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| -rw-r--r-- | ChangeLog | 12087 |
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diff --git a/.lastclean b/.lastclean new file mode 100644 index 000000000..8f92bfdd4 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +35 diff --git a/.version b/.version new file mode 100644 index 000000000..787b12a4f --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.6.1-beta1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..7cf94eee4 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,12087 @@ +2008-10-03 - Russell Bryant <russell@digium.com> + + * Asterisk 1.6.1-beta1 Released + +2008-10-02 19:31 +0000 [r145960-145964] Russell Bryant <russell@digium.com> + + * CHANGES, /: Merged revisions 145962 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r145962 | + russell | 2008-10-02 14:30:45 -0500 (Thu, 02 Oct 2008) | 2 lines + The 'P' command for ExternalIVR was also added in 1.6.0 ........ + + * CHANGES, /: Merged revisions 145959 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r145959 | + russell | 2008-10-02 14:27:37 -0500 (Thu, 02 Oct 2008) | 2 lines + TCP support for ExternalIVR went in to 1.6.1, not 1.6.0 ........ + +2008-10-02 15:30 +0000 [r145781] Sean Bright <sean.bright@gmail.com> + + * /, configure, configure.ac: Merged revisions 145771 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r145771 | seanbright | 2008-10-02 11:28:48 -0400 (Thu, 02 Oct + 2008) | 1 line This is much cleaner, methinks. ........ + +2008-10-02 15:19 +0000 [r145754] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c, /: Merged revisions 145752 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r145752 | tilghman | 2008-10-02 10:17:16 -0500 (Thu, 02 Oct 2008) + | 10 lines Merged revisions 145751 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r145751 | tilghman | 2008-10-02 10:13:21 -0500 (Thu, 02 Oct 2008) + | 3 lines Some sanity checks that may have led to prior crashes, + found by codefreeze-lap (murf) on IRC. Also some cleanup of + incorrectly-used constants. ........ ................ + +2008-10-01 23:54 +0000 [r145694] Sean Bright <sean.bright@gmail.com> + + * /, configure, configure.ac: Merged revisions 145692 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r145692 | seanbright | 2008-10-01 19:48:16 -0400 (Wed, 01 Oct + 2008) | 7 lines Try a test compile using the GMime library. Some + distros install gmime-config in the base package instead of the + -devel package. Now we print a notice and disable GMime support + instead of bombing during the main compilation. (closes issue + #13583) Reported by: arkadia ........ + +2008-10-01 22:24 +0000 [r145557-145609] Mark Michelson <mmichelson@digium.com> + + * /, main/features.c: Merged revisions 145606 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r145606 | + mmichelson | 2008-10-01 17:23:50 -0500 (Wed, 01 Oct 2008) | 11 + lines Okay, this should really do it now. While I did manage to + fix blind transfers with my last commit here, I also caused an + unwanted side-effect. That is, only the first priority of the 'h' + extension would be executed when a blind transfer occurred + instead of all priorities. Essentially, my last commit corrected + the return value of ast_bridge_call. However, the implementation + still was not 100% correct. Now it is. ........ + + * /, main/features.c: Merged revisions 145579 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r145579 | + mmichelson | 2008-10-01 16:33:11 -0500 (Wed, 01 Oct 2008) | 4 + lines if (!(x) == 0) is the same as if (x). ........ + + * /, main/features.c: Merged revisions 145553 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r145553 | + mmichelson | 2008-10-01 16:06:26 -0500 (Wed, 01 Oct 2008) | 13 + lines The logic surrounding the return value of + ast_spawn_extension within ast_bridge_call was reversed. This + problem was observed when a blind transfer placed from the callee + channel of a test call failed. While the problem I am solving + here is exactly the same as what was reported in issue #13584, + the difference is that this fix I am applying is trunk-only. + Issue #13584 was reported against the 1.4 branch, and my tests of + 1.4's blind transfers appear to work fine. ........ + +2008-10-01 17:33 +0000 [r145517] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 145487 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r145487 | lmadsen | 2008-10-01 13:26:20 -0400 + (Wed, 01 Oct 2008) | 12 lines Merged revisions 145479 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r145479 | lmadsen | 2008-10-01 13:18:30 -0400 (Wed, 01 Oct 2008) + | 6 lines Update the realtime_pgsql.sql script to create the + setinterfacevar column. (closes issue #13549) Reported by: fiddur + ........ ................ + +2008-10-01 15:45 +0000 [r145430] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_sms.c: Merged revisions 145428 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r145428 | + tilghman | 2008-10-01 10:44:06 -0500 (Wed, 01 Oct 2008) | 7 lines + Initializing buffer prevents a segfault when arguments are + incomplete. (closes issue #13471) Reported by: alecdavis Patches: + 20080916__bug13471.diff.txt uploaded by Corydon76 (license 14) + Tested by: alecdavis ........ + +2008-09-30 22:26 +0000 [r145262] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 145249 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r145249 | + jpeeler | 2008-09-30 17:21:19 -0500 (Tue, 30 Sep 2008) | 6 lines + (closes issue #13337) Reported by: pj Tested by: pj Set transport + to SIP_TRANSPORT_UDP mode if not specified which fixes calls to + get_transport returning UNKNOWN. ........ + +2008-09-27 16:49 +0000 [r144993] Kevin P. Fleming <kpfleming@digium.com> + + * main/ast_expr2.c, Makefile, agi/Makefile, utils/Makefile, + include/asterisk/astmm.h, main/ast_expr2f.c, pbx/pbx_ael.c, + utils/ael_main.c, main/astmm.c, main/stdtime/localtime.c, + utils/extconf.c, main/ast_expr2.fl, include/asterisk.h, /, + main/Makefile, main/ast_expr2.y, Makefile.moddir_rules, + utils/astman.c: Merged revisions 144949-144951 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r144949 | kpfleming | 2008-09-27 10:52:56 -0500 (Sat, 27 Sep + 2008) | 17 lines Merged revisions 144924-144925 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep + 2008) | 6 lines improve header inclusion process in a few small + ways: - it is no longer necessary to forcibly include + asterisk/autoconfig.h; every module already includes asterisk.h + as its first header (even before system headers), which serves + the same purpose - astmm.h is now included by asterisk.h when + needed, instead of being forced by the Makefile; this means + external modules will build properly against installed headers + with MALLOC_DEBUG enabled - simplify the usage of some of these + headers in the AEL-related stuff in the utils directory ........ + r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep + 2008) | 2 lines fix some minor issues with rev 144924 ........ + ................ r144950 | kpfleming | 2008-09-27 11:10:33 -0500 + (Sat, 27 Sep 2008) | 2 lines fix bugs caused by r144949 when + MALLOC_DEBUG is defined ................ r144951 | kpfleming | + 2008-09-27 11:17:43 -0500 (Sat, 27 Sep 2008) | 1 line remove + incorrect comment ................ + +2008-09-27 01:08 +0000 [r144881] Michiel van Baak <michiel@vanbaak.info> + + * apps/app_queue.c, channels/chan_dahdi.c, /: Merged revisions + 144879 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r144879 | + mvanbaak | 2008-09-27 02:49:24 +0200 (Sat, 27 Sep 2008) | 2 lines + fix a couple of CLI commands that did not have a help + description. ........ + +2008-09-26 23:16 +0000 [r144832] Joshua Colp <jcolp@digium.com> + + * /, configs/rtp.conf.sample: Merged revisions 144829 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r144829 | file | 2008-09-26 20:12:13 -0300 (Fri, 26 Sep 2008) | 2 + lines Update documentation to include default setting. This is + for you jtodd! ........ + +2008-09-26 18:09 +0000 [r144484-144684] Steve Murphy <murf@digium.com> + + * /, pbx/pbx_lua.c: Merged revisions 144681 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r144681 | + murf | 2008-09-26 12:02:06 -0600 (Fri, 26 Sep 2008) | 14 lines + (closes issue #13564) Reported by: mnicholson Patches: + pbx_lua9.diff uploaded by mnicholson (license 96) Many thanks to + Matt for his upgrade to the lua dialplan option! the Description + from the bug: This patch adds a stack trace to errors encountered + while executing lua extensions. The patch also handles out of + memory errors reported by lua. ........ + + * main/pbx.c, /: Merged revisions 144678 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r144678 | murf | 2008-09-26 11:50:35 -0600 (Fri, 26 Sep 2008) | + 20 lines Merged revisions 144677 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) | + 12 lines (closes issue #13563) Reported by: mnicholson Patches: + found1.diff uploaded by mnicholson (license 96) This patch was + mainly meant to apply to trunk and 1.6.x, but I'm applying it to + 1.4 also, which should be a perfectly harmless fix to the vast + majority of users who are not using external switches, but the + few who might be affected will not have to go to the pain of + filing a bug report. ........ ................ + + * utils/build-extensions-conf.lua (removed), /: Merged revisions + 144635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r144635 | + murf | 2008-09-26 10:51:30 -0600 (Fri, 26 Sep 2008) | 1 line Matt + suggests we remove utils/build-extensions-conf.lua, as per bug + 12961, it is no longer necessary. ........ + + * channels/chan_oss.c, apps/app_playback.c, main/pbx.c, /, + funcs/func_cut.c: Merged revisions 144569 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r144569 | + murf | 2008-09-25 16:21:28 -0600 (Thu, 25 Sep 2008) | 14 lines + (closes issue #13557) Reported by: nickpeirson The user attached + a patch, but the license is not yet recorded. I took the liberty + of finding and replacing ALL index() calls with strchr() calls, + and that involves more than just main/pbx.c; chan_oss, + app_playback, func_cut also had calls to index(), and I changed + them out. 1.4 had no references to index() at all. ........ + + * /, pbx/pbx_lua.c: Merged revisions 144563 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r144563 | + murf | 2008-09-25 15:54:11 -0600 (Thu, 25 Sep 2008) | 7 lines + (closes issue #13559) Reported by: mnicholson Patches: + pbx_lua8.diff uploaded by mnicholson (license 96) ........ + + * include/asterisk/hashtab.h, /, pbx/pbx_lua.c, + configs/extensions.lua.sample: Merged revisions 144523 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r144523 | murf | 2008-09-25 15:18:12 -0600 (Thu, 25 Sep + 2008) | 13 lines I added a little verbage to hashtab for the + hashtab_destroy func. It was pretty sparsely documented. This + update fleshes out the pbx_lua module, to add the switch + statements to the extensions in the extensions.lua file, as well + as removing them when the module is unloaded. Many thanks to Matt + Nicholson for his fine contribution! ........ + + * /, pbx/pbx_lua.c: Merged revisions 144482 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r144482 | + murf | 2008-09-25 11:51:11 -0600 (Thu, 25 Sep 2008) | 14 lines + (closes issue #13558) Reported by: mnicholson Considering that + the example extensions.lua used nothing but ["12345"] notation, + and that the resulting error message: [Sep 24 17:01:16] + ERROR[12393]: pbx_lua.c:1204 exec: Error executing lua extension: + attempt to call a nil value is not very informative as to the + nature of the problem, I think this bug fix is a big win! + ........ + +2008-09-23 23:36 +0000 [r144151] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 144149 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r144149 | + mmichelson | 2008-09-23 18:33:33 -0500 (Tue, 23 Sep 2008) | 3 + lines Fix a conflict in flag values ........ + +2008-09-23 17:00 +0000 [r144069] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 144067 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r144067 | murf | 2008-09-23 10:52:32 -0600 (Tue, 23 Sep 2008) | + 37 lines Merged revisions 144066 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) | + 29 lines (closes issue #13489) Reported by: DougUDI Tested by: + murf (closes issue #13490) Reported by: seanbright Tested by: + murf (closes issue #13467) Reported by: edantie Tested by: murf, + edantie, DougUDI This crash happens because we are unsafely + handling old pointers. The channel whose cdr is being handled, + has been hung up and destroyed already. I reorganized the code a + bit, and tried not to lose the fork-cdr-chain concepts of the + previous code. I now verify that the 'previous' channel (the + channel we had when the bridge was started), still exists, by + looking it up by name in the channel list. I also do not try to + reset the CDR's of channels involved in bridges. Testing shows it + solves the crash problem, and should not negatively impact + previous fixes involving CDR's generated during/after blind + transfers. (The reason we need to reset the CDR's on the + "beginning" channels in the first place). ........ + ................ + +2008-09-23 15:39 +0000 [r144027] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 144025 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r144025 | + mmichelson | 2008-09-23 10:37:00 -0500 (Tue, 23 Sep 2008) | 16 + lines When a promiscuous redirect contained both a user and host + portion in the Contact URI and specifies a transport, the parsing + done in parse_moved_contact resulted in a malformed URI. This + commit fixes the parsing so that a proper Dial string may be + formed when the forwarded call is placed. (closes issue #13523) + Reported by: mattdarnell Patches: 13523v2.patch uploaded by + putnopvut (license 60) Tested by: mattdarnell ........ + +2008-09-22 22:52 +0000 [r143906] Sean Bright <sean.bright@gmail.com> + + * /, formats/format_pcm.c: Merged revisions 143904 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r143904 | seanbright | 2008-09-22 18:50:07 -0400 + (Mon, 22 Sep 2008) | 16 lines Merged revisions 143903 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r143903 | seanbright | 2008-09-22 18:49:00 -0400 (Mon, 22 Sep + 2008) | 8 lines Use the advertised header size in .au files + instead of just assuming they are 24 bytes (the minimum). (closes + issue #13450) Reported by: jamessan Patches: pcm-header.diff + uploaded by jamessan (license 246) ........ ................ + +2008-09-21 10:06 +0000 [r143839-143845] Michiel van Baak <michiel@vanbaak.info> + + * /, doc/tex/privacy.tex: Merged revisions 143843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r143843 | + mvanbaak | 2008-09-21 11:53:01 +0200 (Sun, 21 Sep 2008) | 3 lines + fix privacymanager example so it shows how to use the + PRIVACYMRGSTATUS variable ........ + + * /, doc/tex/privacy.tex: Merged revisions 143840 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r143840 | + mvanbaak | 2008-09-21 11:31:54 +0200 (Sun, 21 Sep 2008) | 3 lines + document the new context argument for privacymanager so people + can do pattern matching on the input ........ + + * /, doc/tex/privacy.tex: Merged revisions 143837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r143837 | + mvanbaak | 2008-09-21 11:27:08 +0200 (Sun, 21 Sep 2008) | 2 lines + fix privacy documentation. We no longer do priority jumping +101 + ........ + +2008-09-20 00:55 +0000 [r143739] Sean Bright <sean.bright@gmail.com> + + * /, contrib/scripts/vmail.cgi: Merged revisions 143737 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r143737 | seanbright | 2008-09-19 20:52:20 -0400 + (Fri, 19 Sep 2008) | 17 lines Merged revisions 143736 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r143736 | seanbright | 2008-09-19 20:50:10 -0400 (Fri, 19 Sep + 2008) | 9 lines Make vmail.cgi work with mailboxes defined in + users.conf, too. (closes issue #13187) Reported by: netvoice + Patches: 20080911__bug13187.diff.txt uploaded by Corydon76 + (license 14) (Slightly modified to take alchamist's comments on + mantis into account) Tested by: msales, alchamist, seanbright + ........ ................ + +2008-09-19 15:49 +0000 [r143611] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 143609 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r143609 | mmichelson | 2008-09-19 10:43:28 -0500 (Fri, 19 Sep + 2008) | 11 lines We should only unsubscribe to the device state + event subscription if we have previously subscribed. Otherwise a + segfault will occur. (closes issue #13476) Reported by: jonnt + Patches: 13476.patch uploaded by putnopvut (license 60) Tested + by: jonnt ........ + +2008-09-18 23:55 +0000 [r143561] Steve Murphy <murf@digium.com> + + * /, channels/chan_sip.c: Merged revisions 143559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r143559 | murf | 2008-09-18 17:41:33 -0600 (Thu, 18 Sep 2008) | 9 + lines Merged revisions 143534 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r143534 | murf | 2008-09-18 16:11:51 -0600 (Thu, 18 Sep 2008) | 1 + line A micro-fix, in sip_park_thread, where d is freed before the + func is done using it. ........ ................ + +2008-09-17 20:59 +0000 [r143407] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 143405 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r143405 | tilghman | 2008-09-17 15:57:58 -0500 + (Wed, 17 Sep 2008) | 13 lines Merged revisions 143404 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r143404 | tilghman | 2008-09-17 15:55:47 -0500 (Wed, 17 Sep 2008) + | 6 lines When callerid is blank, we want to use "unknown caller" + in those cases, too. (closes issue #13486) Reported by: tomo1657 + Patches: 20080917__bug13486.diff.txt uploaded by Corydon76 + (license 14) ........ ................ + +2008-09-17 18:30 +0000 [r143349] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /: Merged revisions 143340 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r143340 | mmichelson | 2008-09-17 13:26:35 -0500 (Wed, 17 Sep + 2008) | 14 lines Merged revisions 143337 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep + 2008) | 6 lines Allow for "G.729" if offered in an SDP even + though it is not RFC 3551 compliant. Some Cisco switches will + send this in an SDP, and it doesn't hurt to be able to accept + this. ........ ................ + +2008-09-15 21:33 +0000 [r143143] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 143141 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r143141 | tilghman | 2008-09-15 16:31:36 -0500 + (Mon, 15 Sep 2008) | 13 lines Merged revisions 143140 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r143140 | tilghman | 2008-09-15 16:29:32 -0500 (Mon, 15 Sep 2008) + | 6 lines Set the raw formats at the same time as the other + formats. (closes issue #13240) Reported by: jvandal Patches: + 20080813__bug13240.diff.txt uploaded by Corydon76 (license 14) + ........ ................ + +2008-09-14 22:24 +0000 [r143086] Michiel van Baak <michiel@vanbaak.info> + + * /, channels/chan_skinny.c: Merged revisions 143082 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r143082 | mvanbaak | 2008-09-15 00:16:34 +0200 (Mon, 15 Sep 2008) + | 11 lines plug a couple of memleaks in chan_skinny. (closes + issue #13452) Reported by: pj Patches: memleak5.diff uploaded by + wedhorn (license 30) Tested by: wedhorn, pj, mvanbaak (closes + issue #13294) Reported by: pj ........ + +2008-09-13 13:58 +0000 [r143033] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c, channels/iax2-parser.c, apps/app_dial.c, /: + Merged revisions 143031 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r143031 | + tilghman | 2008-09-13 08:54:15 -0500 (Sat, 13 Sep 2008) | 8 lines + Repair IAXVAR implementation so that it works again (regression?) + (closes issue #13354) Reported by: adomjan Patches: + 20080828__bug13354.diff.txt uploaded by Corydon76 (license 14) + 20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license + 14) Tested by: Corydon76, adomjan ........ + +2008-09-12 22:25 +0000 [r142935] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_local.c, /: Merged revisions 142929 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r142929 | jpeeler | 2008-09-12 17:24:13 -0500 + (Fri, 12 Sep 2008) | 14 lines Merged revisions 142927 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142927 | jpeeler | 2008-09-12 17:22:28 -0500 (Fri, 12 Sep 2008) + | 6 lines (closes issue #12965) Reported by: rlsutton2 Prevents + local channels from playing MOH at each other which was causing + ast_generic_bridge to loop much faster. ........ ................ + +2008-09-12 20:52 +0000 [r142743-142868] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 142866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008) + | 18 lines Merged revisions 142865 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) + | 11 lines Create rules for disallowing contacts at certain + addresses, which may improve the security of various + installations. As this does not change any default behavior, it + is not classified as a direct security fix for anything within + Asterisk, but may help PBX admins better secure their SIP + servers. (closes issue #11776) Reported by: ibc Patches: + 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76, blitzrage ........ ................ + + * /, main/app.c: Merged revisions 142748 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r142748 | + tilghman | 2008-09-12 11:54:44 -0500 (Fri, 12 Sep 2008) | 3 lines + When checking for an encoded character, make sure the string + isn't blank, first. (Closes issue #13470) ........ + + * apps/app_voicemail.c, /: Merged revisions 142745 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r142745 | tilghman | 2008-09-12 11:38:55 -0500 + (Fri, 12 Sep 2008) | 12 lines Merged revisions 142744 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142744 | tilghman | 2008-09-12 11:38:02 -0500 (Fri, 12 Sep 2008) + | 4 lines Missing merge from 1.2 fixes errant exit on DTMF, only + when language is Italian (cf commit 34242) (Closes issue #7353) + ........ ................ + + * /, main/file.c: Merged revisions 142741 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142741 | tilghman | 2008-09-12 11:29:01 -0500 (Fri, 12 Sep 2008) + | 12 lines Merged revisions 142740 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142740 | tilghman | 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008) + | 4 lines Don't return a free'd pointer, when a file cannot be + opened. (closes issue #13462) Reported by: wackysalut ........ + ................ + +2008-09-12 05:03 +0000 [r142632-142678] Steve Murphy <murf@digium.com> + + * apps/app_queue.c, apps/app_dial.c, main/pbx.c, /, + main/features.c, include/asterisk/channel.h: Merged revisions + 142676 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) | + 40 lines Merged revisions 142675 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | + 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is + a "second attempt" to restore the previous "endbeforeh" behavior + in 1.4 and up. In order to capture information concerning all the + legs of transfers in all their infinite combinations, I was + forced to this particular solution by a chain of logical + necessities, the first being that I was not allowed to rewrite + the CDR mechanism from the ground up! This change basically + leaves the original machinery alone, which allows IVR and local + channel type situations to generate CDR's as normal, but a + channel flag can be set to suppress the normal running of the h + exten. That flag would be set by the code that runs the h exten + from the ast_bridge_call routine, to prevent the h exten from + being run twice. Also, a flag in the ast_bridge_config struct + passed into ast_bridge_call can be used to suppress the running + of the h exten in that routine. This would happen, for instance, + if you use the 'g' option in the Dial app. Running this routine + 'early' allows not only the CDR() func to be used in the h + extension for reading CDR variables, but also allows them to be + modified before the CDR is posted to the backends. While I dearly + hope that this patch overcomes all problems, and introduces no + new problems, reality suggests that surely someone will have + problems. In this case, please re-open 13251 (or 13289), and + we'll see if we can't fix any remaining issues. ** trunk note: + some code to suppress the h exten being run from app_queue was + added; for the 'continue' option available only in trunk/1.6.x. + ........ ................ + + * /, main/features.c: Merged revisions 142576 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142576 | murf | 2008-09-11 17:12:53 -0600 (Thu, 11 Sep 2008) | + 28 lines Merged revisions 142575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) | + 20 lines (closes issue #13364) Reported by: mdu113 Well, + fundamentally, the problems revealed in 13364 are because of the + ForkCDR call that is done before the dial. When the bridge is in + place, it's dealing with the first (and wrong) cdr in the list. + So, I wrote a little func to zip down to the first non-locked cdr + in the chain, and thru-out the ast_bridge_call, these results are + used instead of raw chan->cdr and peer->cdr pointers. This + shouldn't affect anyone who isn't forking cdrs before a dial, and + should correct the cdr's of those that do. So, this change ends + up correcting the dstchannel and userfield; the disposition was + fixed by a previous patch, it was OK coming into this problem. + ........ ................ + +2008-09-10 22:18 +0000 [r142478] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 142475 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142475 | murf | 2008-09-10 16:11:27 -0600 (Wed, 10 Sep 2008) | + 38 lines Merged revisions 142474 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) | + 30 lines (closes issue #12318) Reported by: krtorio I made a + small change to the code that handles local channel situations. + In that code, I copy the answer time from the peer cdr, to the + bridge_cdr, but I wasn't also copying the disposition from the + peer cdr. So, Now I copy the disposition, and I've tested against + these cases: 1. phone 1 never answers the phone; no cdr is + generated at all. this should show up as a manager command + failure or something. 2. phone 2 never answers. CDR is generated, + says NO ANSWER 3. phone 2 is busy. CDR is generated, says BUSY 4. + phone 2 answers: CDR is generated, times are correct; disposition + is ANSWERED, which is correct. The start time is the time that + the manager dialed the first phone. The answer time is the time + the second phone picks up. I purposely left the cid and src + fields blank; since this call really originates from the manager, + there is no 'easy' data to put in these fields. If you feel + strongly that these fields should be filled in, re-open this bug + and I'll dig further. ........ ................ + +2008-09-10 19:14 +0000 [r142419] Sean Bright <sean.bright@gmail.com> + + * /, configure, acinclude.m4: Merged revisions 142417 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r142417 | seanbright | 2008-09-10 15:09:03 -0400 + (Wed, 10 Sep 2008) | 17 lines Merged revisions 142416 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142416 | seanbright | 2008-09-10 15:05:46 -0400 (Wed, 10 Sep + 2008) | 9 lines Fix detection of PWLIB and OpenH323 version when + spacing in the headers isn't consistent. (closes issue #13426) + Reported by: bamby Patches: detect_openh323.diff uploaded by + bamby (license 430) (Modified by me to use sed instead of tr) + ........ ................ + +2008-09-10 16:57 +0000 [r142361] Tilghman Lesher <tlesher@digium.com> + + * sounds/Makefile, /: Merged revisions 142359 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142359 | tilghman | 2008-09-10 11:55:31 -0500 (Wed, 10 Sep 2008) + | 10 lines Merged revisions 142358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142358 | tilghman | 2008-09-10 11:54:29 -0500 (Wed, 10 Sep 2008) + | 2 lines Publish new extra sounds version. ........ + ................ + +2008-09-10 16:42 +0000 [r142357] Russell Bryant <russell@digium.com> + + * main/sched.c, /: Merged revisions 142355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142355 | russell | 2008-09-10 11:41:55 -0500 (Wed, 10 Sep 2008) + | 15 lines Merged revisions 142354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008) + | 7 lines It is a normal situation that a task gets put in the + scheduler that should run as soon as possible. Accept "0" as an + acceptable time to run, and also treat negative as "run now", and + don't print a debug message about it. (inspired by a message + asking about the "request to schedule in the past" debug message + on the -dev list) ........ ................ + +2008-09-09 19:18 +0000 [r142082-142221] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 142219 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142219 | mmichelson | 2008-09-09 14:16:30 -0500 (Tue, 09 Sep + 2008) | 22 lines Merged revisions 142218 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142218 | mmichelson | 2008-09-09 14:15:28 -0500 (Tue, 09 Sep + 2008) | 14 lines Make sure that the branch sent in CANCEL + requests matches the branch of the INVITE it is cancelling. + (closes issue #13381) Reported by: atca_pres Patches: + 13381v2.patch uploaded by putnopvut (license 60) Tested by: + atca_pres (closes issue #13198) Reported by: rickead2000 Tested + by: rickead2000 ........ ................ + + * apps/app_queue.c: Merging Revision 142090 from 1.6.0. + + * /, channels/chan_sip.c: Merged revisions 142080 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142080 | mmichelson | 2008-09-09 11:20:41 -0500 (Tue, 09 Sep + 2008) | 29 lines Merged revisions 142079 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep + 2008) | 21 lines When determining if codecs used by SIP peers + allow the media to be natively bridged, use the jointcapability + instead of the peercapability. It seems that the intent of using + the peercapability was to expand the choice of codecs for the + call to increase the chances of being able to native bridge the + channels. The problem is that if a codec were settled on for the + native bridge and that wasn't a codec that was configured to be + used by Asterisk for that peer, then Asterisk would send a + REINVITE with no codecs in the SDP which is a bug no matter how + you slice it. (closes issue #13076) Reported by: ramonpeek + Patches: 13076.patch uploaded by putnopvut (license 60) Tested + by: tbelder ........ ................ + +2008-09-09 15:46 +0000 [r142066] Russell Bryant <russell@digium.com> + + * /, main/features.c: Merged revisions 142064 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008) + | 13 lines Merged revisions 142063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) + | 5 lines Ensure that the stored CDR reference is still valid + after the bridge before poking at it. Also, keep the channel + locked while messing with this CDR. (fixes crashes reported in + issue #13409) ........ ................ + +2008-09-09 12:34 +0000 [r141997-142001] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 141998 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r141998 | + mmichelson | 2008-09-09 07:32:38 -0500 (Tue, 09 Sep 2008) | 7 + lines Use ast_debug for debug messages. I was wondering why debug + messages weren't showing up when I had set the debug level high + for just app_queue.c. It's because we were only checking the + global option_debug variable instead of using the awesome macro + which checks both the global and file-specific value ........ + + * channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 | + mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8 + lines Fix a memory leak in chan_oss (closes issue #13311) + Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel + (license 64) ........ + +2008-09-09 01:51 +0000 [r141951-141952] Russell Bryant <russell@digium.com> + + * main/pbx.c, /: Merged revisions 141807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008) + | 15 lines Merged revisions 141806 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) + | 7 lines When doing an async goto, detect if the channel is + already in the middle of a masquerade. This can happen when + chan_local is trying to optimize itself out. If this happens, + fail the async goto instead of bursting into flames. (closes + issue #13435) Reported by: geoff2010 ........ ................ + + * main/channel.c, /: Merged revisions 141949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 | + russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines + Modify ast_answer() to not hold the channel lock while calling + ast_safe_sleep() or when calling ast_waitfor(). These are + inappropriate times to hold the channel lock. This is what has + caused "could not get the channel lock" messages from chan_sip + and has likely caused a negative impact on performance results of + SIP in Asterisk 1.6. Thanks to file for pointing out this section + of code. (closes issue #13287) (closes issue #13115) ........ + +2008-09-08 22:15 +0000 [r141812-141870] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 141868 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r141868 | + mmichelson | 2008-09-08 17:14:40 -0500 (Mon, 08 Sep 2008) | 4 + lines Um, apparently I didn't actually finish merging before + committing. Bad bad bad ........ + + * /, channels/chan_sip.c: Merged revisions 141810 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141810 | mmichelson | 2008-09-08 16:18:49 -0500 (Mon, 08 Sep + 2008) | 22 lines Merged revisions 141809 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep + 2008) | 14 lines Fix pedantic mode of chan_sip to only check the + remote tag of an endpoint once a dialog has been confirmed. Up + until that point, it is possible and legal for the far-end to + send provisional responses with a different To: tag each time. + With this patch applied, these provisional messages will not + cause a matching problem. (closes issue #11536) Reported by: ibc + Patches: 11536v2.patch uploaded by putnopvut (license 60) + ........ ................ + +2008-09-08 20:20 +0000 [r141747] Jason Parker <jparker@digium.com> + + * Makefile, /, redhat (removed): Merged revisions 141745 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r141745 | qwell | 2008-09-08 15:18:17 -0500 + (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) | + 8 lines Remove RPM package targets from Makefile (and all + associated parts). This has never worked in 1.4, and we decided + that it makes no sense to be done here. There are many distros + out there that already have "proper" spec files that can be + (re)used. Closes issue #13113 Closes issue #10950 Closes issue + #10952 ........ ................ + +2008-09-08 17:15 +0000 [r141684] Sean Bright <sean.bright@gmail.com> + + * build_tools/make_buildopts_h, /: Merged revisions 141682 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon, + 08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on + various platforms doesn't choke on the special characters (like + ^). (closes issue #13417) Reported by: dougm Patches: + 13417.make_buildopts_h.patch uploaded by seanbright (license 71) + Tested by: dougm ........ + +2008-09-06 20:23 +0000 [r141572] Steve Murphy <murf@digium.com> + + * /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 + lines Merged revisions 141565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 + line This fix comes from Joshua Colp The Brilliant, who, given + the trace, came up with a solution. This will most likely will + close 13235 and 13409. I'll wait till Monday to verify, and then + close these bugs. ........ ................ + +2008-09-06 15:28 +0000 [r141506] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 141504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008) + | 12 lines Merged revisions 141503 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008) + | 4 lines Reverting behavior change (AGI should not exit non-zero + on SUCCESS) (closes issue #13434) Reported by: francesco_r + ........ ................ + +2008-09-05 21:13 +0000 [r141369] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 141367 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500 + (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep + 2008) | 7 lines Agent's should not try to call a channel's + indicate callback if the channel has been hung up. It will likely + crash otherwise ABE-1159 ........ ................ + +2008-09-05 14:25 +0000 [r141117-141159] Steve Murphy <murf@digium.com> + + * main/channel.c, /: Merged revisions 141157 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 + lines Merged revisions 141156 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 + line A small change to prevent double-posting of CDR's; thanks to + Daniel Ferrer for bringing it to our attention ........ + ................ + + * res/ael/ael.flex, pbx/ael/ael-test/ref.ael-vtest25 (added), /, + pbx/ael/ael-test/ael-vtest25/extensions.ael, + pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c, + pbx/ael/ael-test/ref.ael-test6: Merged revisions 141115 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, + 04 Sep 2008) | 78 lines Merged revisions 141094 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | + 70 lines (closes issue #13357) Reported by: pj Tested by: murf + (closes issue #13416) Reported by: yarns Tested by: murf If you + find this message overly verbose, relax, it's probably not meant + for you. This message is meant for probably only two people in + the whole world: me, or the poor schnook that has to maintain + this code because I'm either dead or unavailable at the moment. + This fix solves two reports, both having to do with embedding a + function call in a ${} construct. It was tricky because the + funccall syntax has parenthesis () in it. And up till now, the + 'word' token in the flex stuff didn't allow that, because it + would tend to steal the LP and RP tokens. To be truthful, the + "word" token was the trickiest, most unstable thing in the whole + lexer. I was lucky it made this long without complaints. I had to + choose every character in the pattern with extreme care, and I + knew that someday I'd have to revisit it. Well, the day has come. + So, my brilliant idea (and I'm being modest), was to use the + surrounding ${} construct to make a state machine and capture + everything in it, no matter what it contains. But, I have to now + treat the word token like I did with comments, in that I turn the + whole thing into a state-machine sort of spec, with new contexts + "curlystate", "wordstate", and "brackstate". Wait a minute, + "brackstate"? Yes, well, it didn't take very many regression + tests to point out if I do this for ${} constructs, I also have + to do it with the $[] constructs, too. I had to create a separate + pcbstack2 and pcbstack3 because these constructs can occur inside + macro argument lists, and when we have two state machines + operating on the same structures we'd get problems otherwise. I + guess I could have stopped at pcbstack2 and had the brackstate + stuff share it, but it doesn't hurt to be safe. So, the pcbpush + and pcbpop routines also now have versions for "2" and "3". I had + to add the {KEYWORD} construct to the initial pattern for "word", + because previously word would match stuff like "default7", + because it was a longer match than the keyword "default". But, + not any more, because the word pattern only matches only one or + two characters now, and it will always lose. So, I made it the + winner again by making an optional match on any of the keywords + before it's normal pattern. I added another regression test to + make sure we don't lose this in future edits, and had to fix just + one regression, where it no longer reports a 'cascaded' error, + which I guess is a plus. I've given some thought as to whether to + apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I + decided to put it in 1.4 because one of the bug reports was + against 1.4; and it is unexpected that AEL cannot handle this + situation. It actually reduced the amount of useless "cascade" + error messages that appeared in the regressions (by one line, + ehhem). There is a possible side-effect in that it does now do + more careful checking of what's in those ${} constructs, as far + as matching parens, and brackets are concerned. Some users may + find a an insidious problem and correct it this way. This should + be exceedingly rare, I hope. ........ ................ + +2008-09-04 17:28 +0000 [r141042] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c, res/res_agi.c: Merged revisions 141039 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500 + (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) + | 7 lines (closes issue #11979) Fixes multiple parking problems: + Crash when executing a park on an extension dialed by AGI due to + not returning the proper return code. Crash when using a builtin + feature that was a subset of a enabled dynamic feature. Crash due + to always hanging up the peer despite the fact that the peer was + supposed to be parked. ........ ................ + +2008-09-03 20:18 +0000 [r140888-140977] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 140975 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 | + mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4 + lines Fix some locking order issues in app_queue. This was + brought up by atis on IRC a while ago. ........ + + * apps/app_voicemail.c, /: Merged revisions 140887 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r140887 | mmichelson | 2008-09-03 09:41:54 -0500 (Wed, 03 Sep + 2008) | 3 lines Fix compilation ........ + +2008-09-03 14:39 +0000 [r140886] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y, + res/ael/ael.tab.h: Merged revisions 140824 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r140824 | + murf | 2008-09-03 08:01:27 -0600 (Wed, 03 Sep 2008) | 21 lines In + these changes, I have added some explanation of changes to the + Set and MSet apps, so people aren't so shocked and surprised when + they upgrade from 1.4 to 1.6. Also, for the sake of those + upgrading from 1.4 to 1.6 with AEL, I provide automatic support + for the "old" way of using Set(), that still does the exact same + old thing with quotes and backslashes and so on as 1.4 did, by + having AEL compile in the use of MSet() instead of Set(), + everywhere it inserts this code. But, if the app_set var is set + to 1.6 or higher, it uses the "new", non-evaluative Set(). This + only usually happens if the user manually inserts this into the + asterisk.conf file, or runs the "make samples" command. (closes + issue #13249) Reported by: dimas Patches: ael-MSet.diff uploaded + by murf (license 17) Tested by: dimas, murf ........ + +2008-09-03 14:32 +0000 [r140867] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 140860 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r140860 | mmichelson | 2008-09-03 09:31:33 -0500 + (Wed, 03 Sep 2008) | 17 lines Merged revisions 140850 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140850 | mmichelson | 2008-09-03 09:29:15 -0500 (Wed, 03 Sep + 2008) | 9 lines Fix voicemail forwarding when using ODBC storage. + (closes issue #13387) Reported by: moliveras Patches: 13387.patch + uploaded by putnopvut (license 60) Tested by: putnopvut, + moliveras ........ ................ + +2008-09-03 13:27 +0000 [r140819] Russell Bryant <russell@digium.com> + + * main/poll.c, /: Merged revisions 140817 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008) + | 12 lines Merged revisions 140816 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008) + | 4 lines Don't freak out if the poll emulation receives NULL for + the pollfds array (closes issue #13307) Reported by: jcovert + ........ ................ + +2008-09-02 23:51 +0000 [r140755] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 140752 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r140752 | mmichelson | 2008-09-02 18:48:25 -0500 + (Tue, 02 Sep 2008) | 14 lines Merged revisions 140751 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140751 | mmichelson | 2008-09-02 18:47:49 -0500 (Tue, 02 Sep + 2008) | 6 lines After adding the context checking to + app_voicemail for IMAP storage, I left out a crucial place to + copy the context to the vm_state structure. This is the + correction. ........ ................ + +2008-09-02 23:46 +0000 [r140693-140750] Steve Murphy <murf@digium.com> + + * main/cdr.c, /: Merged revisions 140749 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) | + 11 lines Merged revisions 140747 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 + line I am turning the warnings generated in ast_cdr_free and + post_cdr into verbose level 2 messages. Really, they matter + little to end users. You either get the CDR's you wanted, or you + don't, and it is a bug. For trunk, I am going one step further. + These messages were pretty worthless even for debug, so I'm + completely removing them. ........ ................ + + * main/channel.c, /: Merged revisions 140692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | + 13 lines Merged revisions 140690 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 + line After reconsidering, with respect to 13409, ast_cdr_detach + should be OK, better in fact, than ast_cdr_free, which generates + lots of useless warnings that will undoubtably generate + complaints. Hmmm. It doesn't hush the useless warnings, but it + does allow control of posting via the detach and post routines, + for those possible situations, where you'd want to post + single-channel cdrs. ........ ................ + + * main/channel.c, main/pbx.c, /: Merged revisions 140691 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue, + 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | + 14 lines (closes issue #13409) Reported by: tomaso Patches: + asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license + 564) I basically spent the day, verifying that this patch solves + the problem, and doesn't hurt in non-problem cases. Why valgrind + did not plainly reveal this leak absolutely mystifies and stuns + me. Many, many thanks to tomaso for finding and providing the + fix. ........ ................ + +2008-09-02 18:18 +0000 [r140608] Sean Bright <sean.bright@gmail.com> + + * channels/chan_iax2.c, /: Merged revisions 140606 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400 + (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep + 2008) | 8 lines Make sure to use the correct length of the + mohinterpret and mohsuggest buffers when copying configuration + values. (closes issue #13336) Reported by: + decryptus_proformatique Patches: + chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded + by decryptus (license 555) ........ ................ + +2008-09-02 15:13 +0000 [r140565-140568] Russell Bryant <russell@digium.com> + + * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions + 140566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 | + russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines + Update instructions for getting libresample ........ + + * res/ais/amf.c (removed), res/ais/lck.c (removed), /, + res/ais/ckpt.c (removed): Merged revisions 140563 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r140563 | russell | 2008-09-02 10:09:20 -0500 (Tue, 02 Sep 2008) + | 3 lines I'm not sure how these files got to trunk (probably my + fault), but they should not be here ........ + +2008-08-29 17:55 +0000 [r140492] Jeff Peeler <jpeeler@digium.com> + + * CHANGES, /, main/features.c: Merged revisions 140491 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r140491 | jpeeler | 2008-08-29 12:53:32 -0500 (Fri, 29 Aug 2008) + | 2 lines Added the option s to the Park application which will + silence the announcement of the parking space number. Also, fixes + the bug of just clearing the flags instead of actually parsing + the arguments to Park. ........ + +2008-08-29 17:48 +0000 [r140420-140490] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, channels/chan_iax2.c, main/config.c, + main/manager.c, res/ais/lck.c, /, channels/chan_sip.c, + funcs/func_dialgroup.c, res/res_timing_pthread.c, + main/features.c, res/res_phoneprov.c, utils/hashtest2.c, + channels/chan_console.c, main/taskprocessor.c: Merged revisions + 140489 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140489 | mmichelson | 2008-08-29 12:47:17 -0500 (Fri, 29 Aug + 2008) | 30 lines Merged revisions 140488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug + 2008) | 22 lines After working on the ao2_containers branch, I + noticed something a bit strange. In all cases where we provide a + callback function to ao2_container_alloc, the callback function + would only return 0 or CMP_MATCH. After inspecting the + ao2_callback() code carefully, I found that if you're only + looking for one specific item, then you should return CMP_MATCH | + CMP_STOP. Otherwise, astobj2 will continue traversing the current + bucket until the end searching for more matches. In cases like + chan_iax2 where in 1.4, all the peers are shoved into a single + bucket, this makes for potentially terrible performance since the + entire bucket will be traversed even if the peer is one of the + first ones come across in the bucket. All the changes I have made + were for cases where the callback function defined was passed to + ao2_container_alloc so that calls to ao2_find could find a unique + instance of whatever object was being stored in the container. + ........ ................ + + * /, main/file.c: Merged revisions 140433 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r140433 | + mmichelson | 2008-08-29 11:24:37 -0500 (Fri, 29 Aug 2008) | 10 + lines Allow for video files to be opened as well as audio files. + (closes issue #13372) Reported by: epicac Patches: 13372.patch + uploaded by putnopvut (license 60) Tested by: epicac ........ + + * apps/app_voicemail.c, /: Merged revisions 140422 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r140422 | mmichelson | 2008-08-29 11:06:09 -0500 + (Fri, 29 Aug 2008) | 20 lines Merged revisions 140421 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140421 | mmichelson | 2008-08-29 11:01:07 -0500 (Fri, 29 Aug + 2008) | 12 lines Add context checking when retrieving a vm_state. + This was causing a problem for people who had identically named + mailboxes in separate voicemail contexts. This commit affects + IMAP storage only. (closes issue #13194) Reported by: moliveras + Patches: 13194.patch uploaded by putnopvut (license 60) Tested + by: putnopvut, moliveras ........ ................ + + * /, channels/chan_sip.c: Merged revisions 140418 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140418 | mmichelson | 2008-08-29 10:32:02 -0500 (Fri, 29 Aug + 2008) | 18 lines Merged revisions 140417 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140417 | mmichelson | 2008-08-29 10:26:52 -0500 (Fri, 29 Aug + 2008) | 10 lines Fix SIP's parsing so that if a port is specified + in a string to Dial(), it is not ignored. (closes issue #13355) + Reported by: acunningham Patches: 13355v2.patch uploaded by + putnopvut (license 60) Tested by: acunningham ........ + ................ + +2008-08-27 20:14 +0000 [r140303] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 140301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug + 2008) | 19 lines Merged revisions 140299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug + 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when + in pedantic mode. The problem was that the wrong tags would be + compared depending on the direction of the call. (closes issue + #13353) Reported by: flefoll Patches: + chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll + (license 244) ........ ................ + +2008-08-26 18:50 +0000 [r140206] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 140205 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r140205 | jpeeler | 2008-08-26 13:48:55 -0500 + (Tue, 26 Aug 2008) | 17 lines Merged revisions 140056 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26 Aug 2008) + | 9 lines (closes issue #12071) Reported by: tzafrir Patches: + dahdi_close.diff uploaded by tzafrir (license 46) Tested by: + tzafrir, jpeeler This patch fixes closing open file descriptors + in the case of an error. ........ ................ + +2008-08-26 18:12 +0000 [r140055-140171] Russell Bryant <russell@digium.com> + + * Makefile, /: Merged revisions 140169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 | + russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines + Fix building menuselect-tree with PRINT_DIR set. We _must_ use + the --quiet flag here, or else some arbitrary text will end up in + the resulting menuselect-tree file and things will explode. + ........ + + * /, channels/chan_sip.c: Merged revisions 140061 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140061 | russell | 2008-08-26 11:10:06 -0500 (Tue, 26 Aug 2008) + | 14 lines Merged revisions 140060 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140060 | russell | 2008-08-26 11:07:58 -0500 (Tue, 26 Aug 2008) + | 6 lines Fix some bogus scheduler usage in chan_sip. This code + used the return value of a completely unrelated function to + determine whether the scheduler should be run or not. This would + have caused the scheduler to not run in cases where it should + have. Also, leave a note about another scheduler issue that needs + to be addressed at some point. ........ ................ + + * channels/chan_iax2.c, /: Merged revisions 140053 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r140053 | russell | 2008-08-26 10:29:25 -0500 + (Tue, 26 Aug 2008) | 23 lines Merged revisions 140051 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140051 | russell | 2008-08-26 10:27:23 -0500 (Tue, 26 Aug 2008) + | 15 lines Fix a race condition with the IAX scheduler thread. A + lock and condition are used here to allow newly scheduled tasks + to wake up the scheduler just in case the new task needs to run + sooner than the current wakeup time when the thread is sleeping. + However, there was a race condition such that a newly scheduled + task would not properly wake up the scheduler or affect the wake + up period. The order of execution would have been: 1) Scheduler + thread determines wake up time of N ms. 2) Another thread + schedules a task and signals the condition, with an execution + time of < N ms. 3) Scheduler thread locks and goes to sleep for N + ms. By moving the sleep time determination to inside the critical + section, this possibility is avoided. ........ ................ + +2008-08-25 21:49 +0000 [r139929] Jeff Peeler <jpeeler@digium.com> + + * main/manager.c, /: Merged revisions 139928 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139928 | jpeeler | 2008-08-25 16:48:51 -0500 (Mon, 25 Aug 2008) + | 11 lines Merged revisions 139927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139927 | jpeeler | 2008-08-25 16:47:33 -0500 (Mon, 25 Aug 2008) + | 3 lines Fix a typo I made. Lesson learned, apply the patch if + one exists. ........ ................ + +2008-08-25 21:34 +0000 [r139919] Sean Bright <sean.bright@gmail.com> + + * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged + revisions 139915 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug + 2008) | 17 lines Merged revisions 139909 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug + 2008) | 9 lines Some versions of awk (nawk, for example) don't + like empty regular expressions so be slightly more verbose. + (closes issue #13374) Reported by: dougm Patches: 13374.diff + uploaded by seanbright (license 71) Tested by: dougm ........ + ................ + +2008-08-25 21:13 +0000 [r139874] Terry Wilson <twilson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008) + | 10 lines Merged revisions 139869 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008) + | 2 lines Make SIPADDHEADER() propagate indefinitely ........ + ................ + +2008-08-25 16:05 +0000 [r139778] Steve Murphy <murf@digium.com> + + * main/pbx.c, /, main/features.c: Merged revisions 139770 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon, + 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 + lines This patch reverts the changes made via 139347, and 139635, + as users are seeing adverse difference. I will un-close 13251. + Back to the drawing board/ concept/ beginning/ whatever! ........ + ................ + +2008-08-24 16:36 +0000 [r139709] Tilghman Lesher <tlesher@digium.com> + + * /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 | + tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines + Memory leak ........ + +2008-08-22 22:37 +0000 [r139629-139674] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 139662 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) | + 14 lines Merged revisions 139635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 + lines I found some problems with the code I committed earlier, + when I merged them into trunk, so I'm coming back to clean up. + And, in the process, I found an error in the code I added to + trunk and 1.6.x, that I'll fix using this patch also. ........ + ................ + + * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions + 139627 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | + 59 lines Merged revisions 139347 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | + 47 lines (closes issue #13251) Reported by: sergee Tested by: + murf THis is a bold move for a static release fix, but I wouldn't + have made it if I didn't feel confident (at least a *bit* + confident) that it wouldn't mess everyone up. The reasoning goes + something like this: 1. We simply cannot do anything with CDR's + at the current point (in pbx.c, after the __ast_pbx_run loop). + It's way too late to have any affect on the CDRs. The CDR is + already posted and gone, and the remnants have been cleared. 2. I + was very much afraid that moving the running of the 'h' extension + down into the bridge code (where it would be now practical to do + it), would result in a lot more calls to the 'h' exten, so I + implemented it as another exten under another name, but found, to + my pleasant surprise, that there was a 1:1 correspondence to the + running of the 'h' exten in the pbx_run loop, and the new spot at + the end of the bridge. So, I ifdef'd out the current 'h' loop, + and moved it into the bridge code. The only difference I can see + is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this + is still an important decision point, I can replicate it if there + are complaints. To be perfectly honest, the KEEPALIVE situation + is not totally clear to me, and how it relates to a post-bridge + situation is less clear. I suspect the users will point out + everything in total clarity if this steps on anyone's toes! 3. I + temporarily swap the bridge_cdr into the channel before running + the 'h' exten, which makes it possible for users to edit the cdr + before it goes out the door. And, of course, with the + endbeforehexten config var set, the users can also get at the + billsec/duration vals. After the h exten finishes, the cdr is + swapped back and processing continues as normal. Please, all who + deal with CDR's, please test this version of Asterisk, and file + bug reports as appropriate! ........ I also made a little fix to + the app_dial's 'e' option, that is related to my updates. + ................ + +2008-08-22 21:58 +0000 [r139623-139625] Jeff Peeler <jpeeler@digium.com> + + * main/manager.c, /: Merged revisions 139624 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139624 | jpeeler | 2008-08-22 16:57:32 -0500 (Fri, 22 Aug 2008) + | 13 lines Merged revisions 139621 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008) + | 5 lines (closes issue #13359) Reported by: Laureano Patches: + originate_channel_check.patch uploaded by Laureano (license 265) + ........ ................ + + * /, main/features.c: Merged revisions 139622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r139622 | + jpeeler | 2008-08-22 16:52:20 -0500 (Fri, 22 Aug 2008) | 1 line + remove extra comma typo ........ + +2008-08-22 20:21 +0000 [r139459-139565] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 139563 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r139563 | + mmichelson | 2008-08-22 15:20:58 -0500 (Fri, 22 Aug 2008) | 15 + lines The -1 return value from incomplete or improper headers for + the SipNotify manager command was causing the current manager + session to become disconnected. Change the return value to 0 for + these cases. Also change a test for a NULL pointer to be + ast_strlen_zero instead. (closes issue #13351) Reported by: + Laureano Patches: sipnotify_action_fix.patch uploaded by Laureano + (license 265) ........ + + * /, main/features.c: Merged revisions 139558 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r139558 | + mmichelson | 2008-08-22 15:02:35 -0500 (Fri, 22 Aug 2008) | 9 + lines Add missing unique id to ParkedCallGiveUp and + ParkedCallTimeOut manager events (closes issue #13358) Reported + by: srt Patches: 13358_parking_events.diff uploaded by srt + (license 378) ........ + + * include/asterisk/threadstorage.h, /: Merged revisions 139554 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500 + (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug + 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is + selected (closes issue #13298) Reported by: snuffy Patches: + bug13298_20080822.diff uploaded by snuffy (license 35) ........ + ................ + + * channels/chan_iax2.c, /: Merged revisions 139469 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500 + (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug + 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........ + ................ + + * channels/chan_iax2.c, /: Merged revisions 139457 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500 + (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug + 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from + incorrect locking order between iax2_pvt and ast_channel + structures. AST-13 ........ ................ + +2008-08-21 23:44 +0000 [r139399] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500 + (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008) + | 3 lines Fixes loop that could possibly never exit in the event + of a channel never being able to be opened or specify after a + restart. (closes issue #11017) ........ ................ + +2008-08-20 22:19 +0000 [r139217] Russell Bryant <russell@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008) + | 19 lines Merged revisions 139213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) + | 11 lines Fix a crash in the ChanSpy application. The issue here + is that if you call ChanSpy and specify a spy group, and sit in + the application long enough looping through the channel list, you + will eventually run out of stack space and the application with + exit with a seg fault. The backtrace was always inside of a + harmless snprintf() call, so it was tricky to track down. + However, it turned out that the call to snprintf() was just the + biggest stack consumer in this code path, so it would always be + the first one to hit the boundary. (closes issue #13338) Reported + by: ruddy ........ ................ + +2008-08-20 22:07 +0000 [r139212] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c: Merged revisions 139210 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r139210 | + qwell | 2008-08-20 17:06:40 -0500 (Wed, 20 Aug 2008) | 7 lines + Fix output of sipshowpeer manager response. (closes issue #13346) + Reported by: srt Patches: + 13346_malformed_sip_show_peer_response.diff uploaded by srt + (license 378) ........ + +2008-08-20 17:34 +0000 [r139106] Steve Murphy <murf@digium.com> + + * main/cdr.c, /: Merged revisions 139083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) | + 20 lines Merged revisions 139074 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | + 12 lines (closes issue #13263) Reported by: brainy Tested by: + murf The specialized reset routine is tromping on the flags field + of the CDR. I made a change to not reset the DISABLED bit. This + should get rid of this problem. ........ ................ + +2008-08-20 15:39 +0000 [r138888-139018] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug + 2008) | 14 lines Merged revisions 139015 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug + 2008) | 6 lines sip_read should properly handle a NULL return + from sip_rtp_read. (closes issue #13257) Reported by: travishein + ........ ................ + + * /, channels/chan_agent.c: Merged revisions 138943 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r138943 | mmichelson | 2008-08-19 18:19:40 -0500 + (Tue, 19 Aug 2008) | 19 lines Merged revisions 138942 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138942 | mmichelson | 2008-08-19 18:17:17 -0500 (Tue, 19 Aug + 2008) | 11 lines Reset agent_pvt variables back to the values in + agents.conf (from what the corresponding channel variables were + set to) when the agent logs out. (closes issue #13098) Reported + by: davidw Patches: + 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by + bbryant (license 36) Tested by: davidw ........ ................ + + * apps/app_chanspy.c, /: Merged revisions 138887 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r138887 | mmichelson | 2008-08-19 13:52:04 -0500 (Tue, 19 Aug + 2008) | 31 lines Merged revisions 138886 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug + 2008) | 23 lines Add a lock and unlock prior to the destruction + of the chanspy_ds lock to ensure that no other threads still have + it locked. While this should not happen under normal + circumstances, it appears that if the spyer and spyee hang up at + nearly the same time, the following may occur. 1. + ast_channel_free is called on the spyee's channel. 2. The chanspy + datastore is removed from the spyee's channel in + ast_channel_free. 3. In the spyer's thread, the spyer attempts to + remove and destroy the datastore from the spyee channel, but the + datastore has already been removed in step 2, so the spyer + continues in the code. 4. The spyee's thread continues and calls + the datastore's destroy callback, chanspy_ds_destroy. This + involves locking the chanspy_ds. 5. Now the spyer attempts to + destroy the chanspy_ds lock. The problem is that in step 4, the + spyee has locked this lock, meaning that the spyer is attempting + to destroy a lock which is currently locked by another thread. + The backtrace provided in issue #12969 supports the idea that + this is possible (and has even occurred). This commit does not + close the issue, but should help in preventing one type of crash + associated with the use of app_chanspy. ........ ................ + +2008-08-19 17:01 +0000 [r138853-138855] Steve Murphy <murf@digium.com> + + * utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y, + res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug + 2008) | 1 line Oops. put a decl in a generated file. My bad, but + fixed now. ........ + + * main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y, + res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 | + murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines + These changes are in regards to bug 13249, where users are being + surprised by the changes made to the Set app in trunk/1.6.x, as + they come from the 1.4 world. They are only bitten if they write + their AEL dialplan in the 1.4 world, and then carry it over to a + trunk/1.6.x installation where a "make samples" was executed, or + where they hand-edited the asterisk.conf file and added the + [compat] category with app_set = 1.6 (or higher). (this commit + does not totally solve 13249, at least not yet) The change + involves issueing a single warning while the AEL file is loading, + if: 1. app_set is present in the config file, and set to 1.6 or + higher. 2. there are double quotes in an assignment statement (eg + x = "hi there";) 3. the warning was not already issued. The + standalone app, aelparse, does not (yet) issue this warning. I'd + have to have it read in the asterisk.conf file, and that's a bit + of hassle. I'll add it if users request it, tho. ........ + +2008-08-19 00:17 +0000 [r138777-138782] Sean Bright <sean.bright@gmail.com> + + * /, channels/chan_sip.c: Merged revisions 138778-138780 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, + 18 Aug 2008) | 1 line While we're at it, make this machine + parseable too. ........ r138779 | seanbright | 2008-08-18 + 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we + don't need anymore. ........ r138780 | seanbright | 2008-08-18 + 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now, + too (woops) ........ + + * /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 | + seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3 + lines Change event header to RegistrationTime to be more + consistent (and avoid breaking existing frameworks). Pointed out + by Laureano on #asterisk-dev. ........ + +2008-08-18 20:24 +0000 [r138689-138698] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions + 138694 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138694 | + mmichelson | 2008-08-18 15:23:11 -0500 (Mon, 18 Aug 2008) | 10 + lines Change the queue timeout priority logic into less ugly and + confusing code pieces. Clarify the logic within + queues.conf.sample. (closes issue #12690) Reported by: atis + Patches: queue_timeoutpriority.patch uploaded by atis (license + 242) ........ + + * apps/app_queue.c, /: Merged revisions 138687 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug + 2008) | 18 lines Merged revisions 138685 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug + 2008) | 10 lines Change the inequalities used in app_queue with + regards to timeouts from being strict to non-strict for more + accuracy. (closes issue #13239) Reported by: atis Patches: + app_queue_timeouts_v2.patch uploaded by atis (license 242) + ........ ................ + +2008-08-18 15:55 +0000 [r138633] Jason Parker <jparker@digium.com> + + * Makefile, /: Merged revisions 138631 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 | + qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line + Remove option that isn't valid here. ........ + +2008-08-18 02:14 +0000 [r138520] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008) + | 1 line add missing define for SS7 in dahdi_restart ........ + +2008-08-17 14:27 +0000 [r138444-138498] Sean Bright <sean.bright@gmail.com> + + * /, main/features.c: Merged revisions 138482 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 | + seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6 + lines Move Uniqueid to the end of the event for those that rely + on the position of the name/value pairs, pointed out by + snuffy-home on #asterisk-commits. For those of you who rely on + the position of name/value pairs in manager events... stop... + that is why associative arrays were invented. ........ + + * /, main/features.c: Merged revisions 138479 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 | + seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7 + lines Add Uniqueid header to ParkedCall manager event. (closes + issue #13323) Reported by: srt Patches: + 13323_unique_id_for_parkedcalls_event.diff uploaded by srt + (license 378) ........ + + * main/rtp.c, /: Merged revisions 138476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 | + seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7 + lines Add missing colons to RTCPReceived and RTCPSent manager + events. (closes issue #13319) Reported by: srt Patches: + 13319_rtcp_manager_event_headers.diff uploaded by srt (license + 378) ........ + + * channels/chan_iax2.c, /: Merged revisions 138473 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug + 2008) | 7 lines Fix the output of the JitterBufStats manager + event. (closes issue #13324) Reported by: srt Patches: + 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt + (license 378) ........ + + * configs/cdr_tds.conf.sample, /: Merged revisions 138442 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat, + 16 Aug 2008) | 4 lines Since it's introduction in revision 3497, + cdr_tds has *never* read the port configuration option from + cdr_tds.conf. So go ahead and remove it from the sample config. + ........ + +2008-08-16 13:08 +0000 [r138411-138414] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008) + | 2 lines Fix compilation warnings (found with dev-mode) ........ + + * main/pbx.c, /: Merged revisions 138409 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138409 | + tilghman | 2008-08-16 07:52:06 -0500 (Sat, 16 Aug 2008) | 3 lines + Also make sure hinting won't crash on reload. (Closes issue + #13312) ........ + +2008-08-16 01:14 +0000 [r138359-138363] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500 + (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15 + Aug 2008) | 1 line fixes use count to properly decrement if an + active dahdi channel is destroyed allowing module to be unloaded + ........ ................ + + * channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500 + (Fri, 15 Aug 2008) | 20 lines Merged revisions + 138119,138151,138238 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) + | 4 lines Fixes the dahdi restart functionality. Dahdi restart + allows one to restart all DAHDI channels, even if they are + currently in use. This is different from unloading and then + loading the module since unloading requires the use count to be + zero. Reloading the module is different in that the signalling is + not changed from what it was originally configured. Also, this + fixes not closing all the file descriptors for D-channels upon + module unload (which would prevent loading the module + afterwards). (closes issue #11017) ........ r138151 | jpeeler | + 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared + static mutexes using AST_MUTEX_DEFINE_STATIC macro ........ + r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) + | 1 line initialize condition variable ss_thread_complete using + ast_cond_init ........ ................ + +2008-08-15 22:56 +0000 [r138208-138261] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 138260 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) + | 16 lines Merged revisions 138258 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) + | 8 lines More fixes for realtime peers. (closes issue #12921) + Reported by: Nuitari Patches: 20080804__bug12921.diff.txt + uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt + uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ + ................ + + * /: Merged revisions 138207 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ ........ + +2008-08-15 20:28 +0000 [r138184] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 138155 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r138155 | jpeeler | 2008-08-15 15:12:19 -0500 (Fri, 15 Aug 2008) + | 1 line rename all zfd instances in chan_dahdi to dfd to match + 1.4 (left over from DAHDI transition) ........ + +2008-08-15 20:21 +0000 [r138158] Russell Bryant <russell@digium.com> + + * main/autoservice.c, /: Merged revisions 138028 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008) + | 17 lines Merged revisions 138027 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) + | 9 lines Ensure that when a hangup occurs in autoservice, that a + hangup frame gets properly deferred to be read from the channel + owner when it gets taken out of autoservice. (closes issue + #12874) Reported by: dimas Patches: v1-12874.patch uploaded by + dimas (license 88) ........ ................ + +2008-08-15 19:37 +0000 [r138026-138150] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 138148 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138148 | + tilghman | 2008-08-15 14:36:11 -0500 (Fri, 15 Aug 2008) | 2 lines + Change free to ast_free_ptr, too ........ + + * main/pbx.c, /: Merged revisions 138124 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138124 | + tilghman | 2008-08-15 14:22:48 -0500 (Fri, 15 Aug 2008) | 4 lines + e->data can be NULL, so use the safe version of ast_strdup() + (closes issue #13312) Reported by: pj ........ + + * /, channels/chan_sip.c: Merged revisions 138086 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138086 | + tilghman | 2008-08-15 13:02:15 -0500 (Fri, 15 Aug 2008) | 2 lines + regseconds is actually stored as the epoch time, not registration + length ........ + + * /, funcs/func_strings.c: Merged revisions 138024 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500 + (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008) + | 8 lines Additional check for more string specifiers than + arguments. (closes issue #13299) Reported by: adomjan Patches: + 20080813__bug13299.diff.txt uploaded by Corydon76 (license 14) + func_strings.c-sprintf.patch uploaded by adomjan (license 487) + Tested by: adomjan ........ ................ + +2008-08-14 22:43 +0000 [r137989] Russell Bryant <russell@digium.com> + + * /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 | + russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines + Fix a bashism that causes an error when trying to build the pdf + on ubuntu ........ + +2008-08-14 18:50 +0000 [r137935] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug + 2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes + issue #13304) Reported by: eliel Patches: sqlite.patch uploaded + by eliel (license 64) (Slightly modified by me) ........ + +2008-08-14 18:15 +0000 [r137904] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500 + (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) + | 9 lines When creating the secondary subchannel name, it is + necessary to compare to the existing channel name without the + "Zap/" or "DAHDI/" prefix, since our test string is also without + that prefix. (closes issue #13027) Reported by: dferrer Patches: + chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525) + (Slightly modified by me, to compensate for both names) ........ + ................ + +2008-08-14 15:39 +0000 [r137815] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 | + qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines + Make sure we set the socket port, so we don't try to use <ip + address>:0. (closes issue #13255) Reported by: falves11 Patches: + 13255-socketport.diff uploaded by qwell (license 4) Tested by: + falves11 ........ + +2008-08-14 15:35 +0000 [r137813] Russell Bryant <russell@digium.com> + + * /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r137732 | russell | 2008-08-14 09:15:50 -0500 + (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) + | 4 lines Comments in this config file were aligned only if your + tab size was set to 8. So, convert tabs to spaces so that things + should be aligned regardless of what tab size you use in your + editor. ........ ................ + +2008-08-14 15:06 +0000 [r137782] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 | + seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8 + lines If we detect that we are no longer connected, try to + reconnect a few times before giving up. This relies on the + timeout settings in the freetds.conf file and, unfortunately, on + a recent version of FreeTDS (0.82 or newer). I either need to + change the current execs to be non-blocking (which I do not want + to do) or we have to force people to run with the latest and + greatest of FreeTDS. I'm on the fence... ........ + +2008-08-14 02:08 +0000 [r137648-137683] Kevin P. Fleming <kpfleming@digium.com> + + * /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug + 2008) | 9 lines Merged revisions 137679 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug + 2008) | 1 line forgot one module name that changed ........ + ................ + + * /: configure for merging from trunk + + * / (added): now that 1.6.0 has reached the 'release candidate' + stage, it's time to branch 1.6.1 + +2008-08-13 23:00 +0000 [r137627-137640] Kevin P. Fleming <kpfleming@digium.com> + + * build_tools/prep_tarball: make this script actually work + + * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions + 137530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug + 2008) | 1 line add document describing what users will need to be + aware of when upgrading to this version and using DAHDI ........ + +2008-08-13 21:08 +0000 [r137496-137532] Jason Parker <jparker@digium.com> + + * channels/chan_sip.c: Correctly end locally ended calls. (closes + issue #12170) Reported by: pj Patches: + 20080702__issue12170_clear_pendinginvite.diff uploaded by bbryant + (license 36) Tested by: bbryant, pabelanger + + * apps/app_fax.c: Add FAXMODE variable with what fax transport was + used. (closes issue #13252) Patches: v1-13252.patch uploaded by + dimas (license 88) + +2008-08-13 17:36 +0000 [r137456] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c: Convert deprecated routines to the new names. + (closes issue #13297) Reported by: snuffy Patches: + bug13297_20080814.diff uploaded by snuffy (license 35) + +2008-08-13 14:41 +0000 [r137403-137406] Sean Bright <sean.bright@gmail.com> + + * /, doc/tex/cdrdriver.tex: Merged revisions 137405 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, + 13 Aug 2008) | 1 line Update docs to reflect the change to + cdr_tds ........ + + * cdr/cdr_tds.c: Use the ast_vasprintf macro instead of vasprintf + directly. + +2008-08-12 19:48 +0000 [r137299-137301] Russell Bryant <russell@digium.com> + + * doc/tex/asterisk.tex: Grammar hax from Qwell + + * doc/tex/asterisk.tex: Note that developer documentation belongs + in doxygen, and not integrated with the user manual stuff in + doc/tex/. + +2008-08-11 16:14 +0000 [r137239] Russell Bryant <russell@digium.com> + + * Makefile: Make PRINT_DIR work as advertised. + +2008-08-11 14:25 +0000 [r137203] Sean Bright <sean.bright@gmail.com> + + * UPGRADE.txt, cdr/cdr_tds.c: Log the userfield CDR variable like + the other CDR backends, assuming the column is actually there. If + it's not, we still log everything else as before. (closes issue + #13281) Reported by: falves11 + +2008-08-11 00:25 +0000 [r137150] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_odbc.c: Merged revisions 137138 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008) + | 5 lines Deallocate database connection handle on disconnect, as + we allocate another one on connect. (closes issue #13271) + Reported by: dveiga ........ + +2008-08-10 21:10 +0000 [r137028-137112] Sean Bright <sean.bright@gmail.com> + + * include/asterisk/channel.h: Fix this again so we can compile with + shadow warnings enabled and IMAP chosen in voicemail. + + * main/udptl.c, main/say.c, main/taskprocessor.c, main/sched.c: + That's all, folks. Not going to update the Makefile until + res_jabber is converted (snuffy, you there? :)) + + * main/channel.c, main/pbx.c, main/frame.c, main/logger.c, + apps/app_queue.c, main/indications.c, main/asterisk.c, + main/rtp.c, apps/app_voicemail.c, main/cli.c: Another batch of + files from RSW. The remaining apps and a few more files from + main/ + + * main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_delete.c, + main/jitterbuf.c, main/acl.c, main/db1-ast/recno/rec_put.c, + main/astobj2.c, main/config.c, main/rtp.c, main/channel.c, + main/cdr.c, main/manager.c, main/tdd.c, main/features.c, + main/abstract_jb.c, main/file.c, main/http.c, main/callerid.c, + main/app.c, main/event.c, main/audiohook.c, + main/db1-ast/btree/bt_delete.c, main/asterisk.c: Another big + chunk of changes from the RSW branch. Bunch of stuff from main/ + + * apps/app_dial.c, apps/app_dahdibarge.c, apps/app_meetme.c, + apps/app_festival.c, apps/app_record.c, apps/app_dahdiscan.c, + apps/app_disa.c, apps/app_waituntil.c, apps/app_playback.c, + apps/app_forkcdr.c, apps/app_osplookup.c, apps/app_minivm.c, + apps/app_macro.c, apps/app_sms.c, apps/app_directory.c, + apps/app_rpt.c, apps/app_while.c, apps/app_adsiprog.c: More RSW + merges. Everything from apps/ except for the big offenders + app_voicemail and app_queue. + + * res/res_config_pgsql.c, res/res_smdi.c, res/res_timing_pthread.c, + res/res_adsi.c, res/res_agi.c, res/res_phoneprov.c, + res/ael/ael_lex.c, res/res_musiconhold.c, res/ael/ael.flex, + res/res_config_ldap.c, res/res_odbc.c: All of the res/ stuff + (other than res_jabber) from the RSW branch. + +2008-08-09 15:26 +0000 [r136947] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged + revisions 136946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500 + (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) + | 2 lines Regression fixes for Solaris ........ ................ + +2008-08-09 14:12 +0000 [r136888-136917] Sean Bright <sean.bright@gmail.com> + + * channels/chan_unistim.c, channels/chan_sip.c, + channels/chan_skinny.c, codecs/codec_dahdi.c, + channels/chan_iax2.c, channels/xpmr/xpmr.c, + channels/iax2-parser.c, channels/chan_mgcp.c: More RSW merges. + This should do it for the channels/ dir. + + * channels/chan_dahdi.c: Biggest offender? chan_dahdi.c! More RSW + merging. + + * channels/chan_jingle.c, channels/chan_phone.c, + channels/chan_agent.c, channels/chan_features.c, + channels/chan_alsa.c, channels/chan_console.c: Merge more changes + from the resolve-shadow-warnings branch (henceforth known as RSW + since i am too lazy to keep typing it all out). This time a few + of the channels. + +2008-08-09 01:15 +0000 [r136859] Tilghman Lesher <tlesher@digium.com> + + * res/res_agi.c: Update documentation as to the behavior of AGI in + 1.6.0 and higher. Also, add an OOB message that answers the + question of, if AGI no longer shuts down the connection on + hangup, how will FastAGI know when to stop processing the call? + +2008-08-08 18:19 +0000 [r136819] Sean Bright <sean.bright@gmail.com> + + * configure, configure.ac, makeopts.in: Bring in the configure and + makeopts jazz for -Wshadow, but don't add it to the Makefile yet. + +2008-08-08 15:58 +0000 [r136787] Dwayne M. Hubbard <dhubbard@digium.com> + + * channels/chan_dahdi.c: use ARRAY_LEN + +2008-08-08 15:31 +0000 [r136784] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Fix compilation for ODBC voicemail + +2008-08-08 02:34 +0000 [r136751] Tilghman Lesher <tlesher@digium.com> + + * /: Removing bad properties + +2008-08-08 00:48 +0000 [r136746] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, /, pbx/ael/ael-test/ref.ael-ntest10, + include/asterisk/ael_structs.h, pbx/ael/ael-test/ref.ael-test8, + pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 136726 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | + 32 lines (closes issue #13236) Reported by: korihor Wow, this one + was a challenge! I regrouped and ran a new strategy for setting + the ~~MACRO~~ value; I set it once per extension, up near the + top. It is only set if there is a switch in the extension. So, I + had to put in a chunk of code to detect a switch in the pval + tree. I moved the code to insert the set of ~~exten~~ up to the + beginning of the gen_prios routine, instead of down in the switch + code. I learned that I have to push the detection of the switches + down into the code, so everywhere I create a new exten in + gen_prios, I make sure to pass onto it the values of the + mother_exten first, and the exten next. I had to add a couple + fields to the exten struct to accomplish this, in the + ael_structs.h file. The checked field makes it so we don't repeat + the switch search if it's been done. I also updated the + regressions. ........ + +2008-08-07 23:39 +0000 [r136715-136722] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Remove one last batch of debug messages + + * apps/app_voicemail.c: Fix build for non-IMAP storage and get rid + of some debug messages. Thanks to eliel for alerting me. No + thanks to buildbot. + + * /, apps/app_voicemail.c: Merging the imap_consistency_trunk + branch to trunk. For an explanation of what "imap_consistency" + is, please see svn revision 134223 to the 1.4 branch. + Coincidentally, this also fixes a recent bug report regarding the + inability to save messages to the new folder when using IMAP + storage since they will would be flagged as "seen" and not be + recognized as new messages. (closes issue #13234) Reported by: + jaroth + +2008-08-07 21:19 +0000 [r136679] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: show correct called party id and also + store this to the 'placed calls' list once the call is connected. + (closes issue #13180) Reported by: pj Patches: + 2008080700_skinny_calledpartyid.diff uploaded by mvanbaak + (license 7) Tested by: mvanbaak, pj + +2008-08-07 20:54 +0000 [r136676] Shaun Ruffell <sruffell@digium.com> + + * codecs/codec_dahdi.c: Updating codec_dahdi to the new transcoder + interface. + +2008-08-07 20:25 +0000 [r136631-136660] Mark Michelson <mmichelson@digium.com> + + * main/features.c: Bump a LOG_NOTICE message to LOG_DEBUG since it + appears once for every bridged call + + * main/pbx.c: Don't allow Answer() to accept a negative argument. + Negative argument means an infinite delay and we don't want that. + + * main/channel.c: Fix a calculation error I had made in the poll. + The poll would reset to 500 ms every time a non-voice frame was + received. The total time we poll should be 500 ms, so now we save + the amount of time left after the poll returned and use that as + our argument for the next call to poll + + * main/channel.c: Scrap the 500 ms delay when Asterisk auto-answers + a channel. Instead, poll the channel until receiving a voice + frame. The cap on this poll is 500 ms. The optional delay is + still allowable in the Answer() application, but the delay has + been moved back to its original position, after the call to the + channel's answer callback. The poll for the voice frame will not + happen if a delay is specified when calling Answer(). (closes + issue #12708) Reported by: kactus + +2008-08-07 19:01 +0000 [r136594] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /, configs/misdn.conf.sample, + channels/misdn_config.c: Merged revisions 136241 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 + Aug 2008) | 5 lines * The allowed_bearers setting in misdn.conf + misspelled one of its options: digital_restricted. * Fixed some + other spelling errors and typos. ........ + +2008-08-07 17:44 +0000 [r136504-136542] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/doxyref.h, /: Merged revisions 136541 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ ........ + + * apps/app_jack.c: stop using deprecated API call + +2008-08-07 16:55 +0000 [r136489] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_queue.c: Merged revisions 136488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008) + | 7 lines Update persistent state on all exit conditions. (closes + issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt + uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon + ........ + +2008-08-07 16:29 +0000 [r136477] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_voicemail.c: fix some format strings to actually compile + without errors + +2008-08-07 15:16 +0000 [r136408] Sean Bright <sean.bright@gmail.com> + + * codecs/Makefile, utils/muted.c, utils/astman.c, utils/smsq.c, + codecs/codec_dahdi.c, formats/msgsm.h, utils/extconf.c, + utils/frame.c: More merges from resolve-shadow warnings: utils/ + codecs/ and a change I missed from formats/ + +2008-08-07 15:10 +0000 [r136406] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_tds.c: Fix runtime symbol error + +2008-08-07 14:36 +0000 [r136298-136402] Sean Bright <sean.bright@gmail.com> + + * include/asterisk/callerid.h, include/asterisk/strings.h: Merge in + a few more changes. This time the include/ directory. + + * funcs/func_config.c, funcs/func_timeout.c, funcs/func_odbc.c, + funcs/func_strings.c: Continue merging in changes from + resolve-shadow-warnings. funcs/ this time. + + * cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, + cdr/cdr_tds.c, cdr/cdr_csv.c: More from the + resolve-shadow-warnings branch. This time the cdr/ directory. + + * formats/format_pcm.c, pbx/pbx_dundi.c, formats/msgsm.h, + pbx/dundi-parser.c, pbx/pbx_config.c: Start moving in changes + from my resolve-shadow-warnings branch. Going to do this in + pieces so the diffs are a little bit smaller and more reviewable. + pbx/ and formats/ first. + +2008-08-06 21:22 +0000 [r136245] Dwayne M. Hubbard <dhubbard@digium.com> + + * main/taskprocessor.c: move taskprocessor CLI commands into the + core namespace + +2008-08-06 20:15 +0000 [r136112-136191] Tilghman Lesher <tlesher@digium.com> + + * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136190 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008) + | 4 lines -C option takes a filename, not a directory path. + (closes issue #13007) Reported by: klaus3000 ........ + + * apps/app_meetme.c: Janitor ast_str project (closes issue #13058) + Reported by: pputman Patches: app_meetme_aststr2.patch uploaded + by pputman (license 81) + + * funcs/func_dialgroup.c: Persist DIALGROUP() values in astdb + (closes issue #13138) Reported by: Corydon76 Patches: + 20080725__bug13138.diff.txt uploaded by Corydon76 (license 14) + Tested by: pj + +2008-08-06 15:59 +0000 [r136063] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_skinny.c, main/rtp.c: Merged revisions 136062 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug + 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame + type, there are places where ast_rtp_new_source may be called + where the tech_pvt of a channel may not yet have an rtp structure + allocated. This caused a crash in chan_skinny, which was fixed + earlier, but now the same crash has been reported against + chan_h323 as well. It seems that the best solution is to modify + ast_rtp_new_source to not attempt to set the marker bit if the + rtp structure passed in is NULL. This change to + ast_rtp_new_source also allows the removal of what is now a + redundant pointer check from chan_skinny. (closes issue #13247) + Reported by: pj ........ + +2008-08-06 14:51 +0000 [r136034] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_odbc.c: Use a dynamic buffer for rendered SQL, instead + of hardcoding 2048 bytes. Also, switch to using RWLISTs for the + linked list of queries. + +2008-08-06 13:34 +0000 [r136005] Olle Johansson <oej@edvina.net> + + * res/res_jabber.c: - Formatting - Changing debug messages from + VERBOSE to DEBUG channel - Adding a few todo's - Adding a few + more "XMPP"'s to compliment Jabber... + +2008-08-06 03:55 +0000 [r135900-135950] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 135949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) + | 4 lines Fix a longstanding bug in channel walking logic, and + fix the explanation to make sense. (Closes issue #13124) ........ + + * /, main/translate.c: Merged revisions 135915 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008) + | 4 lines Since powerof() can return an error condition, it's + foolhardy not to detect and deal with that condition. (Related to + issue #13240) ........ + + * include/asterisk/utils.h, /, include/asterisk/threadstorage.h: + Merged revisions 135899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) + | 4 lines 1) Bugfix for debugging code 2) Reduce compiler + warnings for another section of debugging code (Closes issue + #13237) ........ + +2008-08-06 00:30 +0000 [r135851] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /, main/abstract_jb.c, main/fixedjitterbuf.h, + include/asterisk/abstract_jb.h: Merged revisions + 135841,135847,135850 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug + 2008) | 27 lines Merging the issue11259 branch. The purpose of + this branch was to take into account "burps" which could cause + jitterbuffers to misbehave. One such example is if the L option + to Dial() were used to inject audio into a bridged conversation + at regular intervals. Since the audio here was not passed through + the jitterbuffer, it would cause a gap in the jitterbuffer's + timestamps which would cause a frames to be dropped for a brief + period. Now ast_generic_bridge will empty and reset the + jitterbuffer each time it is called. This causes injected audio + to be handled properly. ast_generic_bridge also will empty and + reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE + frame since the change in audio source could negatively affect + the jitterbuffer. All of this was made possible by adding a new + public API call to the abstract_jb called ast_jb_empty_and_reset. + (closes issue #11259) Reported by: plack Tested by: putnopvut + ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, + 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel + that occurred when I was testing for a memory leak ........ + r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug + 2008) | 3 lines Remove properties that should not be here + ........ + +2008-08-05 23:45 +0000 [r135821] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c, + include/asterisk/cdr.h: Merged revisions 135799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | + 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf + I discovered that also, in the previous bug fixes and changes, + the cdr.conf 'unanswered' option is not being obeyed, so I fixed + this. And, yes, there are two 'answer' times involved in this + scenario, and I would agree with you, that the first answer time + is the time that should appear in the CDR. (the second 'answer' + time is the time that the bridge was begun). I made the necessary + adjustments, recording the first answer time into the peer cdr, + and then using that to override the bridge cdr's value. To get + the 'unanswered' CDRs to appear, I purposely output them, using + the dial cmd to mark them as DIALED (with a new flag), and + outputting them if they bear that flag, and you are in the right + mode. I also corrected one small mention of the Zap device to + equally consider the dahdi device. I heavily tested 10-sec-wait + macros in dial, and without the macro call; I tested hangups + while the macro was running vs. letting the macro complete and + the bridge form. Looks OK. Removed all the instrumentation and + debug. ........ + +2008-08-05 21:37 +0000 [r135717-135748] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 135747 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 + Aug 2008) | 9 lines In a conversion to use ast_strlen_zero, the + meaning of the flag IAX_HASCALLERID was perverted. This change + reverts IAX2 to the original meaning, which was, that the + callerid set on the client should be overridden on the server, + even if that means the resulting callerid is blank. In other + words, if you set "callerid=" in the IAX config, then the + callerid should be overridden to blank, even if set on the + client. Note that there's a distinction, even on realtime, + between the field not existing (NULL in databases) and the field + existing, but set to blank (override callerid to blank). ........ + + * include/asterisk/config.h, UPGRADE.txt, CHANGES, main/config.c: + Add '+=' append operator to configuration files. + +2008-08-05 17:05 +0000 [r135680-135681] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/datastore.h, include/asterisk/channel.h: + datastore inheritance is a channel feature, so move this + definition back + + * apps/app_dial.c, funcs/func_speex.c, main/pbx.c, main/Makefile, + funcs/func_lock.c, pbx/pbx_lua.c, include/asterisk/channel.h, + apps/app_queue.c, channels/chan_iax2.c, + include/asterisk/manager.h, funcs/func_global.c, + apps/app_speech_utils.c, main/channel.c, funcs/func_enum.c, + main/manager.c, res/res_smdi.c, funcs/func_odbc.c, + funcs/func_volume.c, res/res_agi.c, include/asterisk/datastore.h + (added), pbx/pbx_dundi.c, main/audiohook.c, apps/app_chanspy.c, + apps/app_stack.c, main/datastore.c (added): make datastore + creation and destruction a generic API since it is not really + channel related, and add the ability to add/find/remove + datastores to manager sessions + +2008-08-05 15:30 +0000 [r135648] Tilghman Lesher <tlesher@digium.com> + + * build_tools/make_version: Always output a version string, even + when we can't figure out what we are. (Closes issue #13223) + +2008-08-05 13:26 +0000 [r135598] Sean Bright <sean.bright@gmail.com> + + * /, main/cli.c: Merged revisions 135597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug + 2008) | 1 line Use PATH_MAX for filenames ........ + +2008-08-04 20:15 +0000 [r135537] Russell Bryant <russell@digium.com> + + * configs/chan_dahdi.conf.sample, /: Merged revisions 135536 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) + | 2 lines fix a config sample typo ........ + +2008-08-04 17:12 +0000 [r135476-135485] Tilghman Lesher <tlesher@digium.com> + + * contrib/init.d/rc.mandriva.asterisk (added), Makefile, + contrib/init.d/rc.mandrake.asterisk (removed), + contrib/init.d/rc.mandriva.zaptel (added), + contrib/init.d/rc.mandrake.zaptel (removed): Rename Mandrake + scripts to Mandriva (Closes issue #13221) + + * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135482 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008) + | 2 lines Define ASTSBINDIR for script (Closes issue #13221) + ........ + + * /, apps/app_voicemail.c: Merged revisions 135479 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 + Aug 2008) | 6 lines Memory leak on unload (closes issue #13231) + Reported by: eliel Patches: app_voicemail.leak.patch uploaded by + eliel (license 64) ........ + + * include/asterisk/http.h, main/http.c, res/res_http_post.c: HTTP + module memory leaks (closes issue #13230) Reported by: eliel + Patches: res_http_post_leak.patch uploaded by eliel (license 64) + +2008-08-04 16:28 +0000 [r135439-135474] Russell Bryant <russell@digium.com> + + * configs/chan_dahdi.conf.sample, /: Merged revisions 135473 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) + | 2 lines Add a minor clarification to the documentation of + mohinterpret and mohsuggest ........ + + * channels/chan_console.c: Be explicit that we don't want a result + from this callback. The callback would never indicate a match, so + nothing would have been returned anyway, but it was still a poor + example of proper usage. + +2008-08-03 16:14 +0000 [r135405] Sean Bright <sean.bright@gmail.com> + + * build_tools/cflags.xml, doc/hoard.txt (added), + build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, + CHANGES, makeopts.in: Merge in changes that allow Asterisk to be + built against the Hoard memory allocator. See doc/hoard.txt for + more details. + +2008-08-03 00:03 +0000 [r135332-135373] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: whitespace fixes only. + + * channels/chan_skinny.c: Dont coredump on register of + non-configured devices (closes issue #13224) Reported by: + mvanbaak Patches: noncon.diff uploaded by wedhorn (license 30) + with whitespace fixes by me Tested by: wedhorn, mvanbaak + + * channels/chan_skinny.c: make this work again, and not segfault on + device registration + +2008-08-02 13:21 +0000 [r135302] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_skinny.c: --enable-dev-mode is your friend :-) + +2008-08-02 12:29 +0000 [r135300] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: pass device instead of session to + transmit_ functions. (closes issue #10396) Reported by: wedhorn + Patches: transmit3a.diff uploaded by wedhorn (license 30) Tested + by: wedhorn, mvanbaak + +2008-08-02 04:51 +0000 [r135265] Steve Murphy <murf@digium.com> + + * main/pbx.c, main/features.c: (closes issue #13202) Reported by: + falves11 Tested by: murf falves11 == The changes I introduce here + seem to clear up the problem for me. However, if they do not for + you, please reopen this bug, and we'll keep digging. The root of + this problem seems to be a subtle memory corruption introduced + when creating an extension with an empty extension name. While + valgrind cannot detect it outside of DEBUG_MALLOC mode, when + compiled with DEBUG_MALLOC, this is certain death. The code in + main/features.c is a puzzle to me. On the initial module load, + the code is attempting to add the parking extension before the + features.conf file has even been opened! I just wrapped the + offending call with an if() that will not try to add the + extension if the extension name is empty. THis seems to solve the + corruption, and let the "memory show allocations" work as one + would expect. But, really, adding an extension with an empty name + is a seriously bad thing to allow, as it will mess up all the + pattern matching algorithms, etc. So, I added a statement to the + add_extension2 code to return a -1 if this is attempted. + +2008-08-01 21:56 +0000 [r135235] Terry Wilson <twilson@digium.com> + + * main/http.c, res/res_http_post.c: Fix mime parsing by re-adding + support for passing headers to callback functions + +2008-08-01 19:29 +0000 [r135197] Sean Bright <sean.bright@gmail.com> + + * channels/chan_mgcp.c: Remove some code that used to do something + but does not anymore, mainly to get rid of a shadow warning (but + this seemed legitimate enough to fix here instead of in my + branch). Thanks to putnopvut for taking a look as well. + +2008-08-01 18:16 +0000 [r135158] Russell Bryant <russell@digium.com> + + * configs/iax.conf.sample, channels/iax2.h, CHANGES, + channels/chan_iax2.c, channels/iax2-parser.c: Merge changes from + team/bbryant/keyrotation This set of changes enhances IAX2 + encryption support by adding key rotation to provide enhanced + security. The key used for encryption is rotated right after the + call gets set up, and then again every few minutes. This was + discussed at the last AstriDevCon. For interoperability with + older versions of Asterisk, there is an option that disables key + rotation. (closes issue #13018) Reported by: bbryant Patches: + 07072008__iax2_key_rotation.diff uploaded by bbryant (license 36) + Tested by: russell, bbryant + +2008-08-01 17:09 +0000 [r135126-135128] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Picky, picky, buildbot + + * channels/chan_sip.c, configs/sip.conf.sample: SIP should use the + transport type set in the Moved Temporarily for the next invite. + (closes issue #11843) Reported by: pestermann Patches: + 20080723__issue11843_302_ignores_transport_16branch.diff uploaded + by bbryant (license 36) + 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by + bbryant (license 36) Tested by: pabelanger + +2008-08-01 14:42 +0000 [r135067-135068] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: IMAP-specific items must go in IMAP_STORAGE + defines... + + * configs/voicemail.conf.sample, apps/app_voicemail.c: IMAP storage + functioned under the assumption that folders such as "Work" and + "Family" would be subfolders of the INBOX. This is an invalid + assumption to make, but it could be desirable to set up folders + in this manner, so a new option for voicemail.conf, + "imapparentfolder" has been added to allow for this. (closes + issue #13142) Reported by: jaroth Patches: parentfolder.patch + uploaded by jaroth (license 50) + +2008-08-01 12:17 +0000 [r135056-135061] Michiel van Baak <michiel@vanbaak.info> + + * contrib/scripts/safe_asterisk: Make safe_asterisk work on + dash/sh/bash etc. (closes issue #13111) Reported by: pabelanger + Patches: 2008071901_issue13111_safe_asterisk.diff uploaded by + mvanbaak (license 7) Tested by: mvanbaak, pabelanger + + * /, apps/app_ices.c: Merged revisions 135058 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008) + | 2 lines make app_ices compile on OpenBSD. ........ + + * /, channels/chan_skinny.c: Merged revisions 135055 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 + Aug 2008) | 8 lines fix some potential deadlocks in chan_skinny + (closes issue #13215) Reported by: qwell Patches: + 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7) + Tested by: mvanbaak ........ + +2008-07-31 22:28 +0000 [r135016] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/http.c: Merged revisions 134983 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul + 2008) | 3 lines accomodate users who seem to lack a sense of + humor :-) ........ + +2008-07-31 21:53 +0000 [r134925-134977] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_sqlite.c: Switch command order, to meet with + current specs + + * res/res_config_pgsql.c: Increase column size beyond the minimum + required, since PostgreSQL won't let us modify existing columns. + +2008-07-31 19:48 +0000 [r134922] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 134883 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | + 51 lines (closes issue #11849) Reported by: greyvoip Tested by: + murf OK, a few days of debugging, a bunch of instrumentation in + chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook + pages of notes later, I have made the small tweek necc. to get + the start time right on the second CDR when: A Calls B B answ. A + hits Xfer button on sip phone, A dials C and hits the OK button, + A hangs up C answers ringing phone B and C converse B and/or C + hangs up But does not harm the scenario where: A Calls B B answ. + B hits xfer button on sip phone, B dials C and hits the OK + button, B hangs up C answers ringing phone A and C converse A + and/or C hangs up The difference in start times on the second CDR + is because of a Masquerade on the B channel when the xfer number + is sent. It ends up replacing the CDR on the B channel with a + duplicate, which ends up getting tossed out. We keep a pointer to + the first CDR, and update *that* after the bridge closes. But, + only if the CDR has changed. I hope this change is specific + enough not to muck up any current CDR-based apps. In my defence, + I assert that the previous information was wrong, and this change + fixes it, and possibly other similar scenarios. I wonder if I + should be doing the same thing for the channel, as I did for the + peer, but I can't think of a scenario this might affect. I leave + it, then, as an exersize for the users, to find the scenario + where the chan's CDR changes and loses the proper start time. + ........ and as to 1.4 to trunk; have I expressed my feelings + about code shifting from one file to another? Good. + +2008-07-31 19:43 +0000 [r134919] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_sqlite.c: Two errors: 1) If a function returns + SQLITE_LOCKED, no recovery is possible. 2) An error message can + be allocated, even when no error is signalled. (closes issue + #13109) Reported by: gknispel_proformatique + +2008-07-31 19:39 +0000 [r134916-134917] Russell Bryant <russell@digium.com> + + * /, apps/app_ices.c: Merged revisions 134915 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008) + | 9 lines Get app_ices working again (closes issue #12981) + Reported by: dlogan Patches: + 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant + (license 36) 20080709__app_ices_v2_update_14.diff uploaded by + bbryant (license 36) Tested by: bbryant ........ + + * channels/iax2-parser.c: fix the potential use of an uninitialized + variable + +2008-07-31 19:03 +0000 [r134867] Tilghman Lesher <tlesher@digium.com> + + * channels/iax2-parser.c: Optimize frame cache by realloc'ing the + smallest frame when the cache is full. This ensures that we don't + just keep a cache of tiny frames, continually doing an alloc/free + for each data frame, thus negating the point of having a cache. + +2008-07-31 16:50 +0000 [r134803-134815] Russell Bryant <russell@digium.com> + + * channels/iax2-parser.c: Merged revisions 134814 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008) + | 7 lines In case we have some processing threads that free more + frames than they allocate, do not let the frame cache grow + forever. (closes issue #13160) Reported by: tavius Tested by: + tavius, russell ........ + + * doc/tex/app-sms.tex, doc/tex/queuelog.tex: Fix some tex errors + (closes issue #13211) Reported by: eliel Patches: + fixtexerrors.patch uploaded by eliel (license 64) + +2008-07-31 16:05 +0000 [r134759] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 134758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul + 2008) | 16 lines Add more timeout checks into app_queue, + specifically targeting areas where an unknown and potentially + long time has just elapsed. Also added a check to try_calling() + to return early if the timeout has elapsed instead of potentially + setting a negative timeout for the call (thus making it have *no* + timeout at all). (closes issue #13186) Reported by: + miquel_cabrespina Patches: 13186.diff uploaded by putnopvut + (license 60) Tested by: miquel_cabrespina ........ + +2008-07-30 22:38 +0000 [r134703] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/sched.h, main/sched.c: Oops, wrong define + +2008-07-30 22:04 +0000 [r134653] Steve Murphy <murf@digium.com> + + * /: blocking 134652 from trunk because this problem only applies + to 1.4 + +2008-07-30 21:40 +0000 [r134650] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 134649 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 + Jul 2008) | 4 lines Qwell pointed out, via IRC, that the previous + fix only worked when explicitly set. When nothing is set, and the + option is implied, it breaks, because configure sets the prefix + to 'NONE'. Fixing. ........ + +2008-07-30 21:05 +0000 [r134598] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Merged revisions 134556 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | + mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 + lines Fix the parsing of the "reason" parameter in the Diversion: + header. (closes issue #13195) Reported by: woodsfsg ........ + +2008-07-30 20:38 +0000 [r134596] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 134595 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008) + | 6 lines Reduce stack consumption by 12.5% of the max stack size + to fix a crash when compiled with LOW_MEMORY. (closes issue + #13154) Reported by: edantie ........ + +2008-07-30 20:24 +0000 [r134556] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Fix the parsing of the "reason" parameter in + the Diversion: header. (closes issue #13195) Reported by: + woodsfsg + +2008-07-30 19:55 +0000 [r134541] Russell Bryant <russell@digium.com> + + * funcs/func_curl.c, /: Merged revisions 134540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008) + | 4 lines Fix a memory leak in func_curl. Every thread that used + this function leaked an allocation the size of a pointer. + (reported by jmls in #asterisk-dev) ........ + +2008-07-30 19:48 +0000 [r134481-134538] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 134536 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 + Jul 2008) | 4 lines Only override sysconfdir and mandir when + prefix=/usr (closes issue #13093) Reported by: pabelanger + ........ + + * apps/app_queue.c: Let "roundrobin" also reference rrmemory, for + the 1.6 release (as described in UPGRADE-1.4.txt) (Closes issue + #13181) + + * /, res/res_agi.c: Merged revisions 134480 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008) + | 5 lines launch_netscript sometimes returns -1, which fails to + set AGISTATUS. Map failure to -1, so that AGISTATUS is always + set. (closes issue #13199) Reported by: smw1218 ........ + +2008-07-30 18:33 +0000 [r134476] Mark Michelson <mmichelson@digium.com> + + * /, main/app.c: Merged revisions 134475 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul + 2008) | 4 lines Fix a spot where a function could return without + bringing a channel out of autoservice. ........ + +2008-07-30 17:36 +0000 [r134401-134443] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Document adaptive capabilities + + * res/res_config_sqlite.c: Add adaptive capabilities to the sqlite + realtime driver (closes issue #13097) Reported by: + gknispel_proformatique Patches: 20080730__bug13097.diff.txt + uploaded by Corydon76 (license 14) Tested by: Corydon76 + + * configs/iax.conf.sample, configs/chan_dahdi.conf.sample, + channels/chan_sip.c, main/features.c, configs/sip.conf.sample, + configs/skinny.conf.sample, CHANGES: Move implementation of an + attended-transfer-complete sound from one channel driver into a + common place for multiple channel drivers. (closes issue #13152) + Reported by: caio1982 Patches: atxfer_complete_sound3.diff + uploaded by caio1982 (license 22) + +2008-07-30 15:32 +0000 [r134355] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /: Merged revisions 134352 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul + 2008) | 2 lines use the proper method for building version.h + ........ + +2008-07-30 15:30 +0000 [r134312-134353] Tilghman Lesher <tlesher@digium.com> + + * doc/tex/cliprompt.tex, main/asterisk.c: Add %u and %g to the + ASTERISK_PROMPT settings, for username and group, respectively. + Also, take the opportunity to clean up the CLI prompt generation + code. (closes issue #13175) Reported by: eliel Patches: + cliprompt.patch uploaded by eliel (license 64) + + * Makefile: Minor changes to reduce packaging changes made by the + Fedora maintainer. (closes issue #12974) Reported by: jcollie + Patches: 0001-Don-t-override-duplicate-optimization-flags.patch + uploaded by jcollie (license 412) + +2008-07-29 22:22 +0000 [r134260] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_dahdibarge.c, channels/chan_dahdi.c, /, + apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c, + apps/app_rpt.c: build against the now-typedef-free dahdi/user.h, + and remove some #ifdefs for features that will always be present + in DAHDI + +2008-07-29 21:23 +0000 [r134253] Brett Bryant <bbryant@digium.com> + + * main/http.c: Fix deadlock when unloading res_http_post because + the uris lock was still locked. + +2008-07-28 22:07 +0000 [r134162-134163] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 134161 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 + Jul 2008) | 7 lines Detect when sox fails to raise the volume, + because sox can't read the file. (closes issue #12939) Reported + by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by + Corydon76 (license 14) Tested by: rickbradley ........ + + * /: Restore properties mistakenly removed (broke merging) + +2008-07-28 19:53 +0000 [r134125] Mark Michelson <mmichelson@digium.com> + + * configure, main/Makefile, configure.ac, CHANGES: This commit + compensates for buggy poll(2) implementations. Asterisk has, for + a long time, had its own implementation of poll(2) which just + used the input arguments to call select(2). In 1.4, this internal + implementation was used for Darwin systems. This was removed in + Asterisk trunk at some point, but it seems as though this was not + the right move to make. On Mac OS X, it appears as though the + poll used to gather CLI input does not respond properly when + connecting via a remote Asterisk console. Reverting to the use of + Asterisk's poll fixed the issue. Also, there is now an option for + the configure script, --enable-internal-poll, which will allow + for anyone to use Asterisk's internal poll implementation in case + they suspect that their system's poll implementation is buggy. + closes issue #11928) Reported by: adriavidal Patches: + 1.6.0-configurev2.patch uploaded by putnopvut (license 60) + +2008-07-28 16:49 +0000 [r134088] Tilghman Lesher <tlesher@digium.com> + + * UPGRADE.txt, apps/app_image.c, CHANGES: Change SendImage() to + output a more consistent status variable. (closes issue #13134) + Reported by: eliel Patches: app_image.c.patch uploaded by eliel + (license 64) UPGRADE.patch uploaded by eliel (license 64) + +2008-07-28 16:42 +0000 [r134086] Kevin P. Fleming <kpfleming@digium.com> + + * build_tools/cflags.xml, main/channel.c, apps/app_dahdibarge.c, + channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + doc/ss7.txt, apps/app_dahdiscan.c, apps/app_dahdiras.c, + doc/osp.txt, pbx/pbx_config.c, apps/app_parkandannounce.c, + main/loader.c, sample.call, contrib/scripts/autosupport: remove + remaining Zaptel references in various places + +2008-07-28 16:00 +0000 [r134050] Mark Michelson <mmichelson@digium.com> + + * apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_chanspy.c, + main/asterisk.c: merging the zap_and_dahdi_trunk branch up to + trunk + +2008-07-27 21:12 +0000 [r134005] Russell Bryant <russell@digium.com> + + * funcs/func_config.c, /: Add a missing unlock within error + handling (closes issue #13176) Reported by: pj + +2008-07-26 15:16 +0000 [r133941-133946] Russell Bryant <russell@digium.com> + + * main/devicestate.c: actually use the cache_cache argument + + * main/devicestate.c: ast_device_state() gets called in two + different ways. The first way is when called from elsewhere in + Asterisk to find the current state of a device. In that case, we + want to use the cached value if it exists. The other way is when + processing a device state change. In that case, we do not want to + check the cache because returning the last known state is counter + productive. + + * main/devicestate.c: Re-work comment about how device state + changes are processed to be a bit more clear + + * main/devicestate.c: Remove the code that decided when device + state changes should be cached or not. It is no longer needed. + +2008-07-25 22:08 +0000 [r133860-133904] Tilghman Lesher <tlesher@digium.com> + + * doc/lang/hebrew.ods, apps/app_voicemail.c: Hebrew syntax for + voicemail prompts (closes issue #13155) Reported by: + greenfieldtech Patches: app_voicemail.c.patch uploaded by + greenfieldtech (license 369) hebrew.ods uploaded by + greenfieldtech (license 369) + + * contrib/scripts/asterisk.ldif: Update version (closes issue + #13163) Reported by: suretec Patches: asterisk.ldif uploaded by + suretec (license 70) + + * main/channel.c, channels/chan_dahdi.c, + include/asterisk/devicestate.h, channels/chan_sip.c, + channels/chan_skinny.c, channels/chan_agent.c, + main/devicestate.c, channels/chan_iax2.c: Deprecate + *_device_state_* APIs in favor of *_devstate_* APIs + +2008-07-25 20:56 +0000 [r133819] Kevin P. Fleming <kpfleming@digium.com> + + * main/logger.c: minor change to test automerge + +2008-07-25 19:12 +0000 [r133770] Brandon Kruse <bkruse@digium.com> + + * main/manager.c: Revert tilghman and pari's code changes, as we do + NOT need to uri_decode in manager. (if I sent + core%20show%20channels from a telnet session, it should be + interpreted literally, however, if I send that from an http + session, it should be decoded, which is the behaivor now) + +2008-07-25 17:24 +0000 [r133665] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /, channels/chan_agent.c, main/devicestate.c: + Merged revisions 133649 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) + | 8 lines Fix some errant device states by making the devicestate + API more strict in terms of the device argument (only without the + unique identifier appended). (closes issue #12771) Reported by: + davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 + (license 14) Tested by: davidw, jvandal, murf ........ + +2008-07-25 17:21 +0000 [r133651] Brandon Kruse <bkruse@digium.com> + + * main/http.c: Committing a fix that was introduced a long time ago + (does not affect 1.4), where you would pass a pointer to the end + of a character array, and ast_uri_decode would do no good. + +2008-07-25 15:00 +0000 [r133575-133579] Russell Bryant <russell@digium.com> + + * /, LICENSE: Merged revisions 133578 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r133578 | russell | 2008-07-25 10:00:31 -0500 + (Fri, 25 Jul 2008) | 10 lines Merged revisions 133577 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) + | 2 lines Fix the IAX2 URI for calling Digium ........ + ................ + + * include/asterisk/doxyref.h, main/asterisk.c: Modify the main page + of the doxygen documentation to link to a new page dedicated to + Asterisk licensing information. The licensing page includes the + Asterisk license, as well as a (not yet complete) list of 3rd + party libraries that may be used, as well as what license we + receive them under. Help filling out this list in the format that + I have started in doxyref.h would be much appreciated. :) + +2008-07-25 14:40 +0000 [r133570-133573] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133572 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul + 2008) | 7 lines We need to make sure to null-terminate the "name" + portion of SIP URI parameters so that there are no bogus + comparisons. Thanks to bbryant for pointing this out. ........ + + * apps/app_queue.c: Add a missing unlock. Pointed out by Atis + Lezdins in #asterisk-dev + +2008-07-25 13:01 +0000 [r133566-133568] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Minor coding guidelines tweaks ... - Use + ast_strlen_zero in one place - check for successful string + comparison the way most of Asterisk code does it + + * main/devicestate.c: When the ast_device_state() function is + called to retrieve device state, and the code checks to see if + there is a cached state available, use the aggregate cached state + across all servers, and not just the local state. + +2008-07-24 21:27 +0000 [r133509] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) + | 3 lines Fix rtautoclear and rtcachefriends (Closes issue + #12707) ........ + +2008-07-24 20:40 +0000 [r133486] Russell Bryant <russell@digium.com> + + * channels/chan_agent.c: I made this change from DEVICE_STATE to + DEVICE_STATE_CHANGE, but I had it backwards, this is the right + event to subscribe to ... + +2008-07-24 19:53 +0000 [r133448] Mark Michelson <mmichelson@digium.com> + + * main/logger.c: Print the correct PID in log messages. Prior to + this commit, only the logger thread's PID would be printed. + (closes issue #13150) Reported by: atis Patches: log_pid.diff + uploaded by putnopvut (license 60) Tested by: eliel + +2008-07-24 05:21 +0000 [r133391-133400] Tilghman Lesher <tlesher@digium.com> + + * Makefile, contrib/scripts/asterisk.logrotate: Build the logrotate + script according to paths (Closes issue #13147) + + * Makefile: Optionally install logrotate file (Closes issue #13148) + +2008-07-23 22:03 +0000 [r133299] Steve Murphy <murf@digium.com> + + * main/pbx.c: (closes issue #13144) Reported by: murf Tested by: + murf For: J. Geis The 'data' field in the ast_exten struct was + being 'moved' from the current dialplan to the replacement + dialplan. This was not good, as the current dialplan could have + problems in the time between the change and when the new dialplan + is swapped in. So, I modified the merge_and_delete code to strdup + the 'data' field (the args to the app call), and then it's freed + as normal. I improved a few messages; I added code to limit the + number of calls to the + context_merge_incls_swits_igps_other_registrars() to one per + context. I don't think having it called multiple times per + context was doing anything bad, but it was inefficient. I hope + this fixes the problems Mr. Geiss was noting in asterisk-users, + see + http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html + +2008-07-23 21:50 +0000 [r133296] Jason Parker <jparker@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 133295 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul + 2008) | 1 line inbandrelease is gone - it's now inbanddisconnect + ........ + +2008-07-23 20:33 +0000 [r133197] Brett Bryant <bbryant@digium.com> + + * channels/chan_sip.c: Fix issue where tcp in sip is enabled by + default, despite what it says in the config sample file. Also fix + "sip show settings" for tcp connections. + +2008-07-23 19:48 +0000 [r133041-133171] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_chanspy.c, include/asterisk/options.h, + main/asterisk.c: Merged revisions 133169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul + 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN + in app_chanspy should be set at load time, not at compile time, + since dahdi_chan_name is determined at load time. Also changed + the next_unique_id_to_use to have the static qualifier. Also + added the dahdi_chan_name_len variable so that + strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for + the suggestion. ........ + + * /, apps/app_chanspy.c: Merged revisions 133104 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul + 2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is + twelve. The strncmp call in next_channel should account for this. + ........ + + * /, apps/app_chanspy.c: Merged revisions 133101 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul + 2008) | 6 lines Update the "last" channel in next_channel in + app_chanspy so that the same pseudo channel isn't constantly + returned. related to issue #13124 ........ + + * channels/chan_dahdi.c, /: Merged revisions 133038 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, + 23 Jul 2008) | 7 lines Small cleanup. Move the declaration of the + DAHDI_SPANINFO variable to the block where it is used. This + allows one less #ifdef HAVE_PRI to clutter things up. Thanks to + Tzafrir for pointing this out on #asterisk-dev ........ + +2008-07-23 17:20 +0000 [r132977-132981] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c: Yet another conversion of '|' to ',' + (closes issue #13137) Reported by: eliel Patches: + chan_iax2trunk-IAXPEER.patch uploaded by eliel (license 64) + + * contrib/scripts/asterisk.logrotate (added): Add logrotate script + for Asterisk (closes issue #13085) Reported by: pabelanger + Patches: logrotate uploaded by pabelanger (license 224) + +2008-07-23 16:38 +0000 [r132964-132966] Kevin P. Fleming <kpfleming@digium.com> + + * channels/misdn/isdn_lib.c: use correct function name... please + compile with --enable-dev-mode + + * /, main/utils.c, include/asterisk/stringfields.h: Merged + revisions 132872 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul + 2008) | 2 lines minor optimization for stringfields: when a field + is being set to a larger value than it currently contains and it + happens to be the most recent field allocated from the currentl + pool, it is possible to 'grow' it without having to waste the + space it is currently using (or potentially even allocate a new + pool) ........ + +2008-07-23 12:07 +0000 [r132883] Christian Richter <christian.richter@beronet.com> + + * /, channels/misdn/isdn_lib.c: Merged revisions 132826 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 Jul 2008) + | 1 line another Fix because of r119585, this commit has broken + high frequented BRI Ports, there was a possibility that a + channel, that was marked as in_use would be reused later, the + corresponding port could got stuck then. So it is recommended to + upgrade for chan_misdn users. ........ + +2008-07-23 11:40 +0000 [r132827] Kevin P. Fleming <kpfleming@digium.com> + + * /: remove bogus property that is breaking automerges + +2008-07-23 08:13 +0000 [r132823] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Well, the content of a channel variable may + be longer than the size of a pointer... Thanks, eliel! Reported + by: eliel Patches: chan_siptrunk.SIPPEER.patch uploaded by eliel + (license 64) (closes issue #13135) + +2008-07-22 22:17 +0000 [r132795] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 132777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ Allow + Spiraled INVITEs to work correctly within Asterisk. Prior to this + change, a spiraled INVITE would cause a 482 Loop Detected to be + sent to the caller. With this change, if a potential loop is + detected, the Request-URI is inspected to see if it has changed + from what was originally received. If pedantic mode is on, then + this inspection is fully RFC 3261 compliant. If pedantic mode is + not on, then a string comparison is used to test the equality of + the two R-URIs. This has been tested by using OpenSER to rewrite + the R-URI and send the INVITE back to Asterisk. (closes issue + #7403) Reported by: stephen_dredge Modified: + branches/1.4/channels/chan_sip.c ........ + +2008-07-22 22:14 +0000 [r132791] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c: correct fix made in r132777... the code + *did* compile in dev-mode, as long as libpri was installed and + enabled + +2008-07-22 21:53 +0000 [r132778] Tilghman Lesher <tlesher@digium.com> + + * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged + revisions 132713 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500 + (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) + | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ + ................ + +2008-07-22 21:52 +0000 [r132777] Mark Michelson <mmichelson@digium.com> + + * channels/chan_dahdi.c: Get chan_dahdi to compile in devmode + +2008-07-22 21:21 +0000 [r132705-132721] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132712 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 + Jul 2008) | 6 lines ensure that if any alarms exist at channel + creation time, they are handled identically to if they occurred + later, so that later alarm clearing will work properly and 'make + sense' (closes issue #12160) Reported by: tzafrir ........ + + * /, configure, configure.ac, acinclude.m4: Merged revisions 132704 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul + 2008) | 2 lines make AST_C_COMPILE_CHECK able to print a 'pretty' + description of what it is doing ........ + +2008-07-22 20:46 +0000 [r132703] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged + revisions 132645 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 + lines The most common question on the #asterisk iRC channel and + on mailing lists seems to be in regards to an error message when + retransmit fails. This is frequently misunderstood as a failure + of Asterisk, not a failure of the network to reach the other + party. This document tries to assist the Asterisk user in sorting + out these issues by explaining the logic and pointing at some + possible causes. Hopefully, we will get other questions now :-) + ........ + +2008-07-22 19:59 +0000 [r132573-132643] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, + configure, include/asterisk/autoconfig.h.in, configure.ac: Merged + revisions 132641 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul + 2008) | 2 lines use renamed libpri API call for controlling this + feature (was improperly named before) ........ + + * channels/chan_dahdi.c, /: Merged revisions 132571 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 + Jul 2008) | 2 lines teach chan_dahdi how to find the D-channel on + BRI spans, and don't attempt to use channel 24 as a D-channel on + spans of unexpected sizes ........ + +2008-07-21 22:49 +0000 [r132514-132572] Brett Bryant <bbryant@digium.com> + + * channels/chan_iax2.c: Add autocompletion to "iax2 set debug + peer". (closes issue #13129) Reported by: eliel Patches: + chan_iax2.c.patch uploaded by eliel (license 64) + + * configs/gtalk.conf.sample, configs/jingle.conf.sample, + configs/manager.conf.sample, configs/features.conf.sample: Update + configuration files to add missing options for jingle, gtalk, + manager.conf, and features.conf. (closes issue #13128) Reported + by: caio1982 Patches: missing_options1.diff uploaded by caio1982 + (license 22) + +2008-07-21 21:00 +0000 [r132510-132511] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/fskmodem.h (added), main/fskmodem.c (added): + (Step 2 of 2) + + * include/asterisk/fskmodem_int.h (added), build_tools/cflags.xml, + main/fskmodem_float.c (added), main/tdd.c, + include/asterisk/fskmodem.h (removed), main/fskmodem_int.c + (added), main/callerid.c, include/asterisk/fskmodem_float.h + (added), main/fskmodem.c (removed): Optionally build + integer-based routines for FSK tone decoding (but default to the + more accurate float-based routines). (Closes issue #11679) (Step + 1 of 2) + +2008-07-21 20:54 +0000 [r132466-132508] Brett Bryant <bbryant@digium.com> + + * apps/app_sendtext.c: Fix a bug where SENDTEXTSTATUS isn't set + properly when it isn't supported on a channel (yet _another_ + useful patch by eliel). (closes issue #13081) Reported by: eliel + Patches: app_sendtext.c.patch uploaded by eliel (license 64) + Tested by: eliel + + * channels/chan_iax2.c: Add "iax2 set debug peer" command and + remove deprecated iax2 debug commands that conflicted with adding + new features to the newer debug commaands. (closes issue #13103) + Reported by: mvanbaak Patches: + 2008071901__issue13103_iax2_set_debug_peer.diff uploaded by + mvanbaak (license 7) Tested by: bbryant, mvanbaak + + * channels/chan_sip.c: Fix bug where ast_parse_arg would + inadvertantly enable sip tcp when parsing a tcpbindaddr if it was + disabled. (closes issue #13117) Reported by: pj + + * channels/chan_iax2.c: Fix an issue in iax2 where a call that's + been rejected still kept an open channel on the side that + attempted to make the call (not the side of the call that + rejected the call). Changes were load tested and also approved by + Russell. + +2008-07-21 15:33 +0000 [r132425] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: make buffers config option + (chan_dahdi.conf) parsing safer and added logging in case of + failure + +2008-07-21 14:47 +0000 [r132388-132390] Russell Bryant <russell@digium.com> + + * include/asterisk/libresample.h (removed), + build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, main/Makefile, main/libresample + (removed), configure.ac, codecs/codec_resample.c, makeopts.in, + apps/app_jack.c: Remove libresample from the Asterisk source + tree. It is now available in its own repository, and must be + installed like any other library for Asterisk to use. The two + modules that require it are codec_resample and app_jack. To + install libresample: $ svn co + http://svn.digium.com/svn/libresample/trunk libresample $ cd + libresample $ ./configure $ make $ sudo make install This code is + currently in our own repository because the build system did not + include the appropriate targets for building a dynamic library or + for installing the library. + + * codecs/codec_resample.c, apps/app_jack.c: Enable higher quality + resampling, as it doesn't have a noticeable performance impact on + my machine .. + +2008-07-19 16:46 +0000 [r132312] Kevin P. Fleming <kpfleming@digium.com> + + * /, LICENSE: Merged revisions 132311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul + 2008) | 2 lines grant a license exception to allow distribution + of Asterisk binaries that use the UW IMAP Toolkit (which is + licensed under a non-GPL-compatible license) ........ + +2008-07-19 10:46 +0000 [r132277] Michiel van Baak <michiel@vanbaak.info> + + * res/res_config_sqlite.c: fix a couple of comments in sqlite + resource driver. (closes issue #13110) Reported by: + gknispel_proformatique Patches: res_config_sqlite_comments.patch + uploaded by gknispel (license 261) + +2008-07-18 22:19 +0000 [r132242] Brett Bryant <bbryant@digium.com> + + * main/manager.c: Fixes problem where manager users loaded from + users.conf would be removed early (before the routine to load the + configuration was finished) because a variable wasn't + initialized. + +2008-07-18 20:57 +0000 [r132203-132206] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c: Russell pointed out that using ast_strdupa() + within a loop like this is probably not a good idea, as we might + run out of stack space. Therefore, changing this over to use the + ast_str infrastructure for buffers is probably a good idea. + + * main/manager.c: Fix trunk devmode + +2008-07-18 20:14 +0000 [r132169] Pari Nannapaneni <paripurnachand@digium.com> + + * main/manager.c: updateconfig is not uri decoding variables,values + from the GET url + +2008-07-18 19:09 +0000 [r132109-132113] Tilghman Lesher <tlesher@digium.com> + + * /, main/say.c: Merged revisions 132112 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008) + | 6 lines Fix for Taiwanese number syntax (closes issue #12319) + Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch + uploaded by CharlesWang (license 444) ........ + + * /, main/config.c: Merged revisions 132107 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008) + | 6 lines Textual clarification (closes issue #13106) Reported + by: flefoll Patches: config.c.br14.120173.patch-unknown-directive + uploaded by flefoll (license 244) ........ + +2008-07-18 18:50 +0000 [r132108] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c: Make sure we break the poll so that + messages queued will be sent on the SS7 when using CLI commands + for blocking and blocking of CICs and linksets. + +2008-07-18 17:55 +0000 [r132050] Brett Bryant <bbryant@digium.com> + + * main/hashtab.c, cdr/cdr_radius.c: Fix magic Revision keywords in + hashtab.c and change cdr_radius.c to use the same keyword as the + other files (patch by eliel). (closes issue #13104) Reported by: + eliel Patches: revision.patch uploaded by eliel (license 64) + +2008-07-18 17:10 +0000 [r131982-131989] Tilghman Lesher <tlesher@digium.com> + + * /, main/sched.c: Merged revisions 131988 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008) + | 2 lines Oops ........ + + * /, include/asterisk/sched.h, main/sched.c: Merged revisions + 131985 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008) + | 2 lines Preserve ABI compatibility with last change ........ + + * /, include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c: + Merged revisions 131970 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008) + | 2 lines Make the ast_assert call within ast_sched_del report + something useful. ........ + +2008-07-18 16:16 +0000 [r131923] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/Makefile, include/asterisk/dlfcn-compat.h (removed), + main/dlfcn.c (removed), main/loader.c: Merged revisions 131921 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul + 2008) | 2 lines remove the dlfcn compatibility stuff, because no + platforms that Asterisk currently runs on it use it, and it + doesn't build anyway ........ + +2008-07-18 15:38 +0000 [r131916] Brett Bryant <bbryant@digium.com> + + * /, main/features.c: Merged revisions 131915 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131915 | bbryant | 2008-07-18 10:34:42 -0500 (Fri, 18 Jul 2008) + | 4 lines Fix a bug in blind transfers where the BLINDTRANSFER + variable isn't always set to the other end of the blind transfer. + (closes issue #12586) ........ + +2008-07-17 22:40 +0000 [r131868] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: Add configuration option to + chan_dahdi.conf to allow buffering policy and number of buffers + to be configured per channel. Syntax: buffers=<num of + buffers>,<policy> Where the number of buffers is some + non-negative integer and the policy is either "full", "half", or + "immediate". + +2008-07-17 21:26 +0000 [r131824] Mark Michelson <mmichelson@digium.com> + + * apps/app_senddtmf.c: Document that the duration of dtmf may be + passed to the SendDTMF application. Also correct the default + pause between digits. (closes issue #13102) Reported by: eliel + Patches: app_senddtmf.c.patch uploaded by eliel (license 64) + +2008-07-17 20:37 +0000 [r131753-131791] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 131790 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 + Jul 2008) | 7 lines Revert part of issue #5620 (revision 6965) as + it appears that it was in error. This should fix talk call + progress on analog lines. (closes issue #12178) Reported by: + michael-fig Patches: 20080717__bug12178.diff.txt uploaded by + Corydon76 (license 14) ........ + + * res/res_config_sqlite.c: Fix memory leaks (closes issue #13099) + Reported by: gknispel_proformatique Patches: + res_config_sqlite_leak_on_error.patch uploaded by gknispel + (license 261) + +2008-07-17 18:14 +0000 [r131717] Brett Bryant <bbryant@digium.com> + + * main/features.c: Fix a memory leak in register_group_feature when + attempting to register a feature without specifying a group or + feature to register. (closes issue #13101) Reported by: eliel + Patches: features.c.patch uploaded by eliel (license 64) + +2008-07-17 15:45 +0000 [r131681] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_sqlite.c: Fix memory leak. (Closes issue #13096) + Reported by gknispel_proformatique + +2008-07-17 14:46 +0000 [r131643] Russell Bryant <russell@digium.com> + + * channels/chan_agent.c: Instead of attempting to pass through + AST_EVENT_DEVICE_STATE, use DEVICE_STATE_CHANGE instead. + DEVICE_STATE is a state change on one server, and + DEVICE_STATE_CHANGE is the "real" state of that device across all + servers sharing state. This would have only been a problem with + distributed device state. + +2008-07-17 14:00 +0000 [r131606] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, UPGRADE.txt, CHANGES: Change several 'core' commands + to be 'dialplan' commands (with appropriate deprecation, of + course) (closes issue #13016) Reported by: caio1982 Patches: + dialplan_globals6.diff uploaded by caio1982 (license 22) + +2008-07-16 23:53 +0000 [r131570] Steve Murphy <murf@digium.com> + + * include/asterisk/lock.h: (closes issue #13089) Reported by: murf + Most of this bug was already fixed by Tilghman before I opened + it; Many thanks to Tilghman for his fix in svn version 125794. + That fix cleared up some of the fields in the lock_info. This + commit changes the address that is stored for the lock in the + lock_info struct, so that it is the same as that passed into the + locking macros. This makes searching for a lock_info (as in + log_show_lock()) by its lock addr possible. The lock_addr field + is infinitely more useful if it is the same as what is 'publicly' + available outside the lock_info code. Many thanks to kpfleming, + putnopvut, and Russell for their invaluable insights earlier + today. + +2008-07-16 22:28 +0000 [r131445-131529] Brett Bryant <bbryant@digium.com> + + * apps/app_rpt.c: Janitor project: convert free to ast_free (closes + issue #13082) Reported by: eliel Patches: app_rpt.c.patch + uploaded by eliel (license 64) + + * /, channels/chan_iax2.c: Merged revisions 131491 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 + Jul 2008) | 6 lines Fix a bug in iax2 registration that allowed + peers to register with case-insensitive names (user_cmp_cb and + peer_cmp_cb are now both case-sensitive). (closes issue #13091) + ........ + + * funcs/func_sysinfo.c: Fixes sysinfo operator issue also fixed + elsewhere in r131445. (issue #13057) + + * main/asterisk.c: Fixes an issue with "core show sysinfo" that + used the wrong operator to calculate the number of bytes from a + sysinfo structure. unsigned long. (closes issue #13057) Reported + by: eliel Patches: asterisk.c.patch uploaded by eliel (license + 64) + +2008-07-16 20:48 +0000 [r131422] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 131421 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 + Jul 2008) | 7 lines Always ensure that the channel's tech_pvt + reference is NULL after calling the destroy callback. (closes + issue #13060) Reported by: jpgrayson Patches: + chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license + 492) ........ + +2008-07-16 20:24 +0000 [r131300-131375] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 131369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul + 2008) | 14 lines Move the init_queue call back to where it used + to be (changed Sept 12 last year). It was moved then to prevent a + memory leak. Since then, the same memory leak recurred and was + fixed in a better way. Now it has been found that the placement + of this init_queue call can cause problems if a realtime queue + has values changed to an empty string. The problem is that the + default value for that queue parameter would not be set. (closes + issue #13084) Reported by: elbriga ........ + + * res/res_config_sqlite.c: Don't try to dereference the dbfile + pointer if we know that it's NULL. (closes issue #13092) Reported + by: gknispel_proformatique Patches: + trunk_sqlite_check_vars_null.patch uploaded by gknispel (license + 261) + + * /, apps/app_queue.c: Merged revisions 131357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul + 2008) | 6 lines Apparently, "thread safety" is important, + whatever that means. :P (Thanks Russell!) ........ + + * /, apps/app_queue.c: Merged revisions 131299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul + 2008) | 13 lines Make absolutely certain that the transfer + datastore is removed from the calling channel once the caller is + finished in the queue. This could have weird con- sequences when + dialing local queue members when multiple transfers occur on a + single call. Also fixed a memory leak that would occur when an + attended transfer occurred from a queue member. (closes issue + #13047) Reported by: festr ........ + +2008-07-16 17:59 +0000 [r131243] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, /: Merged revisions 131242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) | + 19 lines (closes issue #13090) Reported by: murf The problem was + that, esoteric as it is, because the hangerupper context + immediately preceded the std-priv-extent macro, that the checking + code accidentally would fall from traversing hangerupper into the + std-priv-exten macro, where it would hit the hangerupper in the + 'includes', and proceed into an infinite recursion. A small fix + to traverse into the statements of the context instead of the + context solves this issue. I also added some commented out + printfs for debug, which were pretty handy in the face of a dorky + gdb. This was a problem around since the package was first + written; but evidently pretty rare in turning up in the field. + ........ + +2008-07-16 15:08 +0000 [r131207] Russell Bryant <russell@digium.com> + + * channels/chan_agent.c: Add missing terminator to + ast_event_subscribe to fix a crash. (from rev 131206 in the 1.6.0 + branch) + +2008-07-16 00:52 +0000 [r131166] Tilghman Lesher <tlesher@digium.com> + + * main/logger.c: Fix rotate strategy (Closes issue #13086) + +2008-07-15 23:36 +0000 [r131129] Steve Murphy <murf@digium.com> + + * main/pbx.c: (closes issue #12960) Reported by: mnicholson Spent + most of the day on this bug, and the solution was so simple. Just + had to find and understand the problem. The problem was, that the + routine to copy the existing switches, includes, and ignorepats + from the old context to the new one, wasn't getting called when + the context is already existent. (In other words, if AEL is + adding a new context to the mix, they get copied, but if + pbx_config already defined a context, then the copy wasn't + happening. This made no sense, so I moved the call to copy the + includes & etc, no matter the case. + +2008-07-15 18:46 +0000 [r131072] Russell Bryant <russell@digium.com> + + * res/res_agi.c: Fix a couple of places in res_agi where the + agi_commands lock would not be released, causing a deadlock. + (Reported by mvanbaak in #asterisk-dev, discovered by bbryant's + change to the lock tracking code to yell at you if a thread exits + with a lock still held) + +2008-07-15 18:25 +0000 [r131044] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, main/manager.c, /, channels/chan_sip.c, + apps/app_voicemail.c: Merged revisions 130959 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) + | 8 lines astman_send_error does not need a newline appended -- + the API takes care of that for us. (closes issue #13068) Reported + by: gknispel_proformatique Patches: + asterisk_1_4_astman_send.patch uploaded by gknispel (license 261) + asterisk_trunk_astman_send.patch uploaded by gknispel (license + 261) ........ + +2008-07-15 18:14 +0000 [r131015] Brett Bryant <bbryant@digium.com> + + * apps/app_queue.c: Fix memory leak in app_queue when a device + state is changed but it isn't a member of any queue. (closes + issue #13073) Reported by: eliel Patches: app_queue.c.patch + uploaded by eliel (license 64) + +2008-07-15 17:49 +0000 [r131013] Michiel van Baak <michiel@vanbaak.info> + + * main/cdr.c, /: Merged revisions 131012 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008) + | 7 lines remove 4 lines of redundant code. (closes issue #13080) + Reported by: gknispel_proformatique Patches: + trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261) + ........ + +2008-07-15 16:20 +0000 [r130890-130951] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Additional + option for videosupport (always) that disables the optimization + to fail to setup video RTP if the two endpoints will not support + it. This assists with call files and certain transfers to ensure + that if two video phones are ever connected, they will always + share a video feed. + + * /, channels/chan_iax2.c: Merged revisions 130889 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 + Jul 2008) | 8 lines Override the callerid in all cases when the + callerid is set in the user, not just when a remote callerid is + set. Also, if not set in the user, allow the remote CallerID to + pass through. (closes issue #12875) Reported by: dimas Patches: + 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14) + ........ + +2008-07-14 22:22 +0000 [r130794-130854] Mark Michelson <mmichelson@digium.com> + + * main/asterisk.c: Fix a memory leak in the case that /dev/null + cannot be opened when running startup commands from cli.conf + (closes issue #13066) Reported by: eliel Patches: + asterisk.c.patch uploaded by eliel (license 64) + + * apps/app_dial.c, /: Merged revisions 130792 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul + 2008) | 8 lines Add a check to the CAN_EARLY_BRIDGE macro in + app_dial to be sure there are no audiohooks present on the + channels involved. This fixed a one-way audio situation I had in + my test setup. I couldn't find any open issues that suggested + one-way audio with regards to mixmonitor (or other audiohook) + usage, though. ........ + +2008-07-14 17:21 +0000 [r130744] Michiel van Baak <michiel@vanbaak.info> + + * main/dnsmgr.c, /: Merged revisions 130735 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008) + | 10 lines notify the user that dnsmgr refresh wont work when + dnsmgr is not enabled. Previously this command would + automagically appear and disappear. This was confusing. (closes + issue #12796) Reported by: chappell Patches: + dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by: + russell, chappell, mvanbaak ........ + +2008-07-14 16:50 +0000 [r130732-130733] Luigi Rizzo <rizzo@icir.org> + + * channels/vgrabbers.c: free memory used by the x11 grabber when + closing it. + + * channels/console_video.c: use + ast_pthread_create_detached_background() instead of redoing it + with separate calls + +2008-07-14 15:44 +0000 [r130697] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_unistim.c, channels/h323/ast_h323.cxx, + include/asterisk/module.h, channels/misdn/isdn_lib.c: Swap + "static" and "const", so that "static" appears at the beginning + of each declaration (suppresses a warning). (closes issue #13070) + Reported by: gknispel_proformatique Patches: + asterisk_trunk_const_static.patch uploaded by gknispel (license + 261) + +2008-07-14 10:39 +0000 [r130635] Russell Bryant <russell@digium.com> + + * /, main/audiohook.c: Merged revisions 130634 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) + | 2 lines Bump up the debug level for a message. ........ + +2008-07-13 23:14 +0000 [r130574-130578] Michiel van Baak <michiel@vanbaak.info> + + * doc/tex/Makefile, build_tools/prep_tarball, res/Makefile: Make + all sed calls Posix sed compatible. To make sure nobody commits + script-modified files we first make a backup of asterisk.tex, run + the script, generate the pdf and / or html, and put the original + asterisk.tex back. This will guard us for the stuff that happened + before that someone committed a locally modified asterisk.tex, + with changes done by this script. (closes issue #13062) Reported + by: mvanbaak Patches: sed_without-i-v3.diff uploaded by mvanbaak + (license 7) Tested by: mvanbaak Feedback from Corydon. Thanks for + taking the time to go through this. + + * channels/chan_skinny.c: Convert chan_skinny's open-coded linked + lists to the list macros (closes issue #12956) Reported by: DEA + Patches: chan_skinny-linkedlists-v2.txt uploaded by DEA (license + 3) Tested by: DEA, mvanbaak + + * main/manager.c, /: Merged revisions 130573 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008) + | 8 lines fix memory leak when originate from manager cannot + create a thread (closes issue #13069) Reported by: + gknispel_proformatique Patches: + asterisk_trunk_action_originate.patch uploaded by gknispel + (license 261) Tested by: gknispel_proformatique, mvanbaak + ........ + +2008-07-13 17:58 +0000 [r130515] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 130514 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 + Jul 2008) | 4 lines Reverting 2 changesets, as it breaks incoming + IAX2 calls (Related to issue #12963) Reported by: mvanbaak + ........ + +2008-07-13 14:58 +0000 [r130479] Michiel van Baak <michiel@vanbaak.info> + + * doc/tex/asterisk.tex: restore ASTERISKVERSION marker to + asterisk.tex. This got lost in commit 97634 + +2008-07-13 02:34 +0000 [r130444] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_agent.c: Unlock list before returning + +2008-07-11 22:23 +0000 [r130320] Kevin P. Fleming <kpfleming@digium.com> + + * /: not needed here + +2008-07-11 22:03 +0000 [r130296-130297] Steve Murphy <murf@digium.com> + + * main/pbx.c: (closes issue #13041) Reported by: eliel OK, now the + context registrar slot is strdup'd. It is freed on destruction. I + don't see the need to do this with all the structs' registrar + fields, but if some wild case proves they should also be handled + this way, then we can put in the extra work at that time. + + * res/res_odbc.c: a small change to make things compile + +2008-07-11 21:36 +0000 [r130293] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c: Support new TRANSPORT definitions in + libss7 + +2008-07-11 20:03 +0000 [r130237] Mark Michelson <mmichelson@digium.com> + + * /, main/audiohook.c: Merged revisions 130236 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul + 2008) | 3 lines Remove redundant logic ........ + +2008-07-11 19:56 +0000 [r130230-130234] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c: Don't copy on NULL. + + * include/asterisk/res_odbc.h, res/res_odbc.c: Add some debug code + and add a missing release + + * channels/chan_dahdi.c, utils/astman.c: Fix trunk breakage + +2008-07-11 19:14 +0000 [r130174] Mark Michelson <mmichelson@digium.com> + + * /, main/audiohook.c: Merged revisions 130173 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul + 2008) | 7 lines Fix a typo in audiohook_read_frame_both. While + this change has not been proven to fix any specific issue, it is + incorrect and could cause unforeseen problems. ........ + +2008-07-11 18:52 +0000 [r130170] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 130169 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 + Jul 2008) | 7 lines Ensure that a destination callno of 0 will + not match for frames that do not start a dialog (new, lagrq, and + ping). (closes issue #12963) Reported by: russellb Patches: + chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492) + ........ + +2008-07-11 18:32 +0000 [r130167] Sean Bright <sean.bright@gmail.com> + + * channels/chan_sip.c: Missed one. Formatting only. + +2008-07-11 18:24 +0000 [r130145] Steve Murphy <murf@digium.com> + + * main/pbx.c: (closes issue #13041) Reported by: eliel Tested by: + murf (closes issue #12960) Reported by: mnicholson In this + 'omnibus' fix, I **think** I solved both the problem in 13041, + where unloading pbx_ael.so caused crashes, or incomplete removal + of previous registrar'ed entries. And I added code to completely + remove all includes, switches, and ignorepats that had a matching + registrar entry, which should appease 12960. I also added a lot + of seemingly useless brackets around single statement if's, which + helped debug so much that I'm leaving them there. I added a + routine to check the correlation between the extension tree lists + and the hashtab tables. It can be amazingly helpful when you have + lots of dialplan stuff, and need to narrow down where a problem + is occurring. It's ifdef'd out by default. I cleaned up the code + around the new CIDmatch code. It was leaving hanging extens with + bad ptrs, getting confused over which objects to remove, etc. I + tightened up the code and changed the call to remove_exten in the + merge_and_delete code. I added more conditions to check for empty + context worthy of deletion. It's not empty if there are any + includes, switches, or ignorepats present. If I've missed + anything, please re-open this bug, and be prepared to supply + example dialplan code. + +2008-07-11 18:09 +0000 [r130129] Brett Bryant <bbryant@digium.com> + + * codecs/codec_g722.c, channels/chan_sip.c, main/threadstorage.c, + utils/astman.c, main/utils.c, channels/chan_gtalk.c, + pbx/dundi-parser.c, main/cli.c, channels/chan_jingle.c, + channels/chan_dahdi.c, channels/chan_skinny.c, + main/abstract_jb.c, apps/app_minivm.c, codecs/codec_resample.c, + codecs/codec_dahdi.c, apps/app_chanspy.c, apps/app_milliwatt.c, + main/asterisk.c, main/dsp.c: Janitor patch to change uses of + sizeof to ARRAY_LEN (closes issue #13054) Reported by: pabelanger + Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) + Tested by: seanbright + +2008-07-11 17:29 +0000 [r130126] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_agent.c: Merged revisions 130102 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 + Jul 2008) | 9 lines Pass the devicestate from an underlying + channel up through the Agent channel. This should make the Agent + always report the correct device state, even when the underlying + channel is used for other purposes. (closes issue #12773) + Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded + by Corydon76 (license 14) Tested by: davidw ........ + +2008-07-11 16:18 +0000 [r130040-130044] Kevin P. Fleming <kpfleming@digium.com> + + * doc/ss7.txt, contrib/utils/zones2indications.c, CHANGES: clean up + a bunch more Zaptel-related references + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, + configure, include/asterisk/autoconfig.h.in, configure.ac: Merged + revisions 130039 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul + 2008) | 4 lines add support for a configuration parameter for + 'inband audio during RELEASE', which is currently mandatory in + libpri-1.4.4 but will become configurable in libpri-1.4.5 later + today (related to issue #13042) ........ + +2008-07-11 14:22 +0000 [r129985-129987] Russell Bryant <russell@digium.com> + + * /, include/asterisk/astobj.h: Merged revisions 129970 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008) + | 2 lines add a simple ASTOBJ_TRYWRLOCK macro ... ........ + + * /: remove space in property value + +2008-07-11 14:16 +0000 [r129916-129968] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/astmm.c: Merged revisions 129966 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul + 2008) | 5 lines fix a flaw found while experimenting with + structure alignment and padding; low-fence checking would not + work properly on 64-bit platforms, because the compiler was + putting 4 bytes of padding between the fence field and the + allocation memory block added a very obvious runtime warning if + this condition reoccurs, so the developer who broke it can be + chastised into fixing it :-) ........ r129967 | kpfleming | + 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines simplify + calculation ........ + + * /, sounds/Makefile: Merged revisions 129907 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r129907 | kpfleming | 2008-07-11 07:15:42 -0500 (Fri, 11 Jul + 2008) | 2 lines don't attempt to set user/group ownership of + extracted sound files (reported on asterisk-users) ........ + +2008-07-11 00:55 +0000 [r129864] Sean Bright <sean.bright@gmail.com> + + * res/res_config_pgsql.c, res/res_config_ldap.c: Fix some usages of + snprintf, and clarify a couple variable names. + +2008-07-10 22:06 +0000 [r129758-129804] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 129803 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r129803 | tilghman | 2008-07-10 16:57:05 -0500 (Thu, 10 + Jul 2008) | 8 lines Correctly deal with duplicate NEW frames (due + to retransmission). Also, fixup the destination call number + matching to be more strict and reliable. (closes issue #12963) + Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch + uploaded by jpgrayson (license 492) Tested by: jpgrayson, + Corydon76 ........ + + * res/res_config_odbc.c, /: Merged revisions 129741 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r129741 | tilghman | 2008-07-10 16:19:48 -0500 (Thu, 10 + Jul 2008) | 2 lines Oops ........ + +2008-07-10 20:56 +0000 [r129738] Terry Wilson <twilson@digium.com> + + * Makefile: Move phoneprov config files to be installed with 'make + samples' so changes aren't inadvertently lost on a 'make install' + +2008-07-10 20:33 +0000 [r129734] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Removed the fn2 field from the vm_state + structure. fn2 was used in three functions. In every case, it was + initialized in the function it was used in. This meant there was + no need to have it in a malloc'd structure just taking up space. + Furthermore two of the functions it was used in were completely + unnecessary since fn2 was set to exactly the same value as the + vm_state's fn string. fn2 was a char array sized at PATH_MAX. On + my system, PATH_MAX is 4096. This equates to a 4K memory savings + per vm_state allocated. Since there is a vm_state malloc'd for + every voicemail user on the system, this could potentially add up + nicely if there are lots of users. In addition, a vm_state is + allocated on the stack each time a caller calls the VoiceMailMain + application, meaning that there is a significant stack savings + with this patch too. Of course, a single vm_state struct still + takes up approximately 20K on my system (when using IMAP storage. + Without IMAP storage, there would be about another 300 bytes + fewer usage), even with this removal. Further optimizations are + probably possible, but most likely not as easy as this one. + +2008-07-10 19:13 +0000 [r129684] Brett Bryant <bbryant@digium.com> + + * apps/app_queue.c: Fixes a bug where the interface for a queue + member gets reloaded as the state_interface, if a state_interface + was set, on reload because the state_interface isn't stored in + the ast_db. (closes issue #13043) Reported by: jvandal Patches: + app_queue.patch uploaded by jvandal (license 413) + +2008-07-10 18:19 +0000 [r129638-129642] Sean Bright <sean.bright@gmail.com> + + * channels/chan_sip.c: A couple more minor text changes + + * channels/chan_sip.c: Remove extraneous \n. Pointed out by eliel + on #asterisk-dev. + +2008-07-10 16:21 +0000 [r129581] Michiel van Baak <michiel@vanbaak.info> + + * main/features.c: Remove deprecated 'show parkedcalls' CLI command + (closes issue #13038) Reported by: eliel Patches: + finish.deprecate.patch uploaded by eliel (license 64) + +2008-07-10 16:12 +0000 [r129569] Russell Bryant <russell@digium.com> + + * /, sample.call: Merged revisions 129567 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008) + | 3 lines Note that pbx_spool.so is the module used for call + files (inspired by a question in #asterisk) ........ + +2008-07-10 13:54 +0000 [r129503] Sean Bright <sean.bright@gmail.com> + + * main/editline: Update svn:ignore + +2008-07-09 19:40 +0000 [r129437] Mark Michelson <mmichelson@digium.com> + + * /, main/rtp.c: Merged revisions 129436 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul + 2008) | 13 lines Fix a problem where inbound rfc2833 audio would + be sent to the core instead of being P2P bridged. When the core + regenerated the rfc2833 packet for the outbound leg, the SSRC + would be different than the RTP audio on the call leg causing + DTMF detection issues on the far end. (closes issue #12955) + Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by + tsearle (license 373) Tested by: tonyredstone ........ + +2008-07-09 15:57 +0000 [r129399] Matthew Fredrickson <creslin@digium.com> + + * main/pbx.c: Add Proceeding() application (#13025) + +2008-07-09 13:44 +0000 [r129344] Sean Bright <sean.bright@gmail.com> + + * main/editline/makelist.in (added), /, main/editline/configure, + main/editline/Makefile.in, main/editline/configure.in, + main/editline/makelist (removed): Merged revisions 129343 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul + 2008) | 4 lines Look for the system installed awk instead of + assuming it's at /usr/bin/awk. Pointed out by jmls via + #asterisk-dev. ........ + +2008-07-09 03:39 +0000 [r129307] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, main/manager.c, res/res_agi.c, pbx/pbx_realtime.c, + include/asterisk/channel.h, include/asterisk/pbx.h, main/cli.c: + Code wasn't ready to be merged - see -dev list discussion + +2008-07-08 22:56 +0000 [r129270] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Fix compilation error when IMAP storage is + enabled + +2008-07-08 21:00 +0000 [r129156] Brett Bryant <bbryant@digium.com> + + * main/dnsmgr.c, main/srv.c, main/dns.c: Fix a bug in SRV lookups + where dnsmgr would discard everything but the first SRV result + from DNS before processing weights and priorities and + dns_parse_answer wouldn't report that there were no records in + DNS unless a failure occured. Also fixed a bug where + dnsmgr_refresh would report that a entry was being changed when + ast_gethostbyname had failed. + +2008-07-08 20:30 +0000 [r129048-129152] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, /, channels/chan_sip.c, + include/asterisk/causes.h: Merged revisions 129149 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 + Jul 2008) | 8 lines Cause SIP to return a 480 instead of a 404 + when a sip peer exists, but is not registered. (closes issue + #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt + uploaded by Corydon76 (license 14) Tested by: ibc ........ + + * main/asterisk.c: Reduce length of time that 'asterisk -rx' waits. + (closes issue #13001) Reported by: eliel Patches: + 20080708__bug13001.diff.txt uploaded by Corydon76 (license 14) + 20080708__bug13001.diff.txt.fixed uploaded by eliel (license 64) + Tested by: Corydon76, eliel + + * /, channels/chan_iax2.c: Merged revisions 129047 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 + Jul 2008) | 7 lines Timestamp decoding for video mini-frames is + bogus, because the timestamp only includes 15 bits, unlike voice + frames, which contain a 16-bit timestamp. (closes issue #13013) + Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch + uploaded by jpgrayson (license 492) ........ + +2008-07-08 16:40 +0000 [r129045] Brett Bryant <bbryant@digium.com> + + * main/pbx.c, main/frame.c, channels/chan_sip.c, apps/app_meetme.c, + channels/h323/ast_h323.cxx, res/res_limit.c, main/acl.c, + channels/iax2-provision.c, pbx/dundi-parser.c, + channels/chan_iax2.c, main/rtp.c, main/channel.c, + channels/chan_dahdi.c, main/manager.c, formats/format_pcm.c, + main/callerid.c, main/logger.c, apps/app_parkandannounce.c, + apps/app_adsiprog.c: Janitor project to convert sizeof to + ARRAY_LEN macro. (closes issue #13002) Reported by: caio1982 + Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) + +2008-07-08 14:17 +0000 [r129006] Russell Bryant <russell@digium.com> + + * apps/app_fax.c: Update app_fax for better compatibility with + spandsp 0.0.5. Add a call to t38_terminal_release, and make sure + that the phase E handler gets called with proper status. (closes + issue #13020) Reported by: dimas Patches: v1-appfax.patch + uploaded by dimas (license 88) + +2008-07-08 10:02 +0000 [r128927-128951] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 128950 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 + lines Don't hangup the call if we can't resolve the Contact if + there's a proxy route set for the call. ---- This comment was + added a while ago and today it hit me badly. /* OEJ: Possible + issue that may need a check: If we have a proxy route between us + and the device, should we care about resolving the contact or + should we just send it? */ ........ + + * /, channels/chan_sip.c: Merged revisions 128912 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 + lines Fix issues where repeated messages where ignored, but + retransmitted reliably instead of unreliably. Reported by: johan + Patches: 12746.txt uploaded by oej (license 306) Tested by: johan + (issue #12746) ........ + +2008-07-08 00:02 +0000 [r128830-128857] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 128856 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 + Jul 2008) | 7 lines Check for non-NULL before stripping + characters. (closes issue #12954) Reported by: bfsworks Patches: + 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14) + Tested by: deti ........ + + * /, apps/app_voicemail.c: Merged revisions 128812 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 + Jul 2008) | 2 lines Stop using deprecated method, as requested by + Kevin. ........ + +2008-07-07 22:42 +0000 [r128796] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 128795 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 + Jul 2008) | 8 lines Fix handling of when a pvt disappears. + Properly return the pvt locked and don't hold the pvt lock while + destroying the ast_channel. (closes issue #13014) Reported by: + jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by + jpgrayson (license 492) ........ + +2008-07-07 20:50 +0000 [r128738] Sean Bright <sean.bright@gmail.com> + + * /, channels/chan_iax2.c: Merged revisions 128737 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, + 07 Jul 2008) | 9 lines Remove spurious trailing whitespace from + log messages and fix a spelling error in a log message. (closes + issue #13017) Reported by: jpgrayson Patches: + chan_iax2_space_after_newline.patch uploaded by jpgrayson + (license 492) chan_iax2_spelling.patch uploaded by jpgrayson + (license 492) ........ + +2008-07-07 20:30 +0000 [r128599-128733] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Crap + + * apps/app_voicemail.c: If imapfolder=foo were set in + voicemail.conf, then when calling VoiceMailMain, app_voicemail + would attempt to play a file called vm-foo instead of playing + vm-INBOX to play the "new" sound file. This commit fixes that + issue. This may fix one of the problems reported in issue #12987 + + * apps/app_voicemail.c: Get app_voicemail compiling when IMAP + storage is used. Brought up by reporter on issue #12987 + + * /, channels/chan_iax2.c: Merged revisions 128639 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, + 07 Jul 2008) | 10 lines By using the iaxdynamicthreadcount to + identify a thread, it was possible for thread identifiers to be + duplicated. By using a globally-unique monotonically- increasing + integer, this is now avoided. (closes issue #13009) Reported by: + jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by + jpgrayson (license 492) ........ + + * doc/tex/extensions.tex, configs/extensions.conf.sample: Update a + few instances of "extensions reload" to "dialplan reload" in the + documentation. Patch provided by caio1982 (license 22) + +2008-07-07 11:53 +0000 [r128564] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: As pointed out on the -dev list, actually + use the result of find_peer() so that a peer reference is not + leaked. + +2008-07-06 20:19 +0000 [r128274-128525] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, configs/sip.conf.sample: - Adding alias + "udpbindaddr" for the UDP port to comply with "tcpbindaddr" and + "tlsbindaddr". Note: I don't think we can start properly without + UDP port open, that needs to be tested. - Removing "bindport" + from configuration example, not needed to mention this any more I + suggest we deprecate "bindaddr" and "bindport" in trunk (for + 1.6.1) + + * channels/chan_sip.c, configs/sip.conf.sample: - Fixing issues + with "sip show settings" - Adding IP address for TCP and/or TLS + too if auto-domain is enabled and binding to a different IP + address - Fixing documentation in sip.conf.sample + + * channels/chan_sip.c: - Remove unused variable "expiry" - Set + global_outboundproxy.force at reload. + + * channels/chan_sip.c: More doxygen comments. + + * channels/chan_sip.c: - Formatting changes - Doxygen changes - + Replacing a doxygen description that was copied from another + function + + * channels/chan_sip.c: Adding note about incorrect manager + registration... + + * doc/realtimetext.txt (added): Adding documentation on the T.140 + support in Asterisk. This is a function that we're the reference + implementation on now. :-) + + * channels/chan_sip.c: Remove comments that doesn't make sense. The + deprecation of type=user will come at a later stage, as indicated + by previous commit message + + * channels/chan_sip.c: Fix severe problem with my previous commit + of "kill-the-user". Russell saw a problem with this code, but not + the correct problem. Thanks, anyway! ;-) + + * main/pbx.c, main/manager.c, pbx/pbx_realtime.c, + include/asterisk/pbx.h: Changing name of global api call to ast_* + My mistake, pointed out by Russell. + + * channels/chan_sip.c: Disabling code used by dumpdb with #ifdef, + since I believe we might use it sometime in the future, but also + want to avoid compiler warnings now. + + * channels/chan_sip.c: Removing the CLI dumpdb command (see + asterisk-dev discussion and decision) + + * channels/chan_sip.c: Adding a few reminders + + * channels/chan_sip.c: Adding doxygen comments to missing parts, + moving some #define ...trying to get my head around the thoughts + behind the TCP/TLS stuff and figure out what needs to be done to + make it useful... + + * channels/chan_sip.c: Adding TCP and TLS to "sip show settings". + TLS needs to have one configuration per configured domain at some + point. + + * channels/chan_sip.c: Add some comments... + + * channels/chan_sip.c: Set tls setting to default in reload_config. + +2008-07-05 21:20 +0000 [r128254] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_sip.c: fix compiling of chan_sip.c + +2008-07-05 21:11 +0000 [r128247] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: I like it when the tree is not broken. + +2008-07-05 20:59 +0000 [r128201-128242] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: KILL THE USER! Actually, kill the in-memory + structure for type=user and start using the sip_peer structure + for every object. Have only one in-memory list and use them + different ways depending on type=user, type=peer and type=friend + - like always. The idea with this first patch is that + configurations should work as before. Some additional features + for realtime peers. By adding a type= field, you can now have + multiple functions. Let's test this for a while. Won't be + integrated in 1.6.0, only in trunk, for testing. There's propably + more to clean up and simplify here. Help is welcome and + encouraged! + + * main/pbx.c, main/manager.c, res/res_agi.c, pbx/pbx_realtime.c, + include/asterisk/channel.h, include/asterisk/pbx.h, main/cli.c: + Implement flags for AGI in the channel structure so taht "show + channels" and AMI commands can display that a channel is under + control of an AGI. Work inspired by work at customer site, but + paid for by Edvina AB + + * configs/sip.conf.sample: Make TCP disabled by default (it's + considered experimental) + + * configs/sip.conf.sample: Reformatting the config sample + + * channels/chan_sip.c: Stop cli command completion with tabs + +2008-07-05 19:52 +0000 [r128198] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Make this actually evaluate how it was intended to + be. + +2008-07-05 19:27 +0000 [r128197] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: Add new + SIP cli command "sip show channelstats" that displays some QoS + data (if we have RTCP reports and not use the p2p rtp bridge). I + could not find a way to detect us using the p2p bridge, which + would be nice. + +2008-07-05 15:17 +0000 [r128160] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif: LDAP schema updates (closes issue + #12860) Reported by: flyn Patches: asterisk.ldif uploaded by + suretec (license 70) asterisk.schema uploaded by suretec (license + 70) + +2008-07-05 03:39 +0000 [r128122-128125] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c: It would help if we actually parsed the + ss7_explicitacm option in the config file... + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add option + to wait to be able to explicitly send ACM via the Proceeding() + application in the dialplan. Also minor documentation update + explaining how to setup multiple signalling links within a + linkset + +2008-07-04 16:41 +0000 [r128027-128082] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Fullcontact needs more than 20 characters, + even for the simplest case + + * main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: Merged + revisions 127973 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) + | 8 lines Fix the 'dialplan remove extension' logic, so that it + a) works with cidmatch, and b) completes contexts correctly when + the extension is ambiguous. (closes issue #12980) Reported by: + licedey Patches: 20080703__bug12980.diff.txt uploaded by + Corydon76 (license 14) Tested by: Corydon76 ........ + +2008-07-04 14:36 +0000 [r127995] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: - reorganize SIP extensions alphabetically, + to make it easier to synch with the IANA list - add a few new + registered and well-known extension names + +2008-07-03 22:47 +0000 [r127931-127934] Brett Bryant <bbryant@digium.com> + + * channels/iax2-parser.c: Fix one more file that got changed. + + * channels/iax2.h, channels/chan_iax2.c: Remove commit that somehow + got mergeed into trunk. + + * channels/iax2.h, channels/chan_iax2.c, channels/iax2-parser.c: + Update these files with transfer code. + +2008-07-03 22:23 +0000 [r127903] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /, apps/Makefile, main/editline/np/vis.c: Merged + revisions 127892,127895 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul + 2008) | 6 lines a couple of small Solaris-related fixes (closes + issue #11885) Reported by: snuffy, asgaroth ........ r127895 | + kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3 + lines remove this, it has been moved to the main Makefile + ........ + +2008-07-03 20:59 +0000 [r127831-127857] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c: Make change proposed by andrew53 on + bugtracker + + * apps/app_chanspy.c: Thanks to a suggestion from seanbright, print + a warning if the attachment of the whisper or barge audiohooks + fails. + + * apps/app_chanspy.c: Fix build + + * apps/app_chanspy.c: Fix a crash when attempting to spy on an + unbridged channel. (closes issue #12986) Reported by: andrew53 + +2008-07-03 17:16 +0000 [r127793] Steve Murphy <murf@digium.com> + + * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, /, + channels/chan_sip.c, main/features.c, include/asterisk/cdr.h: + Merged revisions 127663 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | + 30 lines The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) + Reported by: murf Tested by: murf, deeperror (closes issue + #12907) Reported by: falves11 Tested by: murf, falves11 (closes + issue #11849) Reported by: greyvoip As to 11849, I think these + changes fix the core problems brought up in that bug, but perhaps + not the more global problems created by the limitations of CDR's + themselves not being oriented around transfers. Reopen if necc, + but bug reports are not the best medium for enhancement + discussions. We need to start a second-generation CDR + standardization effort to cover transfers. (closes issue #11093) + Reported by: rossbeer Tested by: greyvoip, murf ........ + +2008-07-03 16:48 +0000 [r127779-127791] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Make sure we stop session timers as soon as + we start hanging up an active call. May fix issue 12919. + + * channels/chan_sip.c: Revert some logic for session timers. We do + send in-dialog requests that should not have session-timer + require headers, like MESSAGE and REFER. So in the future, only + add them on requests and responses that are related to INVITEs + and re-INVITEs. + +2008-07-03 16:22 +0000 [r127767] Kevin P. Fleming <kpfleming@digium.com> + + * /, configure, configure.ac, acinclude.m4: some minor fixes found + while working on issue #12911 (and block the rev from 1.4 since + the equivalent is already here) + +2008-07-03 14:34 +0000 [r127720] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Added a + new option, "timeoutpriority" to queues.conf. A detailed + explanation of the change may be found in + configs/queues.conf.sample (closes issue #12690) Reported by: + atis + +2008-07-03 09:59 +0000 [r127685] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Fix bad formatting in a very confusing + function. Who added the sipdb sql output? It's mixing peers and + users in a strange way and should really not be a CLI command, + since it's not meant for human output. It should be done with an + app connecting to manager. + +2008-07-02 22:17 +0000 [r127622] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Oops + +2008-07-02 22:16 +0000 [r127621] Brett Bryant <bbryant@digium.com> + + * channels/chan_sip.c: Update transport= in sip so that the option + is not broken from a recent commit. + +2008-07-02 21:27 +0000 [r127609] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_unistim.c, include/asterisk/app.h, main/manager.c, + channels/chan_sip.c, main/app.c, channels/chan_iax2.c, + apps/app_voicemail.c: Keep ast_app_inboxcount API compatible with + 1.6.0. + +2008-07-02 21:09 +0000 [r127566] Mark Michelson <mmichelson@digium.com> + + * doc/janitor-projects.txt: Add a janitor project to use ARRAY_LEN + instead of in-line sizeof() and division. + +2008-07-02 20:52 +0000 [r127564] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Fix some crashlike bugs because flag could + be NULL in play_record_review(). (Closes issue #12892) Reported + by: jaroth Patch originally by jaroth, fixed by me. + +2008-07-02 20:49 +0000 [r127558-127562] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 127560 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, + 02 Jul 2008) | 3 lines Fix thread-safety of some of the + pbx_builtin_getvar_helper calls ........ + + * configs/agents.conf.sample, channels/chan_agent.c, CHANGES: The + ackcall and endcall options in agents.conf now have supplemental + options acceptdtmf and enddtmf. These allow for the DTMF pressed + to be configurable instead of being hardcoded to '#' and '*'. + (AST-86) + +2008-07-02 20:28 +0000 [r127545] Terry Wilson <twilson@digium.com> + + * include/asterisk/http.h, main/http.c: Expose the prefix variable + so that it can be used by modules depending on http support + +2008-07-02 18:31 +0000 [r127466] Tilghman Lesher <tlesher@digium.com> + + * main/acl.c: Solaris fix (closes issue #12949) Reported by: snuffy + Patches: bug_12949.diff uploaded by snuffy (license 35) + +2008-07-02 17:27 +0000 [r127434] Brett Bryant <bbryant@digium.com> + + * channels/chan_sip.c: Fix to sip_parse_host so that it passes the + correct information to sip_registry. + +2008-07-02 14:50 +0000 [r127401] Russell Bryant <russell@digium.com> + + * include/asterisk/logger.h, include/asterisk/devicestate.h, + include/asterisk/astobj2.h, include/asterisk/timing.h, + include/asterisk/strings.h, include/asterisk/dnsmgr.h, + include/asterisk/threadstorage.h, include/asterisk/slinfactory.h, + main/libresample/include/libresample.h, include/asterisk/time.h: + Fix a bunch of places where \arg was used instead of \param. + Using \arg to document arguments seems logical, and does work, + but is not the best thing to use. \arg in doxygen is simply for + creating non-nested unordered lists. \param is the correct tag to + use to document function parameters, and will come out better in + the generated documentation. + +2008-07-02 14:30 +0000 [r127398] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_tds.c: Fix a bug I noticed while doing the previous merge + +2008-07-02 12:08 +0000 [r127363] Russell Bryant <russell@digium.com> + + * doc/CODING-GUIDELINES: Add a locking section to the coding + guidelines document. This section covers some locking + fundamentals, as well as some information on locking as it is + used in Asterisk. It describes some of the ways that are used and + could be used to achieve deadlock avoidance. It also demonstrates + the unfortunate conclusion that with the use of recursive locks, + none of the constructs in use today are failsafe from deadlocks. + Finally, it makes some recommendations for new code being + written. As proper locking strategies is a complex subject, this + section still has room for expansion and improvement. This is a + result of collaboration between Luigi Rizzo and myself on the + asterisk-dev mailing list. + +2008-07-02 12:06 +0000 [r127330-127362] Luigi Rizzo <rizzo@icir.org> + + * channels/console_video.c: plug another panic when the gui cannot + be started. We can still send video, just don't try to use what + is not available. + + * channels/console_video.c: prevent a segfault when trying to start + the gui without any specific configuration in oss.conf (reported + by Klaus Darillion on the -video mailing list). + +2008-07-02 02:48 +0000 [r127297] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Change the global timer B to be dependent on + the value of the T1 timer, as recommended in RFC 3261, instead of + being hardcoded to 32 seconds. This is important for LANs, as it + allows autocongestion to occur much more quickly, if desired by + the local PBX administrator. It also corrects a bug: if the T1 + timer was increased beyond 500ms, then timer B would have been + set at a much lower value than recommended. (closes issue #12544) + Reported by: kactus Patches: 20080616__bug12544.diff.txt uploaded + by Corydon76 (license 14) Tested by: kactus + +2008-07-01 23:38 +0000 [r127245] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 127244 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, + 01 Jul 2008) | 5 lines Add error message to failed open(2) calls + inside the copy() function of app_voicemail. This idea came as + part of my work in helping to resolve issue #12764. ........ + +2008-07-01 21:43 +0000 [r127210] Russell Bryant <russell@digium.com> + + * funcs/func_devstate.c: Add a \todo + +2008-07-01 21:21 +0000 [r127169] Tilghman Lesher <tlesher@digium.com> + + * res/res_musiconhold.c: Add AMI events for start/stop of MOH + (closes issue #12909) Reported by: chris-mac Patches: + res_musiconhold-event.patch uploaded by chris-mac (license 506) + +2008-07-01 21:16 +0000 [r127157] Mark Michelson <mmichelson@digium.com> + + * main/channel.c: Place the delay in __ast_answer prior to the + channel-specific answer callback. This change differs from commit + 127113 in that now the channel is not set to AST_STATE_UP until + after the answer callback. (closes issue #12924) Reported by: + snyfer + +2008-07-01 21:03 +0000 [r127154] Brett Bryant <bbryant@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample: Add a configuration + option so the global outboundproxy can use tcptls without it + being defined by each sip user. + +2008-07-01 20:51 +0000 [r127152] Jason Parker <jparker@digium.com> + + * Makefile: Fix a typo that caused this asterisk.conf to not get + correctly generated. (closes issue #12966) Reported by: ibc + Patches: 12966.patch uploaded by bkruse (license 132) + +2008-07-01 20:28 +0000 [r127143] Tilghman Lesher <tlesher@digium.com> + + * build_tools/cflags.xml, /, channels/chan_iax2.c: Merged revisions + 127133 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r127133 | tilghman | 2008-07-01 15:25:37 -0500 (Tue, 01 Jul 2008) + | 2 lines Disable the old, slow search for matching callno in + chan_iax2 (but allow it to be reenabled for debugging) ........ + +2008-07-01 19:53 +0000 [r127113] Kevin P. Fleming <kpfleming@digium.com> + + * main/channel.c: change the process of inserting a delay into the + ast_answer() path so that we don't tell the calling channel that + it has been answered unutil after the delay; for a single-thread + call this won't matter all, but for a dual-thread call (using + chan_local) this may fix the problem in issue 12924 + +2008-07-01 19:20 +0000 [r127074] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 127068 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r127068 | tilghman | 2008-07-01 13:52:53 -0500 (Tue, 01 + Jul 2008) | 8 lines Change around how we schedule pings and + lagrqs, and fix a reason why the jobs were not getting properly + cancelled. (closes issue #12903) Reported by: stevedavies + Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76 + (license 14) Tested by: stevedavies ........ + +2008-07-01 17:22 +0000 [r127017] Kevin P. Fleming <kpfleming@digium.com> + + * res/res_ais.c, build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, res/Makefile, + res/ais/ais.h, makeopts.in: make the AIS checking a little more + generic, and have a more useful configure script command line + option for OpenAIS + +2008-07-01 16:52 +0000 [r127000] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 126999 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r126999 | tilghman | 2008-07-01 11:50:46 -0500 (Tue, 01 + Jul 2008) | 2 lines Suppress annoying warning by finding the + remaining cases where the callno is not in the hash. ........ + +2008-07-01 16:28 +0000 [r126991] Luigi Rizzo <rizzo@icir.org> + + * images/kpad2.jpg: even uglier gui with more buttons + +2008-07-01 16:16 +0000 [r126960] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_dial.c, include/asterisk/channel.h, apps/app_queue.c: + another minor ast_channel memory size decrease... for nearly all + channels, 'dialcontext' is only going to be set once during the + channel's lifetime, so make it a string field instead of a char + array + +2008-07-01 16:14 +0000 [r126959] Luigi Rizzo <rizzo@icir.org> + + * doc/video.txt, doc/video_console.txt (added): add documentation + on video console support + +2008-07-01 15:03 +0000 [r126845-126903] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 126902 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r126902 | oej | 2008-07-01 16:59:31 +0200 (Tis, 01 Jul 2008) | 7 + lines Use domain part of SIP uri in register= configuration as + fromdomain. Reported by: one47 Patches: sip-reg-fromdom2.dpatch + uploaded by one47 (license 23) (closes issue #12474) ........ + + * /, channels/chan_sip.c: Merged revisions 126899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r126899 | oej | 2008-07-01 16:27:33 +0200 (Tis, 01 Jul 2008) | 8 + lines Handle escaped URI's in call pickups. Patch by oej and + IgorG. Reported by: IgorG Patches: bug12299-11062-v2.patch + uploaded by IgorG (license 20) Tested by: IgorG, oej (closes + issue #12299) ........ + + * /, configs/sip.conf.sample: Merged revisions 126844 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul + 2008) | 5 lines Clear up documentation on "domain=" setting in + sip.conf Reported by: davidw (closes issue #12413) ........ + +2008-07-01 12:29 +0000 [r126835] Luigi Rizzo <rizzo@icir.org> + + * main/logger.c: use %p to print a pointer + +2008-07-01 11:58 +0000 [r126755-126790] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 126789 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6 + lines Report 200 OK to all in-dialog OPTIONs requests (to confirm + that the dialog exist). Don't bother checking the request URI. + (closes issue #11264) Reported by: ibc ........ + + * /, channels/chan_sip.c: Merged revisions 126735 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r126735 | oej | 2008-07-01 09:49:15 +0200 (Tis, 01 Jul 2008) | 7 + lines Fix bad XML for hold notification. Reported by: gowen72 + Patches: hold.patch uploaded by gowen72 (license 432) (closes + issue #12942) ........ + +2008-06-30 22:34 +0000 [r126675] Jeff Peeler <jpeeler@digium.com> + + * configs/chan_dahdi.conf.sample (added), + configs/zapata.conf.sample (removed): rename zapata.conf.sample + to chan_dahdi.conf.sample + +2008-06-30 20:25 +0000 [r126637] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c: Add support to see MTP2 down events when + the link layer drops in SS7 + +2008-06-30 16:07 +0000 [r126574] Russell Bryant <russell@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 126573 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 + Jun 2008) | 10 lines Fix a typo in the non-DEBUG_THREADS version + of the recently added DEADLOCK_AVOIDANCE() macro. This caused the + lock to not actually be released, and as a result, not avoid + deadlocks at all. This resolves the issues reported in the last + while about Asterisk locking up all over the place (and most + commonly, in chan_iax2). (closes issue #12927) (closes issue + #12940) (closes issue #12925) (potentially closes others ...) + ........ + +2008-06-30 15:45 +0000 [r126571-126572] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c, channels/console_video.c, + channels/chan_oss.c, channels/console_video.h: implement the + 'freeze' function for incoming frames; fix a bug which caused a + crash when a videodevice was specified after startgui=1 in the + config file. This also involves a slightly different method to + determine if the gui is active or not. + + * apps/app_voicemail.c: fix an uninitialized variable + +2008-06-30 13:03 +0000 [r126517] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: The following patch with some changes for + trunk... Merged revisions 126516 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | + 10 lines Send all responses to an INVITE reliably, so that we + retransmit if we don't get an ACK and also fail if we don't get + the very same precious ACK. Based on patch by tsearle, with my + own additions. (closes issue #12951) Reported by: tsearle + Patches: busy_retransmit.patch uploaded by tsearle (license 373) + ........ + +2008-06-30 12:49 +0000 [r126515] Russell Bryant <russell@digium.com> + + * doc/CODING-GUIDELINES: a few minor updates and typo fixes + +2008-06-30 11:57 +0000 [r126513] Sean Bright <sean.bright@gmail.com> + + * doc/tex/freetds.tex, cdr/cdr_tds.c: Cast a few more strings to + char *, so that we can compile cleanly against FreeTDS 0.60. + Update the docs to reflect that we can now compile and run + against all modern releases of FreeTDS (0.60 through 0.82) + +2008-06-29 21:17 +0000 [r126448-126480] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c, channels/console_video.c, + channels/console_board.c: import the recent additions for video + console into trunk, giving you support for up to 9 video sources + (e.g. webcams, or X11 grabbers, etc.) active at once, displaying + thumbnails for each of them in the main GUI window, and with the + ability to switch between them on the fly during a conversation. + The code also implements a 'Picture in Picture' feature, allowing + you to select any source as primary or secondary, and move the + PiP window by just dragging it with the mouse. The window looks + like this: + ________________________________________________________________ + | ______ ______ ______ ______ ______ ______ ______ | | | tn.1 | | + tn.2 | | tn.3 | | tn.4 | | tn.5 | | tn.6 | | tn.7 | | | |______| + |______| |______| |______| |______| |______| |______| | | ______ + ______ ______ ______ ______ ______ ______ | | |______| |______| + |______| |______| |______| |______| |______| | | + _________________ __________________ _________________ | | | | | + | | | | | | | | | | | | | | | | | | | | | | remote video | | | | + local video | | | | | | | | ______ | | | | | | keypad | | | PIP + || | | | | | | | |______|| | | |_________________| | | + |_________________| | | | | | | | | | | |__________________| | + |________________________________________________________________| + + * channels/console_gui.c, channels/console_board.c, + channels/console_video.h: fix wrong argument in checking + boundaries for a rectangle some whitespace fixes + +2008-06-29 16:19 +0000 [r126356] Kevin P. Fleming <kpfleming@digium.com> + + * configure, configure.ac, pbx/pbx_lua.c, pbx/Makefile, + pbx/pbx_gtkconsole.c: various minor fixes created while i worked + on getting *every* Asterisk module to build on laptop in dev + mode: remove weird pre-setting of LUA paths; they are not + necessary; also use the proper path for including the files in + pbx_lua.c add searching for OpenAIS libraries in /usr/lib/openais + if a path is not specified; not sure if this is really the + optimal solution, but it works make the compiler shut up about + some ignored function results in pbx_gtkconsole; this module is + badly coded anyway + +2008-06-29 13:20 +0000 [r126312-126319] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_tds.c: This was bogus, need to find a better way. + + * cdr/cdr_tds.c: While we're at it, escape all the columns in our + TDS queries as well. Double quotes seems to be more standard than + square brackets (Sybase and SQL Server both support them). + +2008-06-29 13:02 +0000 [r126308-126311] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_oss.c: implement a 'toggle' option for 'console + mute' and 'console unmute' + + * channels/console_video.h: add some defines and fields in + preparation for the import of the video source switching support + + * channels/vgrabbers.c: accept any name starting with X11 for X11 + grabbers - this lets you have multiple active instances of this + grabber; require v4l device names to start with '/dev/' - + prevents some useless attempt to open a file as a device. + + * channels/vcodecs.c, channels/console_video.c: make this compile + after ast_frame's data field changed to a union + +2008-06-29 12:06 +0000 [r126226-126274] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_pgsql.c: Quote column names when inserting CDRs into + postgres to avoid conflicts with reserved words. (closes issue + #12947) Reported by: panolex + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + UPGRADE.txt, cdr/cdr_tds.c: Merge in changes from my + cdr-tds-conversion branch. This changes the internal + implementation from using the volatile libtds, to using the + db-lib front end. The unintended side effect of this is that we + support (at least) versions 0.62 through 0.82 of the FreeTDS + distribution without any #ifdef ugliness. (closes issue #12844) + Reported by: jcollie + +2008-06-28 15:54 +0000 [r126152-126187] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/channel.h: yay for airplane ride + optimizations... sort the fields in ast_channel by alignment + requirements, saving 36 bytes per instance on a 64-bit platform + + * Makefile: fix silly syntax error + + * Makefile: add message when no UI for menuselect is present + + * Makefile: use batch-mode (no user interface) menuselect for + --check-deps operations move automatic user interface selection + for menuselect to this Makefile + +2008-06-27 23:29 +0000 [r126115] Sean Bright <sean.bright@gmail.com> + + * main/cdr.c: Pretty up the 'cdr show status' output. + +2008-06-27 22:10 +0000 [r126021-126057] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 126056 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008) + | 4 lines When we get a 408 Timeout, don't stop trying to + re-register. (closes issue #12863) Reported by: ricvil ........ + + * contrib/scripts/dbsep.cgi: Separate multiple items encoded into a + single field with ';' + +2008-06-27 19:19 +0000 [r125988] Russell Bryant <russell@digium.com> + + * doc/siptls.txt: Fix a typo. Someone on IRC copied this literally + and then wondered why it wasn't working. :) + +2008-06-27 19:05 +0000 [r125980-125984] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c: Revert this part of the fix. We'll fix it + in libss7 + + * channels/chan_dahdi.c: Obviously somebody didn't compile with + libss7 support when doing the DAHDI conversion. + + * channels/chan_dahdi.c: Add support for new commands to + block/unblock all CICs on a linkset + +2008-06-27 17:35 +0000 [r125947] Brett Bryant <bbryant@digium.com> + + * channels/chan_sip.c: Small error in the function that converts + peer transports to a string. + +2008-06-27 17:02 +0000 [r125895] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/lock.h: Document DLA_UNLOCK and DLA_LOCK + +2008-06-27 16:28 +0000 [r125891] Brett Bryant <bbryant@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample: Change the way that + the transport option works for sip users. transport will now take + multiple arguments, the first one listed will be the one used for + new dialogs, and the rest listed will be acceptable ways for that + peer to contact us. This fixes a minor bug where, because SIP + TCP/UDP run on the same port, could cause a TCP peer to be saved + in the ast_db. There will also be warnings when a transport is + changed for an unexpected reason. (issue #12799) + +2008-06-27 16:23 +0000 [r125855-125880] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/lock.h: Optimization suggested by Russell to + cache the value of pthread_self() so that it isn't evaluated + every time through the loop. + + * apps/app_queue.c: Remove debug message + + * apps/app_queue.c: Ensure the thread-safety of the monexec + variable in app_queue. Thanks to Russell for pointing out the + problem + +2008-06-27 16:00 +0000 [r125853] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c: Revert half of the fix, as this part may + have been unnecessary (related to issue #12914) Requested here: + http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html + +2008-06-27 14:14 +0000 [r125799] Mark Michelson <mmichelson@digium.com> + + * utils/Makefile: Remove an unneeded target from the Makefile + +2008-06-27 14:08 +0000 [r125741-125796] Tilghman Lesher <tlesher@digium.com> + + * Makefile: Push relatively unused compiler options down the list, + keeping the popular options at the top. + + * /, main/utils.c, include/asterisk/lock.h: Merged revisions 125793 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008) + | 2 lines In this debugging function, copy to a buffer instead of + using potentially unsafe pointers. ........ + + * channels/chan_local.c, /: Merged revisions 125740 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 + Jun 2008) | 7 lines Add proper deadlock avoidance. (closes issue + #12914) Reported by: ozan Patches: 20080625__bug12914.diff.txt + uploaded by Corydon76 (license 14) Tested by: ozan ........ + +2008-06-27 07:28 +0000 [r125703] Philippe Sultan <philippe.sultan@gmail.com> + + * include/asterisk/jabber.h, res/res_jabber.c: Fix a compile time + error that occurs if OpenSSL is not installed. Reported by Noel + Morais on the users mailing list + +2008-06-27 00:22 +0000 [r125647-125666] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Make this compile with dev-mode on + + * apps/app_queue.c: The monitor-join option for queues was + deprecated in favor of using MixMonitor to mix audio. However, it + was pointed out to me that because of this, the command set for + the MONITOR_EXEC variable is ignored as well. This means that + people can't do their own custom mixing commands at the end of + recordings in order to make, for instance, stereo recordings of + calls. With this patch, app_queue will set the "joinfiles" + variable for the channel's monitor if MONITOR_EXEC is not + zero-length. This means that for normal audio mixing, MixMonitor + is still the preferred choice, but we allow custom mixing to be + done with the two Monitor streams if desired. (closes issue + #12923) Reported by: snyfer + + * apps/app_dial.c, CHANGES: Improve consistency between app_dial + and app_queue with regards to how language is handled between two + channels whose native language is different. Prior to this patch, + app_dial would have the callee inherit the caller's language, and + app_queue would not. After this patch, app_dial no longer has the + language inheritance capability. This seems to make the most + sense since it seems more natural for a person to hear files + played back in his/her native language instead of the language of + the person on the far end of the call. See the CHANGES file for + hints on how to keep the previous behavior of app_dial if + desired. (closes issue #12489) Reported by: bcnit + +2008-06-26 23:18 +0000 [r125593-125596] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: remove block of commented code to set + __ourip This is now handled in skinny_register and load_config. + part two of chan_skinny cleanup + + * channels/chan_skinny.c: remove paging device from chan_skinny. + This has never been used, and noone could give us info about what + it is used for. + +2008-06-26 23:06 +0000 [r125591] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix a really stupid mistake + +2008-06-26 23:04 +0000 [r125589] Jason Parker <jparker@digium.com> + + * /, main/utils.c: Merged revisions 125587 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r125587 | qwell | 2008-06-26 18:03:15 -0500 (Thu, 26 Jun 2008) | + 1 line Make sure to unlock the lock_info lock (huh?). Possible + deadlock? ........ + +2008-06-26 23:01 +0000 [r125586] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 125585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun + 2008) | 11 lines Add the interface of a queue member to the + output of the "queue show" command so that it can easily be + associated with a queue member's name. This helps so that the + appropriate queue member can be removed or paused since the + interface is required, not the member's name. (closes issue + #12783) Reported by: davevg Patches: app_queue.diff uploaded by + davevg (license 209) with small mod from me ........ + +2008-06-26 22:49 +0000 [r125583] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/astcli: Don't hang if the command is blank + +2008-06-26 20:57 +0000 [r125477] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Merged revisions 125476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun + 2008) | 11 lines Prior to this patch, the "queue show" command + used cached information for realtime queues instead of giving + up-to-date info. Now realtime is queried for the latest and + greatest in queue info. (closes issue #12858) Reported by: bcnit + Patches: queue_show.patch uploaded by putnopvut (license 60) + ........ + +2008-06-26 17:40 +0000 [r125386-125438] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Don't play "your message has been saved" + twice. (closes issue #12893) Reported by: jaroth Patches: + duplicate_saved.patch uploaded by jaroth (license 50) + + * codecs/codec_lpc10.c, codecs/codec_a_mu.c, codecs/codec_g722.c, + codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c, + main/translate.c, codecs/codec_g726.c, codecs/codec_gsm.c, + codecs/codec_resample.c, codecs/codec_ulaw.c, + codecs/codec_ilbc.c, include/asterisk/translate.h: Convert casts + to unions, to fix alignment issues on Solaris (closes issue + #12932) Reported by: snuffy Patches: bug_12932_20080627.diff + uploaded by snuffy (license 35) + +2008-06-26 16:54 +0000 [r125385] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 125384 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3 + lines Add support for peer realm based auth (a few missing lines, + the rest is well documented but never worked) ........ + +2008-06-26 15:50 +0000 [r125333] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 125327 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 + Jun 2008) | 5 lines ensure that (whenever possible) if we + generate a log message because an ioctl() call to DAHDI/Zaptel + failed, that we include the reason it failed by including the + stringified error number (issue AST-80) ........ + +2008-06-26 15:37 +0000 [r125332] Russell Bryant <russell@digium.com> + + * main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c, + include/asterisk/timing.h, main/timing.c: - add get_max_rate + timing API call - change ast_settimeout() to honor max rate in + edge cases of file playback (this will make some warning messages + go away at the end of playing back a file) + +2008-06-26 12:09 +0000 [r125279] Kevin P. Fleming <kpfleming@digium.com> + + * res/res_musiconhold.c: fix compile failure found by buildbot (go, + buildbot!) + +2008-06-26 11:02 +0000 [r125191-125277] Tilghman Lesher <tlesher@digium.com> + + * /, main/rtp.c: Merged revisions 125276 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008) + | 7 lines Check for rtcp structure before trying to delete + schedule. (closes issue #12872) Reported by: destiny6628 Patches: + 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14) + Tested by: destiny6628 ........ + + * configs/agents.conf.sample, /: Merged revisions 125218 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008) + | 4 lines Document ackcall=always. (closes issue #12852) Reported + by: davidw ........ + + * configs/http.conf.sample: Update sample configuration to match + what are now the defaults for the prefix. (closes issue #12838, + related to issue #12198) Reported by: pabelanger Patches: + http.conf.diff2 uploaded by pabelanger (license 224) + +2008-06-25 23:05 +0000 [r125138] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_dahdibarge.c, /, apps/app_meetme.c, main/Makefile, + apps/app_dahdiscan.c, apps/app_dahdiras.c, configure.ac, + res/res_timing_dahdi.c, include/asterisk/dahdi.h (removed), + res/res_musiconhold.c, main/channel.c, channels/chan_dahdi.c, + apps/app_flash.c, configure, codecs/codec_dahdi.c, + apps/app_rpt.c, main/asterisk.c: Merged revisions 125132 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun + 2008) | 10 lines allow tonezone to live in a different place than + DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate + packages and can be installed in different places don't include + tonezone.h in dahdi_compat.h, because only a couple of modules + need it get app_rpt building again after the DAHDI changes + (closes issue #12911) Reported by: tzafrir ........ + +2008-06-25 22:40 +0000 [r125133-125135] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/lock.h: Fix indentation + + * include/asterisk/lock.h: Fix a bug in the rwlock tracking. + ast_rwlock_unlock did not take into account that multiple threads + could hold the same rdlock at the same time. As such, it expected + that when a thread released a lock that it must have been the + last to acquire the lock as well. Erroneous error messages would + be sent to the console stating that a thread was attempting to + unlock a lock it did not own. Now all threads are examined to be + sure that the message is only printed when it is supposed to be + printed. + +2008-06-25 19:37 +0000 [r125096] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: implement transfer functionality in + chan_skinny (closes issue #9939) Reported by: wedhorn Patches: + transfer_v6.diff uploaded by wedhorn (license 30) + chan_skinny-transfer-trunk-v10.txt uploaded by DEA (license 3) + chan_skinny-transfer-trunk-v12.txt uploaded by mvanbaak (license + 7) Tested by: DEA, wedhorn, mvanbaak + +2008-06-25 16:00 +0000 [r124912-125055] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_curl.c, funcs/func_curl.c, res/res_curl.c (added): + Separate the global initialization routines for cURL into its own + separate module. + + * channels/chan_dahdi.c, channels/chan_local.c, + channels/chan_features.c, channels/chan_h323.c, + include/asterisk/lock.h, channels/chan_iax2.c: More expansion of + the deadlock avoidance macro, including a macro to do locking of + the channel lock + + * channels/chan_dahdi.c, /, include/asterisk/lock.h: Merged + revisions 124965 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r124965 | tilghman | 2008-06-24 19:46:24 -0500 (Tue, 24 Jun 2008) + | 7 lines Pvt deadlock causes some channels to get stuck in + Reserved status. (closes issue #12621) Reported by: + fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by + Corydon76 (license 14) Tested by: fabianoheringer ........ + + * /, apps/app_voicemail.c: Merged revisions 124910 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 + Jun 2008) | 8 lines Occasionally control characters find their + way into CallerID. These need to be stripped prior to placing + CallerID in the headers of an email. (closes issue #12759) + Reported by: RobH Patches: 20080602__bug12759__2.diff.txt + uploaded by Corydon76 (license 14) Tested by: RobH ........ + +2008-06-24 17:50 +0000 [r124870-124872] Philippe Sultan <philippe.sultan@gmail.com> + + * res/res_jabber.c: Subscribe to buddy's presence only if we really + need to. That is, if the corresponding roster item has a + subscription value set to "none" or "from". Make the code more + readable. + + * res/res_jabber.c: Code simplification + +2008-06-24 11:02 +0000 [r124835] Sean Bright <sean.bright@gmail.com> + + * UPGRADE.txt, CHANGES: Update CHANGES and UPGRADE.txt per + kpfleming's mail to #asterisk-dev. + +2008-06-24 02:16 +0000 [r124798] Russell Bryant <russell@digium.com> + + * res/res_timing_pthread.c: fix a memory leak. (inspired by, and + potentially fixes issue #12917) + +2008-06-23 15:24 +0000 [r124707] Dwayne M. Hubbard <dhubbard@digium.com> + + * main/taskprocessor.c: make solaris happy...pointed out by + snuff-home on IRC + +2008-06-22 17:36 +0000 [r124596-124669] Sean Bright <sean.bright@gmail.com> + + * configs/meetme.conf.sample: Revert my change to the sample meetme + conf file as it was incorrect. + + * configs/meetme.conf.sample: Fix a comment in meetme.conf.sample + per jmls via #asterisk-dev (And this time, do it in the correct + repository :-)) + + * apps/app_rpt.c: Let app_rpt compile. + +2008-06-22 02:58 +0000 [r124541] Steve Murphy <murf@digium.com> + + * apps/app_forkcdr.c, /: Merged revisions 124540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r124540 | murf | 2008-06-21 20:54:52 -0600 (Sat, 21 Jun 2008) | 9 + lines (closes issue #12910) Reported by: chris-mac Sorry, my + testing did not contain the simple case of forkCDR(v), I am much + embarrassed to admit. If I had, I would have more solidly + initialized the opts element for varset. ........ + +2008-06-21 12:53 +0000 [r124396-124505] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_ldap.c: Reduce warning to debug, otherwise we + flood the log when we (legitimately) can't find a record. (Closes + issue #12908) + + * /, apps/app_rpt.c: Merged revisions 124450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r124450 | tilghman | 2008-06-20 18:12:33 -0500 (Fri, 20 Jun 2008) + | 6 lines usleep with a value over 1,000,000 is nonportable. + Changing to use sleep() instead. (closes issue #12814) Reported + by: pputman Patches: app_rtp_sleep.patch uploaded by pputman + (license 81) ........ + + * /, main/app.c: Merged revisions 124395 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r124395 | tilghman | 2008-06-20 17:02:55 -0500 (Fri, 20 Jun 2008) + | 3 lines If the last character in a string to be parsed is the + delimiter, then we should count that final empty string as an + additional argument. ........ + +2008-06-20 21:43 +0000 [r124392-124393] Jeff Gehlbach <jeffg@opennms.org> + + * /: (Missed committing . on previous commit.....) Merged revisions + 124372 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | + 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to + placate smilint - bug 12905 ........ ................ + + * doc/asterisk-mib.txt, doc/digium-mib.txt: Merged revisions 124372 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | + 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to + placate smilint - bug 12905 ........ + +2008-06-20 20:17 +0000 [r124316] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /: Merged revisions 124315 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 + Jun 2008) | 8 lines When using a Local channel, started by a call + file, with a destination of an AGI script, the AGI script does + not always get notified of a hangup if the underlying channel + hangs up early. (closes issue #11833) Reported by: IgorG Patches: + local_hangup-v1.diff uploaded by IgorG (license 20) ........ + +2008-06-20 16:30 +0000 [r124243-124278] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/doxyref.h, main/ast_expr2f.c, main/ast_expr2.fl: + Change references to doc/channelvariables.txt to + doc/tex/channelvariables.tex. This issue came up on the + asterisk-dev mailing list. + + * channels/chan_sip.c: Add a missing "ChannelType" header to one of + the "PeerStatus" manager events in chan_sip (closes issue #12904) + Reported by: eliel Patches: chan_sip.c.patch uploaded by eliel + (license 64) + +2008-06-19 22:59 +0000 [r124183] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 124182 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r124182 | tilghman | 2008-06-19 17:53:22 -0500 (Thu, 19 + Jun 2008) | 7 lines It's possible for a hangup to be received, + even just after the initial cid spill. (closes issue #12453) + Reported by: Alex728 Patches: 20080604__bug12453.diff.txt + uploaded by Corydon76 (license 14) ........ + +2008-06-19 22:34 +0000 [r124180] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Fix attachment behavior when using IMAP + storage for voicemails 1. Filenames had an extra "msg" in the + attachment name 2. The attachment was being saved twice (closes + issue #12894) Reported by: jaroth Patches: imap_attach.patch + uploaded by jaroth (license 50) + +2008-06-19 20:48 +0000 [r124127] Michiel van Baak <michiel@vanbaak.info> + + * doc/CODING-GUIDELINES, channels/chan_sip.c, apps/app_minivm.c, + main/logger.c, pbx/pbx_realtime.c, res/res_realtime.c, + res/res_musiconhold.c, apps/app_directory.c, apps/app_queue.c, + channels/chan_iax2.c, include/asterisk/compiler.h, + apps/app_voicemail.c, funcs/func_realtime.c: Older versions of + GNU gcc do not allow 'NULL' as sentinel. They want (char *)NULL + as sentinel. An example is OpenBSD (confirmed on 4.3) that ships + with gcc 3.3.4 This commit introduces a contstant SENTINEL which + is declared as: #define SENTINEL ((char *)NULL) All places I + could test compile on my openbsd system are converted. Update + CODING-GUIDELINES to tell about this constant. + +2008-06-19 20:35 +0000 [r124125] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Oops + +2008-06-19 20:30 +0000 [r124121] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 124112 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r124112 | mmichelson | 2008-06-19 15:28:41 -0500 (Thu, + 19 Jun 2008) | 8 lines Fix IMAP forwarding so that messages are + sent to the proper mailbox. (closes issue #12897) Reported by: + jaroth Patches: destination_forward.patch uploaded by jaroth + (license 50) ........ + +2008-06-19 20:25 +0000 [r124102] Tilghman Lesher <tlesher@digium.com> + + * main/netsock.c: Make OpenBSD compile again (reported by mvanbaak + via IRC -dev) + +2008-06-19 19:48 +0000 [r124064] Brett Bryant <bbryant@digium.com> + + * main/utils.c: Add errors that report any locks held by threads + when they are being closed. + +2008-06-19 19:22 +0000 [r124049] Tilghman Lesher <tlesher@digium.com> + + * configs/users.conf.sample, CHANGES, pbx/pbx_config.c: Allow + alternative extensions to be specified for a user. (closes issue + #12830) Reported by: jcollie Patches: + astertisk-trunk-121496-alternate-extensions.patch uploaded by + jcollie (license 412) + +2008-06-19 18:57 +0000 [r124024] Brett Bryant <bbryant@digium.com> + + * channels/chan_sip.c: Fix bug in sip registration that sets the + default port to 5060 for tls. + +2008-06-19 18:30 +0000 [r124023] Russell Bryant <russell@digium.com> + + * res/res_timing_pthread.c, main/timing.c: - Make + res_timing_pthread allow a max rate of 100/sec instead of 50/sec + - change the "timing test" CLI command to let you specify a + timing rate to test + +2008-06-19 17:55 +0000 [r123870-123988] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/logger.h, configure, + include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/sched.h, include/asterisk/compiler.h: Detect if + the installed gcc version supports the warn_unused_result + attribute. Reported by mvanbaak via IRC -dev. + + * res/res_config_ldap.c: Don't change pointers that need to be + later passed back for deallocation. (closes issue #12572) + Reported by: flyn Patches: 20080613__bug12572.diff.txt uploaded + by Corydon76 (license 14) + + * main/channel.c, /: Merged revisions 123930 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008) + | 5 lines Change informative messages to use the _multiple + variant when multiple formats are possible. (Closes issue #12848) + Reported by klaus3000 ........ + + * /, build_tools/strip_nonapi: Merged revisions 123909 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r123909 | tilghman | 2008-06-19 11:26:03 -0500 (Thu, 19 + Jun 2008) | 5 lines Only process 40 arguments (20 files) at once + with xargs, because some older shells may force xargs to separate + on an odd boundary. (Closes issue #12883) Reported by Nik Soggia + ........ + + * /, configs/sip.conf.sample: Merged revisions 123883 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 + Jun 2008) | 4 lines Correct description of notifyringing option. + (Closes issue #12890) Reported by gminet ........ + + * /, main/asterisk.c: Merged revisions 123869 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123869 | tilghman | 2008-06-19 11:07:23 -0500 (Thu, 19 Jun 2008) + | 6 lines The RDTSC instruction was introduced on the Pentium + line of microprocessors, and is not compatible with certain 586 + clones, like Cyrix. Hence, asking for i386 compatibility was + always incorrect. See http://en.wikipedia.org/wiki/RDTSC (Closes + issue #12886) Reported by tecnoxarxa ........ + +2008-06-19 15:55 +0000 [r123867] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Forwarding non-urgent IMAP messages could + inadvertently cause the messages to be marked urgent. This fixes + that issue. (closes issue #12895) Reported by: jaroth Patches: + urgent_forwarding.patch uploaded by jaroth (license 50) + +2008-06-19 15:52 +0000 [r123865] Tilghman Lesher <tlesher@digium.com> + + * apps/app_externalivr.c: Missing comma (closes issue #12891) + Reported by: chris-mac + +2008-06-19 14:28 +0000 [r123828-123830] Sean Bright <sean.bright@gmail.com> + + * doc/tex/queuelog.tex: Update the queuelog.tex documentation as + well. + + * apps/app_queue.c: Include original position in TRANSFER entries + written to queue_log. (closes issue #12888) Reported by: slavon + Patches: app_queue_transfer_patch_trunk.diff uploaded by slavon + (license 288) + +2008-06-18 22:17 +0000 [r123715-123770] Tilghman Lesher <tlesher@digium.com> + + * /, main/say.c, doc/lang (added), doc/lang/hebrew.ods: Merged + revisions 123769 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008) + | 8 lines Add support for saying numbers in Hebrew. (closes issue + #11662) Reported by: greenfieldtech Patches: say.c.patch-12042008 + uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods + uploaded by greenfieldtech (with signficant changes to the + spreadsheet by me) ........ + + * pbx/pbx_spool.c, /: Merged revisions 123710 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123710 | tilghman | 2008-06-18 15:22:42 -0500 (Wed, 18 Jun 2008) + | 7 lines Set the variables top-down, so that if a script sets a + variable more than once, the last one will take precedence. + (closes issue #12673) Reported by: phber Patches: + 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14) + ........ + +2008-06-18 20:07 +0000 [r123692] Brett Bryant <bbryant@digium.com> + + * main/tcptls.c: Fix a crash in tcp and tls connections related to + reference counts. + +2008-06-18 15:08 +0000 [r123650-123652] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: A portion of the code which handled the 'c' + queue option had been removed. No telling when it happened. + Anyway, it's back in now and works properly. (Based on issue + reported on mailing list) + + * apps/app_queue.c: Silly pointers. This fixes a memory corruption + I introduced with the attended transfer logging. + +2008-06-18 13:09 +0000 [r123648] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c: Channel lock janitor -- add locks around + retrieval of channel variables (closes issue #12840) Reported by: + pputman Patches: app_dial_threadsafe3.patch uploaded by pputman + (license 81) + +2008-06-18 00:33 +0000 [r123609] Sean Bright <sean.bright@gmail.com> + + * res/res_agi.c: Whitespace only + +2008-06-17 22:24 +0000 [r123546-123575] Brett Bryant <bbryant@digium.com> + + * main/astobj2.c: Revert a previous regression in astobj2.c from + merging a branch. + + * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c, + apps/app_externalivr.c, include/asterisk/tcptls.h, + main/astobj2.c: Updates all usages of ast_tcptls_session_instance + to be managed by reference counts so that they only get destroyed + when all threads are done using them, and memory does not get + free'd causing strange issues with SIP. This code was originally + written by russellb in the team/group/issue_11972/ branch. + +2008-06-17 21:42 +0000 [r123544] Tilghman Lesher <tlesher@digium.com> + + * apps/app_talkdetect.c: Add an option, specifying maximum analysis + time for talk detection. (closes issue #12149) Reported by: + davevg Patches: app_talkdetect.c.diff uploaded by davevg (license + 209) (Plus a few additional cleanups by moi) + +2008-06-17 21:33 +0000 [r123456-123541] Mark Michelson <mmichelson@digium.com> + + * main/astobj2.c: Put quotes around "test" + + * main/astobj2.c: _ys pointed out in #asterisk-bugs that he was + experiencing a memory leak when running the astobj2 test CLI + command. After searching, it appears the leak was in the command + handler itself. Each object was allocated (recount = 1) and then + linked into a container (refounct = 2). Then at the end of the + function, the container was unreffed, causing all the objects to + have their refcount decremented by one, leaving the refcount for + all objects allocated in that function at 1. I've now added an + extra unref to the mix so that the refcount equals zero when the + container is unreffed. + + * /, channels/chan_sip.c: Merged revisions 123485 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun + 2008) | 4 lines Make chan_sip build under dev mode with compilers + >= GCC 4.2 Thanks to jpeeler for alerting me of this ........ + + * main/astobj2.c: Add the same fix from revision 123271 to + container_destruct_debug. + +2008-06-17 20:17 +0000 [r123446-123448] Steve Murphy <murf@digium.com> + + * channels/chan_sip.c, CHANGES: Changes to list peers and users in + alpha. order, as per a reasonable request in 12494. Due to + changes in trunk to use the astobj2 i/f in the sip channel + driver, the order of the entries in the config file was lost, + thus the output was in a random order, but no longer. + + * cdr/cdr_tds.c: This solves a crash in the cdr_tds module on 'stop + gracefully', for situations where 'settings' is not set to a + pointer + +2008-06-17 19:00 +0000 [r123393] Russell Bryant <russell@digium.com> + + * res/res_timing_pthread.c: Fix the check against the max supported + rate + +2008-06-17 18:57 +0000 [r123358-123392] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 123391 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 + Jun 2008) | 3 lines Fix 3 more places where failure to lock the + structure could cause the wrong lock to be unlocked. (Closes + issue #12795) ........ + + * main/pbx.c: If we don't match registrar when destroying a + context, it can cause a crash. (closes issue #12835) Reported by: + ys Patches: pbx.c.diff uploaded by ys (license 281) + +2008-06-17 18:09 +0000 [r123275-123334] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 123333 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun + 2008) | 11 lines Cisco BTS sends SIP responses with a tab between + the Cseq number and SIP request method in the Cseq: header. + Asterisk did not handle this properly, but with this patch, all + is well. (closes issue #12834) Reported by: tobias_e Patches: + 12834.patch uploaded by putnopvut (license 60) Tested by: + tobias_e ........ + + * /, apps/app_queue.c: Merged revisions 123274 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun + 2008) | 12 lines davidw pointed out that the holdtime calculation + used by app_queue does not use "boxcar" filtering as the comments + say. The term "boxcar" means that the number of samples used to + calculate stays constant, with new samples replacing the oldest + ones. The queue holdtime calculation uses all holdtime samples + collected since the queue was loaded, so the comment has been + changed to be accurate. (closes issue #12781) Reported by: davidw + ........ + +2008-06-17 15:52 +0000 [r123272] Russell Bryant <russell@digium.com> + + * /, main/astobj2.c: Merged revisions 123271 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) + | 4 lines Fix a memory leak in astobj2 that was pointed out by + seanbright. When a container got destroyed, the underlying bucket + list entry for each object that was in the container at that time + did not get free'd. ........ + +2008-06-16 23:05 +0000 [r123238] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: Fix some (more) variables that were + forgotten to be renamed, related to 117658 + +2008-06-16 21:42 +0000 [r123203] Doug Bailey <dbailey@digium.com> + + * include/asterisk/callerid.h, channels/chan_dahdi.c, + main/callerid.c: Clean up code that handles fsk mwi message + generation by pulling it from do_monitor and creating its own + thread. Added RP-AS mwi message generation using patches from + meneault as a basis. (closes issue #8587) Reported by: meneault + Tested by: meneault + +2008-06-16 21:31 +0000 [r123201] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c: Oopsie, breakage + +2008-06-16 21:15 +0000 [r123166] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: Fix some variables that were forgotten to + be renamed, related to 117658 + +2008-06-16 20:43 +0000 [r123165] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/pbx.c, main/features.c, + include/asterisk/pbx.h, apps/app_queue.c, apps/app_stack.c: + (closes issue #12689) Reported by: ys Many thanks to ys for doing + the research on this problem. I didn't think it would be best to + unlock the contexts and then relock them after the + remove_extension2() call, so I added an extra arg to + remove_extension2() and set it appropriately in each call. There + were not that many. I considered forcing the code to lock the + contexts before the call to remove_extension2(), but that would + require a slightly greater degree of changes, especially since + the find_context_locked is local to pbx.c I did a simple sanity + test to make sure the code doesn't mess things up in general. + +2008-06-16 20:02 +0000 [r123115] Chris Tooley <chris@tooley.com> + + * apps/app_externalivr.c: Changes response to the ExternalIVR() P + command from pipe delimited to comma delimited. closes issue + #12804 + +2008-06-16 19:57 +0000 [r123111-123114] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /, channels/chan_sip.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions + 123113 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) + | 2 lines Port "hasvoicemail" change from SIP to other channel + drivers ........ + + * /, channels/chan_sip.c: Merged revisions 123110 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) + | 8 lines People expect that if "hasvoicemail" is set in + users.conf, even if "mailbox" isn't set, that SIP will detect a + mailbox. (closes issue #12855) Reported by: PLL Patches: + 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14) + Tested by: PLL ........ + +2008-06-16 17:33 +0000 [r123009-123076] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_tds.c: Last commit for a bit, minor cleanups and move the + lock initialization. + + * cdr/cdr_tds.c: Convert to use stringfields. Still some more work + to do on config load/reload. + + * cdr/cdr_tds.c: Remove some unused variables + + * cdr/cdr_tds.c: Coding guidelines stuff only. + +2008-06-16 13:31 +0000 [r122923-122977] Russell Bryant <russell@digium.com> + + * configs/modules.conf.sample: Note that only one timing interface + should get loaded. + + * res/res_timing_pthread.c (added): Merge res_timing_pthread. This + is a timing interface for Asterisk that does not require DAHDI. + It's called "pthread" because it uses a pthread API call in the + timing thread for sleeping and ensuring we wake up at an + appropriate time. I wasn't sure what else to call it. :) The + timing API requires a file descriptor that can be polled on. So, + when you open a timer, this module creates a pipe and returns the + read end of the pipe. There is a background thread that wakes up + every 10ms and checks to see if any of the currently open timers + need a 'tick' and writes to the appropriate pipe. + + * include/asterisk/_private.h, main/asterisk.c, main/timing.c: Add + a "timing test" CLI command. It opens a timer and configures it + for 50 ticks per second, and then counts to see how many ticks it + actually gets in a second. + + * main/channel.c, include/asterisk/timing.h, main/timing.c: - Fix a + typo in a timing API call - Convert the last part of channel.c + over to use the timing API. This would not have made a difference + when using the dahdi timing module. I noticed it when trying to + use another timing source. Oops. :) + +2008-06-16 12:32 +0000 [r122870-122920] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 122919 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 + lines Only compare the first 15 characters so that even if the + charset is specified we still accept it as SDP. (closes issue + #12803) Reported by: lanzaandrea Patches: chan_sip.c.diff + uploaded by lanzaandrea (license 496) ........ + + * /, channels/chan_sip.c: Merged revisions 122869 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 + lines Don't send a BYE on a dialog that is already gone during a + REFER. (closes issue #12865) Reported by: flefoll Patches: + chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll + (license 244) ........ + +2008-06-16 03:33 +0000 [r122834] Sean Bright <sean.bright@gmail.com> + + * apps/app_fax.c (added): Resurrected app_fax + +2008-06-15 15:21 +0000 [r122802] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c, funcs/func_channel.c, UPGRADE.txt, + channels/chan_iax2.c: Add some more IAX2-specific information + about the channel to the CHANNEL() function and begin the + transition from SIPCHANINFO() to just using CHANNEL(). (closes + issue #12856) Reported by: mostyn Patches: + iax_and_sip_channel_info.patch uploaded by mostyn (license 398) + (with some additional cleanup by me) + +2008-06-13 22:52 +0000 [r122716-122766] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/config.h: Document the input for + ast_realtime_require_field() + + * res/res_config_pgsql.c: Properly detect the size of char/varchar + fields + +2008-06-13 21:45 +0000 [r122714] Mark Michelson <mmichelson@digium.com> + + * main/autoservice.c, /: Merged revisions 122713 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun + 2008) | 9 lines Short circuit the loop in autoservice_run if + there are no channels to poll. If we continued, then the result + would be calling poll() with a NULL pollfd array. While this is + fine with POSIX's poll(2) system call, those who use Asterisk's + internal poll mechanism (Darwin systems) would have a failed + assertion occur when poll is called. (related to issue #10342) + ........ + +2008-06-13 14:15 +0000 [r122557] Tilghman Lesher <tlesher@digium.com> + + * main/dial.c: Convert one more delimiter to use comma. (closes + issue #12850) Reported by: bcnit Patches: + 20080613__bug12850.diff.txt uploaded by Corydon76 (license 14) + Tested by: bcnit + +2008-06-13 12:53 +0000 [r122523-122526] Russell Bryant <russell@digium.com> + + * res/res_timing_dahdi.c: Do not allow res_timing_dahdi to be + unloaded. We can re-enable this once we add automatic use count + handling for timing modules. + + * main/channel.c, res/res_timing_dahdi.c (added), main/file.c, + include/asterisk/timing.h, include/asterisk/channel.h, + channels/chan_iax2.c, main/asterisk.c, main/timing.c: Merge + changes from timing branch - Convert chan_iax2 to use the timing + API - Convert usage of timing in the core to use the timing API + instead of using DAHDI directly - Make a change to the timing API + to add the set_rate() function - change the timing core to use a + rwlock - merge a timing implementation, res_timing_dahdi Basic + testing was successful using res_timing_dahdi + +2008-06-13 11:20 +0000 [r122493] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: Implement call parking in chan_skinny. + (closes issue #11342) Reported by: DEA Patches: + chan_skinny-park.txt uploaded by DEA (license 3) + chan_skinny-park-v2.diff.txt uploaded by mvanbaak (license 7) + Tested by: DEA, mvanbaak + +2008-06-12 23:58 +0000 [r122461] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix a segfault by not trying to store a stack + address for long-term use. Instead use the heap. I can't believe + this never happened *once* in my developer branch when I was + testing. + +2008-06-12 23:08 +0000 [r122433] Jeff Peeler <jpeeler@digium.com> + + * main/features.c, apps/app_parkandannounce.c: (closes issue + 0012193) Reported by: davidw Patch by: Corydon76, modified by me + to work properly with ParkAndAnnounce app + +2008-06-12 21:23 +0000 [r122399] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Recommitting revision 122228, which was + accidentally reverted as a result of commit 122234. + +2008-06-12 20:38 +0000 [r122371] Russell Bryant <russell@digium.com> + + * include/asterisk/timing.h: Complete the documentation for the + timing API. + +2008-06-12 18:53 +0000 [r122312] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 122311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r122311 | mmichelson | 2008-06-12 13:50:58 -0500 (Thu, 12 Jun + 2008) | 9 lines Properly play a holdtime message if the + announce-holdtime option is set to "once." (closes issue #12842) + Reported by: ramonpeek Patches: patch001.diff uploaded by + ramonpeek (license 266) ........ + +2008-06-12 18:23 +0000 [r122262] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 122259 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 + Jun 2008) | 3 lines Fix some race conditions that cause + ast_assert() to report that chan_iax2 tried to remove an entry + that wasn't in the scheduler ........ + +2008-06-12 17:49 +0000 [r122243-122244] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: Fix pseudo channel allocation errors on + startup when using SS7. (from mattf r121914, moving from chan_zap + to chan_dahdi) + + * channels/chan_dahdi.c: Make sure we hangup any calls we have and + NULL out the ss7call value when we get a reset circuit message. + Fixes crash bug. (from mattf r121857, moving from chan_zap to + chan_dahdi) + +2008-06-12 17:38 +0000 [r122241] Russell Bryant <russell@digium.com> + + * include/asterisk/network.h: Get default entity ID determination + working on Linux again (closes issue #12839) + +2008-06-12 17:30 +0000 [r122240] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/timing.h: clarify documentation on how timer + intervals should be specified + +2008-06-12 17:27 +0000 [r122234] Jeff Peeler <jpeeler@digium.com> + + * README, apps/app_dahdibarge.c (added), + contrib/init.d/rc.mandrake.asterisk, /, + include/asterisk/autoconfig.h.in, apps/app_dahdiscan.c (added), + apps/app_chanisavail.c, channels/chan_iax2.c, + configs/muted.conf.sample, main/loader.c, + include/asterisk/doxyref.h, channels/chan_dahdi.c (added), + configure, apps/app_zapscan.c (removed), main/features.c, + doc/tex/backtrace.tex, doc/tex/app-sms.tex, apps/app_zapras.c + (removed), configs/extensions.lua.sample, + include/asterisk/options.h, contrib/init.d/rc.suse.asterisk, + apps/app_dial.c, apps/app_page.c, doc/tex/hardware.tex, + apps/app_fax.c (removed), apps/app_dahdiras.c (added), + configs/queues.conf.sample, configure.ac, + include/asterisk/channel.h, doc/tex/configuration.tex, + configs/zapata.conf.sample, Makefile, apps/app_zapbarge.c + (removed), doc/janitor-projects.txt, configs/vpb.conf.sample, + doc/sms.txt, codecs/codec_dahdi.c (added), + contrib/scripts/loadtest.tcl, configs/smdi.conf.sample, + pbx/pbx_config.c, apps/app_chanspy.c, main/asterisk.c, + configs/users.conf.sample, doc/ss7.txt, apps/app_meetme.c, + configs/rpt.conf.sample, doc/backtrace.txt, + doc/tex/queues-with-callback-members.tex, res/res_musiconhold.c, + configs/extensions.ael.sample, include/asterisk/dahdi.h (added), + contrib/init.d/rc.mandrake.zaptel, codecs/codec_zap.c (removed), + configs/meetme.conf.sample, cdr/cdr_csv.c, main/channel.c, + doc/tex/manager.tex, doc/tex/sla.tex, include/asterisk/dsp.h, + doc/tex/localchannel.tex, apps/app_rpt.c, channels/chan_mgcp.c, + contrib/scripts/autosupport, doc/manager_1_1.txt, + channels/chan_zap.c (removed), doc/asterisk.8, + doc/tex/channelvariables.tex, doc/tex/ael.tex, apps/app_queue.c, + doc/tex/enum.tex, apps/app_getcpeid.c, doc/tex/security.tex, + configs/sla.conf.sample, include/asterisk/zapata.h (removed), + build_tools/menuselect-deps.in, doc/tex/privacy.tex, + apps/app_flash.c, doc/osp.txt, main/file.c, + contrib/utils/zones2indications.c, utils/extconf.c, makeopts.in, + doc/asterisk.sgml, configs/extensions.conf.sample: Goodbye + Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. + Configuration file and dialplan backwards compatability has been + put in place where appropiate. Release announcement to follow. + +2008-06-12 17:14 +0000 [r122232] Russell Bryant <russell@digium.com> + + * channels/misdn/isdn_lib.c: Make this build under dev mode + +2008-06-12 16:25 +0000 [r122228] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merging the work done in the + queue-log-atxfer branch. The net result of this work is that + attended transfers made by queue members will now show up in the + queue_log as a TRANSFER message instead of COMPLETECALLER as it + had been. As far as the details go, I created a datastore which + is attached to the calling channel just prior to when the caller + is bridged with the queue member. If the calling channel is + masqueraded, then during the "fixup" portion, the TRANSFER will + be logged and the datastore will be removed. + +2008-06-12 15:26 +0000 [r122131-122174] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_meetme.c: Merged revisions 122137 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008) + | 8 lines Flipflop the sections for two options, since the + section for 'X' (exit context) may otherwise absorb keypresses + meant for 's' (admin/user menu). (closes issue #12836) Reported + by: blitzrage Patches: 20080611__bug12836.diff.txt uploaded by + Corydon76 (license 14) Tested by: blitzrage ........ + + * main/channel.c, /: Merged revisions 122130 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) + | 4 lines Occasionally, the alertpipe loses its nonblocking + status, so detect and correct that situation before it causes a + deadlock. (Reported and tested by ctooley via #asterisk-dev) + ........ + +2008-06-12 14:56 +0000 [r122091-122128] Steve Murphy <murf@digium.com> + + * main/cdr.c, apps/app_forkcdr.c, /, CHANGES: Merged revisions + 122127 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 + line Arkadia tried to warn me, but the code added to + ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot + it until I was resolving conflicts in trunk. Ugh. Redundant code + removed. It wasn't harmful. Just dumb. ........ + + * main/cdr.c, apps/app_forkcdr.c, /, funcs/func_cdr.c, + include/asterisk/cdr.h, CHANGES: Merged revisions 122046 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | + 37 lines (closes issue #10668) Reported by: arkadia Tested by: + murf, arkadia Options added to forkCDR() app and the CDR() func + to remove some roadblocks for CDR applications. The "show + application ForkCDR" output was upgraded to more fully explain + the inner workings of forkCDR. The A option was added to forkCDR + to force the CDR system to NOT change the disposition on the + original CDR, after the fork. This involves ast_cdr_answer, + _busy, _failed, and so on. The T option was added to forkCDR to + force obedience of the cdr LOCKED flag in the ast_cdr_end, all + the disposition changing funcs (ast_cdr_answer, etc), and in the + ast_cdr_setvar func. The CHANGES file was updated to explain ALL + the new options added to satisfy this bug report (and some + requests made verbally and via email, irc, etc, over the past + months/year) The 's' option was added to the CDR() func, to force + it to skip LOCKED cdr's in the chain. Again, the new options + should be totally transparent to existing apps! Current behavior + of CDR, forkCDR, and the rest of the CDR system should not change + one little bit. Until you add the new options, at least! ........ + +2008-06-12 14:21 +0000 [r122062] Kevin P. Fleming <kpfleming@digium.com> + + * main/Makefile, include/asterisk/timing.h (added), main/timing.c + (added): add infrastructure so that timing source can be a + loadable module... next steps are to convert channel.c and + chan_iax2.c to use this new API, and to move all the + DAHDI-specific timing source code into a new res_timing_dahdi + module + +2008-06-12 14:06 +0000 [r122047] Russell Bryant <russell@digium.com> + + * main/netsock.c: Don't log not being able to set a default EID. + Most people don't care, and those that do can check their setup + using CLI commands. (closes issue #12839) + +2008-06-11 21:38 +0000 [r121955] Terry Wilson <twilson@digium.com> + + * main/features.c: Initialize parkingtime to DEFAULT_PARK_TIME + instead of 0 + +2008-06-11 18:53 +0000 [r121914] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Fix pseudo channel allocation errors on + startup when using SS7 + +2008-06-11 18:19 +0000 [r121867] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /, channels/chan_agent.c, main/abstract_jb.c, + main/sched.c: Merged revisions 121861 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) + | 3 lines Make calls to ast_assert() actually test something, so + that the error message printed is not nonsensical (reported by + mvanbaak via #asterisk-bugs). ........ + +2008-06-11 17:50 +0000 [r121857] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Make sure we hangup any calls we have and + NULL out the ss7call value when we get a reset circuit message. + Fixes crash bug + +2008-06-11 17:44 +0000 [r121855] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/realtime_pgsql.sql, UPGRADE.txt, + include/asterisk/cdr.h: Expand CDR uniqueid field to 150 chars, + to account for maximum systemname. (Closes issue #12831) + +2008-06-11 16:11 +0000 [r121805] Jeff Peeler <jpeeler@digium.com> + + * /, doc/backtrace.txt: Merged revisions 121804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008) + | 1 line add instructions for logging gdb output via set logging + on ........ + +2008-06-11 11:52 +0000 [r121770] Christian Richter <christian.richter@beronet.com> + + * /, channels/misdn/isdn_lib.c: Merged revisions 121751 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008) + | 1 line fixed issue with previous commit, the find_free_channel + test for channels which where inuse was broken. ........ + +2008-06-10 21:51 +0000 [r121716] Russell Bryant <russell@digium.com> + + * doc/distributed_devstate.txt: don't refer to asterisk-events, as + that implies that the code was checked out from a branch + +2008-06-10 21:14 +0000 [r121683] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/res_odbc.h, res/res_config_odbc.c, + res/res_odbc.c: Move the table cache routines to res_odbc, so + they can be used from other places (app_voicemail, for example). + (Related to bug #11678) + +2008-06-10 19:52 +0000 [r121649] Mark Michelson <mmichelson@digium.com> + + * main/event.c: Add an additional sanity check in case an event is + passed between Asterisk boxes with mismatched ie_maps. + +2008-06-10 19:03 +0000 [r121599] Donny Kavanagh <donnyk@gmail.com> + + * codecs/codec_ilbc.c: Revision 117802 changed frame.data to + frame.data.ptr however codec_ilbc.c was not updated. This + resolves that oversight. + +2008-06-10 18:35 +0000 [r121597] Sean Bright <sean.bright@gmail.com> + + * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 121596 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r121596 | seanbright | 2008-06-10 14:34:45 -0400 (Tue, 10 Jun + 2008) | 6 lines Fixes a problem with some buggy versions of GNU + awk (3.1.3) not liking carriage returns in scripts. (closes issue + #12749) Reported by: alinux Tested by: Laureano (on + #asterisk-dev), juggie ........ + +2008-06-10 15:12 +0000 [r121555-121559] Russell Bryant <russell@digium.com> + + * main/pbx.c, res/res_ais.c (added), res/ais/clm.c, + doc/distributed_devstate.txt (added), res/ais/evt.c, res/ais + (added), main/devicestate.c, res/Makefile, res/ais/ais.h, + configs/ais.conf.sample (added), CHANGES, apps/app_queue.c: Merge + another big set of changes from team/russell/events This commit + merges in the rest of the code needed to support distributed + device state. There are two main parts to this commit. Core + changes: - The device state handling in the core has been updated + to understand device state across a cluster of Asterisk servers. + Every time the state of a device changes, it looks at all of the + device states on each node, and determines the aggregate device + state. That resulting device state is what is provided to modules + in Asterisk that take actions based on the state of a device. New + module, res_ais: - A module has been written to facilitate the + communication of events between nodes in a cluster of Asterisk + servers. This module uses the SAForum AIS (Service Availability + Forum Application Interface Specification) CLM and EVT services + (Cluster Management and Event) to handle this task. This module + currently supports sharing Voicemail MWI (Message Waiting + Indication) and device state events between servers. It has been + tested with openais, though other implementations of the spec do + exist. For more information on testing distributed device state, + see the following doc: - doc/distributed_devstate.txt + + * include/asterisk/event.h, include/asterisk/event_defs.h, + main/event.c: Merge some more changes from team/russell/events + This commit pulls in a batch of improvements and additions to the + event API. Changes include: - the ability to dynamically build a + subscription. This is useful if you're building a subscription + based on something you receive from the network, or from options + in a configuration file. - Add tables of event types and IE types + and the corresponding string representation for implementing text + based protocols that use these events, for showing events on the + CLI, reading configuration that references event information, + among other things. - Add a table that maps IE types and the + corresponding payload type. - an API call to get the total size + of an event - an API call to get all events from the cache that + match a subscription - a new IE payload type, raw, which I used + for transporting the Entity ID in my code for handling + distributed device state. - Code improvements to reduce code + duplication - Include the Entity ID of the server that originated + the event in every event - an additional event type, + DEVICE_STATE_CHANGE, to help facilitate distributed device state. + DEVICE_STATE is a state change on one server, DEVICE_STATE_CHANGE + is the aggregate device state change across all servers. + +2008-06-10 14:11 +0000 [r121503] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Fix issue where session timer headers were + present when they should not have been. (closes issue #12706) + Reported by: falves11 Patches: chan_sip.c.diff uploaded by rjain + (license 226) Tested by: falves11 + +2008-06-10 14:06 +0000 [r121501] Russell Bryant <russell@digium.com> + + * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c: + Merge another change from team/russell/events This commit breaks + out some logic from pbx.c into a simple API. The hint processing + code had logic for taking the state from multiple devices and + turning that into the state for a single extension. So, I broke + this out and made an API that lets you take multiple device + states and determine the aggregate device state. I needed this + for some core device state changes to support distributed device + state. + +2008-06-10 13:36 +0000 [r121444-121496] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 121495 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r121495 | file | 2008-06-10 10:34:27 -0300 (Tue, 10 Jun 2008) | 4 + lines If we are destroying a dialog only set the MWI dialog + pointer on the related peer to NULL if it is the dialog currently + being destroyed. (closes issue #12828) Reported by: ramonpeek + ........ + + * main/channel.c, /: Merged revisions 121442 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4 + lines Update BRIDGEPEER variable before we do a generic bridge in + case we just broke out of a native bridge and fell through to + generic. (closes issue #12815) Reported by: ramonpeek ........ + +2008-06-10 12:50 +0000 [r121401-121441] Russell Bryant <russell@digium.com> + + * configs/dundi.conf.sample: Update dundi.conf to indicate that the + asterisk.conf entityid option can be used to set the entityid + used in DUNDi, as well. + + * include/asterisk/utils.h, main/pbx.c, include/asterisk/dundi.h, + doc/tex/channelvariables.tex, pbx/pbx_dundi.c, + pbx/dundi-parser.c, main/asterisk.c, main/netsock.c, + doc/tex/asterisk-conf.tex, pbx/dundi-parser.h: Merge another + change from team/russell/events ... DUNDi uses a concept called + the Entity ID for unique server identifiers. I have pulled out + the handling of EIDs and made it something available to all of + Asterisk. There is now a global Entity ID that can be used for + other purposes as well, such as code providing distributed device + state, which is why I did this. The global Entity ID is set + automatically, just like it was done in DUNDi, but it can also be + set in asterisk.conf. DUNDi will now use this global EID unless + one is specified in dundi.conf. The current EID for the system + can be seen in the "core show settings" CLI command. It is also + available in the dialplan via the ENTITYID variable. + + * channels/chan_iax2.c: Bump up the debug level of a couple of + messages + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: + Merge a couple of configure script checks in from + team/russell/events. This adds the checks for the CLM and EVT + services from the SAForum AIS. I'm going to work on merging in + changes from this branch in pieces. + + * main/taskprocessor.c: Properly initialize the cli_ping condition + and lock + + * main/taskprocessor.c: Change system header includes to be like + how it is done in other files + +2008-06-09 22:51 +0000 [r121367] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_curl.c, res/res_config_pgsql.c, + res/res_config_odbc.c, apps/app_meetme.c, channels/chan_sip.c, + include/asterisk/config.h, main/utils.c, apps/app_queue.c, + channels/chan_iax2.c, apps/app_voicemail.c: Expand RQ_INTEGER + type out to multiple types, one for each precision + +2008-06-09 22:42 +0000 [r121365] Terry Wilson <twilson@digium.com> + + * main/taskprocessor.c: Initialize the lock and destroy lock and + cond in the destructor (thanks, mmichelson) + +2008-06-09 19:33 +0000 [r121334] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Add storage of the useragent in the realtime + database. (Closes AST-38) + +2008-06-09 16:55 +0000 [r121282-121286] Russell Bryant <russell@digium.com> + + * main/dsp.c: arbitrary formatting change to test mantis change + (closes issue #12824) + + * main/channel.c: arbitrary formatting change to test a mantis + change (closes issue #12824) + + * main/channel.c: Minor formatting change to test a mantis change + ... (closes issue #12824) + + * main/channel.c, /: Merged revisions 121280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) + | 10 lines Do not attempt to do emulation if an END digit is + received and the length is less than the defined minimum digit + length, and the other end only wants END digits (SIP INFO, for + example). (closes issue #12778) Reported by: tsearle Patches: + 12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle + ........ + +2008-06-09 16:35 +0000 [r121279] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c: Implement FINDLABEL matching for the new extension + matching engine. (closes issue #12800) Reported by: chris-mac + Patches: 20080608__bug12800.diff.txt uploaded by Corydon76 + (license 14) + +2008-06-09 15:08 +0000 [r121230] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 121229 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Note + that this is being merged to trunk/1.6.0 because it may affect + non-callback agents with ackcall set) ........ r121229 | + mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 + lines A unique situation of timeouts brought forth a failure + situation for autologoff in chan_agent. If using + AgentCallbackLogin-style agents, then if the timeout specified by + the Dial() to reach the agent's phone was shorter than the + timeout specified in queues.conf, then autologoff would only work + if the caller hung up while the agent's phone was ringing. This + patch allows autologoff to work in this situation when the call + in queue transfers to the next available agent (as it would have + if the timeout in queues.conf were less than the timeout in the + Dial()). (closes issue #12754) Reported by: Rodrigo Patches: + 12754.patch uploaded by putnopvut (license 60) Tested by: Rodrigo + ........ + +2008-06-08 11:40 +0000 [r121197] Michiel van Baak <michiel@vanbaak.info> + + * apps/app_privacy.c, CHANGES: add a new argument to PrivacyManager + to specify a context where the entered phone number is checked. + You can now define a set of extensions/exten patterns that + describe valid phone numbers. PrivacyManager will check that + context for a match with the given phone number. This way you get + better control. For example people blindly hitting 10 digits just + to get past privacymanager Example line in extensions.conf: exten + => incoming,n,PrivacyManager(3,10,,route-outgoing) + +2008-06-08 01:41 +0000 [r121131-121163] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_console.c: This was accidentally reverted. Fixes a + bug where if a stream monitor thread was not created (caused from + failure of opening or starting the stream) pthread_cancel was + called with an invalid thread ID. + + * apps/app_parkandannounce.c: Fixes segfault when using + ParkAndAnnounce. Also, loop made more efficient as announce + template only needs to be checked until the number of colon + separated arguments run out, not the entire pointer storage + array. Was done in a similiar fashion in 1.4, but here we're + using less variables. + +2008-06-07 14:18 +0000 [r121079] Russell Bryant <russell@digium.com> + + * channels/chan_local.c, /, channels/chan_agent.c: Merged revisions + 121078 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) + | 7 lines Don't run LIST_HEAD_DESTROY on a STATIC list (closes + issue #12807) Reported by: ys Patches: chan_agent_local.diff + uploaded by ys (license 281) ........ + +2008-06-06 20:24 +0000 [r121010-121042] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c, CHANGES: Added a facility for sending + arbitrary SIP notify commands from AMI. (closes issue #12562) + Reported by: michael-fig Patches: 20080515__bug12562.diff.txt + uploaded by Corydon76 (license 14) + + * main/pbx.c: Make extension match characters case-insensitive. + (closes issue #12777) Reported by: jsmith Patches: + lower_case_patterns-trunk-v1.patch uploaded by jsmith (license + 15) + +2008-06-06 18:30 +0000 [r120906-120960] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 120959 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) + | 1 line add another LOW_MEMORY define I forgot ........ + + * /, channels/chan_sip.c: Merged revisions 120908 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) + | 1 line only define thread storage variable if necessary for + LOW_MEMORY ........ + + * /, channels/chan_sip.c, main/features.c: Merged revisions + 120863,120885 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) + | 3 lines This fixes a crash when LOW_MEMORY is turned on. Two + allocations of the ast_rtp struct that were previously allocated + on the stack have been modified to use thread local storage + instead. ........ r120885 | jpeeler | 2008-06-06 11:39:20 -0500 + (Fri, 06 Jun 2008) | 2 lines Correction to commmit 120863, make + sure proper destructor function is called as well define two + thread storage local variables. ........ + +2008-06-06 17:34 +0000 [r120904] Tilghman Lesher <tlesher@digium.com> + + * apps/app_exec.c: For the purpose of making the changed syntax to + ExecIf easier to transition, allow the deprecated syntax (fixed + for jmls on -dev). + +2008-06-05 21:34 +0000 [r120828] Steve Murphy <murf@digium.com> + + * main/pbx.c: a small fix for a crash that occurs when compiling + AEL with global vars + +2008-06-05 19:07 +0000 [r120789] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_curl.c, include/asterisk/res_odbc.h, + res/res_config_pgsql.c, res/res_config_odbc.c, apps/app_meetme.c, + channels/chan_sip.c, include/asterisk/config.h, + contrib/scripts/dbsep.cgi, apps/app_queue.c, + channels/chan_iax2.c, main/config.c, + configs/res_pgsql.conf.sample, apps/app_voicemail.c: Merge the + adaptive realtime branch, which will make adding new required + fields to realtime less painful in the future. + +2008-06-05 18:03 +0000 [r120733-120734] Russell Bryant <russell@digium.com> + + * UPGRADE-1.2.txt: revert 120733, wrong branch + + * UPGRADE-1.2.txt: Update file names + +2008-06-05 17:02 +0000 [r120676] Philippe Sultan <philippe.sultan@gmail.com> + + * /, res/res_jabber.c: Merged revisions 120675 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008) + | 2 lines Ignore appended resource when comparing JIDs. ........ + +2008-06-05 16:41 +0000 [r120635-120673] Brett Bryant <bbryant@digium.com> + + * CHANGES: Update CHANGES file for the things done in revision + 120635. + + * channels/chan_sip.c, funcs/func_channel.c, + include/asterisk/rtp.h, main/rtp.c: This patch adds more detailed + statistics for RTP channels, and provides an API call to access + it, including maximums, minimums, standard deviatinos, and normal + deviations. Currently this is implemented for chan_sip, but could + be added to the func_channel_read callbacks for the CHANNEL + function for any channel that uses RTP. (closes issue #10590) + Reported by: gasparz Patches: chan_sip_c.diff uploaded by gasparz + (license 219) rtp_c.diff uploaded by gasparz (license 219) + rtp_h.diff uploaded by gasparz (license 219) audioqos-trunk.diff + uploaded by snuffy (license 35) rtpqos-trunk-r119891.diff + uploaded by sergee (license 138) Tested by: jsmith, gasparz, + snuffy, marsosa, chappell, sergee + +2008-06-05 15:58 +0000 [r120567-120602] Tilghman Lesher <tlesher@digium.com> + + * res/res_agi.c, apps/app_stack.c, main/loader.c: Conditionally + load the AGI command gosub, depending on whether or not res_agi + has been loaded, fix a return value in the loader, and ensure + that the help workhorse header does not print on load. + + * UPGRADE.txt: Add info on the [compat] section of asterisk.conf. + +2008-06-04 22:07 +0000 [r120514] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 120513 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120513 | mmichelson | 2008-06-04 17:05:33 -0500 (Wed, 04 Jun + 2008) | 6 lines Make sure that the string we set will survive the + unref of the queue member. Thanks to Russell, who pointed this + out. ........ + +2008-06-04 20:34 +0000 [r120426-120477] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c: MSet doesn't necessarily need chan to be set + + * channels/chan_zap.c, /: Merged revisions 120425 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120425 | tilghman | 2008-06-04 13:35:47 -0500 (Wed, 04 Jun 2008) + | 6 lines If we fail to setup the PRI request channel, don't + continue, exit with an error. (closes issue #11989) Reported by: + Corydon76 Patches: 20080213__zap_memleak.diff.txt uploaded by + Corydon76 (license 14) ........ + +2008-06-04 15:38 +0000 [r120337] Joshua Colp <jcolp@digium.com> + + * pbx/pbx_config.c: We like tabs. + +2008-06-04 14:12 +0000 [r120286] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 120285 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120285 | mmichelson | 2008-06-04 09:11:12 -0500 (Wed, 04 Jun + 2008) | 7 lines Tab completion when removing a member should give + the member's interface, not the name, since the interface is what + is expected for the command. (closes issue #12783) Reported by: + davevg ........ + +2008-06-04 13:33 +0000 [r120283] Joshua Colp <jcolp@digium.com> + + * /, pbx/pbx_config.c: Merged revisions 120282 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120282 | file | 2008-06-04 10:31:09 -0300 (Wed, 04 Jun 2008) | 6 + lines Fix a log message and add a message for when the dialplan + is done reloading. (closes issue #12716) Reported by: chappell + Patches: dialplan_reload_2.diff uploaded by chappell (license 8) + ........ + +2008-06-03 23:17 +0000 [r120227-120230] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_channel.c: Add a function, CHANNELS(), which retrieves + a list of all active channels. (closes issue #11330) Reported by: + rain Patches: func_channel-channel_list_function.diff uploaded by + rain (license 327) (with some additional changes by me, mostly to + meet coding guidelines) + + * pbx/pbx_loopback.c, /: Merged revisions 120226 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) + | 8 lines Due to incorrect use of the AST_LIST_INSERT_HEAD() + macro the loopback switch cannot perform any translation on the + extension number before searching for it in the target context. + (closes issue #12473) Reported by: chappell Patches: + pbx_loopback.c.diff uploaded by chappell (license 8) ........ + +2008-06-03 22:17 +0000 [r120174] Jeff Peeler <jpeeler@digium.com> + + * /, main/config.c: Merged revisions 120173 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) + | 6 lines (closes issue #11594) Reported by: yem Tested by: yem + This change decreases the buffer size allocated on the stack + substantially in config_text_file_load when LOW_MEMORY is turned + on. This change combined with the fix from revision 117462 + (making mkintf not copy the zt_chan_conf structure) was enough to + prevent the crash. ........ + +2008-06-03 22:05 +0000 [r120171] Tilghman Lesher <tlesher@digium.com> + + * Makefile, main/pbx.c, res/res_agi.c, pbx/pbx_realtime.c, + configs/pbx_realtime.conf (removed), include/asterisk/options.h, + main/asterisk.c: Move compatibility options into asterisk.conf, + default them to on for upgrades, and off for new installations. + This includes the translation from pipes to commas for + pbx_realtime and the EXEC command for AGI, as well as the change + to the Set application not to support multiple variables at once. + +2008-06-03 21:35 +0000 [r120169] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 120168 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 + Jun 2008) | 4 lines Fix another place where peer->callno could + change at a very bad time, and also fix a place where a peer was + used after the reference was released. (inspired by rev 120001) + ........ + +2008-06-03 21:22 +0000 [r120166] Mark Michelson <mmichelson@digium.com> + + * CHANGES, apps/app_queue.c: Adding two new queue log events. The + ADDMEMBER event is logged when a dynamic realtime queue member is + added to the queue, and the REMOVEMEMBER event is logged when a + dynamic realtime member is removed. Since no calling channel is + associated with these events the string "REALTIME" is placed + where the channel's unique id is normally placed. (closes issue + #12774) Reported by: atis Patches: queue_log_rt_members.patch + uploaded by atis (license 242) + +2008-06-03 19:48 +0000 [r120064-120129] Russell Bryant <russell@digium.com> + + * apps/app_channelredirect.c, apps/app_disa.c, + apps/app_chanisavail.c: Use proper return values for a few + application modules + + * include/asterisk/lock.h: fix build for non debug threads + + * main/channel.c, main/utils.c, include/asterisk/lock.h, + utils/ael_main.c, utils/conf2ael.c: Add lock tracking for + rwlocks. Previously, lock.h only had the ability to hold tracking + information for mutexes. Now, the "core show locks" output will + output information about who is holding a rwlock when a thread is + waiting on it. (closes issue #11279) Reported by: ys Patches: + trunk_lock_utils.v8.diff uploaded by ys (license 281) + +2008-06-03 16:19 +0000 [r120012] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 120001 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 + Jun 2008) | 9 lines Save the callno when we're poking, because + our peer structure could change during destruction (and thus we + unlock the wrong callno, causing a cascade failure). (closes + issue #12717) Reported by: gewfie Patches: + 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14) + Tested by: gewfie ........ + +2008-06-03 15:49 +0000 [r119930-119998] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ref.ael-vtest17, + pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, + pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, + pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-test8, + pbx/ael/ael-test/ref.ael-test18, + pbx/ael/ael-test/ref.ael-vtest21, + pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 119966 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8 + lines Updated the regressions on AEL. Hadn't updated this for the + changes I made to preserve ${EXTEN} in switches, which affected + several tests because it adds extra priorities, and at least one + needed to be updated because of the removal of the empty + extension warning message. ........ + + * res/ael/pval.c, /: Merged revisions 119929 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | + 16 lines as per + http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html, + which is a message from Philipp Kempgen, requesting that the + WARNING that an extension is empty be reduced to a NOTICE or + less, as empty extensions are syntactically possible, and no big + deal. With which I agree, and have removed that WARNING message + entirely. I think it is not necessary to see this message. It + didn't state that a NoOp() was inserted automatically on your + behalf, and really, as users, who cares? Why freak out dialplan + writers with unnecessary warnings? The details of the + machinations a compiler goes thru to produce working assembly + code is of little interest to most programmers-- we will follow + the unix principal of doing our work silently. ........ + +2008-06-03 14:47 +0000 [r119927] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 119926 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 + lines Treat ECONNREFUSED as an error that will stop further + retransmissions. (issue #AST-58, patch from Switchvox) ........ + +2008-06-03 13:29 +0000 [r119744-119892] Russell Bryant <russell@digium.com> + + * main/logger.c: Do a deep copy of file and function strings to + avoid a potential crash when modules are unloaded. (closes issue + #12780) Reported by: ys Patches: logger.diff uploaded by ys + (license 281) -- modified by me for coding guidelines + + * /, channels/chan_iax2.c: Merged revisions 119838 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 + Jun 2008) | 7 lines Revert a change made for issue #12479. This + change caused a regression such that a dial string such as + (IAX2/foo) did not automatically fall back to dialing the 's' + extension anymore. (closes issue #12770) Reported by: dagmoller + ........ + + * apps/app_fax.c (added): Add app_fax from asterisk-addons, with + some additional changes to resolve compiler warnings, as well as + update to the APIs in spandsp 0.0.5. Spandsp 0.0.5 is being + distributed under the LGPL, so we can move this module into the + main tree. + + * configure, include/asterisk/autoconfig.h.in, configure.ac: After + determining that the version of spandsp installed is an + acceptable version, do a build and link test to ensure that the + library is usable, and that libtiff is also available + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add + a configure script check for spandsp + + * main/manager.c, /: Merged revisions 119742 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) + | 5 lines Improve CLI command blacklist checking for the command + manager action. Previously, it did not handle case or whitespace + properly. This made it possible for blacklisted commands to get + executed anyway. (closes issue #12765) ........ + +2008-06-02 14:35 +0000 [r119741] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c, channels/chan_gtalk.c, res/res_jabber.c: + Do not link the guest account with any configured XMPP client (in + jabber.conf). The actual connection is made when a call comes in + Asterisk. Apply this fix to Jingle too. Fix the + ast_aji_get_client function that was not able to retrieve an XMPP + client from its JID. (closes issue #12085) Reported by: junky + Tested by: phsultan + +2008-06-02 12:30 +0000 [r119688] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 119687 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 + Jun 2008) | 3 lines Even of the first PING or LAGRQ doesn't get + sent because it comes up too soon, make sure to reschedule so it + gets sent later. ........ + +2008-06-02 09:35 +0000 [r119586-119637] Christian Richter <christian.richter@beronet.com> + + * /, channels/misdn/isdn_lib.c: Merged revisions 119636 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) + | 1 line fixed compile issue when dev-mode is enabled ........ + + * channels/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: Merged + revisions 119585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) + | 1 line Added counter for unhandled_bmsg Print, this prevents + the logs to be flooded to fast and save CPU in this error + scenario. Added 'last_used' element to bc structure, when a + bchannel changes from used to free this exact time will be marked + in last_used. When a new channel is requested the find_free_chan + function will check if the new empty channel was used within the + last second, if yes it will search for the next channel, if no it + will return this channel. This simple mechanism has prooven to + prevent race conditions where the NT and TE tried to allocate the + exact same channel at the same time (RELEASE cause: 44). ........ + +2008-06-02 01:08 +0000 [r119531-119534] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 119533 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 + Jun 2008) | 2 lines Change a debug message to an actual debug + message ........ + + * apps/app_dial.c, /: Merged revisions 119530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) + | 2 lines Fix another typo in documentation ........ + +2008-06-01 21:06 +0000 [r119479] Michiel van Baak <michiel@vanbaak.info> + + * apps/app_dial.c, /: Merged revisions 119478 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) + | 2 lines small typo fix 'retires' => 'retries' ........ + +2008-05-30 21:51 +0000 [r119423] Russell Bryant <russell@digium.com> + + * main/utils.c: Fix a minor merge issue that caused a function to + not get compiled in with DEBUG_THREADS like it was supposed to + +2008-05-30 21:23 +0000 [r119419] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_queue.c: Merged revisions 119404 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008) + | 6 lines When joinempty=strict, it only failed on join if there + were busy members. If all members were logged out OR paused, then + it (incorrectly) let callers join the queue. (closes issue + #12451) Reported by: davidw ........ + +2008-05-30 19:47 +0000 [r119355] Joshua Colp <jcolp@digium.com> + + * main/autoservice.c, /: Merged revisions 119354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119354 | file | 2008-05-30 16:46:37 -0300 (Fri, 30 May 2008) | 2 + lines Fix a bug I found while testing for another issue. ........ + +2008-05-30 16:47 +0000 [r119302] Michiel van Baak <michiel@vanbaak.info> + + * contrib/init.d/rc.debian.asterisk, + contrib/init.d/rc.mandrake.asterisk, /, + contrib/init.d/rc.redhat.asterisk, + contrib/init.d/rc.gentoo.asterisk, + contrib/init.d/rc.slackware.asterisk, + contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk: + Merged revisions 119301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119301 | mvanbaak | 2008-05-30 18:44:39 +0200 (Fri, 30 May 2008) + | 14 lines dont use a bashism way to check the $VERSION variable. + The rc/init.d scripts, and safe_asterisk work on normal sh now + again. Tested on: OpenBSD 4.2 (me) Debian etch (me) Ubuntu Hardy + (me and loloski) FC9 (loloski) (closes issue #12687) Reported by: + loloski Patches: 20080529-12687-safe_asterisk-fixversion.diff.txt + uploaded by mvanbaak (license 7) Tested by: loloski, mvanbaak + ........ + +2008-05-30 16:40 +0000 [r119296-119299] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c: Suppress warning about pbx structure already + existing + + * apps/app_dial.c, include/asterisk/agi.h, CHANGES, + apps/app_stack.c: Add native AGI command GOSUB, as invoking Gosub + with EXEC does not work properly. (closes issue #12760) Reported + by: Corydon76 Patches: 20080530__bug12760.diff.txt uploaded by + Corydon76 (license 14) Tested by: tim_ringenbach, Corydon76 + +2008-05-30 12:59 +0000 [r119239] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 119238 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r119238 | russell | 2008-05-30 07:55:36 -0500 + (Fri, 30 May 2008) | 15 lines Merged revisions 119237 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) + | 7 lines - Instead of only enforcing destination call number + checking on an ACK, check all full frames except for PING and + LAGRQ, which may be sent by older versions too quickly to contain + the destination call number. (As suggested by Tim Panton on the + asterisk-dev list) - Merge changes from + team/russell/iax2-frame-race, which prevents PING and LAGRQ from + being sent before the destination call number is known. ........ + ................ + +2008-05-30 11:26 +0000 [r119207] Olle Johansson <oej@edvina.net> + + * include/asterisk/frame.h: Prefer T140 with REDundance before T140 + without. + +2008-05-29 22:28 +0000 [r119157] Russell Bryant <russell@digium.com> + + * main/autoservice.c, /: Merged revisions 119156 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008) + | 10 lines Fix a race condition in channel autoservice. There was + still a small window of opportunity for a DTMF frame, or some + other deferred frame type, to come in and get dropped. (closes + issue #12656) (closes issue #12656) Reported by: dimas Patches: + v3-12656.patch uploaded by dimas (license 88) -- with some + modifications by me ........ + +2008-05-29 21:30 +0000 [r119126] Brett Bryant <bbryant@digium.com> + + * include/asterisk/logger.h, main/logger.c, main/asterisk.c: Adds + support for changing logger settingss on remote consoles with a + new command "logger set level". i.e. "logger set level debug off" + (closes issue #10891) + +2008-05-29 20:26 +0000 [r119074] Steve Murphy <murf@digium.com> + + * main/taskprocessor.c: Had to move the ASTERISK_FILE_VERSION decl + to just after the include of "asterisk.h" or you get undefined + variable errors when you are compiling under the influence of + MTX_PROFILE + +2008-05-29 20:25 +0000 [r119072] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_zap.c, /: Merged revisions 119071 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) + | 7 lines Call waiting tone occurs too often, because it's + getting serviced by both subchannels. (closes issue #11354) + Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded + by Corydon76 (license 14) ........ + +2008-05-29 19:10 +0000 [r119015] Michiel van Baak <michiel@vanbaak.info> + + * main/features.c: Make sure the nrfds and nefds are reset to NULL + before we enter manage_parkinglot. This will get rid of CLI + warnings like: __ast_read: Exception flag set on + 'SIP/<NUMBER>-<ID>', but no exception handler (closes issue + #12748) Reported by: nreinartz Patches: + asterisk-multiparking_initialize_filedescr_sets-0.0.1.patch + uploaded by nreinartz (license 452) + +2008-05-29 19:05 +0000 [r118959-119013] Russell Bryant <russell@digium.com> + + * /, apps/app_milliwatt.c: Merged revisions 119012 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r119012 | russell | 2008-05-29 14:04:52 -0500 (Thu, 29 + May 2008) | 4 lines - Fix a typo in the argument to Playtones - + use ast_safe_sleep() instead of calling the wait application + (thanks to tilghman for pointing these out!) ........ + + * /, channels/chan_iax2.c: Merged revisions 119009 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r119009 | russell | 2008-05-29 13:49:12 -0500 + (Thu, 29 May 2008) | 16 lines Merged revisions 119008 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) + | 7 lines Merge changes from + team/russell/iax2-another-fix-to-the-fix As described in the + following post to the asterisk-dev mailing list, only enforce + destination call numbers when processing an ACK. + http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html + (closes issue #12631) ........ ................ + + * /, apps/app_milliwatt.c: Merged revisions 118961 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r118961 | russell | 2008-05-29 12:51:29 -0500 (Thu, 29 + May 2008) | 3 lines - Mark app_milliwatt dependent on + res_indications (thanks to jsmith) - fix a typo in a log message + (thanks to qwell) ........ + + * /, apps/app_milliwatt.c: Merged revisions 118956 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r118956 | russell | 2008-05-29 12:38:38 -0500 (Thu, 29 + May 2008) | 3 lines Change milliwatt to use the proper tone by + default (1004 Hz) instead of 1000 Hz. An option is there to use + 1000 Hz for anyone that might want it. ........ + +2008-05-29 17:39 +0000 [r118955-118957] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 118954 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 + May 2008) | 2 lines Define also when not DEBUG_THREADS ........ + + * channels/chan_zap.c, /, channels/chan_agent.c, + channels/chan_alsa.c, main/utils.c, include/asterisk/lock.h, + channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions + 118953 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) + | 3 lines Add some debugging code that ensures that when we do + deadlock avoidance, we don't lose the information about how a + lock was originally acquired. ........ + +2008-05-29 12:12 +0000 [r118911] Sean Bright <sean.bright@gmail.com> + + * utils/check_expr.c: Avoid build warning when execinfo.h isn't + available. (closes issue #12751) Reported by: ys Patches: + check_expr.diff uploaded by ys (license 281) + +2008-05-29 01:29 +0000 [r118880] Steve Murphy <murf@digium.com> + + * main/cdr.c, apps/app_forkcdr.c, /: Merged revisions 118858 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) | + 46 lines (closes issue #10668) (closes issue #11721) (closes + issue #12726) Reported by: arkadia Tested by: murf These changes: + 1. revert the changes made via bug 10668; I should have known + that such changes, even tho they made sense at the time, seemed + like an omission, etc, were actually integral to the CDR system + via forkCDR. It makes sense to me now that forkCDR didn't + natively end any CDR's, but rather depended on natively closing + them all at hangup time via traversing and closing them all, + whether locked or not. I still don't completely understand the + benefits of setvar and answer operating on locked cdrs, but I've + seen enough to revert those changes also, and stop messing up + users who depended on that behavior. bug 12726 found reverting + the changes fixed his changes, and after a long review and + working on forkCDR, I can see why. 2. Apply the suggested + enhancements proposed in 10668, but in a completely compatible + way. ForkCDR will behave exactly as before, but now has new + options that will allow some actions to be taken that will + slightly modify the outcome and side-effects of forkCDR. Based on + conversations I've had with various people, these small tweaks + will allow some users to get the behavior they need. For + instance, users executing forkCDR in an AGI script will find the + answer time set, and DISPOSITION set, a situation not covered + when the routines were first written. 3. A small problem in the + cdr serializer would output answer and end times even when they + were not set. This is now fixed. ........ + +2008-05-28 22:05 +0000 [r118790-118824] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: formatting changes. A lot of whitespace + issues have been resolved in this commit Also some doc updates, + but that's only 6 lines + + * channels/chan_phone.c, channels/DialTone.h (removed), + channels/chan_phone.h (added): rename DialTone.h to chan_phone.h + because chan_phone.c is the only file using it + +2008-05-28 19:56 +0000 [r118783] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Update to the janitor project for making sure + to be thread-safe when retrieving the value of a channel + variable. This covers app_queue. This commit also incorporates a + logical change. Previously, if MixMonitor is to be used to record + the call, all the arguments were parsed first. Then the + MixMonitor app would be located. Now the order of these + operations has been swapped. Now the app is located first so that + we only go through the work of parsing the arguments if the app + was found. (closes issue #12742) Reported by: snuffy Patches: + bug_12742.diff uploaded by snuffy (license 35) + +2008-05-28 17:58 +0000 [r118750] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: remove unused astobj.h header file from + chan_skinny.c + +2008-05-28 16:01 +0000 [r118702] Brett Bryant <bbryant@digium.com> + + * channels/chan_iax2.c: Fixes a bug in chan_iax that uses + send_command to poke a peer while a channel is unlocked in some + cases, and because it can cause seemingly random failures could + be related to some bugs in the tracker... + +2008-05-28 15:56 +0000 [r118695] Russell Bryant <russell@digium.com> + + * utils/check_expr.c: Fix a linkage error related to the lock + backtrace support + +2008-05-28 14:29 +0000 [r118647] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged + revisions 118646 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 + lines Add an option to use the source IP address of RTP as the + destination IP address of UDPTL when a specific option is + enabled. If the remote side is properly configured (ports + forwarded) then UDPTL will flow. (closes issue #10417) Reported + by: cstadlmann ........ + +2008-05-28 14:10 +0000 [r118614-118644] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c, include/asterisk/jingle.h: Changed to + temporary namespaces to match with latest XEPs. As soon as Jingle + is completely standardized, we can set those namespaces to their + final values. Added two attributes to the jingle_pvt struct to + store the content name attributes. Reported by Robert McQueen on + Telepathy's framework mailing list : + http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html + Keeping working on our Jingle stack! + + * channels/chan_jingle.c: Code simplification + +2008-05-27 19:45 +0000 [r118562] Brett Bryant <bbryant@digium.com> + + * channels/chan_iax2.c: Remove loop from the detection of a + sequence number that acknowledges the receiving of a packet that + we've kept in memory just incase the packet needs to be + retransmitted. + +2008-05-27 19:34 +0000 [r118560] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 118558 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 + lines Fix an issue where codec preferences were not set on + dialogs that were not authenticated via a user or peer and allow + framing to work without rtpmap in the SDP. (closes issue #12501) + Reported by: slimey ........ + +2008-05-27 19:27 +0000 [r118556] Russell Bryant <russell@digium.com> + + * include/asterisk/compat.h: Add printf format attribute for + vasprintf(). (closes issue #12729) Reported by: snuffy Patches: + bug_12729.diff uploaded by snuffy (license 35) + +2008-05-27 19:21 +0000 [r118554] Tilghman Lesher <tlesher@digium.com> + + * /, main/cli.c: Merged revisions 118551 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008) + | 6 lines When showing an error message for a command, don't + shorten the command output, as it tends to confuse the user (it's + fine for suggesting other commands, however). Reported by: + seanbright (on #asterisk-dev) Fixed by: me ........ + +2008-05-27 19:08 +0000 [r118514] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_chanspy.c: Merged revisions 118509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May + 2008) | 11 lines Russell noted to me that in the case that + separate threads use their own addressing system, the fix I made + for issue 12376 does not guarantee uniqueness to the datastores' + uids. Though I know of no system that works this way, I am going + to change this right now to prevent trying to track down some + future bug that may occur and cause untold hours of debugging + time to track down. The change involves using a global counter + which increases with each new chanspy_ds which is created. This + guarantees uniqueness. ........ + +2008-05-27 18:59 +0000 [r118466] Tilghman Lesher <tlesher@digium.com> + + * /, main/asterisk.c: Merged revisions 118465 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008) + | 8 lines NULL character should terminate only commands back to + the core, not log messages to the console. (closes issue #12731) + Reported by: seanbright Patches: 20080527__bug12731.diff.txt + uploaded by Corydon76 (license 14) Tested by: seanbright ........ + +2008-05-27 17:33 +0000 [r118417-118419] Michiel van Baak <michiel@vanbaak.info> + + * apps/app_voicemail.c: Zap is now DAHDI, mkay + + * apps/app_voicemail.c: small update to the g() option of + app_voicemail to note that gain changes only work on zap channels + right now. issue #12578 shows it's not clear right now. + +2008-05-27 16:43 +0000 [r118371] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_chanspy.c: Merged revisions 118365 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May + 2008) | 14 lines Add a unique id to the datastore allocated in + app_chanspy since it is possible that multiple spies may be + listening to the same channel. (closes issue #12376) Reported by: + DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut + (license 60) Tested by: destiny6628 (closes issue #12243) + Reported by: atis ........ + +2008-05-27 15:46 +0000 [r118359] Tilghman Lesher <tlesher@digium.com> + + * /, configs/queues.conf.sample: Merged revisions 118358 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008) + | 3 lines Add a note that pbx_config.so is needed for Local + channels. (Closes issue #12671) ........ + +2008-05-27 14:51 +0000 [r118328] Russell Bryant <russell@digium.com> + + * include/asterisk/compat.h: Add printf attribute to asprintf + +2008-05-27 13:30 +0000 [r118300-118302] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_ldap.c: When binding anonymously, credentials are + still needed. (closes issue #12601) Reported by: suretec Patches: + res_config_ldap.c.patch uploaded by suretec (license 70) + + * pbx/pbx_realtime.c: In compat14 mode, don't translate pipes + inside expressions, as they aren't argument delimiters, but + rather 'or' symbols. (Closes issue #12723) + +2008-05-25 16:17 +0000 [r118223-118252] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 118251 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) + | 12 lines Realtime flag affects construction in multiple ways, + so consulting whether rtcachefriends was set was done too soon + (needed to be done inside build_peer, not just as a flag to + build_peer). Also, fullcontact needed to be reconstructed, + because realtime separates the embedded ';' into multiple fields. + (closes issue #12722) Reported by: barthpbx Patches: + 20080525__bug12722.diff.txt uploaded by Corydon76 (license 14) + Tested by: barthpbx (Much of the discussion happened on + #asterisk-dev for diagnosing this issue) ........ + + * main/pbx.c, UPGRADE.txt: Change space-zero to now evaluate to + false, as is expected by a great many. (Inspired by a post on the + -users list) + +2008-05-24 01:14 +0000 [r118176-118178] Jeff Peeler <jpeeler@digium.com> + + * doc/api-1.6.0-changes.odt (added): add document describing API + changes from 1.4.0 to 1.6.0 + + * main/features.c: Fixes segfault in parking, patch submitted by + bmd. + +2008-05-23 22:41 +0000 [r118173-118175] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/lock.h: Make sure not to include non-existent + headers if they indeed are non-existent + + * include/asterisk/logger.h, utils/refcounter.c, main/logger.c, + utils/hashtest.c, main/utils.c, include/asterisk/lock.h, + utils/ael_main.c, utils/hashtest2.c, CHANGES, utils/conf2ael.c, + utils/check_expr.c: A new feature thanks to the fine folks at + Switchvox! If a deadlock is detected, then the typical lock + information will be printed along with a backtrace of the stack + for the offending threads. Use of this requires compiling with + DETECT_DEADLOCKS and having glibc installed. Furthermore, issuing + the "core show locks" CLI command will print the normal lock + information as well as a backtraces for each lock. This requires + that DEBUG_THREADS is enabled and that glibc is installed. All + the backtrace features may be disabled by running the configure + script with --without-execinfo as an argument + +2008-05-23 21:26 +0000 [r118164] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_zap.c, /: Merged revisions 118163 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008) + | 1 line Fix a few things I missed to ensure zt_chan_conf + structure is not modified in mkintf ........ + +2008-05-23 21:19 +0000 [r118161] Brett Bryant <bbryant@digium.com> + + * main/manager.c, main/http.c, include/asterisk/manager.h: Add new + functionality to http server that requires manager authentication + for any path that includes a directory named 'private'. This + patch also requires manager authentication for any POST's being + sent to the server as well to help secure uploads. + +2008-05-23 20:55 +0000 [r118157-118159] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Get rid of warnings for those silly + compilers which warn when freeing a const pointer + + * apps/app_voicemail.c: Use a deep copy on strings that come from + ast_events. Otherwise it is likely that after the event is freed, + we no longer refer to valid memory. (closes issue #12712) + Reported by: tomo1657 Patches: 12712.patch uploaded by putnopvut + (license 60) Tested by: tomo1657 + +2008-05-23 18:09 +0000 [r118129] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c: Protect the object from changing while the 'odbc + show' CLI command is running (Closes issue #12704) + +2008-05-23 17:12 +0000 [r118101] Michiel van Baak <michiel@vanbaak.info> + + * apps/app_chanisavail.c, CHANGES: add option 'a' to chanisavail. + If you give chanisavail a list of channels, it will only return + the first available channel. When this option is set, it will + return all the available channels from the given list. (closes + issue #12248) Reported by: dagmoller Patches: + app_chanisavail-snv.patch-v2.txt uploaded by dagmoller (license + 436) - major changes by me because russellb pointed out some + buffer overflows and codeguideline issues. Converted it all to + the ast_str_* api Tested by: dagmoller, mvanbaak + +2008-05-23 13:00 +0000 [r118053] Tilghman Lesher <tlesher@digium.com> + + * /, doc/cli.txt (added): Merged revisions 118052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008) + | 3 lines Add information on using the Asterisk console, + including tab command line completion. (Closes issue #12681) + ........ + +2008-05-23 12:37 +0000 [r118049] Russell Bryant <russell@digium.com> + + * include/asterisk/utils.h, /, main/utils.c: Merged revisions + 118048 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r118048 | russell | 2008-05-23 07:30:53 -0500 (Fri, 23 May 2008) + | 9 lines Don't declare a function that takes variable arguments + as inline, because it's not valid, and on some compilers, will + emit a warning. + http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes + issue #12289) Reported by: francesco_r Patches by Tilghman, final + patch by me ........ + +2008-05-23 10:33 +0000 [r118020] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_gtalk.c, res/res_jabber.c: - remove whitespaces + between tags in received XML packets before giving them to the + parser ; - report Gtalk error messages from a buddy to the + console. This patch makes Asterisk "Google Jingle" (chan_gtalk) + implementation work with Empathy. Note that this is only true for + audio streams, not video. Thank you to PH for his great help! + (closes issue #12647) Reported by: PH Patches: trunk-12647-1.diff + uploaded by phsultan (license 73) Tested by: phsultan, PH + +2008-05-22 21:43 +0000 [r117988] Sean Bright <sean.bright@gmail.com> + + * channels/chan_usbradio.c: Split the compile flags out and wire up + some dependencies + +2008-05-22 21:42 +0000 [r117983-117986] Tilghman Lesher <tlesher@digium.com> + + * pbx/pbx_realtime.c, configs/pbx_realtime.conf (added): Add a + compatibility option for upgrading realtime extensions + + * channels/chan_vpb.cc: Fix trunk breakage + +2008-05-22 20:01 +0000 [r117950] Sean Bright <sean.bright@gmail.com> + + * channels/chan_usbradio.c, apps/app_rpt.c: A couple more places + the frame data change was missed. + +2008-05-22 18:54 +0000 [r117900] Tilghman Lesher <tlesher@digium.com> + + * /, main/asterisk.c: Merged revisions 117899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117899 | tilghman | 2008-05-22 13:53:53 -0500 (Thu, 22 May 2008) + | 2 lines Also remove preamble from asynchronous events (reported + by jsmith on #asterisk-dev) ........ + +2008-05-22 17:50 +0000 [r117834-117870] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_nbs.c: one more place I forgot + + * channels/chan_console.c: chan_console fixes because of + ast_frame.data => ast_frame.data.ptr + +2008-05-22 17:10 +0000 [r117828] Jason Parker <jparker@digium.com> + + * funcs/func_speex.c, codecs/codec_speex.c, + formats/format_ogg_vorbis.c, apps/app_jack.c: Fix a few places + where frame data was used directly. + +2008-05-22 17:08 +0000 [r117802-117825] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_misdn.c: oops + + * channels/chan_misdn.c: forgot chan_misdn + + * main/udptl.c, channels/chan_local.c, main/frame.c, + codecs/codec_adpcm.c, apps/app_test.c, apps/app_alarmreceiver.c, + formats/format_sln16.c, formats/format_wav_gsm.c, + apps/app_ices.c, channels/chan_iax2.c, main/indications.c, + channels/chan_skinny.c, formats/format_pcm.c, apps/app_zapscan.c, + main/features.c, channels/chan_alsa.c, formats/format_h263.c, + apps/app_externalivr.c, formats/format_jpeg.c, + apps/app_milliwatt.c, formats/format_gsm.c, apps/app_dial.c, + codecs/codec_g722.c, formats/format_wav.c, codecs/codec_g726.c, + apps/app_disa.c, include/asterisk/channel.h, + channels/iax2-parser.c, apps/app_speech_utils.c, + channels/chan_misdn.c, apps/app_zapbarge.c, main/audiohook.c, + apps/app_chanspy.c, formats/format_g726.c, + channels/chan_unistim.c, apps/app_meetme.c, formats/format_sln.c, + codecs/codec_gsm.c, res/res_musiconhold.c, channels/chan_gtalk.c, + apps/app_followme.c, codecs/codec_zap.c, formats/format_ilbc.c, + main/channel.c, channels/chan_phone.c, res/res_agi.c, + apps/app_mp3.c, main/app.c, codecs/codec_resample.c, + formats/format_h264.c, include/asterisk/frame.h, + channels/chan_mgcp.c, codecs/codec_lpc10.c, apps/app_nbscat.c, + codecs/codec_a_mu.c, channels/chan_zap.c, channels/chan_sip.c, + apps/app_festival.c, codecs/codec_alaw.c, main/slinfactory.c, + main/translate.c, res/res_adsi.c, channels/chan_console.c, + apps/app_queue.c, channels/chan_oss.c, main/rtp.c, + channels/chan_jingle.c, formats/format_vox.c, main/abstract_jb.c, + channels/chan_h323.c, main/file.c, apps/app_sms.c, + formats/format_g723.c, codecs/codec_ulaw.c, main/dsp.c, + formats/format_g729.c: - revert change to ast_queue_hangup and + create ast_queue_hangup_with_cause - make data member of the + ast_frame struct a named union instead of a void Recently the + ast_queue_hangup function got a new parameter, the hangupcause + Feedback came in that this is no good and that instead a new + function should be created. This I did. The hangupcause was + stored in the seqno member of the ast_frame struct. This is not + very elegant, and since there's already a data member that one + should be used. Problem is, this member was a void *. Now it's a + named union so it can hold a pointer, an uint32 and there's a + padding in case someone wants to store another type in there in + the future. This commit is so massive, because all ast_frame.data + uses have to be altered to ast_frame.data.data Thanks russellb + and kpfleming for the feedback. (closes issue #12674) Reported + by: mvanbaak + +2008-05-22 16:05 +0000 [r117794] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Committing a fix pointed out by Atis Lezdins on + the asterisk-dev list. Thanks! + +2008-05-22 15:49 +0000 [r117792] Sean Bright <sean.bright@gmail.com> + + * configs/jabber.conf.sample: Minor text fix. roster -> resource. + +2008-05-22 13:40 +0000 [r117756] Russell Bryant <russell@digium.com> + + * build_tools/make_buildopts_h, main/asterisk.c: Store build-time + options as a string in AST_BUILDOPTS in buildopts.h. Also, + display this information in the "core show settings" CLI command. + This is useful if you want to verify that you're running a build + with DONT_OPTIMIZE, DEBUG_THREADS, etc. + +2008-05-22 05:10 +0000 [r117725] Tilghman Lesher <tlesher@digium.com> + + * doc/externalivr.txt, apps/app_externalivr.c, CHANGES: Enhance + ExternalIVR with new options and commands. (closes issue #12705) + Reported by: ctooley Patches: + new_externalivr_argument_format-v2.diff uploaded by ctooley + (license 136) new_externalivr_documentation.diff uploaded by + ctooley (license 136) and a few additional fixes by me + +2008-05-21 22:34 +0000 [r117693] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/logger.h, utils/refcounter.c, main/logger.c, + utils/hashtest.c, utils/ael_main.c, utils/hashtest2.c: This + change makes it so that logs will report the correct source of + verbose messages. Until this change, all verbose messages in + Asterisk's log files reported logger.c as the source of the + message. + +2008-05-21 21:31 +0000 [r117628-117658] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_zap.c, /: Merged revisions 117582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117582 | jpeeler | 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008) + | 2 lines Ensure that passed in zt_chan_conf structure is not + modified in mkintf. ........ + + * channels/chan_zap.c, /: Merged revisions 117462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008) + | 3 lines Pass a pointer for the conf parameter to the function + mkintf rather than the whole zt_chan_conf structure. Another + commit is following to make sure the zt_chan_conf structure is + not modified. ........ + +2008-05-21 20:27 +0000 [r117625] Mark Michelson <mmichelson@digium.com> + + * doc/manager_1_1.txt, apps/app_queue.c: Add a new manager event, + AgentRingNoAnswer to app_queue. (closes issue #12591) Reported + by: CCHAsteria Patches: app_queue_RNA_event.diff uploaded by + CCHAsteria (license 477) + +2008-05-21 19:39 +0000 [r117575] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 117574 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2 + lines Apply the autoframing setting to dialogs that do not get + matched against a user or peer. ........ + +2008-05-21 18:43 +0000 [r117520] Tilghman Lesher <tlesher@digium.com> + + * /, main/asterisk.c: Merged revisions 117519 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117519 | tilghman | 2008-05-21 13:40:14 -0500 (Wed, 21 May 2008) + | 3 lines Strip the preamble from the output also when -rx is not + being used (Related to issue #12702) ........ + +2008-05-21 18:31 +0000 [r117517] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Optimize the update_realtime_member_field + function by not having to query the database for the member and + instead using a cached uniqueid. Special thanks to atis for + creating this and for keeping it up to date with necessary + changes (closes issue #11896) Reported by: atis Patches: + realtime_uniqueid_v6.patch uploaded by atis (license 242) Tested + by: atis + +2008-05-21 18:29 +0000 [r117481-117515] Russell Bryant <russell@digium.com> + + * /, main/asterisk.c: Merged revisions 117514 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117514 | russell | 2008-05-21 13:28:46 -0500 (Wed, 21 May 2008) + | 4 lines Don't filter the magic character in the network + verboser. It gets filtered once it reaches the client. (related + to issue #12702, pointed out by tilghman) ........ + + * /, main/asterisk.c, pbx/pbx_gtkconsole.c: Merged revisions 117507 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117507 | russell | 2008-05-21 13:19:34 -0500 (Wed, 21 May 2008) + | 7 lines 1) Don't print the verbose marker in front of every + message from ast_verbose() being sent to remote consoles. 2) Fix + pbx_gtkconsole to filter out the verbose marker. (related to + issue #12702) ........ + + * /, main/asterisk.c: Merged revisions 117479 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117479 | russell | 2008-05-21 13:11:51 -0500 (Wed, 21 May 2008) + | 6 lines Don't display the verbose marker for calls to + ast_verbose() that do not include a VERBOSE_PREFIX in front of + the message. (closes issue #12702) Reported by: johnlange Patched + by me ........ + +2008-05-21 13:39 +0000 [r117431] Tilghman Lesher <tlesher@digium.com> + + * apps/app_externalivr.c: On socket-based connections, there is no + error FD, so don't try waiting on one. (closes issue #12697) + Reported by: ctooley Patches: + fix_externalivr_waitfor_nandfds-v3.diff uploaded by ctooley + (license 136) + +2008-05-21 11:24 +0000 [r117401] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c: do not die on SDL_ACTIVEEVENT reporting + lost focus. + +2008-05-21 02:20 +0000 [r117367] Mark Michelson <mmichelson@digium.com> + + * main/config.c: Be sure that we cache included files for each + source file which loads a configuration file. As it was, only the + first did so. This led to a problem if the included file was + changed (but not the configuration file which includes it) and + the second source file attempted to reload the configuration. It + would not see that the included file had changed. In this + particular example, res_phoneprov and chan_sip both loaded + sip.conf, which included a file call sip.peers.conf. Since + res_phoneprov was the first to load sip.conf, only it cached the + fact that sip.conf included sip.peers.conf. If sip.peers.conf + were changed and sip.conf were not and a sip reload were issued + (meaning that chan_sip attempts to reload sip.conf only if it and + its included files have changed) the changes made to + sip.peers.conf would not be seen and therefore no action would be + taken. (closes issue #12693) Reported by: marsosa + +2008-05-21 01:00 +0000 [r117335] Steve Murphy <murf@digium.com> + + * utils/ael_main.c: These changes were made via the comments + atis_work made at 4:30am (Mountain Time zone- US) in + #asterisk-dev on 20 May 2008. He noted that a backslash was being + inserted before commas in app call arguments in the + extensions.conf.aeldump file that you get from aelparse with the + -w arg. This was being generated from code left over from 1.4, + where commas were substituted with '|', and any remaining commas + needed to be escaped. Many thanks to atis for his comment; please + let us know if these changes break anything! + +2008-05-20 18:07 +0000 [r117266-117297] Luigi Rizzo <rizzo@icir.org> + + * main/manager.c: + Implement a variant of astman_get_header() to + return the first or last match, and possibly skip empty fields. + The function is useful (and used here) when a form submits + multiple 'Action' fields to the Manager. This change slightly + modifies the current behaviour, but only in the case the user + supplies multiple 'Action: ' lines and the first ones are empty, + so the change is totally harmless. + Fix style on a couple of "if + (displayconnects)" statements; + Expand a bit the 'Manager Test' + interface, to make it slightly more user friendly. But also + comment that the HTML should not be embedded in the C source. + None of this stuff needs to be applied to 1.4. + + * main/http.c: Document the possible presence of multiple variables + with the same name in http queries, which might confuse the + manager. Replace calls to ast_uri_decode() with a local function + that also replaces '+' with ' ', as this is the normal encoding + for spaces in http requests. This allows passing cli commands to + the manager through the http interface. + + * main/http.c: Reverse the check for Cookie: and remove leftover + code implementing the same thing. Add an ast_debug() call to help + debugging the url matching. + +2008-05-20 16:25 +0000 [r117262-117264] Tilghman Lesher <tlesher@digium.com> + + * CHANGES, res/res_odbc.c: Increase limit of unshared connections + from 1023 to 4.2 billion. (Related to issue #12677) + + * res/res_odbc.c: Revert part of previous fix, and heavily comment + the logic for object destruction, for future users. (Closes issue + #12677) + +2008-05-19 20:45 +0000 [r117212] Russell Bryant <russell@digium.com> + + * main/channel.c: Minor formatting change to test a mantis change + ... (issue #12674) + +2008-05-19 20:06 +0000 [r117182] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Change + the default for the pridialplan parameter to the far more common + case of 'unknown', and better document the use of each parameter. + (closes issue #12633) Reported by: tzafrir Patches: + pridialplan_unknown_2.diff uploaded by tzafrir (license 46) + +2008-05-19 16:53 +0000 [r117133-117136] Joshua Colp <jcolp@digium.com> + + * res/res_smdi.c, /: Merged revisions 117135 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117135 | file | 2008-05-19 13:50:52 -0300 (Mon, 19 May 2008) | 6 + lines Use the right pthread lock and condition when waiting. + (closes issue #12664) Reported by: tomo1657 Patches: + res_smdi.c.patch uploaded by tomo1657 (license 484) ........ + + * res/res_odbc.c: Remove a premature mutex destroy (the destruction + callback will end up destroying it) and use a callback to purge + remaining classes. (closes issue #12677) Reported by: falves11 + +2008-05-19 16:07 +0000 [r117088] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/utils.h, /: Merged revisions 117086 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r117086 | tilghman | 2008-05-19 11:05:05 -0500 (Mon, 19 + May 2008) | 2 lines The addition of usleep(2) within ast_assert + requires the inclusion of the unistd.h header ........ + +2008-05-19 16:03 +0000 [r117085] Joshua Colp <jcolp@digium.com> + + * main/logger.c: The logger closes the files it is logging to when + reloading so we have to read in the logger configuration even if + it has not changed so that the logs get opened again. (closes + issue #12665) Reported by: DennisD + +2008-05-19 15:47 +0000 [r117084] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c: trap potential failures of SDL when + SDL_WINDOWID is pointing to a random window. This commit is + essentially a workaround for some undesirable behaviour of SDL; + we should not be doing this in the application, but in the + library. + +2008-05-19 15:24 +0000 [r117082] Joshua Colp <jcolp@digium.com> + + * /, channels/h323/ast_h323.cxx: Merged revisions 117081 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r117081 | file | 2008-05-19 12:22:10 -0300 (Mon, 19 May 2008) | 6 + lines Make chan_h323 work with pwlib 1.12.0 (closes issue #12682) + Reported by: bamby Patches: pwlib_nopipe.diff uploaded by bamby + (license 430) ........ + +2008-05-19 14:54 +0000 [r117024-117053] Luigi Rizzo <rizzo@icir.org> + + * configs/oss.conf.sample: fix example configuration for video + support in chan_oss + + * channels/console_gui.c: Some fixes to the code to support running + on an externally supplied window. SDL (at least recent 1.2.x + versions) has the ability to run the graphic output into an + externally supplied window, whose ID in the environment variable + SDL_WINDOWID. Ideally, applications should run unchanged + irrespective of who creates the window. Unfortunately, SDL does + not subscribe to mouse, key and resize events on externally + supplied windows, so we need to do ask for these events + explicitly. On passing, also add some code to handle + SDL_ACTIVEEVENT so if the X11 window is killed while we are + active, we call "stop now" to terminate the asterisk instance. + + * channels/console_video.c: Allow users to specify 'startgui=1' in + oss.conf so that the graphic screen for the video console is + activated at startup. + +2008-05-19 03:44 +0000 [r116979] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 116978 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 + May 2008) | 4 lines Avoid access of uninitialized memory. This + caused a bunch of crashes for me while doing load testing of + development branch where I'm working on some performance + improvements. ........ + +2008-05-18 21:15 +0000 [r116948] Tilghman Lesher <tlesher@digium.com> + + * utils/astcanary.c: Add a set of text to the file astcanary uses + to communicate back the main Asterisk process, which explains the + purpose for the file being there. This should assist people who + find the file and wonder why it exists. + +2008-05-18 19:58 +0000 [r116919] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Remove duplicate colon on Reason header + (closes issue #12678) + +2008-05-17 19:39 +0000 [r116800-116884] Joshua Colp <jcolp@digium.com> + + * channels/iax2-parser.h, channels/chan_iax2.c: Improve native + transfers when a chain of IAX2 connections are in use. (closes + issue #7567) Reported by: tjd Patches: bug_7567_update_v2.diff + uploaded by snuffy (license 35) + + * channels/chan_sip.c: Try to fix attended transfers. + + * /, channels/chan_skinny.c: Merged revisions 116799 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May + 2008) | 4 lines Check to make sure an RTP structure exists before + calling ast_rtp_new_source on it. (closes issue #12669) Reported + by: sbisker ........ + +2008-05-16 20:00 +0000 [r116797] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Try to see if we can make our ringback + situation a little better + +2008-05-16 17:08 +0000 [r116765] Sean Bright <sean.bright@gmail.com> + + * channels/xpmr/xpmr.c: Compile under dev-mode, please. + +2008-05-16 00:51 +0000 [r116731] Jim Dixon <telesistant@hotmail.com> + + * channels/chan_usbradio.c, channels/xpmr/xpmr.h, + channels/xpmr/sinetabx.h, channels/Makefile, + channels/xpmr/xpmr.c, apps/app_rpt.c, channels/xpmr/xpmr_coef.h: + Bring all app_rpt and chan_usbradio stuff up to date + +2008-05-15 22:05 +0000 [r116694] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/utils.h, include/asterisk/strings.h: Add an + extra check in ast_strlen_zero, and make ast_assert() not print + the file, line, and function name twice. (Closes issue #12650) + +2008-05-15 21:54 +0000 [r116663] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_sip.c: Fixes a problem I was having with two SIP + phones using Packet2Packet bridging dropping audio nearly + immediately. The problem was that the lock on the SIP dialog was + not being unlocked while the bridge was still active. (Related to + issue #12566) + +2008-05-15 17:58 +0000 [r116631] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_csv.c: Don't unload config on reload, when config has not + changed. (Closes issue #12652) + +2008-05-15 15:40 +0000 [r116590-116594] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: When counting urgent messages when using + IMAP storage, take into account that the urgent messages are not + in their own folder but are actually "flagged" messages in the + INBOX. (closes issue #12659) Reported by: jaroth Patches: + urgentfolder_v2.patch uploaded by jaroth (license 50) Tested by: + jaroth + + * UPGRADE.txt, apps/app_voicemail.c: Modify externnotify to take + the number of urgent voicemails as a final argument instead of + the string "Urgent" (closes issue #12660) Reported by: jaroth + Patches: externnotify.patch uploaded by jaroth (license 50) + + * apps/app_voicemail.c: Prevent crashes from occurring due to a + strcmp of a NULL pointer. (closes issue #12661) Reported by: + jaroth Patches: urgentcompare.patch uploaded by jaroth (license + 50) + +2008-05-15 10:56 +0000 [r116557] Luigi Rizzo <rizzo@icir.org> + + * main/manager.c, funcs/func_timeout.c, main/features.c, + apps/app_waituntil.c, main/utils.c, main/taskprocessor.c, + main/sched.c: Use casts or intermediate variables to remove a + number of platform/compiler-dependent warnings when handing + struct timeval fields, both reading and printing them. It is a + lost battle to handle the different ways struct timeval is + handled on the various platforms and compilers, so try to be + pragmatic and go through int/long which are universally + supported. + +2008-05-14 22:15 +0000 [r116522] Mark Michelson <mmichelson@digium.com> + + * CHANGES, apps/app_chanspy.c: Adding a new option to Chanspy(). + The 'd' option allows for the spy to press DTMF digits to switch + between spying modes. Pressing 4 activates spy mode, pressing 5 + activates whisper mode, and pressing 6 activates barge mode. Use + of this feature overrides the normal operation of DTMF numbers. + This feature is courtesy of Switchvox. + +2008-05-14 21:54 +0000 [r116471] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Fix pedanticness. + +2008-05-14 21:40 +0000 [r116469] Russell Bryant <russell@digium.com> + + * main/channel.c, main/udptl.c, include/asterisk/utils.h, /, + channels/chan_agent.c, main/abstract_jb.c, + include/asterisk/channel.h, main/rtp.c, main/sched.c: Merged + revisions 116463 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) + | 4 lines Add ast_assert(), which can be used to handle fatal + errors. It is only compiled in if dev-mode is enabled, and only + aborts if DO_CRASH is defined. (inspired by issue #12650) + ........ + +2008-05-14 21:39 +0000 [r116467] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 116466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008) + | 7 lines Avoid zombies when the channel exits before the AGI. + (closes issue #12648) Reported by: gkloepfer Patches: + 20080514__bug12648.diff.txt uploaded by Corydon76 (license 14) + Tested by: gkloepfer ........ + +2008-05-14 21:11 +0000 [r116461] Joshua Colp <jcolp@digium.com> + + * main/pbx.c: Add a missing context unlock. (closes issue #12649) + Reported by: ys Patches: pbx.c.diff uploaded by ys (license 281) + +2008-05-14 20:43 +0000 [r116407-116410] Jason Parker <jparker@digium.com> + + * /, configs/voicemail.conf.sample: Merged revisions 116409 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May 2008) | + 1 line Document exitcontext in app_voicemail sample config + ........ + + * apps/app_voicemail.c: Voicemail "* exit" should not require an + exitcontext to be specified. The behavior in 1.4 was that it + would use the current context if an exitcontext existed. (closes + issue #12605) Reported by: kenjreno Patches: 12605-starexit.diff + uploaded by qwell (license 4) Tested by: file + +2008-05-14 18:54 +0000 [r116350-116353] Joshua Colp <jcolp@digium.com> + + * /, main/Makefile: Merged revisions 116352 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4 + lines Add linux-gnueabi in. (closes issue #12529) Reported by: + tzafrir ........ + + * res/res_config_ldap.c: Make the ldap version setting work without + having both version and protocol set. (closes issue #12613) + Reported by: suretec + +2008-05-14 16:53 +0000 [r116298] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_externalivr.c: Merged revisions 116296 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r116296 | tilghman | 2008-05-14 11:46:48 -0500 (Wed, 14 + May 2008) | 2 lines Detect another way for a connection to have + gone away. (closes issue #12618) Reported by: ctooley Patches: + 1.4-externalivr-test_fd.diff uploaded by ctooley (license 136) + trunk-externalivr-test_fd.diff uploaded by ctooley (license 136) + ........ + +2008-05-14 16:52 +0000 [r116297] Jeff Peeler <jpeeler@digium.com> + + * main/pbx.c, main/features.c: Fixed a few problems with + multiparking: call not being parked in the correct parking spot, + caller not being notified of parking spot position, and + improperly hanging up the call during a transfer due to timing + out (not providing the extension in which to transfer). + +2008-05-14 14:16 +0000 [r116179-116240] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Don't add linefeed on received MESSAGE + + * channels/chan_sip.c: Properly declare charset for text messages. + + * CREDITS, main/frame.c, channels/chan_sip.c, + include/asterisk/rtp.h, CHANGES, include/asterisk/frame.h, + main/rtp.c: Adding spport for T.140 RED - Simple RTP redundancy + to prevent packet loss in text stream Work sponsored by Omnitor + AB, Stockholm, Sweden (http://www.omnitor.se) + + * /, channels/chan_sip.c: Merged revisions 116230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 + lines Accept text messages even with Content-Type: + text/plain;charset=Södermanländska ........ + + * main/manager.c, pbx/pbx_spool.c, channels/chan_sip.c, CHANGES, + sample.call: Add support for codec settings in originate via call + file and manager. This is to enable video and text in originated + calls. Development sponsored by Omnitor AB, Sweden. + (http://www.omnitor.se) + + * res/res_agi.c: Formatting changes (coding guidelines) while + thinking about something else... + + * channels/chan_sip.c: Reformatting + + * channels/chan_sip.c: Adding comments + + * pbx/pbx_spool.c: Doxygen formatting change only + +2008-05-14 00:20 +0000 [r116089-116138] Mark Michelson <mmichelson@digium.com> + + * channels/chan_skinny.c: Undo inadvertent changes to chan_skinny + caused by the merging of urgent messaging support. Thanks to + Damien Wedhorn for pointing out the problem. + + * main/channel.c, /, include/asterisk/lock.h: Merged revisions + 116088 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May + 2008) | 12 lines A change to the way channel locks are handled + when DEBUG_CHANNEL_LOCKS is defined. After debugging a deadlock, + it was noticed that when DEBUG_CHANNEL_LOCKS is enabled in + menuselect, the actual origin of channel locks is obscured by the + fact that all channel locks appear to happen in the function + ast_channel_lock(). This code change redefines ast_channel_lock + to be a macro which maps to __ast_channel_lock(), which then + relays the proper file name, line number, and function name + information to the core lock functions so that this information + will be displayed in the case that there is some sort of locking + error or core show locks is issued. ........ + +2008-05-13 21:18 +0000 [r116001-116039] Russell Bryant <russell@digium.com> + + * channels/chan_local.c, /: Merged revisions 116038 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 + May 2008) | 24 lines Fix a deadlock involving channel autoservice + and chan_local that was debugged and fixed by mmichelson and me. + We observed a system that had a bunch of threads stuck in + ast_autoservice_stop(). The reason these threads were waiting + around is because this function waits to ensure that the channel + list in the autoservice thread gets rebuilt before the stop() + function returns. However, the autoservice thread was also + locked, so the autoservice channel list was never getting + rebuilt. The autoservice thread was stuck waiting for the channel + lock on a local channel. However, the local channel was locked by + a thread that was stuck in the autoservice stop function. It + turned out that the issue came down to the local_queue_frame() + function in chan_local. This function assumed that one of the + channels passed in as an argument was locked when called. + However, that was not always the case. There were multiple cases + in which this channel was not locked when the function was + called. We fixed up chan_local to indicate to this function + whether this channel was locked or not. The previous assumption + had caused local_queue_frame() to improperly return with the + channel locked, where it would then never get unlocked. (closes + issue #12584) (related to issue #12603) ........ + + * main/autoservice.c, /: Merged revisions 115990 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008) + | 5 lines Fix an issue that I noticed in autoservice while + mmichelson and I were debugging a different problem. I noticed + that it was theoretically possible for two threads to attempt to + start the autoservice thread at the same time. This change makes + the process of starting the autoservice thread, thread-safe. + ........ + +2008-05-13 20:29 +0000 [r115945] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_alsa.c: Merged revisions 115944 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May + 2008) | 4 lines Use the right flag to open the audio in + non-blocking. (closes issue #12616) Reported by: + nicklewisdigiumuser ........ + +2008-05-13 20:18 +0000 [r115939-115941] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Need to clear calling_party_cat variable + after we retrieve it + + * channels/chan_zap.c: Add support for receiving calling party + category + +2008-05-13 18:38 +0000 [r115886] Tilghman Lesher <tlesher@digium.com> + + * /, main/asterisk.c: Merged revisions 115884 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115884 | tilghman | 2008-05-13 13:36:13 -0500 (Tue, 13 May 2008) + | 3 lines If the socket dies (read returns 0=EOF), return + immediately. (Closes issue #12637) ........ + +2008-05-13 17:42 +0000 [r115847-115850] Russell Bryant <russell@digium.com> + + * funcs/func_speex.c, apps/app_skel.c, apps/app_jack.c: + Re-introduce proper error handling that was removed in recent + commits. + + * res/res_smdi.c: Initialize the start time in smdi_msg_wait. + Somehow this code got lost in trunk. + +2008-05-12 20:34 +0000 [r115813] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/install_prereq (added): Add a script which + installs every package needed for a Debian install of Asterisk, + and includes possible support (to be contributed) for various + other distributions. (closes issue #10523) Reported by: tzafrir + Patches: install_prereq_2 uploaded by tzafrir (license 46) + +2008-05-12 18:39 +0000 [r115784] Olle Johansson <oej@edvina.net> + + * main/features.c, doc/tex/channelvariables.tex: Add support for + playing an audio file for caller and callee at start and stop of + monitoring (one-touch monitor). Keep messages short, since the + other party is waiting while one party hear the message... + +2008-05-12 17:55 +0000 [r115737] Mark Michelson <mmichelson@digium.com> + + * main/utils.c: Merged revisions 115735 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May + 2008) | 7 lines If a thread holds no locks, do not print any + information on the thread when issuing a core show locks command. + This will help to de-clutter output somewhat. Russell said it + would be fine to place this improvement in the 1.4 branch, so + that's why it's going here too. ........ + +2008-05-12 16:35 +0000 [r115705] Jason Parker <jparker@digium.com> + + * apps/app_queue.c: Correctly document state interface for + AddQueueMember. Discovered while looking at issue #12626. + +2008-05-12 15:17 +0000 [r115669] Brett Bryant <bbryant@digium.com> + + * channels/chan_iax2.c: A small change to fix iax2 native bridging. + +2008-05-11 03:23 +0000 [r115598-115600] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, + configure.ac: Add Zap MTP2 support to chan_zap + + * channels/chan_zap.c: Open up audio channel when we get ACM on SS7 + event + +2008-05-10 14:19 +0000 [r115596] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_pgsql.c: Ensure that "calldate" is acceptable for a + column name. + +2008-05-10 03:30 +0000 [r115593-115595] Claude Patry <cpatry@gmail.com> + + * configs/queues.conf.sample: fix a sample since we now required , + and not | for the arguments separator + + * apps/app_skel.c, apps/app_jack.c: ameliorate load and unload to + dont use DECLINED or FAILED, when theres no .conf involved. + + * funcs/func_speex.c: since we unregister, that has not been + properly registered, i standardized this. + +2008-05-09 22:36 +0000 [r115588-115591] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Remove a debug line + + * channels/chan_unistim.c, include/asterisk/app.h, main/manager.c, + channels/chan_sip.c, channels/chan_skinny.c, UPGRADE.txt, + main/app.c, CHANGES, channels/chan_iax2.c, apps/app_voicemail.c: + Adding support for "urgent" voicemail messages. Messages which + are marked "urgent" are considered to be higher priority than + other messages and so they will be played before any other + messages in a user's mailbox. There are two ways to leave an + urgent message. 1. send the 'U' option to VoiceMail(). 2. Set + review=yes in voicemail.conf. This will give instructions for a + caller to mark a message as urgent after the message has been + recorded. I have tested that this works correctly with file and + ODBC storage, and James Rothenberger (who wrote initial support + for this feature) has tested its use with IMAP storage. (closes + issue #11817) Reported by: jaroth Based on branch + http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent + Tested by: putnopvut, jaroth + +2008-05-09 20:05 +0000 [r115584-115586] Brett Bryant <bbryant@digium.com> + + * CHANGES: Update CHANGES file for previous commit of ENUM and + TXCIDNAME changes. + + * funcs/func_enum.c, include/asterisk/enum.h, main/enum.c: The + following patch adds new options and alters the default behavior + of the ENUM* functions. The TXCIDNAME lookup function has also + gotten a new paramater. The new options for ENUM* functions + include 'u', 's', 'i', and 'd' which return the full uri, trigger + isn specific rewriting, look for branches into an infrastructure + enum tree, or do a direct dns lookup of a number respectively. + The new paramater for TXCIDNAME adds a zone-suffix argument for + looking up caller id's in DNS that aren't e164.arpa. This patch + is based on the original code from otmar, modified by snuffy, and + tested by jtodd, me, and others. (closes issue #8089) Reported + by: otmar Patches: 20080508_bug8089-1.diff - original code by + otmar (license 480), - revised by snuffy (license 35) Tested by: + oej, otmar, jtodd, Corydon76, snuffy, alexnikolov, bbryant + +2008-05-09 17:28 +0000 [r115582] Tilghman Lesher <tlesher@digium.com> + + * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: + Allow a password change to be validated by an external script. + (closes issue #12090) Reported by: jaroth Patches: + vm-check-newpassword.diff.txt uploaded by mvanbaak (license 7) + 20080509__bug12090.diff.txt uploaded by Corydon76 (license 14) + +2008-05-09 16:36 +0000 [r115580] Joshua Colp <jcolp@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 115579 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2 + lines Improve res_ninit and res_ndestroy autoconf logic on the + Darwin platform. ........ + +2008-05-08 19:20 +0000 [r115552-115569] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 115568 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 + May 2008) | 2 lines Remove debug output. ........ + + * /, channels/chan_iax2.c: Merged revisions 115565 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r115565 | russell | 2008-05-08 14:15:25 -0500 + (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) + | 25 lines Fix a race condition that bbryant just found while + doing some IAX2 testing. He was running Asterisk trunk running + IAX2 calls through a few Asterisk boxes, however, the audio was + extremely choppy. We looked at a packet trace and saw a storm of + INVAL and VNAK frames being sent from one box to another. It + turned out that what had happened was that one box tried to send + a CONTROL frame before the 3 way handshake had completed. So, + that frame did not include the destination call number, because + it didn't have it yet. Part of our recent work for security + issues included an additional check to ensure that frames that + are supposed to include the destination call number have the + correct one. This caused the frame to be rejected with an INVAL. + The frame would get retransmitted for forever, rejected every + time ... This race condition exists in all versions that got the + security changes, in theory. However, it is really only likely + that this would cause a problem in Asterisk trunk. There was a + control frame being sent (SRCUPDATE) at the _very_ beginning of + the call, which does not exist in 1.2 or 1.4. However, I am + fixing all versions that could potentially be affected by the + introduced race condition. These changes are what bbryant and I + came up with to fix the issue. Instead of simply dropping control + frames that get sent before the handshake is complete, the code + attempts to wait a little while, since in most cases, the + handshake will complete very quickly. If it doesn't complete + after yielding for a little while, then the frame gets dropped. + ........ ................ + + * /, channels/chan_sip.c: Merged revisions 115561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) + | 3 lines Don't give up on attempting an outbound registration if + we receive a 408 Timeout. (closes issue #12323) ........ + + * /, contrib/scripts/postgres_cdr.sql (removed): Merged revisions + 115557 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008) + | 3 lines remove postgres_cdr.sql, as the CDR schema is in + realtime_pgsql.sql, as well (closes issue #9676) ........ + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115554 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008) + | 3 lines Don't exit the script if Asterisk is not running. + (closes issue #12611) ........ + + * main/pbx.c, /: Merged revisions 115551 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008) + | 4 lines Don't use a channel before checking for channel + allocation failure. (closes issue #12609) Reported by: edantie + ........ + +2008-05-08 15:04 +0000 [r115548] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Remove unused code as well as demote an + error message to a debug message + +2008-05-08 14:41 +0000 [r115537-115546] Russell Bryant <russell@digium.com> + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115545 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008) + | 4 lines Use the same method for executing Asterisk as the rest + of the script. (closes issue #12611) Reported by: b_plessis + ........ + + * main/sched.c: Fix up a problem that was introduced into the + scheduler when it was converted to use doubly linked lists. The + schedule() function had an optimization that had it try to guess + which direction would be better for the traversal to insert the + task into the scheduler queue. However, if the code chose the + path where it traversed the queue in reverse, and the result was + that the task should be at the head of the queue, then the code + would actually put it at the tail, instead. (Problem found by + bbryant, debugged and fixed by bbryant and me) + +2008-05-07 20:22 +0000 [r115525-115535] Tilghman Lesher <tlesher@digium.com> + + * sounds/Makefile: Advance to next sounds release + + * res/res_odbc.c: Don't free the object on destroy, as astobj2 + takes care of that for you + +2008-05-07 18:33 +0000 [r115513-115523] Russell Bryant <russell@digium.com> + + * res/res_config_ldap.c: Only save a password if a username exists. + (closes issue #12600) Reported By: suretec Patch by me + + * res/res_config_ldap.c: Use the default that the log output claims + will be used for the basedn (closes issue #12599) Reported by: + suretec Patches: 12599.patch uploaded by juggie (license 24) + + * channels/chan_h323.c: Let chan_h323 build in dev mode + + * include/asterisk/dlinkedlists.h (added): re-add dlinkedlists.h to + trunk, oops! + + * /, include/asterisk/dlinkedlists.h (removed), + channels/chan_iax2.c: Merged revisions 115512 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r115512 | russell | 2008-05-07 11:24:09 -0500 + (Wed, 07 May 2008) | 11 lines Merged revisions 115511 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) + | 3 lines Remove remnants of dlinkedlists. I didn't actually use + them in the final version of my IAX2 improvements. ........ + ................ + +2008-05-07 13:49 +0000 [r115509] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif: Update typos in description fields + (closes issue #12598) Reported by: suretec Patches: + asterisk_schema_changes.patch uploaded by suretec (license 70) + +2008-05-07 13:41 +0000 [r115507] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Remove redundant header getting. (closes + issue #12597) Reported by: hooi + +2008-05-06 20:15 +0000 [r115473] Mark Michelson <mmichelson@digium.com> + + * utils/refcounter.c: Get refcounter to build with LOW_MEMORY + defined + +2008-05-06 19:55 +0000 [r115419-115423] Jason Parker <jparker@digium.com> + + * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115422 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r115422 | qwell | 2008-05-06 14:55:29 -0500 + (Tue, 06 May 2008) | 15 lines Merged revisions 115421 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) | + 7 lines read requires an argument on some non-bash shells (closes + issue #12593) Reported by: bkruse Patches: + getilbc.sh_12593_v1.diff uploaded by bkruse (license 132) + ........ ................ + + * /, res/res_musiconhold.c: Merged revisions 115418 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May + 2008) | 7 lines Switch to using ast_random() rather than just + rand(). This does not fix the bug reported, but I believe it is + correct. (from issue #12446) Patches: bug_12446.diff uploaded by + snuffy (license 35) ........ + +2008-05-06 19:32 +0000 [r115416] Tilghman Lesher <tlesher@digium.com> + + * /, main/asterisk.c: Merged revisions 115415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115415 | tilghman | 2008-05-06 14:31:39 -0500 (Tue, 06 May 2008) + | 2 lines Don't print the terminating NUL. (Closes issue #12589) + ........ + +2008-05-06 15:14 +0000 [r115344] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Change some NOTICE log messages to debug. + +2008-05-06 13:55 +0000 [r115342] Joshua Colp <jcolp@digium.com> + + * /, configure, configure.ac: Merged revisions 115341 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r115341 | file | 2008-05-06 10:54:15 -0300 (Tue, 06 May + 2008) | 2 lines Add in missing argument. ........ + +2008-05-05 23:38 +0000 [r115334-115337] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c: Merge refcounting of res_odbc + + * /, main/logger.c, main/asterisk.c: Merged revisions 115333 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115333 | tilghman | 2008-05-05 17:50:31 -0500 (Mon, 05 May 2008) + | 7 lines Separate verbose output from CLI output, by using a + preamble. (closes issue #12402) Reported by: Corydon76 Patches: + 20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76 + (license 14) 20080501__no_verbose_in_rx_output__1.4.diff.txt + uploaded by Corydon76 (license 14) ........ + +2008-05-05 22:14 +0000 [r115329] Mark Michelson <mmichelson@digium.com> + + * main/config.c: #execing the same file multiple times led to + warning messages saying that the same file was being #included + twice. This was due to the fact that #exec created a temporary + file which was then #included. The name of the temporary file was + the name of the #exec'd file, with the Unix timestamp and thread + ID concatenated. The issue was that if multiple #exec statements + of the same file were reached in the same second, then the result + was that the temporary files would have duplicate names. To + resolve this, the temporary file now has microsecond resolution + for the timestamp portion. (closes issue #12574) Reported by: + jmls Patches: 12574.patch uploaded by putnopvut (license 60) + Tested by: jmls, putnopvut + +2008-05-05 22:13 +0000 [r115328] Joshua Colp <jcolp@digium.com> + + * funcs/func_speex.c, /, build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, codecs/codec_speex.c, + configure.ac: Merged revisions 115327 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2 + lines Make sure that either the main speex library contains + preprocess functions or that speexdsp does. If both fail then + speex stuff can not be built. ........ + +2008-05-05 22:01 +0000 [r115324] Russell Bryant <russell@digium.com> + + * main/event.c: Simplify code by using a taskprocessor for + dispatching events in the Asterisk core. + +2008-05-05 21:43 +0000 [r115321] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 115320 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May + 2008) | 13 lines Don't consider a caller "handled" until the + caller is bridged with a queue member. There was too much of an + opportunity for the member to hang up (either during a delay, + announcement, or overly long agi) between the time that he + answered the phone and the time when he actually was bridged with + the caller. The consequence of this was that if the member hung + up in that interval, then proper abandonment details would not be + noted in the queue log if the caller were to hang up at any point + after the member hangup. (closes issue #12561) Reported by: + ablackthorn ........ + +2008-05-05 20:28 +0000 [r115315] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c: Remove my rant, since I have now replaced + the rant with code. + +2008-05-05 20:22 +0000 [r115309-115313] Tilghman Lesher <tlesher@digium.com> + + * Makefile, /: Merged revisions 115312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115312 | tilghman | 2008-05-05 15:17:55 -0500 (Mon, 05 May 2008) + | 2 lines Reverse order, such that user configs override default + selections ........ + + * include/asterisk/res_odbc.h, /: Merged revisions 115308 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008) + | 2 lines Err, the documentation on the return value of + ast_odbc_backslash_is_escape is exactly backwards. ........ + +2008-05-05 19:50 +0000 [r115305] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 115304 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) + | 5 lines Avoid putting opaque="" in Digest authentication. This + patch came from switchvox. It fixes authentication with Primus in + Canada, and has been in use for a very long time without causing + problems with any other providers. (closes issue AST-36) ........ + +2008-05-05 19:42 +0000 [r115301-115302] Tilghman Lesher <tlesher@digium.com> + + * UPGRADE.txt: Note change for ExecIf syntax (caught by jmls on + IRC) + + * main/manager.c, CHANGES: Optionally display the value of several + variables within the Status command. (Closes issue AST-34) + +2008-05-05 13:52 +0000 [r115290] Joshua Colp <jcolp@digium.com> + + * apps/app_chanspy.c: Document the 'B' option of app_chanspy. + (closes issue #12582) Reported by: IgorG Patches: + app_chanspy_B_option.diff uploaded by IgorG (license 20) + +2008-05-05 10:55 +0000 [r115288] Kevin P. Fleming <kpfleming@digium.com> + + * UPGRADE.txt: clarify wording + +2008-05-05 03:25 +0000 [r115286] Tilghman Lesher <tlesher@digium.com> + + * contrib/init.d/rc.debian.asterisk, + contrib/init.d/rc.mandrake.asterisk, /, + contrib/init.d/rc.redhat.asterisk, + contrib/init.d/rc.gentoo.asterisk, + contrib/init.d/rc.slackware.asterisk, + contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk: + Merged revisions 115285 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115285 | tilghman | 2008-05-04 22:22:25 -0500 (Sun, 04 May 2008) + | 7 lines When starting Asterisk, bug out if Asterisk is already + running. (closes issue #12525) Reported by: explidous Patches: + 20080428__bug12525.diff.txt uploaded by Corydon76 (license 14) + Tested by: mvanbaak ........ + +2008-05-04 02:11 +0000 [r115277-115283] Joshua Colp <jcolp@digium.com> + + * /, configure, acinclude.m4: Merged revisions 115282 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r115282 | file | 2008-05-03 23:09:44 -0300 (Sat, 03 May + 2008) | 2 lines Expand the test function for GCC attributes so + that more complex attributes are properly recognized. ........ + + * /, include/asterisk/compiler.h: Merged revisions 115279 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2 + lines For my next trick I will make these work with what our + autoconf header file gives us. ........ + + * /, configure, acinclude.m4: Merged revisions 115276 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r115276 | file | 2008-05-03 22:43:26 -0300 (Sat, 03 May + 2008) | 2 lines Treat warnings as errors when checking if a GCC + attribute exists. We have to do this as GCC will just ignore the + attribute and pop up a warning, it won't actually fail to + compile. ........ + +2008-05-03 04:23 +0000 [r115268-115274] Dwayne M. Hubbard <dhubbard@digium.com> + + * apps/app_voicemail.c: app_voicemail uses a taskprocessor for mwi + notification subscriptions + + * main/pbx.c: pbx uses a taskprocessor for device state changes + + * apps/app_queue.c: app_queue uses a taskprocessor for device state + changes + + * include/asterisk/taskprocessor.h (added), main/Makefile, + main/taskprocessor.c (added), include/asterisk/_private.h, + main/asterisk.c: A taskprocessor is an object that has a name, a + task queue, and an event processing thread. Modules reference a + taskprocessor, push tasks into the taskprocessor as needed, and + unreference the taskprocessor when the taskprocessor is no longer + needed. A task wraps a callback function pointer and a data + pointer and is managed internal to the taskprocessor subsystem. + The callback function is responsible for releasing task data. + Taskprocessor API * ast_taskprocessor_get(..) - returns a + reference to a taskprocessor * ast_taskprocessor_unreference(..) + - releases reference to a taskprocessor * + ast_taskprocessor_push(..) - push a task into a taskprocessor + queue Check doxygen for more details + +2008-05-02 14:51 +0000 [r115197-115199] Mark Michelson <mmichelson@digium.com> + + * res/snmp/agent.c: Make res/snmp/agent.c build + + * /, include/asterisk/sched.h: Merged revisions 115196 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, + 02 May 2008) | 6 lines Clarify a comment that was, well, just + wrong. It turns out that ignoring the way that macros expand. + Instead, I have clarified in the comment why the macro will work + even if the scheduler id for the task to be deleted changes + during the execution of the macro. ........ + +2008-05-02 02:56 +0000 [r115104-115159] Tilghman Lesher <tlesher@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/config.h, include/asterisk/compiler.h: Okay, + maybe FreeBSD will like this better. + + * include/asterisk/logger.h, channels/chan_sip.c, + include/asterisk/config.h, include/asterisk/sched.h, + main/asterisk.c, main/config.c, main/sched.c, + apps/app_voicemail.c: Add attributes to various API calls, to + help track down bugs (and remove a deprecated function) + + * include/asterisk/res_odbc.h, /: Merged revisions 115102 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008) + | 2 lines Change the comment of deprecated to an actual compiler + deprecation ........ + +2008-05-01 23:09 +0000 [r115078] Brett Bryant <bbryant@digium.com> + + * channels/chan_zap.c, configure, configure.ac, CHANGES: Add two + new console commands "pri show version" and "ss7 show version" + that will show the version of each library respectively. + +2008-05-01 23:06 +0000 [r115076] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, main/pbx.c, apps/app_read.c, + funcs/func_timeout.c, apps/app_readexten.c, apps/app_disa.c, + include/asterisk/channel.h, apps/app_queue.c, CHANGES, + apps/app_speech_utils.c, main/cli.c, main/channel.c, main/dial.c, + main/manager.c, apps/app_dumpchan.c, res/res_agi.c, main/app.c, + include/asterisk/pbx.h, apps/app_rpt.c: Modify TIMEOUT() to be + accurate down to the millisecond. (closes issue #10540) Reported + by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by + Corydon76 (license 14) Tested by: blitzrage + +2008-05-01 19:05 +0000 [r115021] Russell Bryant <russell@digium.com> + + * doc/smdi.txt, res/res_smdi.c, CHANGES: Merge changes from + team/russell/smdi-msg-searching This commit adds some new + features to the SMDI_MSG_RETRIEVE() dialplan function. + Previously, this function only allowed searching by the + forwarding station. I have added some options to allow you to + also search for messages in the queue by the message desk + terminal ID, as well as the message desk number. This originally + came up as a suggestion on the asterisk-dev mailing list. + +2008-05-01 19:00 +0000 [r115018] Tilghman Lesher <tlesher@digium.com> + + * /, main/utils.c: Merged revisions 115017 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008) + | 6 lines '#' is another reserved character for URIs that also + needs to be escaped. (closes issue #10543) Reported by: blitzrage + Patches: 20080418__bug10543.diff.txt uploaded by Corydon76 + (license 14) ........ + +2008-05-01 18:28 +0000 [r114977] Brett Bryant <bbryant@digium.com> + + * funcs/func_speex.c: Add "read" capability to new libspeex + functions in func_speex.c. func_speex.c is based on contributions + from Switchvox. + +2008-05-01 17:28 +0000 [r114931] Russell Bryant <russell@digium.com> + + * UPGRADE.txt: Clarify the deprecation notice about Macro() to note + that it will not be removed for the sake of backwards + compatibility, since it is a non-trivial task to convert existing + large dialplans that depend on Macro() to use GoSub(), instead. + +2008-05-01 16:57 +0000 [r114926] Brett Bryant <bbryant@digium.com> + + * funcs/func_speex.c (added), include/asterisk/audiohook.h, + main/audiohook.c, CHANGES: Add two new dialplan functions from + libspeex for applying audio gain control and denoising to a + channel, AGC() and DENOISE(). Also included, is a change to the + audiohook API to add a new function (ast_audiohook_remove) that + can remove an audiohook from a channel before it is detached. + This code is based on a contribution from Switchvox. + +2008-05-01 16:49 +0000 [r114922] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c: Allow dringXrange to properly default to 10, + as was done in 1.4. dringXrange is a new feature that was added, + and it attempted to default, but only when the option was + specified. (closes issue #12536) Reported by: bjm Patches: + 12536-dringXrange.diff uploaded by qwell (license 4) Tested by: + bjm + +2008-04-30 20:51 +0000 [r114912] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add + support for specifying the registration expiry on a per + registration basis in the register line. This comes from a + Switchvox patch. (issue AST-24) + +2008-04-30 19:30 +0000 [r114906] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding new + configuration options to app_queue. This adds two new values to + announce-position, "limit" and "more," as well as a new option, + announce-position-limit. For more information on the use of these + options, see CHANGES or configs/queues.conf.sample. (closes issue + #10991) Reported by: slavon Patches: app_q.diff uploaded by + slavon (license 288) Tested by: slavon, putnopvut + +2008-04-30 19:21 +0000 [r114904] Tilghman Lesher <tlesher@digium.com> + + * apps/app_meetme.c, apps/app_minivm.c, apps/app_morsecode.c, + apps/app_macro.c, apps/app_externalivr.c, apps/app_chanspy.c, + apps/app_stack.c, apps/app_speech_utils.c, apps/app_voicemail.c, + apps/app_while.c: Lock around variables retrieved, and copy the + values, if they stay persistent, since another thread could + remove them. (closes issue #12541) Reported by: snuffy Patches: + bug_12156_apps.diff uploaded by snuffy (license 35) Several + additional changes by me + +2008-04-30 16:55 +0000 [r114899] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 114890 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 + lines Don't crash on bad SIP replys. Fix created in Huntsville + together with Mark M (putnopvut) (closes issue #12363) Reported + by: jvandal Tested by: putnopvut, oej ........ + +2008-04-30 16:34 +0000 [r114892] Russell Bryant <russell@digium.com> + + * /, channels/chan_console.c, channels/chan_iax2.c: Merged + revisions 114891 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) + | 28 lines Merge changes from team/russell/iax2_find_callno and + iax2_find_callno_1.4 These changes address a critical performance + issue introduced in the latest release. The fix for the latest + security issue included a change that made Asterisk randomly + choose call numbers to make them more difficult to guess by + attackers. However, due to some inefficient (this is by far, an + understatement) code, when Asterisk chose high call numbers, + chan_iax2 became unusable after just a small number of calls. On + a small embedded platform, it would not be able to handle a + single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run + more than about 16 IAX2 channels. Ouch. These changes address + some performance issues of the find_callno() function that have + bothered me for a very long time. On every incoming media frame, + it iterated through every possible call number trying to find a + matching active call. This involved a mutex lock and unlock for + each call number checked. So, if the random call number chosen + was 20000, then every media frame would cause 20000 locks and + unlocks. Previously, this problem was not as obvious since + Asterisk always chose the lowest call number it could. A second + container for IAX2 pvt structs has been added. It is an astobj2 + hash table. When we know the remote side's call number, the pvt + goes into the hash table with a hash value of the remote side's + call number. Then, lookups for incoming media frames are a very + fast hash lookup instead of an absolutely insane array traversal. + In a quick test, I was able to get more than 3600% more IAX2 + channels on my machine with these changes. ........ + +2008-04-30 16:14 +0000 [r114888] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_console.c: Fixes a bug where if a stream monitor + thread was not created (caused from failure of opening or + starting the stream) pthread_cancel was called with an invalid + thread ID. + +2008-04-30 14:49 +0000 [r114876-114884] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/iax2.h, channels/chan_iax2.c: Merged revisions 114880 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr + 2008) | 2 lines use the ARRAY_LEN macro for indexing through the + iaxs/iaxsl arrays so that the size of the arrays can be adjusted + in one place, and change the size of the arrays from 32768 calls + to 2048 calls when LOW_MEMORY is defined ........ + + * /, Makefile.rules: Merged revisions 114875 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr + 2008) | 2 lines pay attention to *all* header files for + dependency tracking, not just the local ones (inspired by r578 of + asterisk-addons by tilghman) ........ + +2008-04-30 05:05 +0000 [r114874] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Document the Incomplete application addition. + +2008-04-29 22:54 +0000 [r114866] Jeff Peeler <jpeeler@digium.com> + + * channels/iax2-provision.c: Fixes a problem where all the + templates were marked as dead no matter what. The templates + should only be marked as dead if a configuration file has been + successfully loaded and has changes. Bug found while making API + documentation for 1.6.0. + +2008-04-29 21:07 +0000 [r114857] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c: Patching app_chanspy to jibe better with what + is documented. This allows for a colon-delimited list of + spygroups to be specified when calling the ChanSpy application + with the 'g' option. Prior to this, you could only specify a + single group when using the 'g' option. I also have upped the + maximum number of spygroups to 128 and added a #define so that + this can be easily increased or decreased later. (closes issue + #12497) Reported by: jsmith Patches: + app_chanspy_multiple_groups_v2.patch uploaded by jsmith (license + 15) Tested by: atis, jvandal + +2008-04-29 20:05 +0000 [r114852] Jason Parker <jparker@digium.com> + + * phoneprov/polycom.xml: Fix formatting + +2008-04-29 19:42 +0000 [r114849] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 114848 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr + 2008) | 14 lines Use the MACRO_CONTEXT and MACRO_EXTEN channel + variables instead of the channel's macrocontext and macroexten + fields. This is needed because if macros are daisy-chained, the + incorrect context and extension are placed on the new channel. I + also added locking to the channel prior to accessing these + variables as noted in trunk's janitor project file. (closes issue + #12549) Reported by: darren1713 Patches: + app_queue.c.macroextenpatch uploaded by darren1713 (license 116) + (with modifications from me) Tested by: putnopvut ........ + +2008-04-29 18:58 +0000 [r114845] Kevin P. Fleming <kpfleming@digium.com> + + * main/features.c: fix this logic to actually be correct... the fd + can't be *both* -1 and an array index to be checked in rfds/efds + (bug found by gcc-4.3) + +2008-04-29 18:48 +0000 [r114832-114841] Mark Michelson <mmichelson@digium.com> + + * UPGRADE.txt, apps/app_directory.c: Make app_directory dependent + on app_voicemail. This is because the function which says the + person's name is handled inside app_voicemail now. + + * apps/app_directory.c, apps/app_voicemail.c: Since there is now a + globally available function for saying someone's name, a LOT of + functions in app_directory can be removed since the ODBC-specific + lookups are accomplished within app_voicemail. This change + greatly reduces the amount of lines in app_directory that were + solely for the purpose of looking up a name when ODBC_STORAGE is + specified for voicemail. This commit also makes the name-saying + interruptable via DTMF. + + * apps/app_directory.c: Fix a crash happening in app_directory. + This crash would occur if a users.conf existed. + +2008-04-29 17:10 +0000 [r114830] Jason Parker <jparker@digium.com> + + * res/res_config_pgsql.c, /: Merged revisions 114829 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr + 2008) | 1 line Change warning message to debug, since there are + cases where 0 results is perfectly fine. ........ + +2008-04-29 12:54 +0000 [r114824] Kevin P. Fleming <kpfleming@digium.com> + + * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114823 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r114823 | kpfleming | 2008-04-29 07:53:12 -0500 + (Tue, 29 Apr 2008) | 10 lines Merged revisions 114822 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr + 2008) | 2 lines stop script from appending source code if run + multiple times ........ ................ + +2008-04-28 22:38 +0000 [r114813] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/app.h, main/app.c, CHANGES, apps/app_chanspy.c, + apps/app_voicemail.c: Adding a new option 'n' to app_chanspy. + This option allows for the name of the spied-on party to be + spoken instead of the channel name or number. This was + accomplished by adding a new function pointer to point to a + function in app_voicemail which retrieves the name file and plays + it. This makes for an easy way that applications may play a + user's name should it be necessary. app_directory, in particular, + can be simplified greatly by this change. This change comes as a + suggestion from Switchvox, which already has this feature. AST-23 + +2008-04-28 17:00 +0000 [r114776] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Fix deadlock issue in chan_zap with libss7 + due to channel variables being set with the channel pvt lock + being held. #12512 + +2008-04-28 16:37 +0000 [r114773] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: Add + incomplete matching to PBX code and app_dial (closes issue + #12351) Reported by: Corydon76 Patches: + 20080402__pbx_incomplete__3.diff.txt uploaded by Corydon76 + (license 14) pbx_incomplete_with_timeout.diff uploaded by fabled + (license 448) Tested by: Corydon76, fabled + +2008-04-28 13:42 +0000 [r114713] Joshua Colp <jcolp@digium.com> + + * configure, configure.ac: Update autoconf logic with latest API + change for libss7. + +2008-04-28 04:53 +0000 [r114706-114709] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged + revisions 114708 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) + | 5 lines When modules are embedded, they take on a different + name, without the ".so" extension. Specifically check for this + name, when we're checking if a module is loaded. (Closes issue + #12534) ........ + + * apps/app_voicemail.c: Fix breakage caused by #12028. (Closes + issue #12535) + +2008-04-27 22:54 +0000 [r114703] Russell Bryant <russell@digium.com> + + * channels/chan_skinny.c: s/chan_zap/chan_skinny/ + +2008-04-27 15:17 +0000 [r114700] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: Make MWI in chan_skinny event based + modeled after chan_zap and chan_mgcp. (closes issue #12214) + Reported by: DEA Patches: chan_skinny-vm-events-v3.txt uploaded + by DEA (license 3) Tested by: DEA and me + +2008-04-27 01:28 +0000 [r114696] Sean Bright <sean.bright@gmail.com> + + * /, configure, configure.ac: Merged revisions 114695 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, + 26 Apr 2008) | 5 lines When we don't explicitly pass a path to + the --with-tds configure option, we may end up finding tds.h in + /usr/local/include instead of /usr/include. If this happens, the + grep that looks for the version (from tdsver.h) will fail and + we'll have some problems during the build. ........ + +2008-04-26 15:08 +0000 [r114683-114692] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Unleak reference + + * /, contrib/scripts/vmail.cgi: Merged revisions 114689 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008) + | 6 lines Clicking forward without selecting a message leaves an + errant .lock file. (closes issue #12528) Reported by: pukepail + Patches: patch.diff uploaded by pukepail (license 431) ........ + + * channels/chan_sip.c: Add 'sip qualify peer <peer>' command (with + AMI SIPqualifypeer) (closes issue #12524) Reported by: ctooley + Patches: sip_qualify_peer.diff.2 uploaded by ctooley (license + 136) some modifications for trunk by Corydon76 Tested by: + Corydon76 + +2008-04-25 22:24 +0000 [r114678] Mark Michelson <mmichelson@digium.com> + + * CHANGES, apps/app_chanspy.c: Adding a new option, 'B' to + app_chanspy. This option allows the spy to barge on the call. It + is like the existing whisper option, except that it allows the + spy to talk to both sides of the conversation on which he is + spying. This feature has existed in Switchvox, and this merges + the functionality into Asterisk. (AST-32) + +2008-04-25 22:04 +0000 [r114674-114676] Russell Bryant <russell@digium.com> + + * pbx/pbx_lua.c: Lock the channel around datastore access (closes + issue #12527) Reported by: mnicholson Patches: pbx_lua4.diff + uploaded by mnicholson (license 96) + + * /, channels/chan_iax2.c: Merged revisions 114673 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 + Apr 2008) | 3 lines Use consistent logic for checking to see if a + call number has been chosen yet. Also, remove some redundant + logic I recently added in a fix. ........ + +2008-04-25 20:20 +0000 [r114665-114667] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, apps/app_waitforring.c, apps/app_minivm.c, + apps/app_zapscan.c, apps/app_sms.c, apps/app_externalivr.c, + apps/app_followme.c, apps/app_queue.c, apps/app_rpt.c, + apps/app_playback.c, apps/app_parkandannounce.c, + apps/app_speech_utils.c: Whitespace changes only + + * main/app.c: Oops, this isn't necessarily AGI that is forking + anymore + +2008-04-25 19:33 +0000 [r114663] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_chanspy.c: Merged revisions 114662 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr + 2008) | 4 lines Move the unlock of the spyee channel to outside + the start_spying() function so that the channel is not unlocked + twice when using whisper mode. ........ + +2008-04-25 18:32 +0000 [r114660] Jason Parker <jparker@digium.com> + + * apps/app_directed_pickup.c, apps/app_pickupchan.c (removed): + Merge app_pickupchan with app_directed_pickup, for AST-27. + Initially, this was to be a new feature, with a patch from + Switchvox, but after discussions, it was noted that this feature + already existed in trunk. The resulting discussions ended in a + comment that was along the lines of "the patch provided here is a + lot smaller than what is already in trunk, because it doesn't + create a new application and duplicate existing code" It was + decided that these two applications could be easily merged to + reduce code duplication. SO, that's what this does. + +2008-04-25 18:18 +0000 [r114656] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: This patch allows for forwarding a message + with a "comment" attachment if using IMAP storage for voicemail. + The comment will be recorded and attached as a second attachment + in addition to the original message. This will be invoked if you + choose to prepend a message the way you would with file or ODBC + storage (closes issue #12028) Reported by: jaroth Patches: + forward_with_comment_v2.patch uploaded by jaroth (license 50) + Tested by: jaroth + +2008-04-25 18:18 +0000 [r114655] Russell Bryant <russell@digium.com> + + * main/features.c: Merge code from team/russell/parking_updates Add + some additional features to the core park_call_full() function, + and expose them as options to the Park() application. The + functionality being added is the ability to specify a custom + return extension/context/priority, a custom timeout, and a couple + of options. The options are to play ringing instead of MOH to the + parked caller, and to randomize parking spot selection. (code + inspired by the patch in AST-17, code from switchvox) + +2008-04-25 16:25 +0000 [r114651] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Fix a memory leak and protect against + potential dereferences of a NULL pointer. + +2008-04-25 13:56 +0000 [r114644] Sean Bright <sean.bright@gmail.com> + + * channels/chan_console.c: Speaking of building... + +2008-04-24 22:16 +0000 [r114637] Michiel van Baak <michiel@vanbaak.info> + + * apps/app_dial.c, channels/chan_unistim.c, channels/chan_local.c, + channels/chan_zap.c, channels/chan_sip.c, apps/app_disa.c, + apps/app_alarmreceiver.c, include/asterisk/channel.h, + channels/chan_gtalk.c, apps/app_followme.c, apps/app_queue.c, + channels/chan_iax2.c, channels/chan_oss.c, main/channel.c, + channels/chan_jingle.c, channels/chan_misdn.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_alsa.c, apps/app_externalivr.c, + channels/chan_mgcp.c: Pass the hangup cause all the way to the + calling app/channel. (closes issue #11328) Reported by: rain + Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt + uploaded by Corydon76 (license 14) brought up-to-date to trunk by + me + +2008-04-24 22:11 +0000 [r114635] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Hey look, it builds. (closes issue #12519) + Reported by: falves11 + +2008-04-24 21:35 +0000 [r114625-114633] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 114632 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr + 2008) | 11 lines Re-invite RTP during a masquerade so that, for + instance, an AMI redirect of two channels which are natively + bridged will preserve audio on both channels. This prevents a + problem with Asterisk not re-inviting due to one of the channels + having being a zombie. (closes issue #12513) Reported by: + mneuhauser Patches: + asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by + mneuhauser (license 425) ........ + + * /, apps/app_queue.c: Merged revisions 114628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr + 2008) | 8 lines Output of channel variables when + eventwhencalled=vars was set was being truncated two characters. + This patch corrects the problem. (closes issue #12493) Reported + by: davidw ........ + + * channels/chan_local.c, /: Merged revisions 114624 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, + 24 Apr 2008) | 10 lines Resolve a deadlock in chan_local by + releasing the channel lock temporarily. (closes issue #11712) + Reported by: callguy Patches: 11712.patch uploaded by putnopvut + (license 60) Tested by: acunningham ........ + +2008-04-24 19:54 +0000 [r114617-114622] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /: Merged revisions 114621 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 + Apr 2008) | 4 lines Ensure that when we set the accountcode, it + actually shows up in the CDR. (Fix for AMI Originate) (Closes + issue #12007) ........ + + * apps/app_meetme.c: Fix DST calculation, and fix bug in + calculation of whether conf has started yet or not (Closes issue + #12292) Reported by: DEA Patches: app_meetme-rt-dst-sched-fix.txt + uploaded by DEA (license 3) + +2008-04-24 16:47 +0000 [r114612] Jason Parker <jparker@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 51989 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #12496) Reported by: daniele Patches: + misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471) + Tested by: daniele Technically, I didn't use the patch above + except to find out what revision to merge - but it's the same + thing as this revision. ........ r51989 | crichter | 2007-01-24 + 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line added fix from #8899 + ........ + +2008-04-24 15:56 +0000 [r114609] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 114608 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 + Apr 2008) | 4 lines Fix a silly mistake in a change I made + yesterday that caused chan_iax2 to blow up very quickly. (issue + #12515) ........ + +2008-04-24 14:59 +0000 [r114606] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 114603 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 + lines Only have one max-forwards header in outbound REFERs. + Discovered in the Asterisk SIP Masterclass in Orlando. Thanks + Joe! ........ + +2008-04-24 14:55 +0000 [r114598-114604] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Change a verbose message to debug. (closes + issue #12514) + + * /, main/http.c: Merged revisions 114600 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008) + | 6 lines Improve some broken cookie parsing code. Previously, + manager login over HTTP would only work if the mansession_id + cookie was first. Now, the code builds a list of all of the + cookies in the Cookie header. This fixes a problem observed by + users of the Asterisk GUI. (closes AST-20) ........ + + * /, apps/app_chanspy.c: Merged revisions 114597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008) + | 10 lines Fix an issue that caused getting the correct next + channel to not always work. Also, remove setting the amount of + time to wait for a digit from 5 seconds back down to 1/10 of a + second. I believe this was so the beep didn't get played over and + over really fast, but a while back I put in another fix for that + issue. (closes issue #12498) Reported by: jsmith Patches: + app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license + 15) ........ + +2008-04-23 18:33 +0000 [r114595] Jason Parker <jparker@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 114594 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114594 | qwell | 2008-04-23 13:28:44 -0500 (Wed, 23 Apr + 2008) | 8 lines Fix reload/unload for res_musiconhold module. + (closes issue #11575) Reported by: sunder Patches: + M11575_14_rev3.diff uploaded by junky (license 177) + bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176) + ........ + +2008-04-23 18:01 +0000 [r114588-114592] Russell Bryant <russell@digium.com> + + * main/manager.c, /, include/asterisk/manager.h: Merged revisions + 114591 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008) + | 5 lines Store the manager session ID explicitly as 4 byte ID + instead of a ulong. The mansession_id cookie is coded to be + limited to 8 characters of hex, and this could break logins from + 64-bit machines in some cases. (inspired by AST-20) ........ + + * /, channels/chan_iax2.c: Merged revisions 114587 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 + Apr 2008) | 2 lines Fix find_callno_locked() to actually return + the callno locked in some more cases. ........ + +2008-04-23 16:53 +0000 [r114585] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 114584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 + lines Add 502 support for both directions, not only one... (see + r114571) ........ + +2008-04-23 14:55 +0000 [r114580] Joshua Colp <jcolp@digium.com> + + * main/pbx.c, /: Merged revisions 114579 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4 + lines Instead of stopping dialplan execution when SayNumber + attempts to say a large number that it can not print out a + message informing the user and continue on. (closes issue #12502) + Reported by: bcnit ........ + +2008-04-23 00:58 +0000 [r114575-114577] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/logger.h, include/asterisk/astobj.h, + apps/app_voicemail.c: Round 2 of IMAP_STORAGE app_voicemail.c + fixes: This fixes a bug that was thought to be fixed already. + app_voicemail, if using IMAP_STORAGE, has a problem because the + IMAP header files include syslog.h, which define LOG_WARNING and + LOG_DEBUG to be different than what Asterisk uses for those same + macros. This was "fixed" in the past by including all the IMAP + header files prior to including asterisk.h. This fix worked... + unless you were to try to compile with MALLOC_DEBUG. MALLOC_DEBUG + prepends the inclusion of astmm.h to every file, which means that + no matter what order the includes are in in app_voicemail, the + unexpected values for LOG_WARNING and LOG_DEBUG will be in place. + The action taken for this fix was to define AST_LOG_* macros in + addition to the LOG_* macros already defined. These new macros + are used in app_voicemail.c, logger.h, and astobj.h right now, + and their use will be encouraged in the future. In consideration + of those who have written third-party modules which use the LOG_* + macros, these will NOT be removed from the source, however future + use of these macros is discouraged. + + * apps/app_voicemail.c: Round 1 of IMAP_STORAGE-related + app_voicemail changes This makes IMAP_STORAGE include the proper + headers if you have specified the "system" option for --with-imap + when running the configure script and your IMAP-related headers + exist in /usr/include/c-client. This change is due to a hasty + merge of a 1.4 change I made. + +2008-04-22 23:58 +0000 [r114572] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 114571 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) + | 2 lines Treat a 502 just like a 503, when it comes to + processing a response code ........ + +2008-04-22 22:17 +0000 [r114559] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 114558 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 + Apr 2008) | 5 lines When we receive a full frame that is supposed + to contain our call number, ensure that it has the correct one. + (closes issue #10078) (AST-2008-006) ........ + +2008-04-22 21:57 +0000 [r114553] Steve Murphy <murf@digium.com> + + * main/pbx.c: (closes issue #12469) Reported by: triccyx I had a + bit a problem reproducing this in my setup (trying not to disturb + my other stuff) but finally, I got it. The problem appears to be + that the extension is being added in replace mode, which kinda + assumes that the pattern trie has been formed, when in fact, in + this case, it was not. The checks being done are not nec. when + the tree is not yet formed, as changes like this will be + summarized when the trie is formed in the future. I tested the + fix, and the crash no longer happens. Feel free to open the bug + again if this fix doesn't cure the problem. + +2008-04-22 20:25 +0000 [r114548] Russell Bryant <russell@digium.com> + + * main/channel.c: re-add a fix that got lost with a recent change + +2008-04-22 18:14 +0000 [r114540] Jason Parker <jparker@digium.com> + + * main/pbx.c, include/asterisk/pbx.h, apps/app_queue.c: Allow + setqueuevar=yes (et al) to work, after changes to + pbx_builtin_setvar() (closes issue #12490) Reported by: bcnit + Patches: 12490-queuevars-3.diff uploaded by qwell (license 4) + Tested by: qwell + +2008-04-22 18:04 +0000 [r114533-114538] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 114537 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 + Apr 2008) | 9 lines If the dial string passed to the call channel + callback does not indicate an extension, then consider the + extension on the channel before falling back to the default. + (closes issue #12479) Reported by: darren1713 Patches: + exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license + 116) ........ + + * CHANGES, apps/app_jack.c: Add a c() option for the Jack() + application and JACK_HOOK() funciton for supplying a custom + client name. Using the channel name is still the default. This + was done at the request of Jared Smith. + +2008-04-22 15:54 +0000 [r114529] Joshua Colp <jcolp@digium.com> + + * configs/sip_notify.conf.sample, channels/chan_sip.c: Add support + for authenticating on a NOTIFY request. This is useful for phones + that require it when sending them a special packet to get them to + do something (such as reload their configuration). (closes issue + #9896) Reported by: IgorG Patches: sipnotify-113980-v14.patch + uploaded by IgorG (license 20) + +2008-04-22 15:46 +0000 [r114527] Russell Bryant <russell@digium.com> + + * main/manager.c: Correct action_ping() and action_events() with + regards to Manager 1.1 documentation. Also, fix a bug in + xml_translate(). (closes issue #11649) Reported by: ys Patches: + trunk_manager.c.diff uploaded by ys (license 281) + +2008-04-22 14:38 +0000 [r114520] Steve Murphy <murf@digium.com> + + * channels/chan_sip.c, main/utils.c: Hopefully, this will resolve + the issues that russellb had with this log_show_lock(). I + gathered the code that filled the string, and put it in a + different func which I cryptically call + "append_lock_information()". Now, both log_show_lock(), and + handle_show_locks() both call this code to do the work. Tested, + seems to work fine. Also, log_show_lock was modified to use the + ast_str stuff, along with checking for successful ast_str + creation, and freeing the ast_str obj when finished. A break was + inserted to terminate the search for the lock; we should never + see it twice. An example usage in chan_sip.c was created as a + comment, for instructional purposes. + +2008-04-21 23:42 +0000 [r114487] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_unistim.c, channels/chan_zap.c, + channels/chan_sip.c, include/asterisk/channel.h, + channels/chan_gtalk.c, channels/chan_console.c, + channels/chan_iax2.c, configs/features.conf.sample, + configs/iax.conf.sample, channels/chan_jingle.c, + channels/chan_skinny.c, funcs/func_channel.c, main/features.c, + apps/app_dumpchan.c, configs/sip.conf.sample, + channels/chan_mgcp.c: (closes issue #6113) Reported by: oej + Tested by: jpeeler This patch implements multiple parking lots + for parked calls. The default parkinglot is used by default, + however setting the channel variable PARKINGLOT in the dialplan + will allow use of any other configured parkinglot. See + configs/features.conf.sample for more details on setting up + another non-default parkinglot. Also, one can (currently) set the + default parkinglot to use in the driver configuration file via + the parkinglot option. Patch initially written by oej, brought up + to date and finalized by mvanbaak, and then stabilized and + converted to astobj2 by me. + +2008-04-21 22:50 +0000 [r114456] Doug Bailey <dbailey@digium.com> + + * phoneprov/polycom.xml: Change the time zone offset from a hard + code to use res_phoneprov variables + +2008-04-21 21:13 +0000 [r114423] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ref.ael-vtest17, main/ast_expr2.y, + doc/tex/channelvariables.tex, doc/tex/ael.tex, CHANGES, + pbx/ael/ael-test/ael-ntest24/extensions.ael (added), + pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ael-ntest24 + (added), pbx/ael/ael-test/ref.ael-test19, main/ast_expr2.c, + pbx/ael/ael-test/ref.ael-ntest10, main/ast_expr2.h, + pbx/ael/ael-test/ref.ael-test1, main/ast_expr2f.c, + pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-ntest24 + (added), pbx/ael/ael-test/ref.ael-test5, + pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-vtest13, + main/ast_expr2.fl: (closes issue #12467) Reported by: atis Tested + by: murf This upgrade adds the ~~ (concatenation) string operator + to expr2. While not needed in normal runtime pbx operation, it is + needed when raw exprs are being syntax checked. This plays into + future syntax- unification plans. By permission of atis, this + addition in trunk and the reason of why things are as they are + will suffice to close this bug. I also added a short note about + the previous addition of "sip show sched" to the CLI in CHANGES, + which I discovered I forgot in a previous commit. + +2008-04-21 18:44 +0000 [r114389] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Add support for generic name transmission + (#12484) on SS7 in chan_zap + +2008-04-21 15:34 +0000 [r114327] Jeff Peeler <jpeeler@digium.com> + + * apps/app_authenticate.c: This removes an invalid warning message + for an incorrectly entered pin, but more importantly removes an + inapplicable check. If the first argument passed to + app_authenticate does not contain a '/', the argument should be + treated as the sole fixed "password" to match against and that is + all. (Previous behavior was attempting to open a file based on + the pin.) + +2008-04-21 15:01 +0000 [r114325] Russell Bryant <russell@digium.com> + + * doc/janitor-projects.txt: Add a simple janitor project + +2008-04-21 14:40 +0000 [r114320-114323] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 114322 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 + lines Only drop audio if we receive it without a progress + indication. We allow other frames through such as DTMF because + they may be needed to complete the call. (closes issue #12440) + Reported by: aragon ........ + + * res/res_config_ldap.c: Only print out the error message if + ldap_modify_ext_s actually returns an error, and not success. + (closes issue #12438) Reported by: gservat Patches: + res_config_ldap.c-patch-code uploaded by gservat (license 466) + +2008-04-20 14:52 +0000 [r114314] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_pgsql.c: Minor logging cleanups + +2008-04-19 16:58 +0000 [r114303] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: SS7:Added - Generic Name / Access Transport + / Redirecting Number handling. #12425 + +2008-04-19 00:15 +0000 [r114295] Sean Bright <sean.bright@gmail.com> + + * utils: Ignore refcounter + +2008-04-18 21:51 +0000 [r114276-114285] Russell Bryant <russell@digium.com> + + * main/manager.c, /: Merged revisions 114284 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114284 | russell | 2008-04-18 16:48:06 -0500 (Fri, 18 Apr 2008) + | 2 lines Don't destroy a manager session if poll() returns an + error of EAGAIN. ........ + + * Makefile, /: Merged revisions 114278 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114278 | russell | 2008-04-18 15:01:09 -0500 (Fri, 18 Apr 2008) + | 2 lines ensure directories are created before we try to install + stuff into them ........ + + * Makefile, /: Merged revisions 114275 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114275 | russell | 2008-04-18 14:58:55 -0500 (Fri, 18 Apr 2008) + | 2 lines SUBDIRS_INSTALL is already listed as a subtarget for + bininstall ........ + +2008-04-18 19:35 +0000 [r114261-114271] Joshua Colp <jcolp@digium.com> + + * channels/chan_unistim.c: Make sure ADSI is marked as unavailable + on Unistim channels so voicemail does not try to do some ADSI + jazz. (closes issue #12460) Reported by: PerryB + + * apps/app_meetme.c, CHANGES: Add MEETME_INFO dialplan function + that allows querying various properties of a Meetme conference. + (closes issue #11691) Reported by: junky Patches: + meetme_info.patch uploaded by jpeeler (license 325) + +2008-04-18 18:03 +0000 [r114259] Mark Michelson <mmichelson@digium.com> + + * channels/chan_zap.c, /, main/callerid.c: Merged revisions 114257 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr + 2008) | 6 lines Clearing up error messages so they make a bit + more sense. Also removing a redundant error message. Issue AST-15 + ........ + +2008-04-18 16:11 +0000 [r114254] Joshua Colp <jcolp@digium.com> + + * res/res_config_ldap.c: If the parsing of the config file fails + make sure we unlock ldap_lock. (closes issue #12477) Reported by: + IgorG + +2008-04-18 16:05 +0000 [r114253] Doug Bailey <dbailey@digium.com> + + * res/res_http_post.c: Add g__object_unref to clean up gmime + message object + +2008-04-18 13:38 +0000 [r114246] Sean Bright <sean.bright@gmail.com> + + * channels/chan_sip.c: Merged revisions 114245 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr + 2008) | 1 line Only complete the SIP channel name once for 'sip + show channel <channel>' ........ + +2008-04-18 06:53 +0000 [r114243] Tilghman Lesher <tlesher@digium.com> + + * apps/app_setcallerid.c, /: Merged revisions 114242 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114242 | tilghman | 2008-04-18 01:49:16 -0500 (Fri, 18 + Apr 2008) | 3 lines For consistency sake, ensure that the values + that ${CALLINGPRES} returns are valid as an input to + SetCallingPres. (Closes issue #12472) ........ + +2008-04-17 22:24 +0000 [r114231-114233] Russell Bryant <russell@digium.com> + + * main/autoservice.c, /: Merged revisions 114230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114230 | russell | 2008-04-17 17:15:43 -0500 (Thu, 17 Apr 2008) + | 6 lines Remove redundant safety net. The check for the + autoservice channel list state accomplishes the same goal in a + better way. (issue #12470) Reported By: atis ........ + + * main/utils.c: Make this file compile. The variable str is never + set anywhere. Furthermore, it duplicates a lot of code. I will + leave it to murf to clean up. + +2008-04-17 21:09 +0000 [r114229] Jeff Peeler <jpeeler@digium.com> + + * CHANGES: added info describing DNS manager + +2008-04-17 21:04 +0000 [r114208-114227] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_chanspy.c: Merged revisions 114226 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr + 2008) | 9 lines Declaration of the peer channel in this scope was + making it so the peer variable defined in the outer scope was + never set properly, therefore making iterating through the + channel list always restart from the beginning. This bug would + have affected anyone who called chanspy without specifying a + first argument. (closes issue #12461) Reported by: stever28 + ........ + + * main/frame.c, /, include/asterisk/dsp.h, + include/asterisk/frame.h, main/dsp.c: Merged revisions 114207 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr + 2008) | 12 lines It was possible for a reference to a frame which + was part of a freed DSP to still be referenced, leading to memory + corruption and eventual crashes. This code change ensures that + the dsp is freed when we are finished with the frame. This change + is very similar to a change Russell made with translators back a + month or so ago. (closes issue #11999) Reported by: destiny6628 + Patches: 11999.patch uploaded by putnopvut (license 60) Tested + by: destiny6628, victoryure ........ + +2008-04-17 16:25 +0000 [r114205] Russell Bryant <russell@digium.com> + + * Makefile, /: Merged revisions 114204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114204 | russell | 2008-04-17 11:23:45 -0500 (Thu, 17 Apr 2008) + | 3 lines Fix the bininstall target to install from subdirs, as + well. (closes issue AST-8, patch from bmd at switchvox) ........ + +2008-04-17 15:12 +0000 [r114202] Tilghman Lesher <tlesher@digium.com> + + * doc/CODING-GUIDELINES: fileio.h does not exist; io.h does, + though. + +2008-04-17 14:45 +0000 [r114201] Steve Murphy <murf@digium.com> + + * channels/chan_sip.c: Thanks to snuff for finding these omissions + +2008-04-17 13:46 +0000 [r114199] Philippe Sultan <philippe.sultan@gmail.com> + + * /, res/res_jabber.c: Merged revisions 114198 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008) + | 2 lines Use keepalives effectively in order diagnose bug + #12432. ........ + +2008-04-17 12:59 +0000 [r114196] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 114195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008) + | 8 lines Add special case for when the agi cannot be executed, + to comply with the documentation that we return failure in that + case. (closes issue #12462) Reported by: fmueller Patches: + 20080416__bug12462.diff.txt uploaded by Corydon76 (license 14) + Tested by: fmueller ........ + +2008-04-17 12:25 +0000 [r114192-114194] Sean Bright <sean.bright@gmail.com> + + * CHANGES: Update the CHANGES file with yesterday's ChanSpy change. + Sorry Kevin, just saw your e-mail. + + * /, apps/app_chanspy.c: Merged revisions 114191 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114191 | seanbright | 2008-04-17 06:51:20 -0400 (Thu, 17 Apr + 2008) | 1 line Make sure we have enough room for the recording's + filename. ........ + +2008-04-16 23:53 +0000 [r114190] Steve Murphy <murf@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + doc/chan_sip-perf-testing.txt (added): This is the scariest + commit I've done in a long time. This is the astobj2-ification of + chan_sip. I've tested a number of scenarios like crazy. It used + to have 4x the call setup/teardown performance of trunk, but now + it's roughly at parity. I will attempt to find the bottlenecks + and get it back to the 4x mark. The changes made were somewhat + invasive, but the value to the community of these upgrades + outweighs waiting further for more testing. Every change being + made to chan_sip was lousing this code up when we tried to merge. + Peers, Users, Dialogs, are all now astobj2 objects, indexed via + hashtables. Refcounting is used to track objects and free them at + the bitter end of their lives. Please file issues on + bugs.digium.com, and PLEASE, please, please be patient. One + natural advantage to all the hash-table work is that loading + large sip.conf files full of thousands of peers now goes much + faster. One more please: PLEASE help thrash this code and test + it. + +2008-04-16 22:57 +0000 [r114188] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/logger.h, apps/app_nbscat.c, + include/asterisk/app.h, apps/app_festival.c, apps/app_mp3.c, + res/res_agi.c, apps/app_zapras.c, main/logger.c, main/app.c, + apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c, + main/asterisk.c: Standardized routines for forking processes + (keeps all the specialized code in one place). + +2008-04-16 20:54 +0000 [r114187] Steve Murphy <murf@digium.com> + + * main/utils.c, include/asterisk/lock.h: A small enhancement-- I + added the routine log_show_lock to utils.c, which if the + mentioned lock has been acquired, this routine will log to the + console the normal info about that lock you'd see from the CLI + when you do a 'core show locks'. It's solely for debug-- if the + lock is NOT acquired, there is no output. I use it to show + 'unexpected' locks, to see where/why a lock is pre-locked. This + command is to be called from points of interest, like just before + a trylock, and helps to spot fleeting, highly temporal locks that + normally are not locked... + +2008-04-16 20:47 +0000 [r114185] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, /: Merged revisions 114184 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr + 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by + initializing the structure to all zeroes in case it contains + fields that we don't write values into (which it does as of + Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian + ........ + +2008-04-16 20:28 +0000 [r114172-114183] Steve Murphy <murf@digium.com> + + * main/event.c: Introducing a small optimization to + event_unsubscribe; events now use a Doubly-Linked list for + events, gives fast deletions, for the sake of channel driver mwi + events. From team/murf/bug11210. + + * include/asterisk/sched.h, CHANGES, main/sched.c: Introducing a + small upgrade to the ast_sched_xxx facility, to keep it from + eating up lots of cpu cycles. See CHANGES. From the + team/murf/bug11210 branch. + + * utils/Makefile, utils/refcounter.c (added), + include/asterisk/astobj2.h, CHANGES, main/astobj2.c: Introducing + various astobj2 enhancements, chief being a refcount tracing + feature, and various documentation updates in astobj2.h, and the + addition of standalone utility, refcounter, that will filter the + trace output for unbalanced, unfreed objects. This comes from the + team/murf/bug11210 branch. + + * tests/test_dlinklists.c (added), include/asterisk/dlinkedlists.h + (added), CHANGES: Introducing doubly linked lists to trunk from + branch team/murf/bug11210. + +2008-04-16 12:23 +0000 [r114165] Sean Bright <sean.bright@gmail.com> + + * apps/app_chanspy.c: Add the ability to disable channel technology + name playback when speaking the current channel name + +2008-04-15 20:51 +0000 [r114152] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_pgsql.c: Oops, buffer wasn't long enough for query + +2008-04-15 20:39 +0000 [r114150-114151] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 114148 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 + lines Handle subscribe queues in all situations... Thanks to + festr_ on irc for telling me about this bug. ........ + + * channels/chan_sip.c: Adding chanvar to SIPPEER from 1.4 branch + +2008-04-15 20:27 +0000 [r114149] Jason Parker <jparker@digium.com> + + * apps/app_directory.c: If somebody enters a digit during + ast_stream_and_wait, the return value is the digit, which we need + to use later. + +2008-04-15 19:59 +0000 [r114146] Steve Murphy <murf@digium.com> + + * main/pbx.c: These changes: a. fix a self-found problem with + SPAWN-ing an extension, where matches were not being found b. + correct some wording in a comment c. Add some debug for future + debugging. + +2008-04-15 17:54 +0000 [r114143] Sean Bright <sean.bright@gmail.com> + + * apps/app_chanspy.c: I'm not sure why, but "this" bothers me. Ba + dum dum. + +2008-04-15 17:21 +0000 [r114131-114141] Jason Parker <jparker@digium.com> + + * channels/chan_unistim.c: Shorten the mac address pattern, since + some phones use different identifiers (such as the i2050 + softphone). (closes issue #12398) Reported by: c_hans Patches: + chan_unistim_svn.diff uploaded by c (license 460) Tested by: + c_hans + + * /, contrib/scripts/autosupport: Merged revisions 114138 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114138 | qwell | 2008-04-15 12:17:18 -0500 (Tue, 15 Apr 2008) | + 7 lines Update Digium autosupport script, for more useful + information. (closes issue #12452) Reported by: angler Patches: + autosupport.diff uploaded by angler (license 106) ........ + + * /, apps/app_queue.c: Merged revisions 114133 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) | + 8 lines Allow autofill to work in the general section of + queues.conf. Additionally, don't try to (re)set options when they + have empty values in realtime (all unset columns would have an + empty value). (closes issue #12445) Reported by: atis Patches: + 12445-autofill.diff uploaded by qwell (license 4) ........ + + * main/channel.c: Convert several DEBUG logs into ast_debug. + (closes issue #12444) Reported by: IgorG Patches: + channel_c_debug.diff uploaded by IgorG (license 20) + +2008-04-14 19:58 +0000 [r114124-114127] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c: Need a new buffer for each loop + + * res/res_phoneprov.c: Don't unref user twice on failure. Also, + when adding sorted list of users, it is best to check the entry + already in the list for a "next" entry instead of the newly + created entry... + +2008-04-14 18:34 +0000 [r114121] Jason Parker <jparker@digium.com> + + * /, channels/chan_h323.c: Merged revisions 114120 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr + 2008) | 7 lines The call_token on the pvt can occasionally be + NULL, causing a crash. If it is NULL, we can skip this channel, + since it can't the one we're looking for. (closes issue #9299) + Reported by: vazir ........ + +2008-04-14 17:42 +0000 [r114118] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 114117 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr + 2008) | 11 lines Increase the retry count when attempting to show + channels. This apparently cleared an issue someone was seeing + when attempting to show channels when the load was high. (closes + issue #11667) Reported by: falves11 Patches: 11677.txt uploaded + by russell (license 2) Tested by: falves11 ........ + +2008-04-14 16:32 +0000 [r114115] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/astcli: Make tab-completion work for all cases + +2008-04-14 16:25 +0000 [r114113] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 114112 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr + 2008) | 9 lines If the datastore has been moved to another + channel due to a masquerade, then freeing the datastore here + causes an eventual double free when the new channel hangs up. We + should only free the datastore if we were able to successfully + remove it from the channel we are referencing (i.e. the datastore + was not moved). (closes issue #12359) Reported by: pguido + ........ + +2008-04-14 15:36 +0000 [r114109] Joshua Colp <jcolp@digium.com> + + * channels/chan_local.c: During hangup it is possible for p->chan + or p->owner to be NULL, so just return what the channel is + bridged to instead of what they are *really* bridged to. Thanks + Matt Nicholson! + +2008-04-14 15:01 +0000 [r114107] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 114106 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr + 2008) | 5 lines Save a local copy of the generate callback prior + to unlocking the channel in case the generate callback goes NULL + on us after the channel is unlocked. Thanks to Russell for + pointing this need out to me. ........ + +2008-04-14 14:53 +0000 [r114101-114104] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 114103 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 + lines It is possible for the remote side to say they want T38 but + not give any capabilities. (closes issue #12414) Reported by: MVF + ........ + + * /, main/rtp.c: Merged revisions 114100 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4 + lines Don't change the SSRC when a new source comes into play, + this might happen quite often and depending on the remote side... + they might not like this. (closes issue #12353) Reported by: + dimas ........ + +2008-04-14 02:55 +0000 [r114096-114098] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/astcli: Add tab command-line completion (Closes + issue #12428) + + * apps/app_meetme.c: Use ast_mkdir instead of mkdir (Closes issue + #12430) + +2008-04-12 16:21 +0000 [r114092-114093] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Make sure linkset is locked exiting + ss7_start_call + + * channels/chan_zap.c: Make sure we start incoming calls on SS7 + with echo cancellation enabled. Also make sure when completing a + COT we call ss7_start_call with the proper locks held. Lastly, + make sure if we fail to get a channel from zt_new that we don't + assume it's there. + +2008-04-11 23:26 +0000 [r114085-114090] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_pgsql.c: If any field is not null, but has no default, + then it must be set or the insert will fail. (Closes issue + #12285) + + * configs/res_ldap.conf.sample: Make the sample config match the + contributed LDAP schema (Closes issue #12421) + + * res/res_config_ldap.c: Use the correct function for free'ing + objects, and maybe we won't crash. (closes issue #12163) Reported + by: gservat Patches: 20080411__bug12163.diff.txt uploaded by + Corydon76 (license 14) Tested by: gservat + +2008-04-11 22:48 +0000 [r114080-114084] Terry Wilson <twilson@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 114083 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 + Apr 2008) | 7 lines Several places in the code called + find_callno() (which releases the lock on the pvt structure) and + then immediately locked the call and did things with it. + Unfortunately, the call can disappear between the find_callno and + the lock, causing Bad Stuff(tm) to happen. Added + find_callno_locked() function to return the callno withtout + unlocking for instances that it is needed. (issue #12400) + Reported by: ztel ........ + + * res/res_phoneprov.c: Make sure that ${LINE} is set even if + linenumber is not set in users.conf + +2008-04-11 22:09 +0000 [r114077] Doug Bailey <dbailey@digium.com> + + * phoneprov/polycom_line.xml: Change the number of line keys per + registration from 2 to 1 + +2008-04-11 21:04 +0000 [r114067] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c: Fix the fact that global_variables 1) + weren't being updated on reload (thanks for the report, Doug), + and 2) weren't actually being appended to the list of profile + variables because build_profile was called before the list was + populated. Also needed to free the contents returned by + load_file(). + +2008-04-11 15:49 +0000 [r114064] Mark Michelson <mmichelson@digium.com> + + * /, main/features.c: Merged revisions 114063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr + 2008) | 11 lines Fix a race condition that may happen between a + sip hangup and a "core show channel" command. This patch adds + locking to prevent the resulting crash. (closes issue #12155) + Reported by: tsearle Patches: show_channels_crash2.patch uploaded + by tsearle (license 373) Tested by: tsearle ........ + +2008-04-11 14:54 +0000 [r114061] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_ldap.c: Errors are all greater than 0 (closes + issue #12422) Reported by: nito Patches: + res_config_ldap_result_check_patch.diff uploaded by nito (license + 340) + +2008-04-10 22:02 +0000 [r114052] Mark Michelson <mmichelson@digium.com> + + * utils/Makefile, main/manager.c, /, utils/astman.c, + utils/hashtest.c, main/utils.c, include/asterisk/lock.h, + utils/ael_main.c, utils/hashtest2.c, utils/conf2ael.c, + utils/check_expr.c: Merged revisions 114051 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr + 2008) | 3 lines Fix 1.4 build when LOW_MEMORY is enabled. + ........ + +2008-04-10 20:28 +0000 [r114049] Joshua Colp <jcolp@digium.com> + + * channels/chan_local.c, CHANGES: A 'b' option has been added which + causes chan_local to return the actual channel that is behind it + when queried. This is useful for transfer scenarios as the actual + channel will be transferred, not the Local channel. If you have + been using Local channels as queue members and having issues when + the agent did a blind transfer this option may solve the issue. + +2008-04-10 19:58 +0000 [r114046] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 114045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr + 2008) | 6 lines Be sure that we're not about to set bridgepvt + NULL prior to dereferencing it. (closes issue #11775) Reported + by: fujin ........ + +2008-04-10 19:04 +0000 [r114042] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/astcli: The hydra grows yet another head... + (closes issue #12401) Reported by: davevg Patches: astcli.diff2 + uploaded by davevg (license 209) Tested by: davevg, Corydon76 + +2008-04-10 17:27 +0000 [r114036] Jason Parker <jparker@digium.com> + + * /, main/file.c: Merged revisions 114035 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | + 10 lines Only try to prefix language if we are not using an + absolute path (suffix it otherwise). + en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes + issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff + uploaded by qwell (license 4) Tested by: kuj, qwell ........ + +2008-04-10 15:10 +0000 [r114022-114030] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 114029 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 + lines Create the directory where name recordings will go if it + does not exist. (closes issue #12311) Reported by: rkeene + Patches: 12311-mkdir.diff uploaded by qwell (license 4) ........ + + * apps/app_voicemail.c: Don't hardcode ru into the digits filename + so that languageprefix can work. (closes issue #12404) Reported + by: IgorG Patches: voicemail_ru_hardcoded-v1.patch uploaded by + IgorG (license 20) + + * channels/chan_unistim.c, channels/chan_skinny.c, main/rtp.c: Fix + spelling of existent in a few places. (closes issue #12409) + Reported by: candlerb + + * /, channels/chan_sip.c: Merged revisions 114021 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 + lines Don't add custom URI options if they don't exist OR they + are empty. (closes issue #12407) Reported by: homesick Patches: + uri_options-1.4.diff uploaded by homesick (license 91) ........ + +2008-04-09 22:32 +0000 [r113928-113980] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix a crash that happened due to accessing + free'd memory (closes issue #12396) Reported by: tcalosi Patches: + 12396.patch uploaded by putnopvut (license 60) Tested by: tcalosi + + * /, channels/chan_sip.c: Merged revisions 113927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr + 2008) | 8 lines We need to set the persistant_route [sic] + parameter for the sip_pvt during the initial INVITE, no matter if + we're building the route set from an INVITE request or response. + (closes issue #12391) Reported by: benjaminbohlmann Tested by: + benjaminbohlmann ........ + +2008-04-09 19:00 +0000 [r113875] Tilghman Lesher <tlesher@digium.com> + + * /, configs/cdr.conf.sample, cdr/cdr_csv.c: Merged revisions + 113874 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) + | 4 lines If the [csv] section does not exist in cdr.conf, then + an unload/load sequence is needed to correct the problem. Track + whether the load succeeded with a variable, so we can fix this + with a simple reload event, instead. ........ + +2008-04-09 18:05 +0000 [r113840] Joshua Colp <jcolp@digium.com> + + * channels/chan_h323.c: Enable enough RTP bridging to allow P2P to + work. (closes issue #11901) Reported by: pj + +2008-04-09 17:56 +0000 [r113838] Jason Parker <jparker@digium.com> + + * contrib/scripts/astcli: Fix a small file handle "leak" pointed + out by jjshoe on #asterisk. + +2008-04-09 17:48 +0000 [r113836] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c: There was a subtle logical difference between 1.4 and + trunk with regards to how timeouts were handled. In 1.4, if the + absolute timeout were reached on a call, no matter what the + return value of ast_spawn_extension was, the pbx would attempt to + go to the 'T' extension or hangup otherwise. The rearrangement of + this function in trunk made this check only happen in the case + that ast_spawn_extension returned 0. If ast_spawn_extension + returned 1, then the fact that the timeout expired resulted in a + no-op, and would cause an infinite loop to occur in + __ast_pbx_run. This change fixes this problem. Now timeouts will + behave as they did in 1.4 (closes issue #11550) Reported by: pj + Tested by: putnopvut + +2008-04-09 17:41 +0000 [r113834] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c: Move all messages wrapped in skinnydebug + from debug to verbose. (closes issue #12224) Reported by: DEA + Patches: chan_skinny-debug-log.txt uploaded by DEA (license 3) + +2008-04-09 16:52 +0000 [r113785] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 113784 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr + 2008) | 4 lines If we receive an AUTHREQ from the remote server + and we are unable to reply (for example they have a secret + configured, but we do not) then queue a hangup frame on the + Asterisk channel. This will cause the channel to hangup and a + HANGUP to be sent via IAX2 to the remote side which is the proper + thing to do in this scenario. (closes issue #12385) Reported by: + viraptor ........ + +2008-04-09 16:23 +0000 [r113731-113752] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Mark recent additions from #11954 and #12254 + + * configs/voicemail.conf.sample, apps/app_voicemail.c: Permit + message wrap-around during message retrieval. (closes issue + #12254) Reported by: andrew Patches: bug-12253.diff uploaded by + snuffy (license 35) + +2008-04-09 14:41 +0000 [r113682] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 113681 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr + 2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a + reinvite), then we should not send a BYE. (closes issue #12392) + Reported by: fnordian Patches: chan_sip.patch uploaded by + fnordian (license 110) with small modification from me ........ + +2008-04-09 13:55 +0000 [r113647-113649] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c: Permit callee to continue in the dialplan, after + caller has hung up. (closes issue #11954) Reported by: johan + Patches: app_dial_rev104031.patch uploaded by johan (license 334) + + * contrib/scripts/astcli: Additional enhancements (closes issue + #12390) Reported by: tzafrir Patches: astcli_fixes.diff uploaded + by tzafrir (license 46) + +2008-04-09 01:36 +0000 [r113597] Terry Wilson <twilson@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 113596 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 + Apr 2008) | 2 lines Initialize fr->cacheable to make valgrind + happy ........ + +2008-04-08 21:33 +0000 [r113559] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/astcli (added): Add commandline tool for doing + CLI commands through AMI (instead of using asterisk -rx) (closes + issue #12389) Reported by: davevg Patches: astcli uploaded by + davevg (license 209) + +2008-04-08 18:49 +0000 [r113403-113505] Jason Parker <jparker@digium.com> + + * /, channels/chan_skinny.c: Merged revisions 113504 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr + 2008) | 1 line Add a little more that is required for previously + added devices. ........ + + * /, channels/chan_skinny.c: Merged revisions 113454 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr + 2008) | 4 lines Add support for several new(ish) devices - most + notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for + providing me the required information. ........ + + * main/features.c, include/asterisk/features.h: Move + AST_FEATURE_FLAG_* and FEATURE_RETURN_* to features.h so that + they can be used by modules. (closes issue #12384) Reported by: + fnordian Patches: features.patch uploaded by fnordian (license + 110) (patch modified by me, to give FEATURE_RETURN_* an AST_ + prefix) + + * /, main/asterisk.c: Merged revisions 113402 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) | + 1 line Work around some silliness caused by sys/capability.h - + this should fix compile errors a number of users have been + experiencing. ........ + +2008-04-08 16:54 +0000 [r113349-113400] Tilghman Lesher <tlesher@digium.com> + + * /, contrib/scripts/astgenkey.8: Merged revisions 113399 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008) + | 6 lines Add security note on astgenkey's manpage. (closes issue + #12373) Reported by: lmamane Patches: 20080406__bug12373.diff.txt + uploaded by Corydon76 (license 14) ........ + + * /, channels/chan_sip.c: Merged revisions 113348 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) + | 7 lines Move check for still-bridged channels out a little + further, to avoid possible deadlocks. (Closes issue #12252) + Reported by: callguy Patches: 20080319__bug12252.diff.txt + uploaded by Corydon76 (license 14) Tested by: callguy ........ + +2008-04-08 15:05 +0000 [r113297] Joshua Colp <jcolp@digium.com> + + * /, main/audiohook.c: Merged revisions 113296 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 + lines If audio suddenly gets fed into one side of a channel after + a lapse of frames flush the other factory so that old audio does + not remain in the factory causing the sync code to not execute. + (closes issue #12296) Reported by: jvandal ........ + +2008-04-07 22:16 +0000 [r113245] Tilghman Lesher <tlesher@digium.com> + + * configs/manager.conf.sample: Additional note + +2008-04-07 21:49 +0000 [r113243] Jason Parker <jparker@digium.com> + + * configs/manager.conf.sample: Document 'originate' permission in + manager sample config. + +2008-04-07 21:35 +0000 [r113241] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_sip.c: Merged revisions 113013 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) + | 15 lines Merged revisions 113012 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) + | 7 lines (closes issue #12362) (closes issue #12372) Reported + by: vinsik Tested by: tecnoxarxa This one line change makes an if + inside a for loop (in realtime_peer) check all the ast_variables + the loop was intending to test rather than just the first one. + ........ ................ + +2008-04-07 20:22 +0000 [r113207] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: This is a "fix" for something that's been + bugging the crap out of me for a while. The variable name "flag" + to distinguish between whether a message is being forwarded or is + new is not a helpful name. The newly added doxygen documentation + to app_voicemail is tremendously helpful, but I still just...hate + this variable name. I think is_new_message is more indicative of + what its purpose is. + +2008-04-07 19:06 +0000 [r113172] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_strings.c: Merged revisions 113117 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 + Apr 2008) | 3 lines Force ast_mktime() to check for DST, since + strptime(3) does not. (Closes issue #12374) ........ + +2008-04-07 18:57 +0000 [r113170] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c: atoi(NULL) is bad + +2008-04-07 18:02 +0000 [r113119] Jason Parker <jparker@digium.com> + + * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged + revisions 113118 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | + 8 lines Allow playback with noanswer (and add earlyrtp option). + (closes issue #9077) Reported by: pj Patches: earlyrtp.diff + uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, + wedhorn ........ + +2008-04-07 16:12 +0000 [r113066] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 113065 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr + 2008) | 13 lines This fix prevents a deadlock that was + experienced in chan_local. There was deadlock prevention in place + in chan_local, but it would not work in a specific case because + the channel was recursively locked. By unlocking the channel + prior to calling the generator's generate callback in + ast_read_generator_actions(), we prevent the recursive locking, + and therefore the deadlock. (closes issue #12307) Reported by: + callguy Patches: 12307.patch uploaded by putnopvut (license 60) + Tested by: callguy ........ + +2008-04-07 15:18 +0000 [r113013] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 113012 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) + | 7 lines (closes issue #12362) (closes issue #12372) Reported + by: vinsik Tested by: tecnoxarxa This one line change makes an if + inside a for loop (in realtime_peer) check all the ast_variables + the loop was intending to test rather than just the first one. + ........ + +2008-04-07 14:54 +0000 [r113009] Joshua Colp <jcolp@digium.com> + + * main/slinfactory.c, include/asterisk/slinfactory.h: Put my + slinfactory changes back in. + +2008-04-05 13:24 +0000 [r112972] Tilghman Lesher <tlesher@digium.com> + + * res/res_agi.c: AsyncAGI should not close the manager session on + error. (closes issue #12370) Reported by: srt Patches: + asterisk-12370.diff uploaded by srt (license 378) + +2008-04-05 07:58 +0000 [r112906-112939] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c: Clean up some memory leak/ref counting + issues + + * phoneprov/000000000000-directory.xml, phoneprov/polycom.xml, + res/res_phoneprov.c, phoneprov/polycom_line.xml (added): + Multi-line support for phoneprov + +2008-04-05 01:33 +0000 [r112874] Steve Murphy <murf@digium.com> + + * channels/chan_sip.c: Found a little problem with the sip request + handling that could lead to a quick crash of asterisk, and a road + to a DOS attack if left unfixed. Attaching to a running asterisk + with "telnet hostname 5060", I would input "something", then hit + return three times, and asterisk crashes. I traced it to + handle_request_do(), which zeroes out the data (an ast_str ptr) + if the string is too short. Instead of freeing the struct and + nulling the pointer, it now just resets it, because this ast_str + is expected by the calling routine to still be there after + handle_request_do() returns. This appears to fix the crash. I + assume that it was introduced with ast_str's being adopted. It's + a subtle and easy-to-miss sort of problem. I also found all the + places where the req.data is freed, and made sure the ptr is + Nulled out as well; no good leaving bad ptrs laying around-- I + didn't need to do this, but it seemed a good thing to do... + +2008-04-04 19:28 +0000 [r112785-112821] Philippe Sultan <philippe.sultan@gmail.com> + + * /, channels/chan_gtalk.c: Merged revisions 112820 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 + Apr 2008) | 1 line Free newly allocated channel before returning + ........ + + * /, channels/chan_gtalk.c: Merged revisions 112766 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 + Apr 2008) | 7 lines Prevent call connections when codecs don't + match. (closes issue #10604) Reported by: keepitcool Patches: + branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested + by: phsultan ........ + +2008-04-04 00:57 +0000 [r112714] Dwayne M. Hubbard <dhubbard@digium.com> + + * main/asterisk.c: sleep long enough for the zaptel timer error + message to display before exit + +2008-04-04 00:53 +0000 [r112712] Joshua Colp <jcolp@digium.com> + + * /, main/Makefile: Merged revisions 112711 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 + lines Pass in the path to Zaptel for systems that install Zaptel + headers in a separate location. ........ + +2008-04-04 00:32 +0000 [r112653-112708] Dwayne M. Hubbard <dhubbard@digium.com> + + * /: blocked for trunk....woot + + * main/asterisk.c: satisfy buildbot + + * main/asterisk.c: add a Zaptel timer check to verify the timer is + responding when Zaptel support is compiled into Asterisk and + Zaptel drivers are loaded. This will help people not waste their + valuable time debugging side effects. + +2008-04-03 14:35 +0000 [r112600] Mark Michelson <mmichelson@digium.com> + + * channels/chan_zap.c, /: Merged revisions 112599 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr + 2008) | 9 lines Fix the testing of the "res" variable so that it + is more logically correct and makes the correct warning and debug + messages print. (closes issue #12361) Reported by: one47 Patches: + chan_zap_deferred_digit.patch uploaded by one47 (license 23) + ........ + +2008-04-03 07:49 +0000 [r112520-112564] Tilghman Lesher <tlesher@digium.com> + + * formats/format_wav.c, main/file.c, include/asterisk/mod_format.h: + Use a 32k file buffer on recordings, which increases the + efficiency of file recording. (closes issue #11962) Reported by: + garlew Patches: recording.patch uploaded by garlew (license 376) + bug-11962.diff uploaded by snuffy (license 35) + + * channels/chan_misdn.c: Make MISDN generate channel rename events + when the name changes. (closes issue #11142) Reported by: + julianjm Patches: chan_misdn_tmpchan_trunk_v1.diff uploaded by + julianjm (license 99) + +2008-04-02 17:36 +0000 [r112469] Mark Michelson <mmichelson@digium.com> + + * main/manager.c, /: Merged revisions 112468 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr + 2008) | 13 lines Fix a race condition in the manager. It is + possible that a new manager event could be appended during a + brief time when the manager is not waiting for input. If an event + comes during this period, we need to set an indicator that there + is an event pending so that the manager doesn't attempt to wait + forever for an event that already happened. (closes issue #12354) + Reported by: bamby Patches: manager_race_condition.diff uploaded + by bamby (license 430) (comments added by me) ........ + +2008-04-02 15:26 +0000 [r112431] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Since the SIP request structure gets reused + multiple times with TCP handling we have to clear the debug state + or else we will keep spitting out debug even after it has been + turned off. (closes issue #12169) Reported by: pj Patches: + 12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj + +2008-04-02 15:25 +0000 [r112426] Terry Wilson <twilson@digium.com> + + * build_tools/cflags.xml, include/asterisk/http.h, main/manager.c, + res/res_phoneprov.c, main/http.c, res/res_http_post.c (added): + Re-add HTTP post support by moving to res_http_post.c + +2008-04-02 14:32 +0000 [r112394] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 112393 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr + 2008) | 6 lines Ensure that there is no timeout if none is + specified. (closes issue #12349) Reported by: johnlange ........ + +2008-04-01 22:55 +0000 [r112360] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_sip.c: Added dnsmgr status output for sip show + registry. + +2008-04-01 22:45 +0000 [r112357] Steve Murphy <murf@digium.com> + + * main/pbx.c: Bumped across another test set for the new exten + pattern matcher, which revealed a problem with the + CANMATCH/MATCHMORE modes. Direct matches were getting in the way. + Fixed. + +2008-04-01 22:25 +0000 [r112351] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c: Fix a typo that prevented configuration of + non-dynamic peers. + +2008-04-01 22:07 +0000 [r112321] Jeff Peeler <jpeeler@digium.com> + + * CHANGES, channels/chan_iax2.c: Existing DNS manager lookups + extended to check for SRV records. + +2008-04-01 20:02 +0000 [r112289] Steve Murphy <murf@digium.com> + + * main/pbx.c: (closes issue #12298) Reported by: falves11 Patches: + 12298.patch1 uploaded by murf (license 17) Tested by: murf I have + hopes that the changes made over the last few days will finalize + and solidify this code. While there are bound to be small tweaks + still needed, I feel that the job (at last) is somewhat + completed. Finally, I had a chance to comprehend how the scoring + of extension patterns was done in the previous version, and I've + come very close to using the exact same criteria in the new + pattern matching code. The left-right sorting is now replicated + in the trie structure itself, such that the first match found + will the 'best' match. Compared the results against 1.4 for + several extensions. Replicated falves11's setup and it works. + Used some devious patterns provided by jsmith, supplemented with + a few of my own. Looks good. + +2008-04-01 18:27 +0000 [r112241-112252] Jason Parker <jparker@digium.com> + + * apps/app_voicemail.c: Minor formatting cleanup. (closes issue + #12343) Reported by: travishein Patches: + app_voicemail_code_convention.patch uploaded by travishein + (license 385) + + * apps/app_voicemail.c: More voicemail doxygen additions/cleanup. + (issue #12343) Reported by: travishein Patches: + app_voicemail_code_documentation.patch uploaded by travishein + (license 385) + +2008-04-01 18:23 +0000 [r112234] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_vpb.cc: Fix last commit + +2008-04-01 18:06 +0000 [r112210] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 112209 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 + lines Disable Packet2Packet bridging when we need to feed DTMF + frames into the core. Some implementations do not like how we + switch between things. (closes issue #12212) Reported by: bamby + ........ + +2008-04-01 17:53 +0000 [r112207] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_zap.c, main/dnsmgr.c, channels/chan_sip.c, + main/slinfactory.c, CHANGES, channels/chan_iax2.c, + include/asterisk/dnsmgr.h, include/asterisk/slinfactory.h: This + adds DNS SRV record support to DNS manager. If there is a SRV + record for a given domain, the hostname and port listed in the + SRV record will be used. If no SRV record exists or a SRV lookup + is not attempted, the DNS lookup on the specified domain will be + performed as normal. Chan_sip has been modified to take advantage + of the new SRV support. + +2008-04-01 17:48 +0000 [r112155-112205] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 112204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 + lines Do not pass audio until the remote side has indicated they + are providing early media, or if the channel has been answered. + (closes issue #11823) Reported by: SDamm ........ + + * channels/chan_sip.c: Demote a log message down to a warning. + (closes issue #12345) Reported by: caio1982 Patches: + limit_msg.diff uploaded by caio1982 (license 22) + +2008-04-01 17:23 +0000 [r112148] Mark Michelson <mmichelson@digium.com> + + * /, main/dns.c: Merged revisions 112138 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr + 2008) | 10 lines Initialize the __res_state structure used for + dns purposes to all 0's prior to using it. This is due to + valgrind's complaints on issue #12284 as well as an excerpt found + in "Description" portion of the online man page found here: + http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV + (pertains to issue #12284 but does not necessarily close it) + ........ + +2008-04-01 16:50 +0000 [r112126] Joshua Colp <jcolp@digium.com> + + * /, main/slinfactory.c, include/asterisk/slinfactory.h: Merged + revisions 112125 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 + lines Ensure that we do not exceed the hold's maximum size with a + single frame. (closes issue #12047) Reported by: fabianoheringer + Tested by: fabianoheringer ........ + +2008-04-01 16:35 +0000 [r112124] Russell Bryant <russell@digium.com> + + * channels/chan_zap.c: Now that zaptel trunk has been removed, add + the PSTN deprecation notice to chan_zap, as well. + +2008-03-31 22:16 +0000 [r112069-112071] Jason Parker <jparker@digium.com> + + * channels/chan_usbradio.c: I missed a place when this define was + changed. (closes issue #12334) Reported by: ovi Patches: + 12334-asterisk.patch uploaded by dimas (license 88) + + * /, apps/app_voicemail.c: Merged revisions 112068 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar + 2008) | 5 lines Fix a silly infinite loop when choosing an + invalid option. (closes issue #12315) Reported by: jmls ........ + +2008-03-31 21:01 +0000 [r112033-112035] Terry Wilson <twilson@digium.com> + + * main/http.c: Yeah, simplify that logic a bit... + + * main/http.c: Handle blank prefix= in http.conf + +2008-03-31 17:14 +0000 [r111996-111998] Russell Bryant <russell@digium.com> + + * Makefile: Ensure configure gets run on a clean checkout. (closes + issue #12197) Reported by: juggie Patches: 12197.diff uploaded by + juggie (license 24) + + * channels/chan_sip.c: This fixes a high fence violation that + MALLOC_DEBUG reported to me. + +2008-03-31 14:20 +0000 [r111961] Joshua Colp <jcolp@digium.com> + + * res/res_config_sqlite.c: Initialize all these here tmp pointers + at declaration. They confused some compilers a wee bit. (closes + issue #12333) Reported by: ovi + +2008-03-28 22:50 +0000 [r111908-111909] Russell Bryant <russell@digium.com> + + * doc/janitor-projects.txt, include/asterisk/pbx.h: Make some notes + about common usage of pbx_builtin_getvar_helper() that is not + thread-safe. + + * main/dnsmgr.c: Note a minor race condition that I noticed while + reviewing Jeff's changes to this code. + +2008-03-28 21:46 +0000 [r111857] Jason Parker <jparker@digium.com> + + * codecs/gsm/inc/private.h, /: Merged revisions 111856 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar + 2008) | 12 lines Allow gsm to compile correctly on x86 with gcc4 + optimizations. (closes issue #11243) Reported by: whiskerp + Patches: 11243-maybe-asm.diff uploaded by qwell (license 4) + Tested by: Seggy (IRC) Note: While I did write this patch, I + would not have found this if fossil had not reported and fixed + issue #12253. A huge thanks to him for helping to (indirectly) + find the problem here. ........ + +2008-03-28 20:03 +0000 [r111777-111811] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: This time the fix is proper for issue 12284. + I have tested it thoroughly and found that valgrind no longer + complains and that calls do complete correctly. The fix is along + the same lines as before: Make sure the final null terminator + gets copied into the new sip_request's data pointer. Without it, + parse_request will read and potentially write past the end of the + string, causing potential crashes. (closes issue #12284...for + real this time!) reported by falves11 + + * channels/chan_sip.c, include/asterisk/strings.h: Temporary revert + of 111662. It's causing lots of trouble and appears to not be the + proper solution to the problem reported anyway. (related to issue + #12884) + +2008-03-28 19:08 +0000 [r111721-111774] Jason Parker <jparker@digium.com> + + * apps/app_voicemail.c: Replace magic number size from msgArray + array with a define. (same patch as before, I just split this + part out) (close issue #12326) Reported by: travishein Patches: + app_voicemail_code_documentation.patch uploaded by travishein + (license 385) + + * apps/app_voicemail.c: Add a bit of doxygen documentation for + app_voicemail. (issue #12326) Reported by: travishein Patches: + app_voicemail_code_documentation.patch uploaded by travishein + (license 385) + + * /, channels/chan_skinny.c: Merged revisions 111720 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar + 2008) | 1 line Remove unimplemented softkeys. Prompted by issue + #12325. ........ + +2008-03-28 16:36 +0000 [r111662] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c, include/asterisk/strings.h: The copy_request + function did not take into account the necessary null terminator + for the string to be copied into. This resulted in parse_request + reading invalid memory beyond the end of the string, and in some + cases led to crashes. Thanks to falves11 for providing the + valgrind output which led to the closure of this issue. (closes + issue #12284) Reported by: falves11 + +2008-03-28 16:20 +0000 [r111659] Jason Parker <jparker@digium.com> + + * /, formats/format_wav_gsm.c: Merged revisions 111658 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar + 2008) | 8 lines The file size of WAV49 does not need to be an + even number. (closes issue #12128) Reported by: mdu113 Patches: + 12128-noevenlength.diff uploaded by qwell (license 4) Tested by: + qwell, mdu113 ........ + +2008-03-28 14:37 +0000 [r111606] Tilghman Lesher <tlesher@digium.com> + + * /, doc/valgrind.txt: Merged revisions 111605 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) + | 3 lines Update debugging text, since Valgrind eliminated the + --log-file-exactly option. (Closes issue #12320) ........ + +2008-03-28 00:55 +0000 [r111565] Joshua Colp <jcolp@digium.com> + + * apps/app_queue.c: Forgetting to unregister a manager action is + bad, mmmk? + +2008-03-28 00:12 +0000 [r111533] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix a crash that would happen when attempting + to unload the app_queue module. The problem was that when the + refcount on the queue hit 0, the destructor was called, and + inside the destructor, another function was called which would + increase the refcount back to 1 again and then decrease it again + back to 0 for every member in the queue. This meant that the + destructor was being recursively called, leading to a double free + of the queue. This is now fixed by making sure to unlink the + queue from the queues container prior to the final unref of the + queue. + +2008-03-27 22:10 +0000 [r111500] Terry Wilson <twilson@digium.com> + + * main/http.c: Fix another little http problem. In making it match + coding guidelines, a comparison was dropped + +2008-03-27 21:25 +0000 [r111497] Steve Murphy <murf@digium.com> + + * main/pbx.c: comment cleanup and iron out a really dumb mistake in + handling the '.'-wildcard in the new exten pattern matcher. + +2008-03-27 19:26 +0000 [r111443] Tilghman Lesher <tlesher@digium.com> + + * /, main/acl.c: Merged revisions 111442 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008) + | 6 lines For FreeBSD, at least, the ifa_addr element could be + NULL. (closes issue #12300) Reported by: festr Patches: + acl.c.patch uploaded by festr (license 443) ........ + +2008-03-27 13:29 +0000 [r111360-111410] Steve Murphy <murf@digium.com> + + * main/pbx.c, /, apps/app_playback.c: Merged revisions 111391 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 + lines These small documentation updates made in response to a + query in asterisk-users, where a user was using Playback, but + needed the features of Background, and had no idea that + Background existed, or that it might provide the features he + needed. I thought the best way to avert these kinds of queries + was to provide "See Also" references in all three of + "Background", "Playback", "WaitExten". Perhaps a project to do + this with all related apps is in order. ........ + + * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c, + include/asterisk/ael_structs.h: Merged revisions 111341 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | + 15 lines (closes issue #12302) Reported by: pj Tested by: murf + These changes will set a channel variable ~~EXTEN~~ just before + generating code for a switch, with the value of ${EXTEN}. The + exten is marked as having a switch, and ever after that, till the + end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~} + instead in application arguments; (and the ${EXTEN: also). The + reason for this, is that because switches are coded using + separate extensions to provide pattern matching, and jumping + to/from these switch extensions messes up the ${EXTEN} value, + which blows the minds of users. ........ + +2008-03-27 00:27 +0000 [r111246-111295] Jason Parker <jparker@digium.com> + + * main/frame.c: But we can change the API here. + + * main/frame.c, /: Merged revisions 111280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | + 1 line Put this flag back so we don't change the API. ........ + + * main/frame.c, /: Merged revisions 111245 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | + 9 lines Remove excessive smoother optimization that was causing + audio glitches (small "pops") after (about 200ms later) an + "incorrectly" sized frame was received. While it would be very + nice to keep this as optimized as possible, it makes no sense for + the smoother to be dropping random bits of audio like this. Isn't + that the whole point of a smoother? Closes issue #12093. ........ + +2008-03-26 21:23 +0000 [r111213] Terry Wilson <twilson@digium.com> + + * main/http.c: Stupid strcasecmp function :-) + +2008-03-26 20:34 +0000 [r111132-111185] Tilghman Lesher <tlesher@digium.com> + + * channels/misdn_config.c: Oops, missed one + + * include/asterisk/linkedlists.h, main/config.c: Simplify new + macro, simplify configfile logic, now that list is sorted + +2008-03-26 19:56 +0000 [r111130] Joshua Colp <jcolp@digium.com> + + * /, contrib/scripts/autosupport: Merged revisions 111129 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6 + lines Update autosupport script. (closes issue #12310) Reported + by: angler Patches: autosupport.diff uploaded by angler (license + 106) ........ + +2008-03-26 19:52 +0000 [r111127] Kevin P. Fleming <kpfleming@digium.com> + + * /, UPGRADE.txt: Merged revisions 111126 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500 + (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar + 2008) | 2 lines update UPGRADE notes to document usage of the + script ........ ................ + +2008-03-26 19:39 +0000 [r111123] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 111121 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, + 26 Mar 2008) | 4 lines This code change is made just for + clarification. It does exactly the same thing as before. It just + doesn't look as wrong. ........ + +2008-03-26 19:29 +0000 [r111083] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Add expiry value to the sip show + subscriptions CLI command. (closes issue #12025) Reported by: agx + +2008-03-26 19:26 +0000 [r111067] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 111049 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, + 26 Mar 2008) | 9 lines Add a lock to the vm_state structure and + use the lock around mail_open calls to prevent concurrent access + of the same mailstream. This, along with trunk's ability to + configure TCP timeouts for IMAP storage will help to prevent + crashes and hangs when using voicemail with IMAP storage. (closes + issue #10487) Reported by: ewilhelmsen ........ + +2008-03-26 19:19 +0000 [r111036] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/linkedlists.h, CHANGES, main/config.c: Add a + linkedlist macro that maintains a sorted list + +2008-03-26 19:16 +0000 [r111028] Jason Parker <jparker@digium.com> + + * main/dsp.c: Only try to detect silence when we actually need to, + instead of...always. If this is wrong, I'd love to hear why. + +2008-03-26 19:08 +0000 [r111025] Kevin P. Fleming <kpfleming@digium.com> + + * /, contrib/scripts/get_ilbc_source.sh (added), codecs/ilbc: + Merged revisions 111024 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r111024 | kpfleming | 2008-03-26 14:06:56 -0500 + (Wed, 26 Mar 2008) | 10 lines Merged revisions 111019 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar + 2008) | 2 lines add a script to make getting the iLBC source code + simple for end users ........ ................ + +2008-03-26 19:05 +0000 [r111022] Jason Parker <jparker@digium.com> + + * channels/chan_usbradio.c, channels/chan_vpb.cc, + channels/chan_zap.c, include/asterisk/dsp.h, main/dsp.c: Large + cleanup of DSP code Per comments from dimas: 1. The code now + generates DTMF_BEGIN frames in addition to DTMF_END ones. 2. + "quelching" rewritten - now each detector (MF/DTMF/generic tone) + may mark fragment of a frame for suppression (squelching, muting) + with a call to mute_fragment. Actual muting happens only once at + the very end of ast_dsp_process where all marked fragments are + zeroed. This way every detector sees original data in the frame + without any piece of a frame being zeroed by a detector which was + run before. 3. DTMF detector tries to "mute" one block before and + one block after the block where actual tone was detected. Muting + of previois block is something new for this patch. Obviously this + operation is not always possible - if current frame does not + contain data for previous block - it is too late. But at least we + make our best. Muting of next block was already done by the old + code but it only affects part of the next block which is in the + frame being processed. New code keeps this information in state + structures so it will mute proper number of samples in the next + frame(s) too. 4. Removed ast_dsp_digitdetect and + ast_dsp_getdigits APIs because these are not used. 5. DSP API + extended a bit - ast_dsp_was_muted() function added which returns + true if DSP code was muting any fragment in the last frame. + chan_zap uses this function to decide it needs to turn on + confmute on the channel. This is to replace AST_FRAME_DTMF + 'm'/'u' (mute/unmute) functionality. (closes issue #11968) + Reported by: dimas Patches: v2-11968-dsp.patch uploaded by dimas + (license 88) v4-11968-zap.patch uploaded by dimas (license 88) + Tested by: dimas, qwell + +2008-03-26 19:05 +0000 [r111017-111021] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 111020 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 + lines If we are requested to authenticate a reinvite make sure + that it contains T38 SDP if need be. (closes issue #11995) + Reported by: fall ........ + + * /, channels/chan_iax2.c: Merged revisions 110628 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar + 2008) | 4 lines Add an option (transmit_silence) which transmits + silence during both Record() and DTMF generation. The reason this + is an option is that in order to transmit silence we have to + setup a translation path. This may not be needed/wanted in all + cases. (closes issue #10058) Reported by: tracinet ........ + +2008-03-26 18:41 +0000 [r111012-111013] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Oops, fix this, too + + * main/udptl.c, main/dnsmgr.c, include/asterisk/config.h, + channels/iax2-provision.c, main/enum.c, main/rtp.c, + main/config.c, main/loader.c, main/cdr.c, main/manager.c, + main/features.c, main/logger.c, main/http.c, + include/asterisk/udptl.h, main/asterisk.c, main/dsp.c: Add the + "config reload <conffile>" command, which allows you to tell + Asterisk to reload any file that references a given configuration + file. + +2008-03-26 17:44 +0000 [r110963] Kevin P. Fleming <kpfleming@digium.com> + + * /, UPGRADE.txt: Merged revisions 110962 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar + 2008) | 2 lines add note that the user will need to enable + codec_ilbc to get it to build ........ + +2008-03-26 17:28 +0000 [r110911-110930] Donny Kavanagh <donnyk@gmail.com> + + * Makefile: revert something dumb, because i was running svn diff + in a subfolder not the root of trunk, before doing my commit and + did not see it + + * Makefile, doc/snmp.txt: update documentation to reflect the + changes in the way configure detects net-snmp. (closes issue + #12067) Reported by: juggie Patches: 12067_snmp_doc.patch + uploaded by juggie (license 24) Tested by: juggie + +2008-03-26 17:10 +0000 [r110881] Kevin P. Fleming <kpfleming@digium.com> + + * codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h + (removed), codecs/ilbc/syntFilter.h (removed), + codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c + (removed), codecs/ilbc/StateConstructW.h (removed), + codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h + (removed), codecs/ilbc/getCBvec.c (removed), + codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c + (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c + (removed), codecs/ilbc/getCBvec.h (removed), + codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/iLBC_define.h + (removed), codecs/ilbc/FrameClassify.c (removed), + codecs/ilbc/enhancer.h (removed), codecs/ilbc/lsf.h (removed), + codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h + (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c + (removed), codecs/ilbc/anaFilter.c (removed), + codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c + (removed), codecs/ilbc/doCPLC.h (removed), + codecs/ilbc/anaFilter.h (removed), UPGRADE.txt, + codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c + (removed), codecs/ilbc/createCB.h (removed), CHANGES, + codecs/ilbc/constants.h (removed), codecs/ilbc/iLBC_decode.h + (removed), codecs/ilbc/iCBSearch.c (removed), codecs/Makefile, + codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed), + codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c + (removed), codecs/ilbc/iCBSearch.h (removed), + codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed), + codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c + (removed), codecs/ilbc/hpOutput.h (removed), + codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c, + codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h + (removed), codecs/ilbc/iCBConstruct.c (removed): Merged revisions + 110880 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700 + (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar + 2008) | 2 lines due to licensing restrictions, we cannot + distribute the source code for iLBC encoding and decoding... so + remove it, and add instructions on how the user can obtain it + themselves ........ ................ + +2008-03-26 00:02 +0000 [r110831] Mark Michelson <mmichelson@digium.com> + + * main/manager.c: This ensures that the manager interface is not + enabled by default. Prior to this change, it was possible to + start Asterisk with the manager interface enabled, then either + comment out the enabled option or make manager.conf unopenable + and the manager interface would still be enabled. + +2008-03-25 22:51 +0000 [r110780] Jason Parker <jparker@digium.com> + + * /, cdr/cdr_custom.c: Merged revisions 110779 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | + 6 lines Make file access in cdr_custom similar to cdr_csv. Fixes + issue #12268. Patch borrowed from r82344 ........ + +2008-03-25 20:02 +0000 [r110726] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_sip.c: This one line change makes an if inside a + for loop (in realtime_peer) check all the ast_variables the loop + was intending to test rather than just the first one. + +2008-03-25 17:46 +0000 [r110689-110691] Tilghman Lesher <tlesher@digium.com> + + * configs/voicemail.conf.sample, configs/extensions.conf.sample: + Update sample configurations to make virtual hosting more + obvious. (closes issue #11969) Reported by: pprindeville Patches: + acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347) + + * configs/extensions.conf.sample: Update the sample configuration, + to use Macro less (since it's now deprecated). (closes issue + #12293) Reported by: pprindeville Patches: + bugid-0012293.1.6.patch uploaded by pprindeville (license 347) + +2008-03-25 15:44 +0000 [r110636-110639] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Oops here too. I need to stop coding for a + while... + + * /, channels/chan_sip.c: Merged revisions 110635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar + 2008) | 7 lines When reverting a commit, I accidentally left in + this bit which was an experiment to see what would happen. It + passed the compile test, and I didn't notice I had left this + change in too. So this is a revert of a revert...sort of. + ........ + +2008-03-25 15:18 +0000 [r110629-110631] Joshua Colp <jcolp@digium.com> + + * main/channel.c, channels/chan_sip.c, configs/sip.conf.sample, + CHANGES: Add a special dialplan variable to chan_sip which will + cause an audio file to be played upon completion of an attended + transfer. (closes issue #9239) Reported by: sunder + + * Makefile, /, main/app.c, include/asterisk/options.h, + main/asterisk.c: Merged revisions 110628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 + lines Add an option (transmit_silence) which transmits silence + during both Record() and DTMF generation. The reason this is an + option is that in order to transmit silence we have to setup a + translation path. This may not be needed/wanted in all cases. + (closes issue #10058) Reported by: tracinet ........ + +2008-03-25 10:54 +0000 [r110625] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Use the "Server" header when responding to + SIP requests. (closes issue #12278) Reported by: rjain Patches: + chan_sip.c.diff uploaded by rjain (license 226) + +2008-03-24 20:14 +0000 [r110619-110621] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Remove the "Event: registration" header from + Asterisk-generated SIP REGISTER requests. rjain points out that + RFC 3265 specifies that the Event: header is not a valid header + for REGISTER requests and that the "registration" value is not + defined at IANA. (closes issue #12279) Reported by: rjain + Patches: chan_sip.c.diff uploaded by rjain (license 226) + + * channels/chan_sip.c: Merged revisions 110618 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar + 2008) | 15 lines This is a revert for revision 108288. The reason + is that that revision was not for an actual bug fix per se, and + so it really should not have been in 1.4 in the first place. + Plus, people who compile with DO_CRASH are more likely to + encounter a crash due to this change. While I think the usage of + DO_CRASH in ast_sched_del is a bit absurd, this sort of change is + beyond the scope of 1.4 and should be done instead in a developer + branch based on trunk so that all scheduler functions are fixed + at once. I also am reverting the change to trunk and 1.6 since + they also suffer from the DO_CRASH potential. (closes issue + #12272) Reported by: qq12345 ........ + +2008-03-24 17:36 +0000 [r110615] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 110614 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 + Mar 2008) | 2 lines Turn a NOTICE into a DEBUG message. ........ + +2008-03-24 15:28 +0000 [r110610] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Only print out the set_address_from_contact + host verbose message if debugging is enabled on the dialog. + (closes issue #12280) Reported by: rjain Patches: chan_sip.c.diff + uploaded by rjain (license 226) + +2008-03-21 21:52 +0000 [r110578] Jason Parker <jparker@digium.com> + + * sounds/Makefile: Update to 1.4.11 core sounds. + +2008-03-21 17:58 +0000 [r110542] Joshua Colp <jcolp@digium.com> + + * include/asterisk/audiohook.h, main/audiohook.c: Merge over + ast_audiohook_volume branch. This adds API calls for use by + developers to adjust the volume on a channel. + +2008-03-21 15:24 +0000 [r110499] Russell Bryant <russell@digium.com> + + * configs/sip.conf.sample, CHANGES: Note that the TCP and TLS + support is currently considered experimental and is subject to + change while we work out the remaining issues. + +2008-03-21 14:36 +0000 [r110475] Jason Parker <jparker@digium.com> + + * /, codecs/gsm/Makefile: Merged revisions 110474 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) | + 7 lines Don't attempt to do optimizations of gsm on mips + platforms either. (closes issue #12270) Reported by: zandbelt + Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33) + ........ + +2008-03-21 01:44 +0000 [r110444] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Add note of the added Directory options, from commit + 110237 (closes issue #7151) + +2008-03-20 23:14 +0000 [r110303-110396] Russell Bryant <russell@digium.com> + + * main/autoservice.c, /: Merged revisions 110395 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) + | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms + in the autoservice thread. This really should not make a + difference except in very rare cases. That case would be that all + of the channels in autoservice are not generating any frames. In + that case, this change reduces the potential amount of time that + a thread waits in ast_autoservice_stop() for the autoservice + thread to wrap back around to the beginning of its loop. (closes + issue #12266, reported by dimas) ........ + + * codecs/codec_g722.c: Use the correct buffer for + g722tolin16_sample. This shouldn't have caused any problems, but + Qwell noticed the typo here. + + * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions + 110336 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r110336 | russell | 2008-03-20 16:54:58 -0500 + (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) + | 6 lines Fix some very broken code that was introduced in 1.2.26 + as a part of the security fix. The dnsmgr is not appropriate + here. The dnsmgr takes a pointer to an address structure that a + background thread continuously updates. However, in these cases, + a stack variable was passed. That means that the dnsmgr thread + would be continuously writing to bogus memory. ........ + ................ + + * main/file.c: Fix a bug when using zaptel timing for playing back + files that have a sample rate other than 8 kHz. The issue here is + that format modules give a "whennext" sample value, which is used + to calculate when to set a timer for to retrieve the next frame. + However, the zaptel timer operates on 8 kHz samples, so this must + be taken into account. (another part of issue #12164, reported by + milazzo and jsmith, patch by me) + +2008-03-20 18:01 +0000 [r110272] Mark Michelson <mmichelson@digium.com> + + * main/dial.c: Add missing unlock + +2008-03-20 17:45 +0000 [r110268-110270] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c, apps/app_minivm.c, include/asterisk/netsock.h, + main/netsock.c: Remove astobj.h from some places where it wasn't + needed + + * main/channel.c, res/res_musiconhold.c: Add some fixes that I made + in regards to wideband codec handling to get G.722 music on hold + working for me. (issue #12164, reported by milazzo and jsmith, + patches by me) res/res_musiconhold.c: - I moved a single line so + that the sample queue update happened before ast_write(). The + reason that this was a bug is that the G.722 frame originally + says it has 320 samples in it (which is correct). However, when + the frame is written to a channel that uses RTP, main/rtp.c + modifies the frame to cut the number of samples in half before it + sends it on the wire. This is to account for the stupid incorrect + G.722 spec that makes it so we have to lie about the number of + samples with RTP. I should probably go and re-work the RTP code + so it doesn't modify the frame so that a bug like this won't + happen in the future. However, this change to MOH is harmless. + main/channel.c: - I made two fixes in regards to generator + timing. Generators use samples for timing. However, this code + assumed 8 kHz samples. In one case, it was a hard coded 160 + samples, that is now written as the sample rate / 50. The other + place was dealing with timing a generator based on frames coming + from the other direction. However, that would have only worked if + the sample rates for the formats in both directions were the + same. The code now takes into account that the sample rates may + differ, and scales the generator samples accordingly. + +2008-03-20 05:06 +0000 [r110211-110237] Tilghman Lesher <tlesher@digium.com> + + * apps/app_directory.c, sounds/Makefile: Upgrade the sounds + version; add several directory enhancements: 1) Number of digits + to enter can now be configured 2) The digits can now match on + both first AND last name, instead of only one or the other + (Closes issue #7151) + + * channels/chan_vpb.cc: Fix recent trunk breakage + +2008-03-19 22:58 +0000 [r110164] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 110163 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) + | 5 lines Fix a bug where when calls on the trunk side hang up + while on hold, the state is not properly reflected. (closes issue + #11990, reported by anakaoka, patched by me) ........ + +2008-03-19 22:25 +0000 [r110132-110161] Jason Parker <jparker@digium.com> + + * channels/chan_misdn.c, channels/chan_zap.c, channels/chan_sip.c, + channels/chan_h323.c, include/asterisk/dsp.h, + channels/chan_mgcp.c, main/dsp.c: Rename DSP_FEATURE_DTMF_DETECT, + because we are *NOT* only detecting DTMF digits. This was very + misleading. Early cleanup for issue #11968 + + * channels/chan_usbradio.c, channels/chan_vpb.cc, + channels/chan_zap.c, channels/chan_sip.c, include/asterisk/dsp.h, + channels/chan_mgcp.c, main/dsp.c: Rename very poorly named + function to reflect what it actually does. This was causing quite + a bit of confusion for me... + +2008-03-19 21:05 +0000 [r110087] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_sip.c, CHANGES: This change adds DNS manager + support for registrations not referencing a peer entry. It looks + like there is support for DNS manager for realtime peers as well, + however it is not implemented correctly. The improper usage + occurs when ast_dnsmgr_lookup is called with one of the arguments + being an address from the stack to be continually updated. The + variable from the stack will go out of scope and dnsmgr will + continue to try and update the memory there, causing possible + stack corruption. This problem will be worked on next as well as + adding DNS manager support for peer entries. + +2008-03-19 20:34 +0000 [r110084] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_chanspy.c: Merged revisions 110083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar + 2008) | 4 lines Add a missing unlock in the case that memory + allocation fails in app_chanspy. Thanks to Russell for confirming + that this was an issue. ........ + +2008-03-19 19:13 +0000 [r110036] Joshua Colp <jcolp@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 110035 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar + 2008) | 4 lines Add sanity checking for position resuming. We + *have* to make sure that the position does not exceed the total + number of files present, and we have to make sure that the + position's filename is the same as previous. These values can + change if a music class is reloaded and give unpredictable + behavior. (closes issue #11663) Reported by: junky ........ + +2008-03-19 18:57 +0000 [r110023] Russell Bryant <russell@digium.com> + + * /: remove svnmerge-blocked property that is not supposed to be + here + +2008-03-19 18:25 +0000 [r110020] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 110019 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 + lines Make sure that the mark bit does not incorrectly cause + video frame timestamps to be calculated as if they are audio + frames. (closes issue #11429) Reported by: sperreault Patches: + 11429-frametype.diff uploaded by qwell (license 4) ........ + +2008-03-19 17:15 +0000 [r109974] Jason Parker <jparker@digium.com> + + * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml + (added), /: Merged revisions 109973 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) | + 5 lines People report bugs about Asterisk crashing with DO_CRASH + enabled was getting a little silly... Now we only show certain + cflags when you run configure with --enable-dev-mode + (corresponding menuselect change to follow) ........ + +2008-03-19 16:54 +0000 [r109970] Joshua Colp <jcolp@digium.com> + + * main/pbx.c, CHANGES: Add the ability to use a pattern match for a + hint. (closes issue #7767) Reported by: Corydon76 Patches: + 20070314__simple_hint_lookup.diff.txt uploaded by Corydon76 + pbx-trunk-98436.diff uploaded by plack (license 365) + +2008-03-19 16:24 +0000 [r109942] Steve Murphy <murf@digium.com> + + * /, main/config.c: Merged revisions 109908 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) | + 72 lines (closes issue #11442) Reported by: tzafrir Patches: + 11442.patch uploaded by murf (license 17) Tested by: murf I + didn't give tzafrir very much time to test this, but if he does + still have remaining issues, he is welcome to re-open this bug, + and we'll do what is called for. I reproduced the problem, and + tested the fix, so I hope I am not jumping by just going ahead + and committing the fix. The problem was with what file_save does + with templates; firstly, it tended to print out multiple options: + [my_category](!)(templateref) instead of + [my_category](!,templateref) which is fixed by this patch. + Nextly, the code to suppress output of duplicate declarations + that would occur because the reader copies inherited declarations + down the hierarchy, was not working. Thus: [master-template](!) + mastervar = bar [template](!,master-template) tvar = value + [cat](template) catvar = val would be rewritten as: ;! ;! + Automatically generated configuration file ;! Filename: + experiment.conf (/etc/asterisk/experiment.conf) ;! Generator: + Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;! + [master-template](!) mastervar = bar + [template](!,master-template) mastervar = bar tvar = value + [cat](template) mastervar = bar tvar = value catvar = val This + has been fixed. Since the config reader 'explodes' inherited vars + into the category, users may, in certain circumstances, see + output different from what they originally entered, but it should + be both correct and equivalent. ........ + +2008-03-19 16:21 +0000 [r109912-109926] Kevin P. Fleming <kpfleming@digium.com> + + * res/res_phoneprov.c: ensure that res_phoneprov's HTTP handler + tells the dispatcher what method it can handle + + * main/manager.c, main/http.c: actually implement HTTP request + dispatching based on both URI and method; reduce duplication of + data when generating responses using ast_http_error() + +2008-03-19 15:45 +0000 [r109910] Russell Bryant <russell@digium.com> + + * main/pbx.c: Fix some more breakage that I introduced when + changing extension state callbacks to the list macros. + +2008-03-19 15:41 +0000 [r109909] Kevin P. Fleming <kpfleming@digium.com> + + * main/http.c: clean up code to conform to coding guidelines + +2008-03-19 15:22 +0000 [r109833-109907] Russell Bryant <russell@digium.com> + + * main/pbx.c: Remove an unneeded variable. This compiled, but I + missed the uninitialized warning because I always compile without + optimizations turned on. Sorry! + + * main/pbx.c: Convert handling of extension state callbacks to the + list macros. + + * main/pbx.c: Minor coding style changes, including adding handling + for memory allocation failure + + * main/http.c: Minor change to use Asterisk macros + + * /, main/utils.c: Merged revisions 109838 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008) + | 2 lines Tweak spacing in a recent change because I'm very + picky. ........ + + * channels/chan_sip.c: Set req->data to NULL after free'ing to + ensure that it never gets accidentally double free'd. (reported + by dhubbard directly to me) + +2008-03-18 23:32 +0000 [r109802] Mark Michelson <mmichelson@digium.com> + + * channels/chan_zap.c: Fix a typo which caused a double free in + chan_zap. This was discovered by Juggie while attempting to load + chan_zap. Apparently this would happen if an error were + encountered while trying to process zapata.conf. + +2008-03-18 23:22 +0000 [r109775] Tilghman Lesher <tlesher@digium.com> + + * configs/res_ldap.conf.sample, res/res_config_ldap.c: Change back + to using ldap_initialize() and let the user specify a URL + directly, instead of trying to piece it together, badly. + +2008-03-18 22:36 +0000 [r109764] Russell Bryant <russell@digium.com> + + * /, apps/app_chanspy.c: Merged revisions 109763 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008) + | 3 lines Fix one place where the chanspy datastore isn't removed + from a channel. (issue #12243, reported by atis, patch by me) + ........ + +2008-03-18 22:32 +0000 [r109762] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/http.h, main/manager.c, res/res_phoneprov.c, + main/http.c, include/asterisk/_private.h: start the process of + changing HTTP request dispatching to do it based on *both* URI + and method, so that POST support can move into a module; move + http.c's private function prototypes into _private.h + +2008-03-18 20:59 +0000 [r109714] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 109713 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar + 2008) | 12 lines This patch makes it so that all queue member + status changes are handled through device state code. This + removes several problems people were seeing where their queue + members would get into an "unknown" state. Huge props go to atis + on this one since he was the one who found the code section that + was causing the problem and proposed the solution. I just wrote + what he suggested :) (closes issue #12127) Reported by: atis + Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested + by: atis, jvandal ........ + +2008-03-18 20:13 +0000 [r109683] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_ldap.c: Set protocol version, port number + correctly. (closes issue #12211, closes issue #12209) Reported + by: sylvain + +2008-03-18 20:02 +0000 [r109681] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Since a sip request's data field is now a + stringfield, we not only have to check if the string is + zero-length, but also if the data field is non-null. (closes + issue #12250) Reported by: caio1982 + +2008-03-18 19:53 +0000 [r109680] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/dbsep.cgi: Comment debug + +2008-03-18 19:24 +0000 [r109651] Jason Parker <jparker@digium.com> + + * /, codecs/log2comp.h: Merged revisions 109648 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) | + 7 lines Allow codecs that use log2comp (g726) to compile + correctly on x86 with gcc4 optimizations. (closes issue #12253) + Reported by: fossil Patches: log2comp.patch uploaded by fossil + (license 140) ........ + +2008-03-18 18:58 +0000 [r109545-109621] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add option + 'randomperiodicannounce' to queues.conf. Setting this will allow + the list of periodic announcments specified to be played in a + random order instead of being played sequentially. (closes issue + #6681) Reported by: alt_phil Tested by: putnopvut + + * /, channels/chan_agent.c: Merged revisions 109575 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, + 18 Mar 2008) | 6 lines Make sure an agent doesn't try to send + dtmf to a NULL channel closes issue #12242 Reported by Yourname + ........ + + * include/asterisk/astmm.h: Add format attribute to printf-style + functions in astmm.h + +2008-03-18 16:23 +0000 [r109451-109475] Kevin P. Fleming <kpfleming@digium.com> + + * channels/misdn/isdn_lib.h, channels/chan_misdn.c, + channels/misdn/isdn_lib.c: fix up various warnings found via the + addition of format string checking... some of these were really, + really bad code + + * configure, include/asterisk/autoconfig.h.in, acinclude.m4: ensure + that dependencies on AST_C_DEFINE_CHECK symbols work properly + +2008-03-18 15:43 +0000 [r109447] Terry Wilson <twilson@digium.com> + + * include/asterisk/utils.h, cdr/cdr_sqlite3_custom.c, + apps/app_meetme.c, channels/chan_sip.c, apps/app_festival.c, + main/translate.c, res/res_phoneprov.c, main/jitterbuf.c, + utils/astman.c, main/utils.c, include/jitterbuf.h, + apps/app_queue.c, channels/chan_iax2.c, utils/frame.c, + main/cli.c, Makefile, funcs/func_enum.c, main/manager.c, + channels/chan_misdn.c, include/asterisk/astobj.h, res/res_agi.c, + main/features.c, apps/app_minivm.c, res/res_realtime.c, + utils/extconf.c, res/res_indications.c, + include/asterisk/strings.h, res/res_config_ldap.c, + main/asterisk.c, utils/check_expr.c, apps/app_voicemail.c: Go + through and fix a bunch of places where character strings were + being interpreted as format strings. Most of these changes are + solely to make compiling with -Wsecurity and -Wformat=2 happy, + and were not actual problems, per se. I also added format + attributes to any printf wrapper functions I found that didn't + have them. -Wsecurity and -Wmissing-format-attribute added to + --enable-dev-mode. + +2008-03-18 15:13 +0000 [r109396] Joshua Colp <jcolp@digium.com> + + * main/manager.c, main/logger.c: Make sure values are interpreted + as character strings and not format strings. (AST-2008-004) + +2008-03-18 15:10 +0000 [r109394] Jason Parker <jparker@digium.com> + + * /: Block this here. Already committed. + +2008-03-18 15:08 +0000 [r109390] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c, main/rtp.c: Merged revisions 109386 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 + lines Put a maximum limit on the number of payloads accepted, and + also make sure a given payload does not exceed our maximum value. + (AST-2008-002) ........ + +2008-03-18 15:07 +0000 [r109389] Jason Parker <jparker@digium.com> + + * channels/chan_sip.c: Do not return with a successful + authentication if the From header ends up empty. (AST-2008-003) + +2008-03-18 14:09 +0000 [r109357] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ael-ntest23/t1, pbx/ael/ael-test/ael-ntest23/t2, + pbx/ael/ael-test/ael-ntest23/t3, /, include/asterisk/extconf.h, + pbx/ael/ael-test/ael-ntest23/extensions.ael, + pbx/ael/ael-test/ael-ntest23 (added), utils/conf2ael.c, + pbx/ael/ael-test/ael-ntest23/t1/a.ael, + pbx/ael/ael-test/ael-ntest23/t1/b.ael, + pbx/ael/ael-test/ael-ntest23/t1/c.ael, + pbx/ael/ael-test/ael-ntest23/t2/d.ael, + pbx/ael/ael-test/ael-ntest23/t2/e.ael, + pbx/ael/ael-test/ael-ntest23/t2/f.ael, res/ael/ael_lex.c, + pbx/ael/ael-test/ref.ael-ntest23 (added), + pbx/ael/ael-test/ael-ntest23/t3/g.ael, + pbx/ael/ael-test/ael-ntest23/t3/h.ael, utils/ael_main.c, + pbx/ael/ael-test/ael-ntest23/t3/i.ael, utils/extconf.c, + pbx/ael/ael-test/ael-ntest23/t3/j.ael, res/ael/ael.flex, + pbx/ael/ael-test/ael-ntest23/qq.ael: Merged revisions 109309 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | + 17 lines (closes issue #11903) Reported by: atis Many thanks to + atis for spotting this problem and reporting it. The fix was to + straighten out how items are placed on and removed from the file + stack. Regressions as well as the provided test case helped to + straighten out all code paths. valgrind was used to make sure all + memory allocated was freed. Sorry for not solving this earlier. I + got distracted. Added the ntest23 regression test, which is + mainly a copy of ntest22, but with a few juicy errors thrown in, + to replicate the kind of error that atis spotted. ........ + +2008-03-18 07:23 +0000 [r109316] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add + manager peerstatus events when peer can't authenticate. (closes + issue #11959) Reported by: mostyn Patches: peerstatus3.patch + uploaded by mostyn (license 398) + +2008-03-18 00:28 +0000 [r109282] Sean Bright <sean.bright@gmail.com> + + * configure, configure.ac: Fix a typo + +2008-03-17 22:10 +0000 [r109229] Terry Wilson <twilson@digium.com> + + * build_tools/cflags.xml, build_tools/menuselect-deps.in, + configure, include/asterisk/autoconfig.h.in, main/Makefile, + configure.ac, main/http.c, main/minimime (removed), + build_tools/make_buildopts_h, makeopts.in: Replace minimime with + superior GMime library so that the entire contents of an http + post are not read into memory. This does introduce a dependency + on the GMime library for handling HTTP POSTs, but it is available + in most distros. If the library is present, then the compile flag + for ENABLE_UPLOADS is enabled by default in menuselect. + +2008-03-17 22:06 +0000 [r109227] Mark Michelson <mmichelson@digium.com> + + * /, main/utils.c: Merged revisions 109226 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar + 2008) | 12 lines Fix a logic flaw in the code that stores lock + info which is displayed via the "core show locks" command. The + idea behind this section of code was to remove the previous lock + from the list if it was a trylock that had failed. Unfortunately, + instead of checking the status of the previous lock, we were + referencing the index immediately following the previous lock in + the lock_info->locks array. The result of this problem, under the + right circumstances, was that the lock which we currently in the + process of attempting to acquire could "overwrite" the previous + lock which was acquired. While this does not in any way affect + typical operation, it *could* lead to misleading "core show + locks" output. ........ + +2008-03-17 17:58 +0000 [r109172] Michiel van Baak <michiel@vanbaak.info> + + * /: block rev 109171 that is already here + +2008-03-17 17:47 +0000 [r109169] Steve Murphy <murf@digium.com> + + * main/pbx.c, include/asterisk/pbx.h: (closes issue #12238) + Reported by: mvanbaak Tested by: murf, mvanbaak Due to a bug that + occurred when merge_contexts_and_delete scanned the "old" or + existing contexts, and found a context that doesn't exist in the + new set, yet owned by a different registrar. The context is + created in the new set, with the old registrar, and and all the + priorities and extens that have a different registrar are copied + into it. But, not the includes, ignorepats, and switches. I added + code to do this immediately after the context is created. This + still leaves a logical hole in the code. If you define a context + in two places, (eg. in extensions.conf and also in + extensions.ael), and they both have includes, but different in + composition, no new context will be generated, and therefore the + 'old' includes, switches, and ignorepats will not be copied. I'd + have added code to simply add any non-duplicates into the 'new' + context that had a different registrar, but there is one big + complication: includes, and switches are definitely order + dependent. (ignorepats I'm not sure about). And we'll have to + develop some sort of policy about how we merge order dependent + lists, especially if the intersection of the two sets is empty. + (in other words, they do not have any elements in common). Do the + new go first, or the old? I've elected to punt this issue until a + user complains. Hopefully, this is pretty rare thing. + +2008-03-17 17:43 +0000 [r109168] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: Update the directory of placed calls on + skinny phones when dialing a channel that does not provide + progress (analog ZAP lines) The phone does handle the double + update on calls to channels that do provide progress and wont + insert duplicate items (closes issue #12239) Reported by: DEA + Patches: chan_skinny-call-log.txt uploaded by DEA (license 3) + +2008-03-17 17:31 +0000 [r109166] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, configure, configure.ac, acinclude.m4: don't define + Zaptel features as libraries, they aren't, and we don't want + '--with-zaptel-<foo>' configure options for them also some minor + cleanups + +2008-03-17 16:47 +0000 [r109113] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Remove something that is never ever used. + +2008-03-17 16:37 +0000 [r109111] Jason Parker <jparker@digium.com> + + * configs/sip_notify.conf.sample: Add sample events for aastra + phones. aastra-check-cfg is the same as the other check-cfg + entries, and aastra-xml is to load a pre-configured xml script. + (closes issue #12229) Reported by: gowen72 Patches: aastra.patch + uploaded by gowen72 (license 432) + +2008-03-17 16:26 +0000 [r109054-109108] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 109107 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4 + lines 200 OKs in response to a reinvite need to be sent reliably. + If the remote side does not receive one the dialog will be torn + down. (closes issue #12208) Reported by: atrash ........ + + * channels/chan_sip.c: Make sure that the temporary sip_request + structure is empty so that copy_request doesn't think it already + has an ast_str. (closes issue #12231) Reported by: IgorG + +2008-03-17 14:21 +0000 [r109024] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_chanspy.c: Merged revisions 109012 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r109012 | mmichelson | 2008-03-17 09:18:26 -0500 (Mon, 17 Mar + 2008) | 6 lines Make sure that we release the lock on the spyee + channel if the spyee or spy has hung up (closes issue #12232) + Reported by: atis ........ + +2008-03-16 21:50 +0000 [r108962] Michiel van Baak <michiel@vanbaak.info> + + * main/dial.c, /: Merged revisions 108961 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008) + | 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes + issue #12228) Reported by: andrew Patches: SRC.patch uploaded by + andrew (license 240) ........ + +2008-03-16 17:55 +0000 [r108927-108929] Russell Bryant <russell@digium.com> + + * apps/app_voicemail.c: Remove an unnecessary thread attribute + instance + + * apps/app_voicemail.c: Fix polling for mailbox changes in + mailboxes that are not in the default vm context. (closes issue + #12223) Reported by: DEA Patches: vm-polled-imap.txt uploaded by + DEA (license 3) + +2008-03-15 16:21 +0000 [r108740-108894] Russell Bryant <russell@digium.com> + + * main/pbx.c: Remove a double write lock of the contexts lock in + ast_wrlock_contexts(). How did this ever work? (closes issue + #12219) Reported by: ys Patches: pbx.c.diff uploaded by ys + (license 281) + + * include/asterisk/dnsmgr.h: Doxygenify dnsmgr.h + + * Makefile: Make sure configure is run before menuselect on a clean + checkout (closes issue #12197) Reported by: juggie Patches: + 12197.diff uploaded by juggie (license 24) + + * /, channels/chan_oss.c: Merged revisions 108796 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) + | 5 lines Fix a channel name issue. chan_oss registers the + "Console" channel type, but it created channels with an "OSS" + prefix. (closes issue #12194, reported by davidw, patched by me) + ........ + + * /, contrib/init.d/rc.suse.asterisk: Merged revisions 108792 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108792 | russell | 2008-03-14 15:04:35 -0500 (Fri, 14 Mar 2008) + | 4 lines Update the SuSE init script to start networking before + asterisk, as well. (closes issue #12200, reported by and change + suggested by reinerotto) ........ + + * configure, acinclude.m4: Do a link test in AST_EXT_TOOL_CHECK() + to ensure we have all the required libs reported by the tool. + (closes issue #12067, reported by Juggie, patched by me) + +2008-03-14 16:52 +0000 [r108738] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 108737 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar + 2008) | 33 lines Fix a race condition in the SIP packet scheduler + which could cause a crash. chan_sip uses the scheduler API in + order to schedule retransmission of reliable packets (such as + INVITES). If a retransmission of a packet is occurring, then the + packet is removed from the scheduler and retrans_pkt is called. + Meanwhile, if a response is received from the packet as + previously transmitted, then when we ACK the response, we will + remove the packet from the scheduler and free the packet. The + problem is that both the ACK function and retrans_pkt attempt to + acquire the same lock at the beginning of the function call. This + means that if the ACK function acquires the lock first, then it + will free the packet which retrans_pkt is about to read from and + write to. The result is a crash. The solution: 1. If the ACK + function fails to remove the packet from the scheduler and the + retransmit id of the packet is not -1 (meaning that we have not + reached the maximum number of retransmissions) then release the + lock and yield so that retrans_pkt may acquire the lock and + operate. 2. Make absolutely certain that the ACK function does + not recursively lock the lock in question. If it does, then + releasing the lock will do no good, since retrans_pkt will still + be unable to acquire the lock. (closes issue #12098) Reported by: + wegbert (closes issue #12089) Reported by: PTorres Patches: + 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested + by: jvandal ........ + +2008-03-14 14:32 +0000 [r108683] Jason Parker <jparker@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 108682 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r108682 | qwell | 2008-03-14 09:29:05 -0500 (Fri, 14 Mar + 2008) | 4 lines Fix a potential segfault if chan (or + chan->music_state) is NULL. Closes issue #12210, credit to + edantie for pointing this out. ........ + +2008-03-13 23:12 +0000 [r108639] Jeff Peeler <jpeeler@digium.com> + + * doc/externalivr.txt, apps/app_externalivr.c, CHANGES: documenting + changes as a result of adding TCP functionality to ExternalIVR + +2008-03-13 21:47 +0000 [r108586] Mark Michelson <mmichelson@digium.com> + + * main/manager.c: Make this compile + +2008-03-13 21:40 +0000 [r108531-108584] Russell Bryant <russell@digium.com> + + * main/channel.c, /, include/asterisk/channel.h, + apps/app_chanspy.c: Merged revisions 108583 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) + | 11 lines Fix another issue that was causing crashes in chanspy. + This introduces a new datastore callback, called chan_fixup(). + The concept is exactly like the fixup callback that is used in + the channel technology interface. This callback gets called when + the owning channel changes due to a masquerade. Before this was + introduced, if a masquerade happened on a channel being spyed on, + the channel pointer in the datastore became invalid. (closes + issue #12187) (reported by, and lots of testing from atis) (props + to file for the help with ideas) ........ + + * /, channels/chan_sip.c: Merged revisions 108530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) + | 10 lines Make a tweak that gets the LEDs on polycom phones to + blink when an extension that has been subscribed to goes on hold. + Otherwise, they just stay on like it does when an extension is in + use. (closes issue #11263) Reported by: russell Patches: + notify_hold.rev1.txt uploaded by russell (license 2) Tested by: + russell ........ + +2008-03-13 20:59 +0000 [r108529] Mark Michelson <mmichelson@digium.com> + + * main/manager.c: Fixing a potential buffer overflow in the manager + command ModuleCheck. Though this overflow is exploitable + remotely, we are NOT issuing a security advisory for this since + in order to exploit the overflow, the attacker would have to + establish an authenticated manager session AND have the system + privilege. By gaining this privilege, the attacker already has + more powerful weapons at his disposal than overflowing a buffer + with a malformed manager header, so the vulnerability in this + case really lies with the authentication method that allowed the + attacker to gain the system privilege in the first place. + +2008-03-13 20:38 +0000 [r108523] Jeff Peeler <jpeeler@digium.com> + + * apps/app_externalivr.c: set variable to NULL to prevent + uninitialized warning + +2008-03-13 20:35 +0000 [r108439-108508] Russell Bryant <russell@digium.com> + + * apps/app_voicemail.c: Fix a place where configuration values + could cause an overflow of a buffer. + + * /, apps/app_followme.c: Merged revisions 108469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) + | 4 lines Fix a couple uses of sprintf. The second one could + actually cause an overflow of a stack buffer. It's not a security + issue though, it only depends on your configuration. ........ + + * channels/chan_sip.c: Merge changes from + team/jamesgolovich/chan_sip-ast_str This set of changes removes + the hard coded maximum packet size of 4kB from chan_sip. It now + starts by allocating 1kB, and growing the buffer as needed to + accommodate large packets. (closes issue #8556, reported by + mikma, patch by jamesgolovich) + +2008-03-13 18:59 +0000 [r108404] Jeff Peeler <jpeeler@digium.com> + + * apps/app_externalivr.c: (closes issue #11827) Reported by: + ctooley Patches: eivr_tcp_generic.patch uploaded by jpeeler + (license 325) This change adds the ability to communicate over a + TCP socket instead of forking a child process. + +2008-03-12 22:49 +0000 [r108295-108346] Russell Bryant <russell@digium.com> + + * main/http.c: Make the default prefix empty, like it was in + Asterisk 1.4. (closes issue #12198, reported by bkruse, patched + by me) + + * include/asterisk/http.h, main/tcptls.c, main/manager.c, + channels/chan_sip.c, res/res_phoneprov.c, main/http.c, + include/asterisk/tcptls.h: Rename ast_tcptls_server_instance to + session_instance, since this pertains to server and client usage. + +2008-03-12 22:09 +0000 [r108289-108293] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Let's get this to compile + + * /, channels/chan_sip.c: Merged revisions 108288 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar + 2008) | 14 lines Change AST_SCHED_DEL use to ast_sched_del for + autocongestion in chan_sip. The scheduler callback will always + return 0. This means that this id is never rescheduled, so it + makes no sense to loop trying to delete the id from the scheduler + queue. If we fail to remove the item from the queue once, it will + fail every single time. (Yes I realize that in this case, the + macro would exit early because the id is set to -1 in the + callback, but it still makes no sense to use that macro in favor + of calling ast_sched_del once and being done with it) This is the + first of potentially several such fixes. ........ + +2008-03-12 21:37 +0000 [r108286] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: add + support for named sections in zapata.conf, and fix a few bugs in + config file parsing (closes issue #9503) Reported by: tzafrir + Patches: fix_cleanups uploaded by tzafrir (license 46) + zapata_sections uploaded by tzafrir (license 46) + skipchannel_options uploaded by tzafrir (license 46) conf_sample + uploaded by tzafrir (license 46) patches updated by me to better + conform to coding guidelines and fix some problems + +2008-03-12 21:19 +0000 [r108238] Mark Michelson <mmichelson@digium.com> + + * /, include/asterisk/sched.h: Merged revisions 108227 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, + 12 Mar 2008) | 12 lines Added a large comment before the + AST_SCHED_DEL macro to explain its purpose as well as when it is + appropriate and when it is not appropriate to use it. I also + removed the part of the debug message that mentions that this is + probably a bug because there are some perfectly legitimate places + where ast_sched_del may fail to delete an entry (e.g. when the + scheduler callback manually reschedules with a new id instead of + returning non-zero to tell the scheduler to reschedule with the + same idea). I also raised the debug level of the debug message in + AST_SCHED_DEL since it seems like it could come up quite + frequently since the macro is probably being used in several + places where it shouldn't be. Also removed the redundant line, + file, and function information since that is provided by ast_log. + ........ + +2008-03-12 21:06 +0000 [r108226] Joshua Colp <jcolp@digium.com> + + * main/slinfactory.c, include/asterisk/slinfactory.h: Doxygenify + slinfactory a bit. + +2008-03-12 20:27 +0000 [r108191] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c: Merged revisions 108086 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar + 2008) | 6 lines if we receive an INVITE with a Content-Length + that is not a valid number, or is zero, then don't process the + rest of the message body looking for an SDP closes issue #11475 + Reported by: andrebarbosa ........ + +2008-03-12 19:59 +0000 [r108137] Russell Bryant <russell@digium.com> + + * main/channel.c, /, apps/app_chanspy.c: Merged revisions 108135 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) + | 40 lines (closes issue #12187, reported by atis, fixed by me + after some brainstorming on the issue with mmichelson) - Update + copyright info on app_chanspy. - Fix a race condition that caused + app_chanspy to crash. The issue was that the chanspy datastore + magic that was used to ensure that spyee channels did not + disappear out from under the code did not completely solve the + problem. It was actually possible for chanspy to acquire a + channel reference out of its datastore to a channel that was in + the middle of being destroyed. That was because datastore + destruction in ast_channel_free() was done near the end. So, this + left the code in app_chanspy accessing a channel that was + partially, or completely invalid because it was in the process of + being free'd by another thread. The following sort of shows the + code path where the race occurred: + ============================================================================= + Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy) + --------------------------------------||------------------------------------- + ast_channel_free() || - remove channel from channel list || - + lock/unlock the channel to ensure || that no references retrieved + from || the channel list exist. || + --------------------------------------||------------------------------------- + || channel_spy() - destroy some channel data || - Lock chanspy + datastore || - Retrieve reference to channel || - lock channel || + - Unlock chanspy datastore + --------------------------------------||------------------------------------- + - destroy channel datastores || - call chanspy datastore d'tor || + which NULL's out the ds' || - Operate on the channel ... + reference to the channel || || - free the channel || || || - + unlock the channel + --------------------------------------||------------------------------------- + ============================================================================= + ........ + +2008-03-12 18:29 +0000 [r108084] Joshua Colp <jcolp@digium.com> + + * /, include/asterisk/audiohook.h, main/audiohook.c, + apps/app_mixmonitor.c: Merged revisions 108083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 + lines Add a trigger mode that triggers on both read and write. + The actual function that returns the combined audio frame though + will wait until both sides have fed in audio, or until one side + stops (such as the case when you call Wait). (closes issue + #11945) Reported by: xheliox ........ + +2008-03-12 17:06 +0000 [r108032-108034] Russell Bryant <russell@digium.com> + + * funcs/func_config.c: - Add Tilghman to the copyright info ... he + wrote the hard part :) - Remove some magic in unload_module that + isn't needed. Module use counts already ensure that the function + isn't going to be in use at this point. + + * main/channel.c, /: Merged revisions 108031 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008) + | 4 lines Destroy the channel lock after the channel datastores. + (inspired by issue #12187) ........ + +2008-03-12 07:43 +0000 [r107878-107998] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Deadlock fixes (closes issue #12143) + Reported by: kactus Patches: 20080312__bug12143__2.diff.txt + uploaded by Corydon76 (license 14) Tested by: kactus + + * apps/app_dumpchan.c, apps/app_zapras.c, main/loader.c: Revert + several changes from revision 102525, as the changes were not + compatible, and, in fact, introduced regressions. (Closes issue + #12190) + + * funcs/func_config.c: Cache config files, when possible, for speed + + * contrib/scripts/iax-friends.sql, /, + contrib/scripts/sip-friends.sql: Merged revisions 107877 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107877 | tilghman | 2008-03-11 20:52:40 -0500 (Tue, 11 Mar 2008) + | 2 lines Document all of the possible realtime fields ........ + +2008-03-11 23:38 +0000 [r107827] Jason Parker <jparker@digium.com> + + * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 107826 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) | + 7 lines Update documentation for pgsql ODBC voicemail. (closes + issue #12186) Reported by: jsmith Patches: + vm_pgsql_doc_update.patch uploaded by jsmith (license 15) + ........ + +2008-03-11 22:55 +0000 [r107791] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_curl.c, res/res_config_pgsql.c, + res/res_config_odbc.c, include/asterisk/config.h, + res/res_config_ldap.c, res/res_config_sqlite.c, main/config.c: An + offhand comment from Russell made me realize that the + configuration file caching would not work properly for users.conf + and any other file read from more than one place. I needed to add + the filename which requested the config file to get it to work + properly. + +2008-03-11 22:54 +0000 [r107787-107790] Russell Bryant <russell@digium.com> + + * funcs/func_config.c: remove documentation of an argument that i + did not implement + + * funcs/func_config.c (added), CHANGES: Add a trivial new dialplan + function, AST_CONFIG(), which allows you to access a variable + from an Asterisk configuration file in the dialplan, or anywhere + else where dialplan functions can be used. (Inspired by a + discussion with Tilghman and Pari) + +2008-03-11 21:10 +0000 [r107721-107722] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_odbc.c: Convert prepare_and_execute to direct_execute for + speed (closes issue #11935) Reported by: falves11 Patches: + 20080208__bug11935.diff.txt uploaded by Corydon76 (license 14) + Tested by: falves11, Corydon76 + + * contrib/scripts/dbsep.cgi (added), configs/dbsep.conf.sample + (added): Add contributed script for separation of database access + from Asterisk + +2008-03-11 20:54 +0000 [r107719] Russell Bryant <russell@digium.com> + + * apps/app_voicemail.c: This patch adds support for extended help + prompts in voicemail. These prompts are in the 1.4.9 sounds + release. (closes issue #11705) Reported by: jaroth Patches: + helpprompts.patch uploaded by jaroth (license 50) + +2008-03-11 20:53 +0000 [r107718] Jason Parker <jparker@digium.com> + + * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged + revisions 107714 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | + 5 lines Copy voicemail dependency logic for res_adsi to + chan_gtalk and chan_jingle (for jabber). (closes issue #12014) + Reported by: junky ........ + +2008-03-11 20:50 +0000 [r107715] Kevin P. Fleming <kpfleming@digium.com> + + * /, Makefile.rules, channels/Makefile: Merged revisions 107713 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar + 2008) | 2 lines get chan_vpb to build properly in dev mode + ........ + +2008-03-11 20:36 +0000 [r107638-107710] Joshua Colp <jcolp@digium.com> + + * apps/app_page.c: Dial a device even if it's state is unknown. + (closes issue #12184) Reported by: bluecrow76 Patches: + asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by + bluecrow76 (license 270) + + * /, main/features.c: Merged revisions 107646 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4 + lines Make sure the visible indication is on the right channel so + when the masquerade happens the proper indication is enacted. + (closes issue #11707) Reported by: iam ........ + + * /, apps/app_meetme.c: Merged revisions 107637 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 + lines Add an additional check for setting conference parameter + when using the marked user options. It was possible for it to + return to a no listen/no talk state if a masquerade happened. + (closes issue #12136) Reported by: aragon ........ + +2008-03-11 16:28 +0000 [r107551] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_pgsql.c: Whitespace changes only + +2008-03-11 15:59 +0000 [r107530] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c: Remove some redundant logic from + wait_for_answer. This also let's us get rid of one of those XXX + comments from the code. The redundancy occurs because the + 'single' flag implies that the 'r' and 'm' flags are not set, so + there's no need to explicitly check them again. + +2008-03-11 15:39 +0000 [r107466-107525] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_vpb.cc: fix another potential bug found by gcc 4.3 + + * /: block fix that is already here + + * codecs/Makefile, /, apps/app_sms.c, apps/app_rpt.c, + channels/misdn/isdn_lib.c: Merged revisions 107464 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 + Mar 2008) | 2 lines fix various other problems found by gcc 4.3 + ........ + +2008-03-11 15:05 +0000 [r107465] Joshua Colp <jcolp@digium.com> + + * main/features.c: Clarify comment about masquerading and playback + of the parking slot. (closes issue #12180) Reported by: davidw + +2008-03-11 14:37 +0000 [r107373-107462] Kevin P. Fleming <kpfleming@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + apps/app_sms.c: Merged revisions 107461 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar + 2008) | 2 lines stop checking for mktime() in the configure + script... we don't use it, and the test is buggy under gcc 4.3 + ........ + + * /, configure, main/Makefile, configure.ac, makeopts.in: Merged + revisions 107408 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar + 2008) | 5 lines check for compiler support for + -fno-strict-overflow before using it (tested with Debian's gcc + 4.3, 4.1 and 3.4) (closes issue #12179) Reported by: Netview + ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 107405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar + 2008) | 2 lines fix small bug in IMAP toolkit testing ........ + + * main/udptl.c, utils/Makefile, /, main/Makefile, + main/editline/readline.c, res/Makefile: Merged revisions 107352 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar + 2008) | 11 lines fix up various compiler warnings found with + gcc-4.3: - the output of flex includes a static function called + 'input' that is not used, so for the moment we'll stop having the + compiler tell us about unused variables in the flex source files + (a better fix would be to improve our flex post-processing to + remove the unused function) - main/stdtime/localtime.c makes + assumptions about signed integer overflow, and gcc-4.3's improved + optimizer tries to take advantage of handling potential overflow + conditions at compile time; for now, suppress these optimizations + until we can fiure out if the code needs improvement - + main/udptl.c has some references to uninitialized variables; in + one case there was no bug, but in the other it was certainly + possibly for unexpected behavior to occur - + main/editline/readline.c had an unused variable ........ + +2008-03-11 01:09 +0000 [r107292] Terry Wilson <twilson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 107290 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008) + | 2 lines If we fail to alloc a channel, we should re-lock the + pvt structure before returning. ........ + +2008-03-10 21:48 +0000 [r107231] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: (closes + issue #6019) Reported by: ssokol Patches: + 20080304__bug6019.diff.txt uploaded by Corydon76 (license 14) + Tested by: putnopvut + +2008-03-10 20:28 +0000 [r107177] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, /: Merged revisions 107173 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) | + 5 lines Make sure to reenable echo can after a "failed" + (canceled, etc) three-way call. (closes issue #11335) Reported + by: rebuild ........ + +2008-03-10 20:17 +0000 [r107159-107162] Russell Bryant <russell@digium.com> + + * main/pbx.c, /: Merged revisions 107161 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) + | 8 lines Fix another bug specifically related to asynchronous + call origination. Once the PBX is started on the channel using + ast_pbx_start(), then the ownership of the channel has been + passed on to another thread. We can no longer access it in this + code. If the channel gets hung up very quickly, it is possible + that we could access a channel that has been free'd. (inspired by + BE-386) ........ + + * main/pbx.c, /: Merged revisions 107158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) + | 9 lines Fix some bugs related to originating calls. If the code + failed to start a PBX on the channel (such as if you set a call + limit based on the system's load average), then there were cases + where a channel that has already been free'd using ast_hangup() + got accessed. This caused weird memory corruption and crashes to + occur. (fixes issue BE-386) (much debugging credit goes to + twilson, final patch written by me) ........ + +2008-03-10 20:00 +0000 [r107157] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: If we receive a 488 on a T38 request + reinvite back to audio. As well reinvite across a bridge back to + audio if one side doesn't negotiate to T38. (closes issue #8677) + Reported by: alex-911 + +2008-03-10 17:13 +0000 [r107100-107103] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 107102 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008) + | 2 lines Resolve a compiler warning. ........ + + * main/channel.c, /: Merged revisions 107099 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008) + | 3 lines Fix a race condition where the generator can go away + (closes issue #12175, reported by edantie, patched by me) + ........ + +2008-03-10 15:45 +0000 [r107068] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: app_queue has now been doxygenified thanks to + snuffy! The ony thing I changed was the way that locks are + referenced, since the old 1.2 names were still used in the + comments. (closes issue #11997) Reported by: snuffy Patches: + bug_11997_queue_doxy.diff uploaded by snuffy (license 35) + +2008-03-10 14:55 +0000 [r107019] Steve Murphy <murf@digium.com> + + * main/cdr.c: way back in July, in r.75706, a fix was made ot the + strftime usages, which was good, but in this case, the check for + a nil time was accidentally removed, and now it is restored, to + keep timevals like '1969-12-31 17:00:00' from showing up in the + cdrs. No idea what databases will do with this. No bugs filed as + yet, but it felt like a bug. + +2008-03-10 14:36 +0000 [r107017] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c, main/cdr.c, /, include/asterisk/cdr.h: Merged + revisions 107016 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 + lines Move where unanswered CDRs are dropped to the CDR core, not + everything uses app_dial. (closes issue #11516) Reported by: ys + Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested + by: anest, jcapp, dartvader ........ + +2008-03-08 16:03 +0000 [r106946] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, /: Merged revisions 106945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar + 2008) | 2 lines don't generate D-Channel "up" and "down" messages + unless the channel state is actually changing; also, generate the + "up" message when an implicit "up" occurs due to reception of a + normal event when we thought the channel was "down" ........ + +2008-03-07 22:52 +0000 [r106896] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 106895 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008) + | 2 lines Only start the SLA thread if SLA has actually been + configured. ........ + +2008-03-07 22:36 +0000 [r106892] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Make sure we don't start a call when we have + already done so in response to a COT message + +2008-03-07 22:15 +0000 [r106843] Jason Parker <jparker@digium.com> + + * /, main/editline/Makefile.in: Merged revisions 106842 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) | + 5 lines Fix hardcoded grep in editline, were GNU grep is + required. (closes issue #12124) Reported by: dmartin ........ + +2008-03-07 19:33 +0000 [r106789] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 106788 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4 + lines Ignore source update control frame. (closes issue #12168) + Reported by: plack ........ + +2008-03-07 18:57 +0000 [r106757] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/pbx.c, include/asterisk/pval.h, + channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y, + apps/app_queue.c, channels/chan_iax2.c, utils/conf2ael.c, + utils/Makefile, res/ael/pval.c, channels/chan_skinny.c, + res/ael/ael.tab.c, main/features.c, pbx/pbx_ael.c, + res/ael/ael_lex.c, utils/ael_main.c, res/ael/ael.tab.h, + utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c, + res/ael/ael.flex: (closes issue #6002) Reported by: rizzo Tested + by: murf Proposal of the changes to be made, and then an + announcement of how they were accomplished: + http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html + and: + http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html + Here is a recap, file by file, of what I have done: + pbx/pbx_config.c pbx/pbx_ael.c All funcs that were passed a ptr + to the context list, now will ALSO be passed a hashtab ptr to the + same set. Why? because (for the time being), the dialplan is + stored in both, to facilitate a quick, low-cost move to + hash-tables to speed up dialplan processing. If it was deemed + necessary to pass the context LIST, well, it is just as necessary + to have the TABLE available. This is because the list/table in + question might not be the global one, but temporary ones we would + use to stage the dialplan on, and then swap into the global + position when things are ready. We now have one external function + for apps to use, "ast_context_find_or_create()" instead of the + pre-existing "find" and "create", as all existing usages used + both in tandem anyway. pbx_config, and pbx_ael, will stage the + reloaded dialplan into local lists and tables, and then call + merge_contexts_and_delete, which will merge (now) existing + contexts and priorities from other registrars into this local set + by copying them. Then, merge_contexts_and_delete will lock down + the contexts, swap the lists and tables, and unlock (real quick), + and then destroy the old dialplan. chan_sip.c chan_iax.c + chan_skinny.c All the channel drivers that would add regcontexts + now use the ast_context_find_or_create now. chan_sip also + includes a small fix to get rid of warnings about removing + priorities that never got entered. apps/app_meetme.c + apps/app_dial.c apps/app_queue.c All the apps that added a + context/exten/priority were also modified to use + ast_context_find_or_create instead. include/asterisk/pbx.h + ast_context_create() is removed. Find_or_create_ is the new + method. ast_context_find_or_create() interface gets the hashtab + added. ast_merge_contexts_and_delete() gets the local hashtab arg + added. ast_wrlock_contexts_version() is added so you can detect + if someone else got a writelock between your readlocking and + writelocking. ast_hashtab_compare_contexts was made public for + use in pbx_config/pbx_ael ast_hashtab_hash_contexts was in like + fashion make public. include/asterisk/pval.h ast_compile_ael2() + interface changed to include the local hashtab table ptr. + main/features.c For the sake of the parking context, we use + ast_context_find_or_create(). main/pbx.c I changed all the "tree" + names to "table" instead. That's because the original + implementation was based on binary trees. (had a free library). + Then I moved to hashtabs. Now, the names move forward too. + refcount field added to contexts, so you can keep track of how + many modules wanted this context to exist. Some log messages that + are warnings were inflated from LOG_NOTICE to LOG_WARNING. Added + some calls to ast_verb(3,...) for debug messages Lots of little + mods to ast_context_remove_extension2, which is now excersized in + ways it was not previously; one definite bug fixed. + find_or_create was upgraded to handle both local lists/tables as + well as the globals. context_merge() was added to do the + per-context merging of the old/present contexts/extens/prios into + the new/proposed local list/tables + ast_merge_contexts_and_delete() was heavily modified. + ast_add_extension2() was also upgraded to handle changes. the + context_destroy() code was re-engineered to handle the new way of + doing things, by exten/prio instead of by context. res/ael/pval.c + res/ael/ael.tab.c res/ael/ael.tab.h res/ael/ael.y + res/ael/ael_lex.c res/ael/ael.flex utils/ael_main.c + utils/extconf.c utils/conf2ael.c utils/Makefile Had to change the + interface to ast_compile_ael2(), to include the hashtab ptr. This + ended up involving several external apps. The main gotcha was I + had to include lock.h and hashtab.h in several places. As a side + note, I tested this stuff pretty thoroughly, I replicated the + problems originally reported by Luigi, and made triply sure that + reloads worked, and everything worked thru "stop gracefully". I + found a and fixed a few bugs as I was merging into trunk, that + did not appear in my tests of bug6002. How's this for verbose + commit messages? + +2008-03-07 17:17 +0000 [r106684-106707] Russell Bryant <russell@digium.com> + + * /, include/asterisk/sched.h: Merged revisions 106704 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 + Mar 2008) | 8 lines Change a warning message to a debug message. + This is happening quite frequently, and it is not worth spamming + users with these messages unless we are pretty confident that it + should never happen. As it stands today, it _will_ and _does_ + happen and until that gets cleaned up a reasonable amount on the + development side, let's not spam the logs of everyone else. + (closes issue #12154) ........ + + * doc/smdi.txt: fix example usage + +2008-03-07 16:26 +0000 [r106553-106654] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 106635 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r106635 | tilghman | 2008-03-07 10:22:11 -0600 (Fri, 07 + Mar 2008) | 3 lines Warn the user when a temporary greeting + exists (Closes issue #11409) ........ + + * /, main/rtp.c: Merged revisions 106606 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) + | 3 lines Properly initialize rtp->schedid (Closes issue #12154) + ........ + + * main/channel.c, funcs/func_enum.c, channels/chan_misdn.c, + main/frame.c, /, channels/chan_sip.c, funcs/func_odbc.c, + funcs/func_strings.c, utils/extconf.c, apps/app_chanspy.c, + apps/app_rpt.c, main/asterisk.c, apps/app_speech_utils.c, + apps/app_voicemail.c: Merged revisions 106552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) + | 6 lines Safely use the strncat() function. (closes issue + #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt + uploaded by Corydon76 (license 14) ........ + +2008-03-07 01:19 +0000 [r106501-106518] Russell Bryant <russell@digium.com> + + * doc/smdi.txt: minor text changes + + * doc/smdi.txt: Add updated SMDI documentation that I had only + sitting in my email ... oops + + * codecs/codec_g722.c, formats/format_pcm.c, main/file.c, + main/rtp.c: Merge changes from team/russell/g722-sillyness ... + Fix a number of other places where the number of samples in a + G722 frame was not properly handled because of various reasons. + main/rtp.c: - When a G722 frame is read from the smoother, the + number of samples in the frame must be divided by 2 before being + sent out over the network. Even though G722 is 16 kHz, an error + in some previous spec has made it so that we have to list the + number of samples such as if it was 8 kHz. main/file.c: - When + scheduling the next time to expect a frame, take into account + that the format of the file we're reading from may not be 8 kHz. + codecs/codec_g722.c: - When converting from G722 to slinear, + g722_decode() expects its samples parameter to be in the silly + (real samples / 2) format. Make it so. - When converting from + slinear to G722, properly set the number of samples in the frame + to be the number of bytes of output * 2. formats/format_pcm.c: - + This format module handles G722, among a number of other formats. + However, the read() and seek() functions did not account for the + fact that G722 has 2 samples per byte. (closes issue #12130, + reported by rickross, patched by me) + +2008-03-06 22:11 +0000 [r106439] Jason Parker <jparker@digium.com> + + * main/file.c: Fix file playback in many cases. (closes issue + #12115) Reported by: pj Patches: v2-fileexists.patch uploaded by + dimas (license 88) (with modifications by me) Tested by: dimas, + qwell, russell + +2008-03-06 22:11 +0000 [r106438] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c, /: Merged revisions 106437 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar + 2008) | 8 lines Quell an annoying message that is likely to print + every single time that ast_pbx_outgoing_app is called. The reason + is that __ast_request_and_dial allocates the cdr for the channel, + so it should be expected that the channel will have a cdr on it. + Thanks to joetester on IRC for pointing this out ........ + +2008-03-06 19:31 +0000 [r106399] Donny Kavanagh <donnyk@gmail.com> + + * res/res_agi.c: trivial fix for an agi error when attempting to + use EAGI on a dead/hungup channel, we now print an error that + makes sense given our removal of deadagi as an actual + application. (closes issue #12161) Reported by: explidous + Patches: res_agi_12161.patch uploaded by juggie (license 24) + Tested by: juggie + +2008-03-06 05:21 +0000 [r106329-106346] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_ldap.c: Missing braces, fix parsing (closes issue + #12112) Reported by: cyrenity Patches: + res_config_ldap.patch-03-03-2008 uploaded by cyrenity (license + 416) Tested by: cyrenity, Corydon76 + + * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 106328 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106328 | tilghman | 2008-03-05 22:40:06 -0600 (Wed, 05 Mar 2008) + | 2 lines Upgrade to the next release of sounds ........ + +2008-03-05 23:21 +0000 [r106250] Jason Parker <jparker@digium.com> + + * Makefile: Add a cmenuselect/cmenuconfig, to force curses. + +2008-03-05 22:50 +0000 [r106240] Russell Bryant <russell@digium.com> + + * Makefile, /: Add the nmenuselect makefile targets. This is a newt + menuselect interface which was written by seanbright. It is much + sexier than my curses one. :) (issue #12139) + +2008-03-05 22:43 +0000 [r106239] Joshua Colp <jcolp@digium.com> + + * channels/chan_unistim.c, apps/app_dial.c, channels/chan_zap.c, /, + channels/chan_sip.c, channels/chan_console.c, + apps/app_followme.c, channels/chan_oss.c, main/rtp.c, + main/channel.c, channels/chan_phone.c, main/dial.c, + channels/chan_skinny.c, main/file.c, channels/chan_h323.c, + channels/chan_alsa.c, include/asterisk/frame.h, + channels/chan_mgcp.c: Merged revisions 106235 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 + lines Add a control frame to indicate the source of media has + changed. Depending on the underlying technology it may need to + change some things. (closes issue #12148) Reported by: jcomellas + ........ + +2008-03-05 22:40 +0000 [r106238] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 106237 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 + Mar 2008) | 3 lines Fix a potential deadlock and a few different + potential crashes. (closes issue #12145, reported by thiagarcia, + patched by me) ........ + +2008-03-05 22:33 +0000 [r106236] Mark Michelson <mmichelson@digium.com> + + * doc/manager_1_1.txt, main/manager.c, CHANGES: Adding the Atxfer + manager command. With this, you may initiate an attended transfer + over AMI (closes issue #10585) Reported by: ornati Patches: + atxfer-trunk-r90428.diff uploaded by ornati (license 210) (with + modifications from me) Tested by: putnopvut + +2008-03-05 21:19 +0000 [r106186] Michiel van Baak <michiel@vanbaak.info> + + * doc/tex/realtime.tex: document var_metric usage to prevent + bugreports that are actually configuration issues (closes issue + #12151) Reported by: caio1982 Patches: DB_metric3.diff uploaded + by caio1982 (license 22) + +2008-03-05 17:40 +0000 [r106139] Tilghman Lesher <tlesher@digium.com> + + * apps/app_talkdetect.c: Should check these values for non-NULL + before scanning. (Closes issue #12147) + +2008-03-05 16:39 +0000 [r106110] Joshua Colp <jcolp@digium.com> + + * main/dsp.c: Fix code up to what it was meant to be. + +2008-03-05 16:23 +0000 [r106072] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, apps/app_meetme.c, apps/app_waitforsilence.c, + apps/app_record.c, UPGRADE.txt, apps/app_followme.c, CHANGES, + main/loader.c, configs/dsp.conf.sample (added), + apps/app_minivm.c, res/res_agi.c, include/asterisk/dsp.h, + main/app.c, apps/app_amd.c, main/asterisk.c, main/dsp.c, + apps/app_voicemail.c: Create a centralized configuration option + for silencethreshold (closes issue #11236) Reported by: philipps + Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 + (license 14) Tested by: philipps + +2008-03-05 15:40 +0000 [r106040] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, /: Merged revisions 106038 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar + 2008) | 7 lines when a PRI call must be moved to a different B + channel at the request of the other endpoint, ensure that any DSP + active on the original channel is moved to the new one (closes + issue #11917) Reported by: mavetju Tested by: mavetju ........ + +2008-03-05 15:23 +0000 [r106036] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c, include/asterisk/sched.h: Merged + revisions 106015 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) + | 7 lines Correctly initialize retransid in SIP, and ensure that + the warning when failing to delete a schedule entry can actually + hit the log. (closes issue #12140) Reported by: slavon Patches: + sch2.patch uploaded by slavon (license 288) (Patch slightly + modified by me) ........ + +2008-03-05 04:34 +0000 [r105899-105984] Russell Bryant <russell@digium.com> + + * apps/app_queue.c: - simplify a few statements with ARRAY_LEN() - + constify the stregy int to string mappings array + + * /, main/translate.c, include/asterisk/frame.h, main/rtp.c: Merged + revisions 105932 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) + | 5 lines Fix a bug that I just noticed in the RTP code. The + calculation for setting the len field in an ast_frame of audio + was wrong when G.722 is in use. The len field represents the + number of ms of audio that the frame contains. It would have set + the value to be twice what it should be. ........ + + * funcs/func_global.c: Fix the SHARED() read callback to properly + unlock the channel. This function could not have worked, as it + left the channel locked in all cases. + +2008-03-04 23:24 +0000 [r105864] Mark Michelson <mmichelson@digium.com> + + * main/manager.c: There are several places in manager.c where + BUFSIZ is used for a buffer which will contain nowhere near that + amount of data. This makes these buffers more reasonably sized. + +2008-03-04 23:10 +0000 [r105840-105841] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_zap.c, channels/console_gui.c, apps/app_queue.c, + main/asterisk.c: Fix minor misuses of snprintf + + * main/frame.c, main/say.c, main/utils.c, main/astobj2.c, + main/enum.c, main/fskmodem.c, main/config.c, main/poll.c, + main/loader.c, main/term.c, main/cli.c, main/channel.c, + main/dial.c, main/manager.c, main/tdd.c, main/strcompat.c, + main/features.c, main/logger.c, main/app.c, main/image.c, + main/dns.c, main/pbx.c, main/translate.c, main/jitterbuf.c, + main/rtp.c, main/netsock.c, main/cryptostub.c, main/callerid.c, + main/file.c, main/alaw.c, main/dlfcn.c, main/dsp.c: Whitespace + changes only + +2008-03-04 22:28 +0000 [r105734-105804] Russell Bryant <russell@digium.com> + + * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c, + include/asterisk/tcptls.h: add a destroy API call for a server + instance + + * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c, + include/asterisk/tcptls.h: More public API name changes to use an + appropriate ast_ prefix + + * include/asterisk/http.h, main/tcptls.c, main/manager.c, + channels/chan_sip.c, res/res_phoneprov.c, main/http.c, + include/asterisk/tcptls.h: Rename public object server_instance + to ast_tcptls_server_instance + + * channels/chan_sip.c: Fix some bugs in the SIP tcp helper thread. + - fix a spot where a lock wouldn't get unlocked in an error + condition - call ast_mutex_destroy() on the lock before freeing + its memory (related to issue #11972) + +2008-03-04 20:32 +0000 [r105733] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c: Set username to default to the category name + if it isn't overridden by a usernmae= setting in users.conf + +2008-03-04 18:11 +0000 [r105675-105677] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 105676 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 + lines In addition to setting the marker bit let's change our ssrc + so they know for sure it is a different source. ........ + + * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: + Merged revisions 105674 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 + lines When a new source of audio comes in (such as music on hold) + make sure the marker bit gets set. (closes issue #10355) Reported + by: wdecarne Patches: 10355.diff uploaded by file (license 11) + (closes issue #11491) Reported by: kanderson ........ + +2008-03-04 16:55 +0000 [r105595-105597] Russell Bryant <russell@digium.com> + + * CHANGES: Update CHANGES heading + + * funcs/func_version.c: Simplify a trivial snprintf() with + ast_copy_string() + +----- + * Changes above include everything done for 1.6.1 since 1.6.0 was branched + off. |