuf, if available. Also, fix a possible resource leak. (closes issue #18943) Reported by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	a1e7bf50b5	remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call ... If the channel condition is one of the following after breaking out of the loop, don't try to update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81 Review: https://reviewboard.asterisk.org/r/1128/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	8fe14985fb	Add \r\n to remaining http headers passed to ast_http_send ... r309204 changed the behavior of ast_http_send. It now requires headers to be passed with trailing \r\n. This change updates the remaining instances in the code that did not pass the \r\n. (closes issue #18186) Reported by: nivaldomjunior Patches: res_phoneprov.c.diff uploaded by lathama (license 1028) manager.diff.txt uploaded by twilson (license 396) Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	d409098f0e	Be more tolerant of what URI we accept for call completion PUBLISH requests. ... (closes issue #18946) Reported by: GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	15641c348e	Merged revisions 310141 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. (closes issue #18295) Reported by: pruiz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ed3e04e831	Returns with an error notice if CHANNEL function of SIP channel is read without arguments. ... (Closes issue #18653) Reported by: wuwu Patches: diff.patch uploaded by jrose (license 1225) Tested by: jrose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	08b05120a9	Spelling fix in "calendar show calendar" ... s/Cartegories/Catagories/ (closes issue #18931) Reported by: pdugas Patches: res_calendar.c.patch uploaded by pdugas (license 1222) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8bfde13607	Make pri parameter description consistent. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	4ad0ddf5e3	Merged revisions 309857 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309857 | jrose | 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | 8 lines Bug fix for MixMonitor involving filenames with '.' not in the extension Closes issue #18391) Reported by: pabelanger Patches:       bugfix.patch uploaded by jrose (license 1225) Tested by: jrose ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	56cd7709a5	Merged revisions 309251 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	c8876e8ca6	Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Moises Silva	3770b4d7cb	Fix caller id passed to openr2_chan_make_call ... (closes issue #18894) Reported by: malufrj Tested by: moy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	e4a3720d49	Merged revisions 309677 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) | 7 lines Missed part of the conversion when we started passing ppid to astcanary. (closes issue #18850) Reported by: viraptor Patches: canary_ppid.patch uploaded by viraptor (license 543) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	d4a55c8fd8	Merged revisions 309584 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar 2011) | 2 lines Restore mysterious lua_pushvalue() call removed in r309494. The mystery has been solved. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	a6f3fd48e0	Merged revisions 309541 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar 2011) | 4 lines Check for errors from fseek() when loading config file, properly abort on errors from fread(), and supply a traceback for errors generated when loading the config file. Also, prepend a newline to traceback output so that the main error message is on it's own line. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	43918cb291	Merged revisions 309494 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar 2011) | 2 lines remove mysterious lua_pushvalue() that is never used ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	4988f8f6e1	Export global symbols from pbx_lua to allow modules to be loaded. Fixes a regression introduced in r278132. ... (closes issue #18671) Reported by: Igels Patches: pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96) Tested by: Igels git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309448 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5b9f9f78ca	Get real channel of a DAHDI call. ... Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Ruggles	d5e1774082	Merged revisions 309356 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines fix small memory leak fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	12d5ae5565	Blocked revisions 309348 via svnmerge ... ........ r309348 | lmadsen | 2011-03-03 14:13:11 -0600 (Thu, 03 Mar 2011) | 5 lines Update PickupChan documentation. The PickupChan uses the ampersand as the argument separator. (closes issue #18905) Reported by: vmikhnevych Tested by: vmikhnevych ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	546b652f1f	Merged revisions 309255 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP. Since it's a duplicate, nothing is going to be done, so delme doesn't need to be set at all. Strangely, when this was added, this was being set to 1 in 1.6, and 0 in trunk. (issue AST-439) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	d3cb8a6dab	Fix consistency of CRLFs on HTTP headers that get sent out. ... (closes issue #18186) Reported by: nivaldomjunior Patches: 18186-httpheadernewline.diff uploaded by qwell (license 4) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5413ea1c8c	Document CHANNEL(keypad_digits) and CHANNEL(no_media_path). ... * Added XML documentation for CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML documentation for CHANNEL(reversecharge). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	10e378b07c	Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal. ... Looks like an unintended change when sig_analog.c was extracted from chan_dahdi.c. Removed useless conditional around needed code and fixed resulting compiler warning. (closes issue #18667) Reported by: enegaard Patches: issue18667.patch uploaded by enegaard (license 1197) Tested by: enegaard JIRA SWP-2965 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	0a577f6cdd	Merged revisions 309083 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines Fixes thread blocking issue in the sip TCP/TLS implementation. (closes issue #18497) Reported by: vois Patches: issues_18497.diff uploaded by dvossel (license 671) Tested by: vois, rossbeer, kowalma, Freddi_Fonet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	1b78442e0d	Merged revisions 309033-309034 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error. Detect whether Flex will compile without the workaround; if so, suppress our workaround code. ........ r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify meaning, removing double negative (stupid!) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	f14ba8fa19	Merged revisions 308990 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) | 7 lines Statements updating zero rows may return SQL_NO_DATA. This is fine; it's handled. (closes issue #18815) Reported by: irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot (license 52) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	73254a28cb	Fix Deadlock with attended transfer of SIP call ... Call path sip_set_rtp_peer (locks chan then pvt) transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (locks p->owner) But by the time p->owner lock was attempted, seems as though chan and p->owner were different. So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods. (closes issue #18837) Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81, cmaj Review: [https://reviewboard.asterisk.org/r/1126/] git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago