diff options
| author | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-08-11 21:32:27 +0000 |
|---|---|---|
| committer | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-08-11 21:32:27 +0000 |
| commit | 875863188836b1670fba9c518342bccf3982e4b5 (patch) | |
| tree | c3c00f17f1de9c54218f19f40de91b389dd03c04 | |
| parent | bfeed9e23f4f80f77c1f7d95e061ddd9b4ea251b (diff) | |
Importing files for 1.6.1.5-rc1 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.1.5-rc1@211724 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | .lastclean | 1 | ||||
| -rw-r--r-- | .version | 1 | ||||
| -rw-r--r-- | ChangeLog | 56863 |
3 files changed, 56865 insertions, 0 deletions
diff --git a/.lastclean b/.lastclean new file mode 100644 index 000000000..7facc8993 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +36 diff --git a/.version b/.version new file mode 100644 index 000000000..42ecaf5dc --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.6.1.5-rc1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..1962982af --- /dev/null +++ b/ChangeLog @@ -0,0 +1,56863 @@ +2009-08-11 Tilghman Lesher <tlesher@digium.com> + + * Asterisk 1.6.1.5-rc1 released + +2009-08-10 19:51 +0000 [r211569-211586] Tilghman Lesher <tlesher@digium.com> + + * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500 + (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10 + Aug 2009) | 1 line Conversion specifiers, not format specifiers + ........ ................ + + * channels/chan_iax2.c, main/channel.c, main/cdr.c, res/ael/pval.c, + apps/app_setcallerid.c, main/manager.c, apps/app_rpt.c, + main/asterisk.c, funcs/func_rand.c, apps/app_dahdibarge.c, + res/res_config_pgsql.c, funcs/func_timeout.c, + codecs/codec_speex.c, apps/app_record.c, apps/app_morsecode.c, + main/acl.c, funcs/func_cut.c, cdr/cdr_pgsql.c, + apps/app_followme.c, main/enum.c, res/res_config_sqlite.c, + agi/eagi-sphinx-test.c, main/config.c, channels/misdn_config.c, + channels/chan_dahdi.c, funcs/func_channel.c, apps/app_macro.c, + apps/app_sms.c, pbx/pbx_config.c, apps/app_verbose.c, main/dsp.c, + apps/app_voicemail.c, apps/app_adsiprog.c, funcs/func_speex.c, + channels/chan_sip.c, res/res_limit.c, agi/eagi-test.c, + funcs/func_math.c, channels/chan_agent.c, main/utils.c, + channels/iax2-provision.c, apps/app_talkdetect.c, + main/indications.c, channels/chan_oss.c, main/cli.c, + pbx/pbx_loopback.c, res/res_config_curl.c, channels/chan_misdn.c, + res/res_smdi.c, apps/app_osplookup.c, channels/chan_skinny.c, + pbx/pbx_dundi.c, utils/extconf.c, apps/app_mixmonitor.c, + channels/chan_mgcp.c, main/timing.c, doc/CODING-GUIDELINES, + main/pbx.c, utils/muted.c, apps/app_readfile.c, + apps/app_meetme.c, /, apps/app_privacy.c, apps/app_waituntil.c, + cdr/cdr_adaptive_odbc.c, res/res_http_post.c, pbx/dundi-parser.c, + res/res_musiconhold.c, apps/app_queue.c, main/netsock.c, + utils/frame.c, channels/chan_usbradio.c, funcs/func_enum.c, + channels/chan_phone.c, pbx/pbx_spool.c, apps/app_waitforring.c, + funcs/func_odbc.c, main/features.c, res/res_agi.c, + apps/app_minivm.c, main/http.c, res/snmp/agent.c, + res/res_config_ldap.c, apps/app_chanspy.c, apps/app_stack.c, + res/res_odbc.c, funcs/func_dialplan.c, main/dnsmgr.c, + main/frame.c, apps/app_waitforsilence.c, funcs/func_strings.c, + apps/app_disa.c, apps/app_alarmreceiver.c: AST-2009-005 + +2009-08-10 14:12 +0000 [r211349] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 | + file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix + retrieval of the port used for the video stream when adding SDP + to a SIP message. (closes issue #15121) Reported by: jsmith + ........ + +2009-08-09 15:43 +0000 [r211234-211277] Tilghman Lesher <tlesher@digium.com> + + * /, main/astfd.c: Merged revisions 211275 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009) + | 9 lines Merged revisions 211274 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009) + | 2 lines Small oops. Clear the flags which have been checked. + ........ ................ + + * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 | + tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines + Check for NULL frame, before dereferencing pointer. (closes issue + #15617) Reported by: rain ........ + +2009-08-07 20:17 +0000 [r211115] Russell Bryant <russell@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009) + | 11 lines Recorded merge of revisions 211112 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009) + | 4 lines Resolve a deadlock involving app_chanspy and + masquerades. (ABE-1936) ........ ................ + +2009-08-07 18:19 +0000 [r211047] Tilghman Lesher <tlesher@digium.com> + + * apps/app_queue.c, /: Merged revisions 211040 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009) + | 21 lines Merged revisions 211038 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) + | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name, + not the membername. This is a partial revert of revision 82590, + which was an attempted cleanup, but in reality, it broke + QUEUE_MEMBER_LIST, which has always been intended as a method by + which component interfaces could be queried from the queue. + Membername isn't useful here, because that field cannot be used + to obtain further information about the member. See the + documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, + QUEUE_MEMBER_PENALTY, and the various AMI commands which take a + member argument for further justification. (closes issue #15664) + Reported by: rain Patches: app_queue-queue_member_list.diff + uploaded by rain (license 327) ........ ................ + +2009-08-07 13:09 +0000 [r210994] Kevin P. Fleming <kpfleming@digium.com> + + * main/udptl.c, /: Merged revisions 210992 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 | + kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13 + lines Workaround broken T.38 endpoints that offer tiny + MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as + the maximum IFP size that should be sent to them, rather than the + maximum packet payload size. If such an endpoint also requests + UDPRedundancy as the error correction mode, we'll end up + calculating a tiny maximum IFP size, so small as to be unusable. + This patch sets a lower bound on what we'll consider the remote's + maximum IFP size to be, assuming that endpoints that do this + really can accept larger packets than they've offered to accept. + (closes issue #15649) Reported by: dazza76 ........ + +2009-08-06 21:47 +0000 [r210910-210916] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 210914 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009) + | 14 lines Merged revisions 210913 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009) + | 7 lines Because channel information can be accessed outside of + the channel thread, we must lock the channel prior to modifying + it. (closes issue #15397) Reported by: caspy Patches: + 20090714__issue15397.diff.txt uploaded by tilghman (license 14) + Tested by: caspy ........ ................ + + * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged + revisions 210908 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 | + tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines + Allow Gosub to recognize quote delimiters without consuming them. + (closes issue #15557) Reported by: rain Patches: + 20090723__issue15557.diff.txt uploaded by tilghman (license 14) + Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ + ........ + +2009-08-06 17:48 +0000 [r210819] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 | + file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines + Accept additional T.38 reinvites after an initial one has been + handled. Discussion of this subject has yielded that it is not + actually acceptable to change T.38 parameters after the initial + reinvite but declining is harsh and can cause the fax to fail + when it may be possible to allow it to continue. This patch + changes things so that additional T.38 reinvites are accepted but + parameter changes ignored. This gives the fax a fighting chance. + (closes issue #15610) Reported by: huangtx2009 ........ + +2009-08-05 20:28 +0000 [r210681] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500 + (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) + | 14 lines Dialplan starts execution before the channel setup is + complete. * Issue 15655: For the case where dialing is complete + for an incoming call, dahdi_new() was asked to start the PBX and + then the code set more channel variables. If the dialplan hungup + before these channel variables got set, asterisk would likely + crash. * Fixed potential for overlap incoming call to erroneously + set channel variables as global dialplan variables if the + ast_channel structure failed to get allocated. * Added missing + set of CALLINGSUBADDR in the dialing is complete case. (closes + issue #15655) Reported by: alecdavis ........ ................ + +2009-08-05 18:57 +0000 [r210567] Leif Madsen <lmadsen@digium.com> + + * doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 + (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) + | 11 lines Update imapstorage.txt documentation. Updated the + imapstorage.txt documentation to reflect that issues with + c-client versions older than 2007 seem to cause crashing issues + that are not seen with more recent versions. Documentation has + been updated to reflect this. (closes issue #14496) Reported by: + vbcrlfuser Patches: __20090727-imap-documentation-patch.txt + uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, + dbrooks ........ ................ + +2009-08-04 14:54 +0000 [r210240] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /: Merged revisions 210238 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug + 2009) | 16 lines Merged revisions 210237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug + 2009) | 10 lines Eliminate spurious compiler warnings from system + headers on *BSD platforms. Ensure that system headers located in + /usr/local/include are actually treated as system headers by the + compiler, and not as local headers which are subject to warnings + from the -Wundef compiler option and others. (closes issue + #15606) Reported by: mvanbaak ........ ................ + +2009-08-01 11:32 +0000 [r209836-209900] Russell Bryant <russell@digium.com> + + * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r209887 | russell | 2009-08-01 06:29:25 -0500 + (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009) + | 5 lines Resolve a valgrind warning about a read from + uninitialized memory. (issue #15396) Reported by: aragon ........ + ................ + + * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r209839 | russell | 2009-08-01 06:02:07 -0500 + (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) + | 13 lines Modify how Playtones() is used in Milliwatt() to + resolve gain issue. When Milliwatt() was changed internally to + use Playtones() so that the proper tone was used, it introduced a + drop in gain in the output signal. So, use the playtones API + directly and specify a volume argument such that the output + matches the gain of the original Milliwatt() code. (closes issue + #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff + uploaded by russell (license 2) Tested by: rue_mohr ........ + ................ + + * /, main/event.c: Merged revisions 209835 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 | + russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines + Fix ast_event_queue_and_cache() to actually do the cache() part. + (closes issue #15624) Reported by: ffossard Tested by: russell + ........ + +2009-08-01 01:25 +0000 [r209781] Kevin P. Fleming <kpfleming@digium.com> + + * channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile, + channels/misdn/ie.c: Merged revisions 209760-209761 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 + (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul + 2009) | 7 lines Minor changes inspired by testing with latest + GCC. The latest GCC (what will become 4.5.x) has a few new + warnings, that in these cases found some either downright buggy + code, or at least seriously poorly designed code that could be + improved. ........ ................ r209761 | kpfleming | + 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert + accidental Makefile change. ................ + +2009-07-31 21:58 +0000 [r209714] Russell Bryant <russell@digium.com> + + * /, main/event.c: Merged revisions 209711 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 | + russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines + Fix some places where ast_event_type was used instead of + ast_event_ie_type. ........ + +2009-07-30 18:46 +0000 [r209593] David Brooks <dbrooks@digium.com> + + * include/asterisk/abstract_jb.h, channels/chan_dahdi.c, + contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, + codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions + 209554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | + dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines + Fixes numerous spelling errors. Patch submitted by alecdavis. + (closes issue #15595) Reported by: alecdavis ........ + +2009-07-30 14:40 +0000 [r209517] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 | + mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8 + lines Fix a crash that can result if text codecs are allowed but + textsupport is disabled. (closes issue #15596) Reported by: + fabled Patches: sip-red.patch uploaded by fabled (license 448) + ........ + +2009-07-28 00:19 +0000 [r209327] Tilghman Lesher <tlesher@digium.com> + + * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009) + | 9 lines Merged revisions 209315 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) + | 2 lines Publish French extra sounds ........ ................ + +2009-07-27 21:44 +0000 [r209262-209281] Kevin P. Fleming <kpfleming@digium.com> + + * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 | + kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7 + lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE + messages about T.38 negotiation in debug level 1 messages, clean + up some looping logic, and correct an improper use of ast_free() + for freeing an ast_frame. ........ + + * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 | + kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10 + lines Make T.38 switchover in ReceiveFAX synchronous. In receive + mode, if the channel that ReceiveFAX is running on supports T.38, + we should *always* attempt to switch T.38, rather than listening + for an incoming CNG tone and only triggering on that. The channel + may be using a low-bitrate codec that distorts the CNG tone, the + sending FAX endpoint may not send CNG at all, or there could be a + variety of other reasons that we don't detect it, but in all + those cases if T.38 is available we certainly want to use it. + ........ + +2009-07-27 20:57 +0000 [r209237] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /: Merged revisions 209235 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 | + mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5 + lines Gracefully handle malformed RTP text packets. AST-2009-004 + ........ + +2009-07-27 20:28 +0000 [r209233] David Brooks <dbrooks@digium.com> + + * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c, + channels/chan_vpb.cc, res/res_smdi.c, /, + include/asterisk/module.h, main/features.c, res/res_agi.c: Merged + revisions 209098 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | + dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines + Fixing typos. Replaces "recieved" with "received" and "initilize" + with "initialize" (closes issue #15571) Reported by: alecdavis + ........ + +2009-07-27 20:17 +0000 [r209134-209199] Mark Michelson <mmichelson@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul + 2009) | 9 lines Honor channel's music class when using realtime + music on hold. (closes issue #15051) Reported by: alexh Patches: + 15051.patch uploaded by mmichelson (license 60) Tested by: alexh + ........ + + * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions + 209132 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul + 2009) | 24 lines Merged revisions 209131 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul + 2009) | 18 lines Allow for UDPTL to use only even-numbered ports + if desired. There are some VoIP providers out there that will not + accept SDP offers with odd numbered UDPTL ports. While it is my + personal opinion that these VoIP providers are misinterpreting + RFC 2327, it really is not a big deal to play along with their + silly little games. Of course, since restricting UDPTL ports to + only even numbers reduces the range of available ports by half, + so the option to use only even port numbers is off by default. A + user can enable the behavior by setting use_even_ports=yes in + udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: + 15182.patch uploaded by mmichelson (license 60) Tested by: + CGMChris ........ ................ + +2009-07-27 15:40 +0000 [r209058] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /: Merged revisions 209056 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 | + kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10 + lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and + underscore-variants to sub-makes. During the recent Makefile + improvements I made, it seemed the 'make' was automatically + carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so + I removed the explict export of them. However, there are some + circumstances where make does this, and some where it does not, + so I've brought them back to ensure they are always exported. I + also removed an extraneous double setting of _ASTLDFLAGS on *BSD + platforms. ........ + +2009-07-27 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.1.2 released + +2009-06-05 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.1.1 released + +2009-06-04 David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: Additional updates for AST-2009-001 + +2009-06-04 David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001 + +2009-04-27 Leif Madsen <lmadsen@digium.com> + + * Create Asterisk 1.6.1.0 + +2009-04-20 Leif Madsen <lmadsen@digium.com> + + * Create Asterisk 1.6.1.0-rc5 + +2009-04-20 17:08 +0000 [r189352] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 | + file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines + Fix a bug with non-UDP connections that caused dialogs to not get + freed. This issue crept up because of a reference count issue on + non-UDP based dialogs. The dialog reference count was increased + when transmitting a packet reliably but never decreased. This + caused the dialog structure to hang around despite being unlinked + from the dialogs container. (closes issue #14919) Reported by: + vrban ........ + +2009-04-20 14:06 +0000 [r189280] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 189278 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr + 2009) | 18 lines Merged revisions 189277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr + 2009) | 12 lines Move the check for chan->fdno == -1 to after the + zombie/hangup check. Many users were finding that their hung up + channels were staying up and causing 100% CPU usage. (issue + #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch + uploaded by mmichelson (license 60) Tested by: falves11, bamby + ........ ................ + +2009-04-18 01:38 +0000 [r189206] David Vossel <dvossel@digium.com> + + * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500 + (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) + | 12 lines Fixed autologoff in agents.conf not working when agent + logs in via AgentLogin app An agent logs in by calling an + extension that calls the AgentLogin app. In agents.conf + ackcall=always is set, so when they get a call they have the + choice to either acknowledge it or ignore it. autologoff=10 is + set as well, so if the agent ignores the call over 10sec one may + assume that the agent should be logged out (and in this case + hungup on as well), but this was not happening. (closes issue + #14091) Reported by: evandro Patches: autologoff.diff uploaded by + dvossel (license 671) Review: + http://reviewboard.digium.com/r/225/ ........ ................ + +2009-04-17 21:55 +0000 [r189139] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 189137 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009) + | 17 lines Merged revisions 188833,189134 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) + | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. + Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | + rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines + Modifed/added some debug messages. JIRA ABE-1835 ........ + ................ + +2009-04-17 20:21 +0000 [r189103] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 | + mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 + lines Prevent a crash when SIP blonde transferring an unbridged + call. If one attempts to use the attended transfer button on a + SIP phone to transfer an unbridged call (such as a call to an + IVR) but hangs up while the target of the transfer is still + ringing, we need to not crash. The problem was that ast_hangup + was called from outside the channel thread. AST-211 ........ + +2009-04-17 19:46 +0000 [r189080] Sean Bright <sean.bright@gmail.com> + + * main/asterisk.c, /: Merged revisions 189077 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 | + seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1 + line Fix copy/paste error with 'transmit silence' flag. ........ + +2009-04-17 17:33 +0000 [r189069] Matthew Nicholson <mnicholson@digium.com> + + * main/pbx.c, /: Merged revisions 189010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr + 2009) | 12 lines Merged revisions 189009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr + 2009) | 5 lines Make Busy() application set the CDR disposition + to BUSY. (closes issue #14306) Reported by: cristiandimache + ........ ................ + +2009-04-17 14:48 +0000 [r188940-188949] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | + 22 lines Merged revisions 188946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | + 15 lines Fix a bug where a value used to create the channel name + was bogus. This commit fixes the scenario where an incoming call + is authenticated using a peer entry. Previously the channel name + was created using either the username setting from the sip.conf + entry or the IP address that the call came from. Now the channel + name will be created using the peer name itself. This commit will + not change the way the channel name is generated for users or + friends. (closes issue #14256) Reported by: Nick_Lewis Patches: + chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: + Nick_Lewis, file ........ ................ + + * channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri, + 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 + lines Fix a situation where the DAHDI channel private structure + lock was not unlocked when it should have been. (issue AST-210) + ........ ................ + +2009-04-16 22:05 +0000 [r188776-188838] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009) + | 14 lines Merged revisions 188835 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) + | 7 lines Only update realtime, if global option rtupdate != + false (closes issue #14885) Reported by: deepesh Patches: + 20090413__bug14885.diff.txt uploaded by tilghman (license 14) + Tested by: deepesh ........ ................ + + * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500 + (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009) + | 4 lines Umask should not be exported into global namespace. + (closes issue #14912) Reported by: jcapp ........ + ................ + +2009-04-15 22:12 +0000 [r188649] David Vossel <dvossel@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500 + (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) + | 12 lines National prefix inserted even when caller ID not + available When the caller ID is restricted, the expected behavior + is for the caller id to be blank. In chan_dahdi, the national + prefix is placed onto the callers number even if its restricted + (empty) causing the caller id to be the national prefix rather + than blank. (closes issue #13207) Reported by: shawkris Patches: + national_prefix.diff uploaded by dvossel (license 671) Review: + http://reviewboard.digium.com/r/220/ ........ ................ + +2009-04-15 20:20 +0000 [r188473-188596] Mark Michelson <mmichelson@digium.com> + + * /, main/file.c: Merged revisions 188585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr + 2009) | 13 lines Merged revisions 188582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr + 2009) | 7 lines Update ast_readvideo_callback to match + ast_readaudio_callback. This fixes potential refcount errors that + may occur on ast_filestreams. AST-208 ........ ................ + + * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 | + mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3 + lines Fix a couple of queue member reference leaks. ........ + +2009-04-14 17:43 +0000 [r188254-188415] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, /: Merged revisions 188413 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 | + file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix + an incorrect clock rate when sending T140 text. (closes issue + #14029) Reported by: epicac ........ + + * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 | + file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix + a bug with the change I made yesterday to outbound proxy support. + Per discussion with oej on IRC we need the actual IP address, not + the outbound proxy IP address, in the sa field. Upon further + inspection this should make the behaviour of all other uses of + the outbound proxy in the code. ........ + +2009-04-14 05:46 +0000 [r188208-188212] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 188210 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 | + tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines + As suggested by Russell, warn users when their dialplan arguments + contain pipes, but not commas. ........ + + * /, utils/smsq.c: Merged revisions 188206 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 | + tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines + Application delimiter is ',', not '|'. (closes issue #14881) + Reported by: stegro Patches: smsq.patch uploaded by stegro + (license 752) ........ + +2009-04-13 19:33 +0000 [r188104] Mark Michelson <mmichelson@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 188102 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr + 2009) | 5 lines Fix another crash related to cached realtime + music on hold. This was another off-by-one problem caused by + moh_register. ........ + +2009-04-13 16:32 +0000 [r188069] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 | + file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines + Fix a bug where using an outbound proxy would cause the local + address to be 127.0.0.1. Copy the outbound proxy IP address into + the SIP dialog structure as the IP address we will be sending to. + This has to be done because the logic that determines what local + IP address to use in the SIP messages is not aware of an outbound + proxy being in place. It only knows what IP address we are + sending to. (closes issue #12006) Reported by: mnicholson + ........ + +2009-04-13 14:20 +0000 [r188038] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 | + mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6 + lines Set all queue variables on both the caller and member + channels. This allows for the variables to be accessed if a + member macro is run. Thanks to Grigoriy Puzankin for bringing + this up on the -dev list. ........ + +2009-04-10 20:28 +0000 [r187914] Jeff Peeler <jpeeler@digium.com> + + * channels/Makefile, /: Merged revisions 187906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 | + jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines + Fix module embedding for chan_h323. Include libchanh323.a in the + modules.link file so that all the symbols can be resolved at link + time. (closes issue #11966) Reported by: dome Patches: + issue_11966.patch uploaded by kpfleming (license 421) Tested by: + jpeeler ........ + +2009-04-10 17:30 +0000 [r187767] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/sip-friends.sql, + contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500 + (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 + Apr 2009) | 2 lines Add lastms column to the contributed table + designs ........ ................ + +2009-04-10 16:54 +0000 [r187723] Kevin P. Fleming <kpfleming@digium.com> + + * /, build_tools/embed_modules.xml: Merged revisions 187721 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10 + Apr 2009) | 5 lines clean up some patterns for files to remove + add embedding support for bridge and test modules ........ + +2009-04-10 16:03 +0000 [r187678] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 | + tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines + Ensure pvt is not NULL before dereferencing it. (closes issue + #14784) Reported by: pj ........ + +2009-04-10 16:00 +0000 [r187676] Russell Bryant <russell@digium.com> + + * tests/test_heap.c, /: Merged revisions 187675 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 | + russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines + Disable test modules by default. ........ + +2009-04-10 03:56 +0000 [r187600] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, main/pbx.c, main/manager.c, /, + include/asterisk/linkedlists.h, main/features.c, main/http.c, + main/app.c, include/asterisk/lock.h, main/audiohook.c: Merged + revisions 187599 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r187599 | + tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines + Modify headers and macros, according to Russell's suggestions on + the -dev list ........ + +2009-04-09 19:14 +0000 [r187495] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 187488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r187488 | mmichelson | 2009-04-09 13:58:41 -0500 (Thu, 09 Apr + 2009) | 24 lines Merged revisions 187484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr + 2009) | 18 lines Handle a SIP race condition (reinvite before an + ACK) properly. RFC 5047 explains the proper course of action to + take if a reINVITE is received before the ACK from a previous + invite transaction. What we are to do is to treat the reINVITE as + if it were both an ACK and a reINVITE and process it normally. + Later, when we receive the ACK we had been expecting, we will + ignore it since its CSeq is less than the current iseqno of the + sip_pvt representing this dialog. (closes issue #13849) Reported + by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson + (license 60) Tested by: mmichelson, klaus3000 ........ + ................ + +2009-04-09 18:54 +0000 [r187486] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /, include/asterisk/linkedlists.h, + include/asterisk/lock.h: Merged revisions 187483 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500 + (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) + | 8 lines Race condition between ast_cli_command() and 'module + unload' could cause a deadlock. Add lock timeouts to avoid this + potential deadlock. (closes issue #14705) Reported by: jamessan + Patches: 20090320__bug14705.diff.txt uploaded by tilghman + (license 14) Tested by: jamessan ........ ................ + +2009-04-09 17:43 +0000 [r187427] Mark Michelson <mmichelson@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 187421,187424 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu, + 09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using + cached realtime moh. The moh_register function links an mohclass + and then immediately unrefs the class since the container now has + a reference. The problem with using realtime music on hold is + that the class is allocated, registered, and started in one fell + swoop. The refcounting logic resulted in the count being off by + one. The same problem did not happen when using a static config + because the allocation and registration of an mohclass is a + separate operation from starting moh. This also did not affect + non-cached realtime moh because the classes are not registered at + all. I also have modified res_musiconhold to use the _t_ variants + of the ao2_ functions so that more info can be gleaned when + attempting to trace the refcounts. I found this to be incredibly + helpful for debugging this issue and there's no good reason to + remove it. (closes issue #14661) Reported by: sum ........ + r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr + 2009) | 3 lines Use safe macro practices even though they really + aren't necessary. ........ + +2009-04-09 17:22 +0000 [r187305-187388] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 | + tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines + Allow '/' in username portion of register; this is a regression. + (closes issue #14668) Reported by: Netview ........ + + * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions + 187363 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) + | 10 lines Merged revisions 187362 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) + | 3 lines Permit zero-length text messages in SIP. (Related to an + issue posted to the -users list, subject "AEL2, BASE64_DECODE and + hexadecimal") ........ ................ + + * main/asterisk.c, agi/Makefile, build_tools/cflags.xml, + utils/Makefile, include/asterisk.h, /, main/Makefile, + main/file.c, main/astfd.c (added): Merged revisions 187302 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500 + (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) + | 3 lines Add debugging mode for diagnosing file descriptor + leaks. (Related to issue #14625) ........ r187301 | tilghman | + 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, + missed this file in the last commit. ........ ................ + +2009-04-08 16:53 +0000 [r186987-187048] Mark Michelson <mmichelson@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 187046 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500 + (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr + 2009) | 10 lines Fix a small logical error when loading moh + classes. We were unconditionally incrementing the number of + mohclasses registered. However, we should actually only increment + if the call to moh_register was successful. While this probably + has never caused problems, I noticed it and decided to fix it + anyway. ........ ................ + + * main/channel.c, /: Merged revisions 186985 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr + 2009) | 30 lines Merged revisions 186984 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr + 2009) | 24 lines Make a couple of changes with regards to a new + message printed in ast_read(). "ast_read() called with no + recorded file descriptor" is a new message added after a bug was + discovered. Unfortunately, it seems there are a bunch of places + that potentially make such calls to ast_read() and trigger this + error message to be displayed. This commit does two things to + help to make this message appear less. First, the message has + been downgraded to a debug level message if dev mode is not + enabled. The message means a lot more to developers than it does + to end users, and so developers should take an effort to be sure + to call ast_read only when a channel is ready to be read from. + However, since this doesn't actually cause an error in operation + and is not something a user can easily fix, we should not spam + their console with these messages. Second, the message has been + moved to after the check for any pending masquerades. ast_read() + being called with no recorded file descriptor should not + interfere with a masquerade taking place. This could be seen as a + simple way of resolving issue #14723. However, I still want to + try to clear out the existing ways of triggering this message, + since I feel that would be a better resolution for the issue. + ........ ................ + +2009-04-08 05:07 +0000 [r186900] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 | + tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines + Add lastms to the require API call. ........ + +2009-04-08 00:10 +0000 [r186835-186844] Mark Michelson <mmichelson@digium.com> + + * /, formats/format_wav.c, formats/format_wav_gsm.c: Merged + revisions 186842 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr + 2009) | 14 lines Merged revisions 186841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr + 2009) | 8 lines Fix a few typos of the word "frequency." (closes + issue #14842) Reported by: jvandal Patches: frequency-typo.diff + uploaded by jvandal (license 413) ........ ................ + + * /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 | + mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 + lines Fix bad merge from fix for issue 13867. (closes issue + #14686) Reported by: davidw ........ + + * main/channel.c, /: Merged revisions 186833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr + 2009) | 15 lines Merged revisions 186832 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr + 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a + p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, + warning sounds will not be properly played to either party of the + bridge. (closes issue #14845) Reported by: adomjan ........ + ................ + +2009-04-07 22:33 +0000 [r186806] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_macro.c: Merged revisions 186799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) + | 10 lines Merged revisions 186775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) + | 3 lines Fix Macro documentation to match current (and intended) + behavior. (See -dev mailing list) ........ ................ + +2009-04-07 20:53 +0000 [r186722] Mark Michelson <mmichelson@digium.com> + + * main/manager.c, /: Merged revisions 186720 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr + 2009) | 12 lines Merged revisions 186719 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr + 2009) | 6 lines Ensure that \r\n is printed after the ActionID in + an OriginateResponse. (closes issue #14847) Reported by: kobaz + ........ ................ + +2009-04-03 20:21 +0000 [r186466] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500 + (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr + 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not + properly switch formats when requested Don't offer + AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could + provide a slight performance benefit, the translation core in + Asterisk has some flaws when a channel driver offers multiple raw + formats. this fix is much simpler than fixing the translation + core to solve that issue (although that will be done later). + ........ ................ + +2009-04-03 20:04 +0000 [r186448] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged + revisions 186444,186447 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) + | 14 lines Merged revisions 186415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) + | 7 lines Distinguish in a sent email between simple sends and + forwards. (closes issue #11678) Reported by: jamessan Patches: + 20090330__bug11678.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman, lmadsen ........ ................ r186447 | + tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines + Merged revisions 186445 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) + | 2 lines Found a conflict in the last commit, due to multiple + targets ........ ................ + +2009-04-03 16:38 +0000 [r186381] David Vossel <dvossel@digium.com> + + * /, main/audiohook.c: Merged revisions 186379 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 | + dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 4 lines + audio_audiohook_write_list() did not correctly update sample size + after ast_translate. audio_audiohook_write_list() did not take + into account that the sample size may change after translation + depending on if the original frame is is 8khz or 16khz. the + sample size is now updated after translating to reflect this + possibility. This caused the audio on the receiving end to sound + terrible. Thanks to jcolp and mmichelson for helping me work this + out. (issue AST-197) ........ + +2009-04-03 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.1.0-rc4 released. + +2009-04-03 15:54 +0000 [r186323] Joshua Colp <jcolp@digium.com> + + * include/asterisk/crypto.h, /: Merged revisions 186321 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, + 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 + lines Fix a problem with the crypto variable definitions not + actually being defined properly. (closes issue #14804) Reported + by: jvandal ........ ................ + +2009-04-03 14:33 +0000 [r186288] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr + 2009) | 20 lines Fix the ability to retrieve voicemail messages + from IMAP. A recent change made interactive vm_states no longer + get added to the list of vm_states and instead get stored in + thread-local storage. In trunk and all the 1.6.X branches, the + problem is that when we search for messages in a voicemail box, + we would attempt to update the appropriate vm_state struct by + directly searching in the list of vm_states instead of using the + get_vm_state_by_imap_user function. This meant we could not find + the interactive vm_state that we wanted. (closes issue #14685) + Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson + (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........ + +2009-04-03 02:06 +0000 [r186232] Russell Bryant <russell@digium.com> + + * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009) + | 29 lines Merged revisions 186229 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) + | 21 lines Fix a memory leak in cdr_radius. I came across this + while doing some testing of my ast_channel_ao2 branch. After + running a test overnight that generated over 5 million calls, + Asterisk had taken up about 1 GB of my system memory. So, I + re-ran the test with MALLOC_DEBUG turned on. However, it showed + no leaks in Asterisk during the test, even though Asterisk was + still consuming it somehow. Instead, I turned to valgrind, which + when run with --leak-check=full, told me exactly where the leak + came from, which was from allocations inside the radiusclient-ng + library. This explains why MALLOC_DEBUG did not report it. After + a bit of analysis, I found that we were leaking a little bit of + memory every time a CDR record was passed to cdr_radius. I don't + actually have a radius server set up to receive CDR records. + However, I always have my development systems compile and install + all modules. In addition to making sure there are not build + errors across modules, always loading modules helps find bugs + like this, too, so it is strongly recommend for all developers. + ........ ................ + +2009-04-02 21:59 +0000 [r186177] Mark Michelson <mmichelson@digium.com> + + * configs/features.conf.sample, /: Merged revisions 186175 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500 + (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr + 2009) | 5 lines Fix instructions in one-step parking comment to + make more sense. Changed a capital K to a lowercase k. ........ + ................ + +2009-04-02 17:27 +0000 [r186108] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500 + (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 + Apr 2009) | 3 lines ensure that the buffer passed to + DAHDI_SET_BUFINFO is fully initialized ........ ................ + +2009-04-02 17:14 +0000 [r186062] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 186060 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) + | 16 lines Merged revisions 186059 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 + (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 + Apr 2009) | 2 lines Fix for AST-2009-003 ........ + ................ ................ + +2009-04-02 13:53 +0000 [r185956] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500 + (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr + 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and + DAHDI_GET_PARAMS ioctls were recently corrected to show that they + do, in fact, read data from userspace as part of their work. due + to this fix, valgrind now reports a number of cases where + chan_dahdi passed an uninitialized (or partially) buffer to these + ioctls, which could lead to unexpected behavior. this patch + corrects chan_dahdi to ensure that buffers passed to these ioctls + are always fully initialized. ........ ................ + +2009-04-01 19:06 +0000 [r185848] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) + | 16 lines Merged revisions 185845 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) + | 10 lines Fixes issue with dropped calles due to re-Invite glare + and re-Invites never executing after a 491 Acknowledgement for + 491 responses were never being processed because it didn't match + our pending invite's seqno. Since the ACK was never processed, + the 491 frame would continue to be retransmitted until eventually + the call was dropped due to max retries. Now during a pending + invite, if we receive another invite, we send an 491 and hold on + to that glare invite's seqno in the "glareinvite" variable for + that sip_pvt struct. When ACK's are received, we first check to + see if it is in response to our pending invite, if not we check + to see if it is in response to a glare invite. In this case, it + is in response to the glare invite and must be dealt with or the + call is dropped. I've changed the wait time for resending the + re-Invite after receving a 491 response to comply with RFC 3261. + Before this patch the scheduled re-Invite would only change a + flag indicating that the re-Invite should be sent out, now it + actually sends it out as well. (closes issue #12013) Reported by: + alx Review: http://reviewboard.digium.com/r/213/ ........ + ................ + +2009-04-01 13:50 +0000 [r185774] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 185772 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009) + | 14 lines Merged revisions 185771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) + | 6 lines Fix a case where DTMF could bypass audiohooks. This + change fixes a situation where an audiohook that wants DTMF would + not actually get it. This is in the code path where we end DTMF + digit length emulation while handling a NULL frame. ........ + ................ + +2009-03-31 22:38 +0000 [r185666] Kevin P. Fleming <kpfleming@digium.com> + + * utils, /: Merged revisions 185664 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 | + kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line + ignore copied (generated) file ........ + +2009-03-31 22:05 +0000 [r185471-185602] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar + 2009) | 12 lines Merged revisions 185599 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar + 2009) | 6 lines Fix crash that would occur if an empty member was + specified in queues.conf. (closes issue #14796) Reported by: pida + ........ ................ + + * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500 + (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar + 2009) | 8 lines Fix Russian voicemail intro to say the word + "messages" properly. (closes issue #14736) Reported by: chappell + Patches: voicemail_no_messages.diff uploaded by chappell (license + 8) ........ ................ + +2009-03-31 17:48 +0000 [r185427] David Brooks <dbrooks@digium.com> + + * /, channels/chan_gtalk.c: Merged revisions 185363 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500 + (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) + | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains + extra whitespaces To drill into the xmpp to find the capabilities + between channels, chan_gtalk calls iks_child() and iks_next(). + iks_child() and iks_next() are functions in the iksemel xml + parsing library that traverse xml nodes. The bug here is that + both iks_child() and iks_next() will return the next iks_struct + node *regardless* of type. chan_gtalk expects the next node to be + of type IKS_TAG, which in most cases, it is, but in this case (a + call being made from the Empathy IM client), there exists + iks_struct nodes which are not IKS_TAG data (they are extraneous + whitespaces), and chan_gtalk doesn't handle that case, so + capabilities don't match, and a call cannot be made. + iks_first_tag() and iks_next_tag(), on the other hand, will not + return the very next iks_struct, but will check to see if the + next iks_struct is of type IKS_TAG. If it isn't, it will be + skipped, and the next struct of type IKS_TAG it finds will be + returned. This assures that chan_gtalk will find the iks_struct + it is looking for. This fix simply changes all calls to + iks_child() and iks_next() to become calls to iks_first_tag() and + iks_next_tag(), which resolves the capability matching. The + following is a payload listing from Empathy, which, due to the + extraneous whitespace, will not be parsed correctly by iksemel: + <iq from='dbrooksjab@235-22-24-10/Telepathy' + to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> + <session xmlns='http://www.google.com/session' + initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' + id='1837267342'> <description + xmlns='http://www.google.com/session/phone'> <payload-type + clockrate='16000' name='speex' id='96'/> <payload-type + clockrate='8000' name='PCMA' id='8'/> <payload-type + clockrate='8000' name='PCMU' id='0'/> <payload-type + clockrate='90000' name='MPA' id='97'/> <payload-type + clockrate='16000' name='SIREN' id='98'/> <payload-type + clockrate='8000' name='telephone-event' id='99'/> </description> + </session> </iq> Review: http://reviewboard.digium.com/r/181/ + ........ ................ + +2009-03-31 14:57 +0000 [r185263] Russell Bryant <russell@digium.com> + + * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 | + russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines + Don't free() an astobj2 object. (closes issue #14672) Reported + by: makoto ........ + +2009-03-31 14:10 +0000 [r185199] Joshua Colp <jcolp@digium.com> + + * /, main/audiohook.c: Merged revisions 185197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) | + 15 lines Merged revisions 185196 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 + lines Fix crash when moving audiohooks between channels. Handle + the scenario where we are called to move audiohooks between + channels and the source channel does not actually have any on it. + (closes issue #14734) Reported by: corruptor ........ + ................ + +2009-03-30 20:50 +0000 [r185126-185127] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged + revisions 185123 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) + | 9 lines Merged revisions 185121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) + | 1 line Update the channel allocation method documentation. + ........ ................ + + * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 + (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) + | 19 lines Make chan_misdn BRI TE side normally defer channel + selection to the NT side. Channel allocation collisions are not + handled by chan_misdn very well. This patch simply avoids the + problem for BRI only. For PRI, allocation collisions are still + possible but less likely since there are simply more channels + available and each end could use a different allocation strategy. + misdn.conf options available: te_choose_channel - Use to force + the TE side to allocate channels. method - Specify the channel + allocation strategy. (closes issue #13488) Reported by: + Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich + Tested by: crich, siepkes, festr ........ ................ + +2009-03-30 16:47 +0000 [r185088] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar + 2009) | 45 lines Merged revisions 185031 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar + 2009) | 39 lines Fix queue weight behavior so that calls in + low-weight queues are not inappropriately blocked. (This is + copied and pasted from the review request I made for this patch) + Asterisk has some odd behavior when queue weights are used. The + current logic used when potentially calling a queue member is: If + the member we are going to call is part of another queue and + _that other queue has any callers in it_ and has a higher weight + than the queue we are calling from, then don't try to contact + that member. The issue here is what I have marked with + underscores. If the higher-weighted queue has any callers in it + at all, then the queue member will be unreachable from the + lower-weighted queue. This has the potential to be really really + bad if using a queue strategy, such as leastrecent or + fewestcalls, with the potential to call the same member + repeatedly. The fix proposed by garychen on issue 13220 is very + simple and, as far as I can see, works well for this situation. + With this set of changes, the logic used becomes: If the member + we are going to call is part of another queue, the other queue + has a higher weight than the queue we are calling from, and the + higher weight queue has at least as many callers as available + members, then do not try to contact the queue member. If the + higher weighted queue has fewer callers than available members, + then there is no reason to deny the call to this member since the + other queue can afford to spare a member. Since the fix involved + writing a generic function for determining the number of + available members in the queue, I also modified the is_our_turn + function to make use of the new num_available_members function to + determine if it is our turn to try calling a member. There is one + small behavior change. Before writing this patch, if you had + autofill disabled, then if you were the head caller in a queue, + you would automatically be told that it was your turn to try + calling a member. This did not take into account whether there + were actually any queue members available to take the call. Now + we actually make sure there is at least one member available to + take the call if autofill is disabled. (closes issue #13220) + Reported by: garychen Review: + http://reviewboard.digium.com/r/202/ ........ ................ + +2009-03-30 14:41 +0000 [r184950] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | + 21 lines Merged revisions 184947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | + 14 lines Improve our handling of T38 in the initial INVITE from a + device. We now answer with matching media streams to what is + requested. If an INVITE is received with both a T38 and RTP media + stream this means we answer with both. For any outgoing calls + created as a result of this inbound one no T38 is requested in + the initial INVITE. Instead if we start receiving udptl packets + we trigger a reinvite on the outbound side. (closes issue #12437) + Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu + Review: http://reviewboard.digium.com/r/208/ ........ + ................ + +2009-03-30 13:57 +0000 [r184912] Russell Bryant <russell@digium.com> + + * channels/h323/Makefile.in, /: Merged revisions 184910 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 + Mar 2009) | 4 lines Fix build error when chan_h323 is not being + built. (reported by cai1982 in #asterisk-dev) ........ + +2009-03-29 05:52 +0000 [r184840-184845] Russell Bryant <russell@digium.com> + + * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009) + | 13 lines Merged revisions 184842 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) + | 5 lines Ensure targs variable is fully initialized. (closes + issue #14758) Reported by: tim_ringenbach ........ + ................ + + * channels/Makefile, /: Merged revisions 184838 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | + russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines + Simplify chan_h323 build to not require a second run of "make". + (closes issue #14715) Reported by: jthurman Patches: + h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license + 614) Tested by: tzafrir, russell ........ + +2009-03-27 19:17 +0000 [r184765] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_iax2.c, main/timing.c, main/channel.c, /, + include/asterisk/timing.h, include/asterisk/channel.h: Merged + revisions 184762 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 | + kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 + lines Improve timing interface to remember which provider + provided a timer The ability to load/unload timing interfaces is + nice, but it means that when a timer is allocated, it may come + from provider A, but later provider B becomes the 'preferred' + provider. If this happens, all timer API calls on the timer that + was provided by provider A will actually be handed to provider B, + which will say WTF and return an error. This patch changes the + timer API to include a pointer to the provider of the timer + handle so that future operations on the timer will be forwarded + to the proper provider. (closes issue #14697) Reported by: moy + Review: http://reviewboard.digium.com/r/211/ ........ + +2009-03-27 18:09 +0000 [r184728] Russell Bryant <russell@digium.com> + + * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 + Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure + we use the best RNG available. ........ + +2009-03-27 15:54 +0000 [r184675] Joshua Colp <jcolp@digium.com> + + * /, res/res_agi.c: Merged revisions 184673 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 | + file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix + speech structure leak in the AGI speech recognition integration. + The AGI dialplan applications did not destroy the speech + structure automatically if it was not destroyed by the running + AGI script. They will now do this. (issue LUMENVOX-15) ........ + +2009-03-27 14:04 +0000 [r184631] Russell Bryant <russell@digium.com> + + * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /, + res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions + 184630 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 | + russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines + Change g_eid to ast_eid_default. ........ + +2009-03-27 13:22 +0000 [r184587] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | + 16 lines Merged revisions 184565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 + lines Fix an issue where nat=yes would not always take effect for + the RTP session on outgoing calls. If calls were placed using an + IP address or hostname the global nat setting was copied over but + was not set on the RTP session itself. This caused the RTP stack + to not perform symmetric RTP actions. (closes issue #14546) + Reported by: acunningham ........ ................ + +2009-03-27 02:25 +0000 [r184513-184547] Russell Bryant <russell@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009) + | 20 lines Fix some issues with rwlock corruption that caused + deadlock like symptoms. When dvossel and I were doing some load + testing last week, we noticed that we could make Asterisk trunk + lock up instantly when we started generating a bunch of calls. + The backtraces of locked threads were bizarre, and many were + stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a + number of places where a backtrace would be loaded into an + invalid index of the backtrace array. It's an off by one error, + which ends up writing over the rwlock itself. 2) Ensure that in + the array of held locks, we NULL out an index once it is not + being used so that it's not confusing when analyzing its + contents. 3) Remove a bunch of logging referring to an rwlock + operating being done with "deep reentrancy". It is normal for + _many_ threads to hold a read lock on an rwlock. ........ + + * /, main/file.c: Merged revisions 184515 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 | + russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines + Don't act surprised if we get a -1 indication. ........ + + * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26 + Mar 2009) | 2 lines Pass more useful information through to lock + tracking when DEBUG_THREADS is on. ........ + +2009-03-26 22:19 +0000 [r184451] Kevin P. Fleming <kpfleming@digium.com> + + * sounds/Makefile, /: Merged revisions 184448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar + 2009) | 9 lines Merged revisions 184447 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar + 2009) | 3 lines use new, improved 8kHz prompts ........ + ................ + +2009-03-26 21:18 +0000 [r184394] David Vossel <dvossel@digium.com> + + * /, apps/app_test.c: Merged revisions 184389 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009) + | 14 lines Merged revisions 184388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) + | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF + 8 app_test was failing when sending the last DTMF digit, 8, + because of the 100ms pause issued after DTMF is sent. During this + pause the other side would hang up causing the test to look like + it failed. Now the other side waits a second before hanging up. + (closes issue #12442) Reported by: tzafrir ........ + ................ + +2009-03-25 22:13 +0000 [r184325-184345] Russell Bryant <russell@digium.com> + + * /, main/event.c: Merged revisions 184344 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 | + russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines + Remove unneeded AST_LIST_ENTRY() and comment on the purpose of + ast_event_ref. ........ + + * channels/chan_iax2.c, channels/chan_dahdi.c, + include/asterisk/event.h, channels/chan_skinny.c, res/ais/evt.c, + main/event.c, include/asterisk/strings.h, main/asterisk.c, + channels/chan_mgcp.c, apps/app_voicemail.c, + channels/chan_unistim.c, include/asterisk/devicestate.h, /, + channels/chan_sip.c, main/devicestate.c, + include/asterisk/_private.h: Merged revisions 184339 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) + | 35 lines Improve performance of the ast_event cache + functionality. This code comes from + svn/asterisk/team/russell/event_performance/. Here is a summary + of the changes that have been made, in order of both invasiveness + and performance impact, from smallest to largest. 1) Asterisk + 1.6.1 introduces some additional logic to be able to handle + distributed device state. This functionality comes at a cost. One + relatively minor change in this patch is that the extra + processing required for distributed device state is now + completely bypassed if it's not needed. 2) One of the things that + I noticed when profiling this code was that a _lot_ of time was + spent doing string comparisons. I changed the way strings are + represented in an event to include a hash value at the front. So, + before doing a string comparison, we do an integer comparison on + the hash. 3) Finally, the code that handles the event cache has + been re-written. I tried to do this in a such a way that it had + minimal impact on the API. I did have to change one API call, + though - ast_event_queue_and_cache(). However, the way it works + now is nicer, IMO. Each type of event that can be cached (MWI, + device state) has its own hash table and rules for hashing and + comparing objects. This by far made the biggest impact on + performance. For additional details regarding this code and how + it was tested, please see the review request. (closes issue + #14738) Reported by: russell Review: + http://reviewboard.digium.com/r/205/ ........ + + * /: add reviewboard:url property. + +2009-03-25 19:26 +0000 [r184282] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 | + file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix + issue with a T38 reinvite being sent even if not configured to do + so. If we receive a T38 request negotiate control frame we should + only attempt to do so if the option is enabled on the dialog. + ........ + +2009-03-25 15:12 +0000 [r184223] Eliel C. Sardanons <eliels@gmail.com> + + * main/asterisk.c, /: Merged revisions 184220 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) | + 19 lines Merged revisions 184188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | + 13 lines Avoid destroying the CLI line when moving the cursor + backward and trying to autocomplete. When moving the cursor + backward and pressing TAB to autocomplete, a NULL is put in the + line and we are loosing what we have already wrote after the + actual cursor position. (closes issue #14373) Reported by: eliel + Patches: asterisk.c.patch uploaded by eliel (license 64) Tested + by: lmadsen ........ ................ + +2009-03-25 01:55 +0000 [r184149] Russell Bryant <russell@digium.com> + + * main/timing.c, utils/Makefile, /, include/asterisk/compat.h: + Merged revisions 184147 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 | + russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines + Fix build issues on Mac OSX. (closes issue #14714) Reported by: + ygor ........ + +2009-03-24 22:42 +0000 [r184081] Mark Michelson <mmichelson@digium.com> + + * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar + 2009) | 15 lines Merged revisions 184078 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar + 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. + The 'digit' variable is guaranteed to be non-NULL, so the if + statement could never evaluate true. Changing to ast_strlen_zero + makes the logic correct. This was found while reviewing + ast_channel_ao2 code review. ........ ................ + +2009-03-24 21:47 +0000 [r184039] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) + | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low + and =medium The default codec configuration for chan_iax2 is + bandwidth=low. I noticed slin16 being negotiated as the codec in + some test calls, but that no longer happens after this change. + ........ + +2009-03-24 15:28 +0000 [r183867-183916] Tilghman Lesher <tlesher@digium.com> + + * /, configs/voicemail.conf.sample: Merged revisions 183914 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500 + (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) + | 3 lines Additionally note that the operator option needs an 'o' + extension. (Related to issue #14731) ........ ................ + + * /, main/http.c: Merged revisions 183865 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 | + tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines + Allow browsers to cache images and other static content. (This is + a regression over 1.4) ........ + +2009-03-23 18:59 +0000 [r183768] Mark Michelson <mmichelson@digium.com> + + * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar + 2009) | 13 lines Merged revisions 183700 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar + 2009) | 7 lines Fix a memory leak in res_monitor.c The only way + that this leak would occur is if Monitor were started using the + Manager interface and no File: header were given. Discovered + while reviewing the ast_channel_ao2 review request. ........ + ................ + +2009-03-23 18:12 +0000 [r183703] Leif Madsen <lmadsen@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) + | 7 lines Fixes a documentation error introduced during the CLI + cleanup at AstriDevCon 2008. (closes issue #14655) Reported by: + ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728) + Tested by: lmadsen ........ + +2009-03-20 17:08 +0000 [r183563] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r183560 | russell | 2009-03-20 12:00:58 -0500 + (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) + | 2 lines Fix a crash in IAX2 registration handling found during + load testing with dvossel. ........ ................ + +2009-03-19 20:33 +0000 [r183438] David Vossel <dvossel@digium.com> + + * include/asterisk/features.h, apps/app_dial.c, /, main/features.c: + Merged revisions 183436 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) + | 13 lines Merged revisions 183386 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) + | 6 lines Cleaning up a few things in detect disconnect patch + Initialized ast_call_feature in detect_disconnect to avoid + accessing uninitialized memory. Cleaned up /param tags in + features.h. No longer send dynamic features in + ast_feature_detect. issue #11583 ........ ................ + +2009-03-19 19:19 +0000 [r183333] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500 + (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) + | 8 lines Delay signalling progress until a PRI channel really + signals progress. (closes issue #13034) Reported by: klaus3000 + Patches: 20090316__bug13034.diff.txt uploaded by tilghman + (license 14) patch_trunk_183progress_klaus3000.txt uploaded by + klaus3000 (license 65) Tested by: klaus3000 ........ + ................ + +2009-03-19 18:14 +0000 [r183249] Russell Bryant <russell@digium.com> + + * main/loader.c, /, configure, include/asterisk/autoconfig.h.in, + configure.ac: Merged revisions 183242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) + | 10 lines Merged revisions 183241 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) + | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving + like expected. ........ ................ + +2009-03-19 18:11 +0000 [r183246] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 183244 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 | + mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 + lines Fix a memory leak associated with queues. For every attempt + that app_queue made to place an outbound call to a queue member, + we would allocate a queue_end_bridge structure. When the bridge + for the call had completed, we would free the structure. + Unfortunately not all call attempts actually end up bridged to a + member, so we need to be more selective of when to allocate the + structure. With this change, the allocation occurs in an area + where we can guarantee that the call will be bridged. (closes + issue #14680) Reported by: caspy Patches: 14680.patch uploaded by + mmichelson (license 60) Tested by: caspy ........ + +2009-03-19 17:08 +0000 [r183198] David Vossel <dvossel@digium.com> + + * include/asterisk/features.h, apps/app_dial.c, /, main/features.c: + Merged revisions 183172 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) + | 20 lines Merged revisions 183126 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) + | 17 lines Allow disconnect feature before a call is bridged + feature.conf has a disconnect option. By default this option is + set to '*', but it could be anything. If a user wishes to + disconnect a call before the other side answers, only '*' will + work, regardless if the disconnect option is set to something + else. This is because features are unavailable until bridging + takes place. The default disconnect option, '*', was hardcoded in + app_dial, which doesn't make any sense from a user perspective + since they may expect it to be something different. This patch + allows features to be detected from outside of the bridge, but + not operated on. In this case, the disconnect feature can be + detected before briding and handled outside of features.c. + (closes issue #11583) Reported by: sobomax Patches: + patch-apps__app_dial.c uploaded by sobomax (license 359) + 11583.latest-patch uploaded by murf (license 17) + detect_disconnect.diff uploaded by dvossel (license 671) Tested + by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ + ........ ................ + +2009-03-19 16:09 +0000 [r183121] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar + 2009) | 20 lines Merged revisions 183115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar + 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls + would erroneously report the device as "in use." A user was + having an issue where if an outgoing SIP call was canceled, the + SIP device would remain in use if we had not received any + response to the initial INVITE we sent out. The SIP device would + remain in use until the autocongestion timer was exhausted. I + tracked down the cause of this to be the section of code I am + removing here. I asked several people what the purpose of this + code was meant to be, but no one could give me any sort of answer + as to why this was here. The person who was having this issue has + been using this patch for several months and it has stopped the + problems they have had. AST-196 ........ ................ + +2009-03-19 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.1.0-rc3 + +2009-03-19 15:43 +0000 [r183067-183110] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | + file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines + Improve our triggering of a T38 switchover internally when + triggered by a received reinvite. Previously we reached across + the channel bridge to get the other party's SIP dialog structure + in order to trigger an outgoing reinvite. This is extremely + dangerous to do and only works if bridged to another SIP channel. + This patch changes this to use the T38 control frame method of + requesting a switchover. This change also causes the SIP channel + driver to propogate back whether the switchover worked or not + instead of blindly accepting the incoming T38 reinvite. Review: + http://reviewboard.digium.com/r/200/ ........ + + * main/channel.c, /: Merged revisions 183057 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | + file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix + an issue where a T38 control frame would get dropped. If two + channels were bridged together using a generic bridge the T38 + control frame would get passed up instead of being indicated on + the other channel. ........ + +2009-03-18 21:19 +0000 [r183030] Jeff Peeler <jpeeler@digium.com> + + * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 + Mar 2009) | 4 lines Add some code removed by mistake from commit + 182722 that works around a file descriptor leak in versions of + PWLib prior to 1.12.0. ........ + +2009-03-18 14:32 +0000 [r182946] Russell Bryant <russell@digium.com> + + * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c, + configure, apps/app_mp3.c, res/res_agi.c, + include/asterisk/poll-compat.h, channels/chan_alsa.c, + main/asterisk.c, apps/app_nbscat.c, /, main/Makefile, + include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/io.h, main/utils.c, include/asterisk/channel.h: + Merged revisions 182847 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) + | 52 lines Merged revisions 182810 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) + | 44 lines Fix cases where the internal poll() was not being used + when it needed to be. We have seen a number of problems caused by + poll() not working properly on Mac OSX. If you search around, + you'll find a number of references to using select() instead of + poll() to work around these issues. In Asterisk, we've had poll.c + which implements poll() using select() internally. However, we + were still getting reports of problems. vadim investigated a bit + and realized that at least on his system, even though we were + compiling in poll.o, the system poll() was still being used. So, + the primary purpose of this patch is to ensure that we're using + the internal poll() when we want it to be used. The changes are: + 1) Remove logic for when internal poll should be used from the + Makefile. Instead, put it in the configure script. The logic in + the configure script is the same as it was in the Makefile. + Ideally, we would have a functionality test for the problem, but + that's not actually possible, since we would have to be able to + run an application on the _target_ system to test poll() + behavior. 2) Always include poll.o in the build, but it will be + empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() + throughout the source tree to ast_poll(). I feel that it is good + practice to give the API call a new name when we are changing its + behavior and not using the system version directly in all cases. + So, normally, ast_poll() is just redefined to poll(). On systems + where AST_POLL_COMPAT is defined, ast_poll() is redefined to + ast_internal_poll(). 4) Change poll() in main/poll.c to be + ast_internal_poll(). It's worth noting that any code that still + uses poll() directly will work fine (if they worked fine before). + So, for example, out of tree modules that are using poll() will + not stop working or anything. However, for modules to work + properly on Mac OSX, ast_poll() needs to be used. (closes issue + #13404) Reported by: agalbraith Tested by: russell, vadim + http://reviewboard.digium.com/r/198/ ........ ................ + +2009-03-17 20:52 +0000 [r182724] Jeff Peeler <jpeeler@digium.com> + + * channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /, + channels/h323/ast_h323.cxx, configure, + autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h, + channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions + 182722 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | + jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines + Allow H.323 Plus library to be used in addition to the OpenH323 + library Chan_h323 can now be compiled against both the previously + supported versions of OpenH323 as well as the current H.323 Plus + (version 1.20.2). The configure script has been modified to look + in the default install location of h323 to hopefully help avoid + using the environment variables OPENH323DIR and PWLIBDIR. Also, + the CLI command "h323 show version" has been added which + indicates which version of h323 is in use. (closes issue #11261) + Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch + uploaded by jthurman (license 614) ........ + +2009-03-17 15:31 +0000 [r182570] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 182553 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | + russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines + Tweak the handling of the frame list inside of ast_answer(). This + does not change any behavior, but moves the frames from the local + frame list back to the channel read queue using an O(n) algorithm + instead of O(n^2). ........ + +2009-03-17 15:00 +0000 [r182527-182533] Kevin P. Fleming <kpfleming@digium.com> + + * main/channel.c, /: Merged revisions 182530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | + kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 + lines correct logic flaw in ast_answer() changes in r182525 + ........ + + * main/channel.c, /, main/features.c, include/asterisk/channel.h: + Merged revisions 182525 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | + kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 + lines Improve behavior of ast_answer() to not lose incoming + frames ast_answer(), when supplied a delay before returning to + the caller, use ast_safe_sleep() to implement the delay. + Unfortunately during this time any incoming frames are discarded, + which is problematic for T.38 re-INVITES and other sorts of + channel operations. When a delay is not passed to ast_answer(), + it still delays for up to 500 milliseconds, waiting for media to + arrive. Again, though, it discards any control frames, or + non-voice media frames. This patch rectifies this situation, by + storing all incoming frames during the delay period on a list, + and then requeuing them onto the channel before returning to the + caller. http://reviewboard.digium.com/r/196/ ........ + +2009-03-17 05:54 +0000 [r182452] Tilghman Lesher <tlesher@digium.com> + + * main/db.c, /: Merged revisions 182450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) + | 14 lines Merged revisions 182449 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) + | 7 lines Fix race in astdb The underlying db1 implementation + does not fully isolate the pages retrieved from astdb, so the + lock protecting accesses needs to be extended until the copy from + the shared memory structure is done. (closes issue #14682) + Reported by: makoto ........ ................ + +2009-03-16 17:53 +0000 [r182284] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 182282 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 + (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) + | 7 lines Randomize IAX2 encryption padding The 16-32 byte random + padding at the beginning of an encrypted IAX2 frame turns out to + not be all that random at all. This patch calls ast_random to + fill the padding buffer with random data. The padding is + randomized at the beginning of every encrypted call and for every + encrypted retransmit frame. Review: + http://reviewboard.digium.com/r/193/ ........ ................ + +2009-03-16 17:38 +0000 [r182280] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /, funcs/func_env.c: Merged revisions + 182211,182278 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009) + | 14 lines Merged revisions 182208 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) + | 7 lines Fixup glare detection, to fix a memory leak of a local + pvt structure. (closes issue #14656) Reported by: caspy Patches: + 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) + Tested by: caspy ........ ................ r182278 | tilghman | + 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines Fix an + off-by-one error in the FILE() function, and extend FILE()'s + length parameter to work like variable substitution. Previously, + FILE() returned one less character than specified, due to the + terminating NULL. Both the offset and length parameters now + behave identically to the way variable substitution offsets and + lengths also work. (closes issue #14670) Reported by: BMC + ................ + +2009-03-16 14:00 +0000 [r182173] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 182171 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 | + file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix + a memory leak in the ast_answer / __ast_answer API call. For a + channel that is not yet answered this API call will wait until a + voice frame is received on the channel before returning. It does + this by waiting for frames on the channel and reading them in. + The frames read in were not freed when they should have been. + ........ + +2009-03-13 21:27 +0000 [r182068-182123] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 182121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 | + mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 + lines Change faulty comparison used when announcing average hold + minutes and seconds (closes issue #14227) Reported by: caspy + ........ + + * /, main/features.c: Merged revisions 182029 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar + 2009) | 41 lines Merged revisions 181990 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar + 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and + peer when interpreting DTMF. Dynamic features defined in the + applicationmap section of features.conf allow one to specify + whether the caller, callee, or both have the ability to use the + feature. The documentation in the features.conf.sample file could + be interpreted to mean that one only needs to set the + DYNAMIC_FEATURES channel variable on the calling channel in order + to allow for the callee to be able to use the features which he + should have permission to use. However, the DYNAMIC_FEATURES + variable would only be read from the channel of the participant + that pressed the DTMF sequence to activate the feature. The + result of this was that the callee was unable to use dynamic + features unless the dialplan writer had taken measures to be sure + that the DYNAMIC_FEATURES variable was set on the callee's + channel. This commit changes the behavior of + ast_feature_interpret to concatenate the values of + DYNAMIC_FEATURES from both parties involved in the bridge. The + features themselves determine who has permission to use them, so + there is no reason to believe that one side of the bridge could + gain the ability to perform an action that they should not have + the ability to perform. Kevin Fleming pointed out on the + asterisk-users list that the typical way that this was worked + around in the past was by setting _DYNAMIC_FEATURES on the + calling channel so that the value would be inherited by the + called channel. While this works, the documentation alone is not + enough to figure out why this is necessary for the callee to be + able to use dynamic features. In this particular case, changing + the code to match the documentation is safe, easy, and will + generally make things easier for people for future installations. + This bug was originally reported on the asterisk-users list by + David Ruggles. (closes issue #14657) Reported by: mmichelson + Patches: 14657.patch uploaded by mmichelson (license 60) ........ + ................ + +2009-03-13 17:29 +0000 [r182042] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | + file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix + an issue with requesting a T38 reinvite before the call is + answered. The code responsible for sending the T38 reinvite did + not check if an INVITE was already being handled. This caused + things to get confused and the call to fail. The code now defers + sending the T38 reinvite until the current INVITE is done being + handled. (issue AST-191) ........ + +2009-03-13 16:58 +0000 [r181987] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c: Merged revisions 181985 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181985 | + kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line + improve a bit of suboptimal code ........ + +2009-03-12 21:45 +0000 [r181771-181849] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 181846 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 | + mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 + lines Run the macro on the queue member's channel when he + answers, not the caller's channel. ........ + + * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar + 2009) | 28 lines Merged revisions 181768 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar + 2009) | 22 lines Properly send a 487 on an INVITE we have not + responded to if we receive a BYE. If we receive an INVITE from an + endpoint and then later receive a BYE from that same endpoint + before we have sent a final response for the INVITE, then we need + to respond to the INVITE with a 487. There was logic in the code + prior to this commit which seemed to exist solely to handle this + situation, but there was one condition in an if statement which + was incorrect. The only way we would send a 487 was if the + sip_pvt had no owner channel. This made no sense since we created + the owner channel when we received the INVITE, meaning that the + majority of the time we would never send the 487. The 487 being + sent should not rely on whether we have created a channel. Its + delivery should be dependent on the current state of the initial + INVITE transaction. With this commit, that logic is now correctly + in place. (closes issue #14149) Reported by: legranjl Patches: + 14149.patch uploaded by mmichelson (license 60) Tested by: + legranjl ........ ................ + +2009-03-12 18:07 +0000 [r181733] Tilghman Lesher <tlesher@digium.com> + + * /, main/translate.c: Merged revisions 181731 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181731 | + tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines + Adjust translation table column widths based upon the translation + times. Previously, only 5 columns were displayed, and if a + translation time exceeded 99,999 useconds, it would be displayed + as 0, instead of its actual time. (closes issue #14532) Reported + by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman + (license 14) Tested by: pj ........ + +2009-03-12 16:58 +0000 [r181614-181667] Joshua Colp <jcolp@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, + 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 + lines Fix incorrect usage of strncasecmp... I really meant to use + strcasecmp. ........ ................ + + * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, + 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 + lines Fix another scenario where depending on configuration the + stream would not get read. For custom commands we don't know + whether the audio is coming from a stream or not so we are going + to have to read the data despite no channels. (closes issue + #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 + 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in + previous commit. ........ ................ + + * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, + 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | + 10 lines Fix issue with streaming MOH failing if nobody is + listening. When a music class is setup to actually provide music + on hold from a stream we need to constantly read audio from it + since it will constantly be providing audio. This is now done + despite there being no channels listening to it. (closes issue + #14416) Reported by: caspy ........ ................ + + * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | + file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix + crash when sleep and retries argument was not given to RetryDial + application. (closes issue #14647) Reported by: sherpya ........ + +2009-03-12 01:05 +0000 [r181544] Richard Mudgett <rmudgett@digium.com> + + * /, build_tools/make_version: Merged revisions 181542 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) + | 1 line Use the correct branch integrated property when + generating the version string ........ + +2009-03-11 23:21 +0000 [r181521] Michiel van Baak <michiel@vanbaak.info> + + * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk Provide + correct hint to debug SIP trouble in the default config (closes + issue #14646) Reported by: strk + +2009-03-11 22:27 +0000 [r181474] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 181465 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181465 | + russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines + Make handling of the BRIDGE_PLAY_SOUND variable thread-safe. + ........ + +2009-03-11 22:23 +0000 [r181457] Jason Parker <jparker@digium.com> + + * /, configure, configure.ac: Merged revisions 181444 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181444 | qwell | 2009-03-11 17:20:13 -0500 + (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | + 4 lines Allow prefix to set localstatedir (when used and + different from the default). This is similar to the /etc change + that was made for the non-FreeBSD case. ........ ................ + +2009-03-11 22:16 +0000 [r181426-181430] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 181428 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | + russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines + Make handling of the BRIDGEPVTCALLID variable thread-safe. + ........ + + * main/channel.c, /: Merged revisions 181424 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) + | 17 lines Merged revisions 181423 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) + | 9 lines Make code that updates BRIDGEPEER variable thread-safe. + It is not safe to read the name field of an ast_channel without + the channel locked. This patch fixes some places in channel.c + where this was being done, and lead to crashes related to + masquerades. (closes issue #14623) Reported by: guillecabeza + ........ ................ + +2009-03-11 17:40 +0000 [r181373] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions + 181371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) + | 17 lines Merged revisions 181340 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) + | 11 lines encrypted IAX2 during packet loss causes decryption to + fail on retransmitted frames If an iax channel is encrypted, and + a retransmit frame is sent, that packet's iseqno is updated while + it is encrypted. This causes the entire frame to be corrupted. + When the corrupted frame is sent, the other side decrypts it and + sends a VNAK back because the decrypted frame doesn't make any + sense. When we get the VNAK, we look through the sent queue and + send the same corrupted frame causing a loop. To fix this, + encrypted frames requiring retransmission are decrypted, updated, + then re-encrypted. Since key-rotation may change the key held by + the pvt struct, the keys used for encryption/decryption are held + within the iax_frame to guarantee they remain correct. (closes + issue #14607) Reported by: stevenla Tested by: dvossel Review: + http://reviewboard.digium.com/r/192/ ........ ................ + +2009-03-11 17:29 +0000 [r181298-181359] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | + 21 lines Merged revisions 181328 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | + 14 lines Fix issue where an attended transfer could not be + completed under a rare scenario. When completing an attended + transfer chan_sip does a check to make sure the extension in the + URI portion of the Refer-To header is a local valid extension. We + don't actually need to check this since we know for sure the + other channel is already up and talking to the extension. Some + devices do not put the extension in the Refer-To header either, + which can cause the extension check to fail. We now no longer do + this check if it is an attended transfer. (closes issue #14628) + Reported by: sverre Patches: 14628.diff uploaded by file (license + 11) ........ ................ + + * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | + 16 lines Merged revisions 181295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 + lines Fix a problem with inband DTMF detection on outgoing SIP + calls when dtmfmode=auto. When dtmfmode was set to auto the + inband DTMF detector was not setup on outgoing SIP calls. This + caused inband DTMF detection to fail. The inband DTMF detector is + now setup for both dtmfmode inband and auto. (closes issue + #13713) Reported by: makoto ........ ................ + +2009-03-11 15:54 +0000 [r181199-181283] Jeff Peeler <jpeeler@digium.com> + + * channels/h323/ast_h323.cxx: add missing header file + + * pbx/pbx_config.c, utils/Makefile, include/asterisk/utils.h, + include/asterisk/astmm.h, /, channels/chan_sip.c, + channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c: + Merged revisions 181135 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | + jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines + Fix malloc debug macros to work properly with h323. The main + problem here was that cstdlib was undefining free thereby causing + the proper debug macros to not be used. ast_h323.cxx has been + changed to call ast_free instead to avoid the issue. A few other + issues were addressed: - There were a few instances of functions + improperly passing ast_free instead of ast_free_ptr. - Some clean + up was done to avoid the debug macros intentionally being + redefined. (copied below from Kevin's commit, appreciate the + help) - disable astmm.h from doing anything when STANDALONE is + defined, which is used by the tools in the utils/ directory that + use parts of Asterisk header files in hackish ways; also ensure + that utils/extconf.c and utils/conf2ael.c are compiled with + STANDALONE defined. (closes issue #13593) Reported by: pj + ........ + +2009-03-11 01:04 +0000 [r181035] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 181032-181033 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 + (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar + 2009) | 9 lines Fix incorrect tag checking on transfers when + pedantic=yes is enabled. (closes issue #14611) Reported by: + klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt + uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ + r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar + 2009) | 3 lines Remove unused variables. ........ + ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 + (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC + 3891 ................ + +2009-03-10 22:07 +0000 [r180947] Jason Parker <jparker@digium.com> + + * /, configure, configure.ac, autoconf/ast_prog_sed.m4, + autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r180944 | qwell | 2009-03-10 17:03:41 -0500 + (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar + 2009) | 1 line Make things happier when using autoconf 2.62+ + ........ ................ + +2009-03-10 14:42 +0000 [r180802] Joshua Colp <jcolp@digium.com> + + * main/manager.c, /: Merged revisions 180800 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | + file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines + Reset the thread local string buffer when handling the UserEvent + action. (closes issue #14593) Reported by: JimDickenson ........ + +2009-03-09 21:22 +0000 [r180740] Jeff Peeler <jpeeler@digium.com> + + * include/asterisk/heap.h, include/asterisk/http.h, + include/asterisk/logger.h, main/tcptls.c, + include/asterisk/res_odbc.h, include/asterisk/doxyref.h, + include/asterisk/event.h, include/asterisk/audiohook.h, + include/asterisk/dsp.h, include/asterisk/lock.h, + include/asterisk/udptl.h, include/asterisk/dnsmgr.h, + include/asterisk/utils.h, include/asterisk/devicestate.h, /, + include/asterisk/taskprocessor.h, include/asterisk/astobj2.h, + include/asterisk/channel.h, include/asterisk/tcptls.h, + include/asterisk/manager.h, main/enum.c, + include/asterisk/callerid.h, include/asterisk/app.h, + include/asterisk/linkedlists.h, include/asterisk/sched.h, + include/asterisk/datastore.h, include/asterisk/timing.h, + include/asterisk/dlinkedlists.h, include/asterisk/pbx.h, + include/asterisk/enum.h, include/asterisk/config.h, + include/asterisk/rtp.h, include/asterisk/extconf.h, + main/devicestate.c: Merged revisions 180719 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r180719 | + jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines + Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 + Copied from my review board description: This is a continuation + of the API changes documentation started for describing changes + between releases. Most of the API changes were pretty simple + needing only to be brought to attention via the new "Asterisk API + Changes" list. However, if you see anything that needs further + explanation feel free to supplement what is there. The current + method of documenting is to add (in the header file): \version + <ver number> <description of changes> and then to add the + function to the change list in doxyref.h on the AstAPIChanges + page. I also made sure all the functions that were newly added + were tagged with \since 1.6.1. I think this is a good habit to + start both for the historical aspect as well as for the future + ability to easily add a "New Asterisk API" page. Review: + http://reviewboard.digium.com/r/190/ ........ + +2009-03-06 18:26 +0000 [r180585] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600 + (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, + 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when + IMAP storage is enabled. ........ ................ + +2009-03-06 17:35 +0000 [r180537] David Vossel <dvossel@digium.com> + + * main/enum.c, /: Merged revisions 180534 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) + | 15 lines Merged revisions 180532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) + | 9 lines Fix handling of backreferences for ENUM lookups enum.c + did not handle regex backtraces correctly. The '\1' in the regex + is a backreference that requires a pattern match to be inserted. + The way the code used to work is that it would find the + backreference and insert the entire input string minus the '+'. + This is incorrect. The regexec() function takes in a variable + called pmatch which is an array of structs containing the start + and end indexes for each backreference substring. The original + code actually passed the pmatch array pointer into regexec but + never did anything with it. Now when a backtrace is found, the + backtrace number is looked up in the pmatch array and the correct + substring is inserted. (closes issue #14576) Reported by: + chris-mac Review: http://reviewboard.digium.com/r/187/ ........ + ................ + +2009-03-05 23:28 +0000 [r180425-180467] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600 + (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar + 2009) | 16 lines [IMAP] Fix message retrieval issues when + identical mailbox names were defined in separate contexts. There + was a fix put in a while back so that an X-Asterisk-VM-Context + message header was added to stored IMAP voicemails. This would + allow for us to differentiate if the same mailbox name was used + in multiple contexts. The problem still left was that not all + places where messages were retrieved actually attempted to use + this header for information when retrieving messages. This commit + fixes that so that MWI and message retrieval from VoiceMailMain + work as expected. (closes issue #13853) Reported by: vicks1 + Patches: 13853_v2.patch uploaded by mmichelson (license 60) + Tested by: lmadsen ........ ................ + + * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged + revisions 180383 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar + 2009) | 31 lines Merged revisions 180380 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar + 2009) | 25 lines Fix broken mailbox parsing when searchcontexts + option is enabled. When using the searchcontexts option in + voicemail.conf, the code made the assumption that all mailbox + names defined were unique across all contexts. However, the code + did nothing to actually enforce this assumption, nor did it do + anything to alert a user that he may have created an ambiguity in + his voicemail.conf file by defining the same mailbox name in + multiple contexts. With this change, we now will issue a nice + long warning if searchcontexts is on and we encounter the same + mailbox name in multiple contexts and ignore any duplicates after + the first box. Whether searchcontexts is enabled or not, if we + come across a duplicate mailbox in the same context, then we will + issue a warning and ignore the duplicated mailbox. I have also + added a small note to voicemail.conf.sample in the explanation + for searchcontexts explaining that you cannot define the same + mailbox in multiple contexts if you have enabled the option. + (closes issue #14599) Reported by: lmadsen Patches: 14599.patch + uploaded by mmichelson (license 60) (with slight modification) + Tested by: lmadsen ........ ................ + +2009-03-05 18:40 +0000 [r180378] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/frame.h, main/rtp.c, main/frame.c, /: Merged + revisions 180373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar + 2009) | 15 lines Merged revisions 180372 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar + 2009) | 9 lines Fix problems when RTP packet frame size is + changed During some code analysis, I found that calling + ast_rtp_codec_setpref() on an ast_rtp session does not work as + expected; it does not adjust the smoother that may on the RTP + session, in fact it summarily drops it, even if it has data in + it, even if the current format's framing size has not changed. + This is not good. This patch changes this behavior, so that if + the packetization size for the current format changes, any + existing smoother is safely updated to use the new size, and if + no smoother was present, one is created. A new API call for + smoothers, ast_smoother_reconfigure(), was required to implement + these changes. Review: http://reviewboard.digium.com/r/184/ + ........ ................ + +2009-03-04 Leif Madsen <lmadsen@digium.com> + + * Released Asterisk 1.6.1.0-rc2 + +2009-03-04 21:09 +0000 [r180263] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 180261 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r180261 | + russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines + Resolve object matching issues related to the removal of the + sip_user object. Previously, chan_sip had both sip_peer and + sip_user objects in memory. A patch went in to remove sip_user to + simplify the code, since everything could be done with just + sip_peer. This patch resolves some regressions found that were + introduced by those changes. This code comes from + svn/asterisk/team/group/sip-object-matching/. Here is a list of + the changes that have been made: 1) When doing a match by name + with the find_peer() function, make it much easier to specify + which objects should be matched by having a parameter that + specifies exactly which object types should be considered. Also, + update find_by_name() to handle this parameter. Finally, update + all code to use the new option values. 2) When looking up an + object for an outbound request by name, consider peers only. + (create_addr()) 3) Only match peers on an incoming registration + request. 4) When doing authentication (except for SUBSCRIBE), + look up users by name, instead of all objects by name. 5) When + doing authentication (except for SUBSCRIBE), after looking for a + user by name, look for a peer by IP address, instead of all + objects by IP address. 6) When handling the SIP qualify CLI + command or manager action, look for a peer by name, instead of + any object by name. 7) When handling the SIP unregister CLI + command, look for a peer by name, instead of any object by name. + 9) In sip_do_debug_peer(), search for a peer by name, instead of + any object by name. 9) When handling the SIPPEER() dialplan + function, search for a peer by name, instead of any object by + name. 10) In the following session timer related functions, + st_get_se(), st_get_refresher(), and st_get_mode(), when looking + for an object for a given sip_pvt using pvt->peername, look for a + peer by name, instead of any object by name. 11) Fix build_peer() + to properly handle the case where separate type=peer and + type=user entries were specified in sip.conf. (closes issue + #14505) Reported by: lmadsen Review: + http://reviewboard.digium.com/r/172/ ........ + +2009-03-04 19:27 +0000 [r180122-180197] Joshua Colp <jcolp@digium.com> + + * /, main/callerid.c: Merged revisions 180195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | + 11 lines Merged revisions 180194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 + lines Look for the number in a callerid string starting from the + end. This way a value using <> can exist in the name portion. + (issue #AST-194) ........ ................ + + * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 | + file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines + Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) + Reported by: alecdavis Patches: app_dial.optionk.diff.txt + uploaded by alecdavis (license 585) ........ + +2009-03-03 23:39 +0000 [r180080] David Vossel <dvossel@digium.com> + + * main/channel.c, include/asterisk/app.h, apps/app_read.c, /, + main/app.c: Merged revisions 180032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | + dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines + app_read does not break from prompt loop with user terminated + empty string In app.c, ast_app_getdata is called to stream the + prompts and receive DTMF input. If ast_app_getdata() receives an + empty string caused by the user inputing the end of string + character, in this case '#', it should break from the prompt loop + and return to app_read, but instead it cycles through all the + prompts. I've added a return value for this special case in + ast_readstring() which uses an enum I've delcared in apps.h. This + enum is now used as a return value for ast_app_getdata(). (closes + issue #14279) Reported by: Marquis Patches: fix_app_read.patch + uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded + by dvossel (license 671) Tested by: Marquis, dvossel Review: + http://reviewboard.digium.com/r/177/ ........ + +2009-03-03 23:31 +0000 [r180077] Steve Murphy <murf@digium.com> + + * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile, + utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y, + main/ast_expr2f.c: Merged revisions 179973 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | + 33 lines Merged revisions 179807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some + work to do to port these changes to trunk; the check_expr stuff + hasn't been updated here for quite some time, it appears. I added + some more tests to the check_expr2 suite. I had to play around + with the makefile a bit, etc. I added STANDALONE2 #ifdefs to + ast_expr2.y so as not to conflict structure with aelparse. + ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar + 2009) | 19 lines These changes allow AEL to better check ${} + constructs within $[...], that are concatenated with text. I + modified and added rules in ast_expr2.fl to better handle the + concatenations. I added some default routines to ast_expr2.y so + the standalone would compile. It also looks like I haven't run + this thru bison since 2.1, so it's good to get this updated. The + Makefile has comments added now for check_expr2 and check_expr to + explain what they are for, and how to run them. The testexpr2s + stuff has been removed, in favor of check_expr2. expr2.testinput + has been updated to include the two expressions that inspired + these changes (from mcnobody on #asterisk this morning) The + regression has been run and all looks well. ........ + ................ + +2009-03-03 22:49 +0000 [r179939-180009] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions + 180007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar + 2009) | 22 lines Merged revisions 180006 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar + 2009) | 17 lines Clarify some documentation of queues.conf.sample + It had always been possible to explicitly specify a "blank" value + for a sound file in queues.conf and have no sound played back. + The problem with this is that it would result in some ugly CLI + warnings from file.c. This commit introduces a check when playing + a file in app_queue to see if the name of the file is zero-length + and return early if that is the case. Also, the ability to + specify the blank sound files in queues.conf is now mentioned + more clearly in queues.conf.sample (closes issue #14227) Reported + by: caspy ........ ................ + + * doc/timing.txt (added), /, res/res_timing_dahdi.c: Merged + revisions 179937 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r179937 | + mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 + lines Add documentation for timing modules used in Asterisk This + document specifies the timing modules available in Asterisk + beginning with Asterisk 1.6.1. The document goes into detail + about the differences between each and gives a general overview + of what timing is used for in Asterisk. There is also a section + which can be used to help customize your setup or to troubleshoot + timing issues you may have. I also added messages to the DAHDI + timing test used in res_timing_dahdi.c that points to this new + documentation if people experience problems. Big thanks to all + who contributed comments on this. (closes issue #14490) Reported + by: mmichelson Patches: timing.txt uploaded by mmichelson + (license 60) Review: http://reviewboard.digium.com/r/164/ + ........ + +2009-03-03 20:09 +0000 [r179905] Russell Bryant <russell@digium.com> + + * /, apps/app_directed_pickup.c: Merged revisions 179903 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar + 2009) | 1 line fix a leaked channel lock (and future deadlock) + when we try to pick up our own channel ........ + +2009-03-03 18:30 +0000 [r179843] Joshua Colp <jcolp@digium.com> + + * /, main/features.c: Merged revisions 179841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | + 16 lines Merged revisions 179840 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 + lines Do not assume that the bridge_cdr is still attached to the + channel when the 'h' exten is finished executing. It is possible + for a masquerade operation to occur when the 'h' exten is + operating. This operation moves the CDR records around causing + the bridge_cdr to no longer exist on the channel where it is + expected to. We can not safely modify it afterwards because of + this, so don't even try. (closes issue #14564) Reported by: meric + ........ ................ + +2009-03-03 16:48 +0000 [r179744] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 179742 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) + | 14 lines Merged revisions 179741 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) + | 6 lines Ensure chan->fdno always gets reset to -1 after + handling a channel fd event. Since setting fdno to -1 had to be + moved, a couple of other code paths that do process an fd event + return early and do not pass through the code path where it was + moved to. So, set it to -1 in a few other places, too. ........ + ................ + +2009-03-03 14:41 +0000 [r179674] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 179672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | + 10 lines Merged revisions 179671 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 + lines Move where fdno is set to the default value to *after* the + read callback of the channel driver is called. We have to do this + as the underlying channel driver may need the fdno value to + determine what to read. ........ ................ + +2009-03-03 13:56 +0000 [r179611] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 179609 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) + | 17 lines Merged revisions 179608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) + | 9 lines Make it easier to detect an improper call to + ast_read(). When you call ast_waitfor() on a channel, the index + into the channel fds array that holds the file descriptor that + poll() determines has input available is stored in fdno. This + patch clears out this value after a call to ast_read() and also + reports errors if ast_read() is called without an fdno set. From + a discussion on the asterisk-dev list. ........ ................ + +2009-03-03 00:04 +0000 [r179539] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /: Merged revisions 179537 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) + | 21 lines Merged revisions 179536 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) + | 15 lines Fix bridging regression from commit 176701 This fixes + a bad regression where the bridge would exit after an attended + transfer was made. The problem was due to nexteventts getting set + after the masquerade which caused the bridge to return + AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: + tim_ringenbach ........ ................ + +2009-03-02 23:39 +0000 [r179535] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) + | 48 lines Merged revisions 179532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) + | 40 lines Move ast_waitfor() down to avoid the results of the + API call becoming stale. This call to ast_waitfor() was being + done way too soon in this section of code. Specifically, there + was code in between the call to waitfor and the code that uses + the result that puts the channel in autoservice. By putting the + channel in autoservice, the previous results of ast_waitfor() + become meaningless, as the autoservice thread will do it's own + ast_waitfor() and ast_read() on the channel. So, when we came + back out of autoservice and eventually hit the block of code that + calls ast_read() on the channel, there may not actually be any + input on the channel available. Even though the previous call to + ast_waitfor() in app_meetme said there was input, the autoservice + thread has since serviced the channel for some period of time. + This bug manifested itself while dvossel was doing some testing + of MeetMe in Asterisk trunk. He was using the timerfd timing + module. When the code hit ast_read() erroneously, it determined + that it must have been called because of input on the timer fd, + as chan->fdno was set to AST_TIMING_FD, since that was the cause + of the last legitimate call to ast_read() done by autoservice. In + this test, an IAX2 channel was calling into the MeetMe + conference. It was _much_ more likely to be seen with an IAX2 + channel because of the way audio is handled. Every audio frame + that comes in results in a call to ast_queue_frame(), which then + uses ast_timer_enable_continuous() to notify the channel thread + that a frame is waiting to be handled. So, the chances of + ast_waitfor() indicating that a channel needs servicing due to a + timer event on an IAX2 event is very high. Finally, it is + interesting to note that if a different timing interface was + being used, this bug would probably not be noticed. When + ast_read() is called and erroneously thinks that there is a timer + event to handle, it calls the ast_timer_ack() function. The + pthread and dahdi timing modules handle the ack() function being + called when there is no event by simply ignoring it. In the case + of the timerfd module, it results in a read() on the timer fd + that will block forever, as there is no data to read. This caused + Asterisk to lock up very quickly. Thanks to dvossel and + mmichelson for the fun debugging session. :-) ........ + ................ + +2009-03-02 23:12 +0000 [r179471] Tilghman Lesher <tlesher@digium.com> + + * /, main/app.c: Merged revisions 179469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) + | 17 lines Merged revisions 179468 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) + | 10 lines When ending a recording with silence detection, + remember to reduce the duration. The end of the recording is + correspondingly trimmed, but the duration was not trimmed by the + number of seconds trimmed, so the saved duration was necessarily + longer than the actual soundfile duration. (closes issue #14406) + Reported by: sasargen Patches: 20090226__bug14406.diff.txt + uploaded by tilghman (license 14) Tested by: sasargen ........ + ................ + +2009-03-02 23:04 +0000 [r179464] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 179462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) + | 16 lines Merged revisions 179461 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) + | 8 lines Ensure that only one thread is calling ast_settimeout() + on a channel at a time. For example, with an IAX2 channel, you + can have both the channel thread and the chan_iax2 processing + threads calling this function, and doing so twice at the same + time is a bad thing. (Found in a debugging session with dvossel + and mmichelson) ........ ................ + +2009-03-02 20:18 +0000 [r179407] Jason Parker <jparker@digium.com> + + * /, main/editline/configure, main/editline/np/unvis.c, + main/editline/sys.h, main/editline/configure.in: Merged revisions + 179396 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | + 9 lines Merged revisions 179395 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | + 1 line Remove several silly warnings in editline. One about a + broken preprocessor directive, and another about strlcpy/strlcat. + (closes issue #14264) Reported by: dimas ........ + ................ + +2009-03-02 17:19 +0000 [r179362] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179361 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r179361 | tilghman | 2009-03-02 11:18:48 -0600 (Mon, 02 Mar 2009) + | 2 lines Backport 1.6.0 fix to trunk (failsafe if db is not + loaded) ........ + +2009-03-02 14:14 +0000 [r179293] Joshua Colp <jcolp@digium.com> + + * /, main/audiohook.c: Merged revisions 179291 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r179291 | + file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix + issue where changing the volume of both directions of audio did + not work. (closes issue #14574) Reported by: KNK Patches: + audiohook_volume_fix.diff uploaded by KNK (license 545) ........ + +2009-03-01 23:28 +0000 [r179221-179256] Mark Michelson <mmichelson@digium.com> + + * apps/app_speech_utils.c, /: Merged revisions 179254 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar + 2009) | 5 lines Swap reversed timevals. This was pointed out by + ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........ + + * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | + mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 + lines Properly free memory and remove scheduler entries when a + transmission failure occurs. Previously, only the "data" field of + the sip_pkt created during __sip_reliable_xmit was freed when + XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was + called, this inevitably resulted in the reading and writing of + freed memory. XMIT_ERROR is a condition meaning that we don't + want to attempt resending the packet at all. The proper action to + take is to remove the scheduler entry we just created, free the + packet's data as well as the packet itself, and unlink it from + the list of packets on the sip_pvt structure. (closes issue + #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by + mmichelson (license 60) Tested by: Nick_Lewis ........ + +2009-02-27 21:48 +0000 [r179166] Russell Bryant <russell@digium.com> + + * configs/ais.conf.sample, res/res_ais.c, /, + doc/distributed_devstate.txt: Merged revisions 179164 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 + Feb 2009) | 2 lines Mark res_ais as experimental, as the binary + event format is subject to change. ........ + +2009-02-27 21:34 +0000 [r179163] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) + | 3 lines If config file is blank, don't load module. (Closes + issue #14563) ........ + +2009-02-27 21:25 +0000 [r179160] Russell Bryant <russell@digium.com> + + * /, UPGRADE.txt: Merged revisions 179154 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r179154 | + russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines + Add a note about the ordering of entries in sip.conf in 1.6.1. + ........ + +2009-02-27 19:06 +0000 [r179059] Jason Parker <jparker@digium.com> + + * /, doc/tex/channelvariables.tex: Merged revisions 179057 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb + 2009) | 8 lines Update documentation for DIALEDTIME and + ANSWEREDTIME variables. (closes issue #14566) Reported by: + klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by + klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by + klaus3000 (license 65) ........ + +2009-02-27 03:56 +0000 [r178988] Steve Murphy <murf@digium.com> + + * configs/features.conf.sample, /, main/features.c: Merged + revisions 178986 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | + 26 lines Merged revisions 178956 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 In this + case, it's just a matter of reducing the default timeouts from + 2000 to 1000 msec, as the max def feature digit timeout is no + longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 + -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default + feature digit timeout to 1000 ms from the previous default of + 500. As per bug 14515, a dev discussion arrived at a "mediated + concensus" of a default feature digit timeout of 1.0 sec. Some + voted for 1300; ctooley thought 1500 for distracted phone users + in phone booths; kpfleming put his foot down at 1.0 sec. Users + who found the previous default max delay of 250 msec perfect, are + welcome to override the new default. Notice that I said that 250 + msec was the default; wait a minute, you might say, the config + file said it was 500 msec!; well, because of the bug fix for + 14515, we found that 500 msec was actually enforcing a max of + 250. The bug fix would restore 500 msec, but we felt even that + was a bit tight for most users... 2000 msec was pushed earlier by + mmichelson, so that reduces to 1000 msec after the bug fix. + Enjoy! ........ ................ + +2009-02-26 17:50 +0000 [r178875] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 178871 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) + | 6 lines IAX2 prune realtime, minor tweak to last fix A return + statement was missing which caused unexpected cli output. issue + #14479 ........ + +2009-02-26 17:38 +0000 [r178869] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 178828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | + 34 lines Merged revisions 178804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | + 28 lines This patch prevents the feature detection timeout from + being cut in half. Because the ast_channel_bridge() call will + return 0 and pass a frame pointer for both DTMF_BEGIN and + DTMF_END, the feature_timer field in hte config struct is getting + decremented twice, which effectively cuts the digittimeout in + half. I added conditions to the if statement to only let DTMF_END + frames to flow thru, which solved the problem. Also, when the + frame pointer is null, let control flow thru-- this usually + happens on timeouts. I added a comment to the code to explain + what's going on and why. Many thanks to sodom for reporting this + problem. Personnally, it always seemed like something was wrong + with the featuredigittimeout, but I never could quite decide + what... and was too busy to investigate. This bug forced the + issue, and now we know. Sodom had other issues in 14515, but I + couldn't reproduce them. If he still has problems, and wants to + get them solved, he is welcome to reopen 14515. (closes issue + #14515) Reported by: sodom Patches: 14515.patch uploaded by murf + (license 17) Tested by: murf, sodom ........ ................ + +2009-02-26 16:44 +0000 [r178803] Joshua Colp <jcolp@digium.com> + + * /, main/file.c: Merged revisions 178801 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 | + file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix + an issue where the timer for file playback would not be stopped + if DAHDI was not installed. (closes issue #14541) Reported by: + grant ........ + +2009-02-26 16:07 +0000 [r178769] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 178767 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) + | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. + If "iax2 prune realtime all" was called, it would appear like the + command was successful, but in reality nothing happened. This is + because the reload that was supposed to take place checks the + config files, sees no changes, and does nothing. If there had + been a change in the the config file, the realtime users would + have been marked for deletion and everything would have been + fine. Now prune_users() and prune_peers() are called instead of + reload_config() to prune all users/peers that are realtime. These + functions remove all users/peers with the rtfriend and delme + flags set. iax2_prune_realtime() also lacked the code to properly + delete a single friend. For example. if iax2 prune realtime + <friend> was called, only the peer instance would be removed. The + user would still remain. (closes issue #14479) Reported by: + mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ + +2009-02-25 12:46 +0000 [r178511] Russell Bryant <russell@digium.com> + + * main/asterisk.c, /: Merged revisions 178509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) + | 10 lines Merged revisions 178508 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) + | 2 lines Update the copyright year for the main page of the + doxygen documentation. ........ ................ + +2009-02-24 23:28 +0000 [r178383-178448] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 178446 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600 + (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) + | 5 lines Add section about the #exec command in configuration + files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, + with additional notes by tilghman (license 14) ........ + ................ + + * main/asterisk.c, /: Merged revisions 178381 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 | + tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines + Apparently, a void cast doesn't override warn_unused_result. + ........ + +2009-02-24 20:44 +0000 [r178379-178380] Russell Bryant <russell@digium.com> + + * Makefile: revert accidental Makefile change. + + * main/rtp.c, Makefile, /: Merged revisions 178374 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r178374 | russell | 2009-02-24 14:39:57 -0600 + (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) + | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset + to 0 properly. (issue #14460) Reported by: moliveras Tested by: + russell ........ ................ + +2009-02-24 20:41 +0000 [r178305-178377] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 178375 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 | + tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines + The 3 possible errors with pipe(2) are all impossible in this + situation. ........ + + * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 + Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of + depending upon the astcanary process being inherited by init. + ........ + + * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 | + tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines + Cause astcanary to exit if Asterisk exits abnormally and doesn't + kill astcanary. Also, add some documentation supporting the use + of astcanary. (closes issue #14538) Reported by: KNK Patches: + asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545) + ........ + +2009-02-24 15:22 +0000 [r178232] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | + 16 lines Merged revisions 178205 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 + lines Skip check for extension when subscribing for MWI. Since + the remote side is not actually subscribing to a specific + extension when subscribing for MWI just skip the check to see if + the extension exists. They can't use it to specify the mailbox + either since we require configuration of that in sip.conf (closes + issue #14531) Reported by: festr ........ ................ + +2009-02-23 23:22 +0000 [r178172] Russell Bryant <russell@digium.com> + + * main/rtp.c, /: Merged revisions 178142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) + | 22 lines Merged revisions 178141 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) + | 14 lines Fix infinite DTMF when a BEGIN is received without an + END. This commit is related to rev 175124 of 1.4 where a previous + attempt was made to fix this problem. The problem with the + previous patch was that the inserted code needed to go _before_ + setting the lastrxts to the current timestamp. Because those were + the same, the dtmfcount variable was never decremented, and so + the END was never sent. In passing, I removed the dtmfsamples + variable which was completed unused. I also removed a redundant + setting of the lastrxts variable. (closes issue #14460) Reported + by: moliveras ........ ................ + +2009-02-21 16:04 +0000 [r177945] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 177944 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177944 | + tilghman | 2009-02-21 09:59:49 -0600 (Sat, 21 Feb 2009) | 2 lines + On update, test against the existence of sipregs. ........ + +2009-02-21 12:51 +0000 [r177851] Michiel van Baak <michiel@vanbaak.info> + + * /, channels/chan_sip.c: Merged revisions 177849 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177849 | + mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines + make chan_sip.c compile on OpenBSD again. ........ + +2009-02-20 23:05 +0000 [r177789] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 177787 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009) + | 16 lines Merged revisions 177786 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) + | 9 lines Don't print the CR-NL combination when we aren't + outputting to the manager. An embedded CR-NL in a CLI command + screws up several AMI parsers that don't expect to see that + combination in the middle of output. (Closes issue #14305) + Reported by: martins Patch by: tilghman ........ ................ + +2009-02-20 22:27 +0000 [r177785] Dwayne M. Hubbard <dhubbard@digium.com> + + * /, apps/app_fax.c: Merged revisions 177699 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177699 | + dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines + Make app_fax compatible with spandsp-0.0.6pre4 Prior to + spandsp-0.0.6pre4 the t30_stats_t structure used a + pages_transferred integer to indicate the number of pages + transferred (so far) during the fax session. The + spandsp-0.0.6pre4 release removed the pages_transferred integer + and replaced it with two different integers - pages_tx and + pages_rx. This revision uses the new integers for + spandsp-0.0.6pre4 while maintaining backwards compatibility for + previous spandsp releases. ........ + +2009-02-20 22:15 +0000 [r177760-177764] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/strings.h: Oops, last merge broke 1.6.1 branch + + * apps/app_system.c, include/asterisk/app.h, /, main/app.c: Merged + revisions 177664 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 | + tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines + Allow semicolons to be escaped, when passing arguments to the + System command. (closes issue #14231) Reported by: jcovert + Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 + (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by + jcovert (license 551) Tested by: jcovert ........ + + * include/asterisk/threadstorage.h, /: Merged revisions 177732 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r177732 | tilghman | 2009-02-20 15:25:37 -0600 + (Fri, 20 Feb 2009) | 10 lines Merged revisions 177701 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) + | 3 lines This exception does not appear to still be true for + Solaris 10, and OpenSolaris definitely needs it to be removed. + Fixed for snuff-home on -dev channel. ........ ................ + +2009-02-20 20:34 +0000 [r177700] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with + undefined audio codecs in chan_iax2 During iax2 call negotiation, + supported codecs are passed in an Information Element containing + a 2 byte field where each bit correlates to a specific codec. In + 1.6 only audio codec bits 0-12 are defined, leaving bits 13-14 + undefined. By default all bits are enabled unless specified + otherwise. Since its a 2 byte field and 13-14 are not defined, + these bits are never turned off. In trunk, bits 13-14 are + defined, which means 1.6 is advertising support for codecs it + does not have when talking to trunk. I fixed this by adding + #define for undefined audio codec bits. These bits are then + removed from iax2's full bandwidth capabilities. (closes issue + #14283) Reported by: jcovert + +2009-02-20 17:28 +0000 [r177663] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 177661 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r177661 | tilghman | 2009-02-20 11:22:19 -0600 (Fri, 20 Feb 2009) + | 2 lines Oops, merge broke trunk ........ + +2009-02-20 00:38 +0000 [r177626] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 177624 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177624 | + jpeeler | 2009-02-19 18:35:53 -0600 (Thu, 19 Feb 2009) | 7 lines + Set sip_request ast_str data to NULL so ast_str_copy allocates + space properly in copy_request (issue #14478) Reported by: + erik_dedecker ........ + +2009-02-20 00:26 +0000 [r177623] Steve Murphy <murf@digium.com> + + * /, main/Makefile, main/ast_expr2f.c: Merged revisions 177595 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r177595 | murf | 2009-02-19 16:56:50 -0700 (Thu, + 19 Feb 2009) | 32 lines Merged revisions 177540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was + already pretty 8-bit clean; but I'm still removing the --full + from the flex command so everything is uniform. ........ r177540 + | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines + This patch fixes a problem with 8-bit input to the ast_expr2 + scanner. The real culprit was the --full argument to flex in the + Makefile! This causes a 7-bit scanner to be generated. I reviewed + the rules and found one rule where I needed to specifically + include 8-bit chars for a token. I tested against the text + supplied by ibercom, and all looks very well. This has been there + a surprisingly long time! (closes issue #14498) Reported by: + ibercom Patches: 14498.patch uploaded by murf (license 17) Tested + by: murf ........ ................ + +2009-02-19 22:35 +0000 [r177539] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 177537 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r177537 | tilghman | 2009-02-19 16:33:00 -0600 + (Thu, 19 Feb 2009) | 14 lines Merged revisions 177536 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) + | 7 lines Fix up potential crashes, by reducing the sharing + between interactive and non-interactive threads. (closes issue + #14253) Reported by: Skavin Patches: 20090219__bug14253.diff.txt + uploaded by Corydon76 (license 14) Tested by: Skavin ........ + ................ + +2009-02-19 16:46 +0000 [r177389] Jeff Peeler <jpeeler@digium.com> + + * /, include/asterisk/channel.h: Merged revisions 177387 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19 + Feb 2009) | 3 lines Fix another merge error from 176708 ........ + +2009-02-19 16:40 +0000 [r177386] Joshua Colp <jcolp@digium.com> + + * apps/app_speech_utils.c, /: Merged revisions 177384 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r177384 | file | 2009-02-19 12:38:41 -0400 (Thu, + 19 Feb 2009) | 10 lines Merged revisions 177383 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 + lines If we are able to create a speech structure unset the ERROR + variable in case it was previously set. (issue #LUMENVOX-13) + ........ ................ + +2009-02-19 15:57 +0000 [r177358] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: Merged revisions 177356 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177356 | + jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines + Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev + on the asterisk-dev mailing list. Thanks! ........ + +2009-02-19 00:17 +0000 [r177294] Steve Murphy <murf@digium.com> + + * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177286 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r177286 | murf | 2009-02-18 16:50:57 -0700 (Wed, + 18 Feb 2009) | 39 lines Merged revisions 177225 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | + 34 lines This patch fixes a regression of sorts that was + introduced in rev 24425. It basically fixes AST-190/ABE-1782. + What was wrong: the user has 6000 extensions in one context; and + then 6000 contexts, one per extension. The parser could only + handle about 4893 of the 6000 extens in the single context. This + was due to the regression I mentioned. To get rid of shift/reduce + conflicts, Luigi set up right-recursive lists for globals, + context elements, switch lists, and statements. Right recursive + lists got rid of the warnings, but instead, they use up a + tremendous amount of stack space when the lists are long. I saw + this a few years back, and resolved not to fix it until someone + complained. That day has arrived! After the changes were made, I + ran the regression test suite, and there were no problems. I took + the test case the user provided, and added 100,000 extensions to + the single context, that already had 6,000 extens in it. (I'll + see your 6, and raise you 100!) It takes a few minutes to read it + all in, check it and generate code for it, but no problems. So, I + think I can say that fundamentally, there are no longer any + limits on the number of items you can place in contexts, + statement blocks, switches, or globals, beyond your virt mem + constraints. ........ ................ + +2009-02-18 23:15 +0000 [r177230] Kevin P. Fleming <kpfleming@digium.com> + + * main/frame.c, /: Merged revisions 177229 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177229 | + kpfleming | 2009-02-18 17:09:58 -0600 (Wed, 18 Feb 2009) | 3 + lines fix two very minor bugs: if anyone ever uses SLINEAR16 as a + format in RTP, ensure that the samples are byte-swapped to + network order if needed. also, when a smoother is operating on a + format that has a sample rate other than 8000 samples per second, + use the proper sample rate for computing delivery timestamps. + ........ + +2009-02-18 23:03 +0000 [r177228] David Vossel <dvossel@digium.com> + + * /, main/features.c: Merged revisions 177226 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177226 | + dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines + Locking issue in action_bridge and bridge_exec action_bridge() + and bridge_exec() both search for the channels to bridge to, and + then immediately drop the lock. Instead, they should hold the + lock until the masquerade is complete. This will guarantee the + channel remains and prevent any other weirdness from occurring. + In action_bridge() some more weirdness comes into play. Both + channels are needlessly locked at the same time and perform the + exact same logic. It makes sense from a coding organizational + standpoint, but could cause a theoretical deadlock so I split the + code up. There is an issue associated with this, but since its a + rather complicated thing to reproduce I'm not certain this alone + will close it. issue# 14296 Review: + http://reviewboard.digium.com/r/167/ ........ + +2009-02-18 20:16 +0000 [r177164] Jeff Peeler <jpeeler@digium.com> + + * channels/h323/chan_h323.h, channels/h323/cisco-h225.cxx, + channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4, + channels/h323/cisco-h225.h, /, channels/h323/caps_h323.cxx, + channels/h323/ast_ptlib.h (added), channels/h323/ast_h323.cxx, + configure, channels/h323/compat_h323.h, configure.ac, + channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4, + channels/h323/ast_h323.h: Merged revisions 177162 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r177162 | jpeeler | 2009-02-18 14:11:57 -0600 (Wed, 18 Feb 2009) + | 14 lines Modify h323 to build against PTLib as well as the + older PWLib Several changes in PTLib have occurred requiring + build time detection. Changes accounted for include the library + name change, config option change, install location change, and a + boolean type change which is handled by ast_ptlib.h. Also, the + sed check has been modified to properly work with autoconf >= + 2.62. (closes issue #14224) Reported by: bergolth Patches: + asterisk-autoconf-sed.patch uploaded by bergolth (license 661) + asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested + by: jpeeler ........ + +2009-02-18 19:30 +0000 [r177158] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 177101 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177101 | + russell | 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) | 8 lines + Re-add 'o' option to MeetMe, reverting rev 62297. Enabling this + option by default proved to be a bad idea, as the talker + detection is not very reliable. So, make it optional again, and + off by default. (issue #13801) Reported by: justdave ........ + +2009-02-18 19:09 +0000 [r177100] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/config.h: Merged revisions 177098 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r177098 | tilghman | 2009-02-18 13:05:15 -0600 + (Wed, 18 Feb 2009) | 9 lines Merged revisions 177096 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 + Feb 2009) | 2 lines Document the return value of the update + method (as requested on -dev list) ........ ................ + +2009-02-18 17:26 +0000 [r177037] Doug Bailey <dbailey@digium.com> + + * /, main/utils.c: Merged revisions 177035 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 | + dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines + Fixed error where a check for an zero length, terminated string + was needed. ........ + +2009-02-18 17:14 +0000 [r177007] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 177005 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r177005 | + file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines Fix + ordering of output for a ChannelUpdate manager event. (closes + issue #14497) Reported by: vinsik Patches: + chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) + ........ + +2009-02-18 16:20 +0000 [r176962] Doug Bailey <dbailey@digium.com> + + * /, main/utils.c: Merged revisions 176948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r176948 | + dbailey | 2009-02-18 10:09:12 -0600 (Wed, 18 Feb 2009) | 2 lines + Need to take into account the \0 terminator of the old string to + determine the amount available. ........ + +2009-02-18 15:59 +0000 [r176946] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 176943 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r176943 | + murf | 2009-02-18 08:35:26 -0700 (Wed, 18 Feb 2009) | 45 lines + This patch fixes merge_contexts_and_delete so it does not + deadlock when hints are present. Reason: when I re-engineered the + merge_and_delete func to reduce its lock time, I failed to notice + that the functions it calls still also do locking as before. This + leads to deadlocks on dialplan reloads, when there are actually + living, subscribed hints registered in the system. While the + reporter come across this problem while using AEL, I might note + that these deadlocks should also happen if extensions.conf were + used. Here I added these routines to pbx.c: + ast_add_extension_nolock add_pri_lockopt + ast_add_extension2_lockopt find_context add_hint_nolock All of + the above routines are static and restricted to be used only + within pbx.c, and more specifically within the + merge_contexts_and_delete routine. They are pretty much the same + as their counterparts except they don't lock contexts or hints. + Most of them now do the real work of their name-alike, with + optional locking via extra arguments, and are called by their + name-alike. The goal was to have the original functions so they + would behave exactly as before. Both PJ and I tested these fixes, + and the deadlocking problem is no longer encountered. (closes + issue #14357) Reported by: pj Patches: 14357.diff uploaded by + murf (license 17) Tested by: pj, murf ........ + +2009-02-18 06:15 +0000 [r176903-176906] Russell Bryant <russell@digium.com> + + * include/asterisk/heap.h, /: Merged revisions 176904 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r176904 | russell | 2009-02-18 00:14:47 -0600 (Wed, 18 Feb 2009) + | 2 lines Add example code for a heap traversal. ........ + + * main/pbx.c, /: Merged revisions 176901 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r176901 | + russell | 2009-02-18 00:00:40 -0600 (Wed, 18 Feb 2009) | 9 lines + Fix a number of incorrect uses of strncpy(). The big problem here + is that the 3rd argument provided in these uses of strncpy() did + not reserve a byte for the null terminator, leaving the potential + for writing one byte past the end of the buffer. Aside from this, + there were coding guidelines violations with regards to spacing, + as well as hard coded lengths being used instead of sizeof(). + ........ + +2009-02-18 00:23 +0000 [r176809] Shaun Ruffell <sruffell@digium.com> + + * /, codecs/codec_dahdi.c: Merged revisions 176760 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009) + | 10 lines Several changes to codec_dahdi to play nice with G723. + This commit brings in the changes that were living out on the + svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. + codec_dahdi.c now always uses signed linear as the simple codec + so that a soft g729 codec will not end up being preferred to the + hardware codec. There are also changes to allow codec_dahdi.c to + feed packets to the hardware in the native sample size of the + codec. This solves problems with choppy audio when using G723. + ........ + +2009-02-17 22:21 +0000 [r176731] Dwayne M. Hubbard <dhubbard@digium.com> + + * /, channels/chan_sip.c: Merged revisions 176705 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r176705 | + dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11 + lines create a UDPTL structure in create_addr_from_peer() if it + does not already exist for T38 This is required to create a UDPTL + structure in create_addr_from_peer() to handle the scenario where + 't38pt_udptl=yes' is not defined in the [general] section of + sip.conf but is defined the peer's context. I tested this patch + by enabling t38pt_udptl in the [general] section on one system + and only enabling t38pt_udptl in a peer's context on the system + sending a fax. Without the patch, the sending system will fail to + initiate T38 negotiation with the warning message, "No way to add + SDP without an UDPTL structure". When this patch is applied the + sending side will successfully initiate T38 negotiation. ........ + +2009-02-17 22:15 +0000 [r176711] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /, main/features.c, include/asterisk/channel.h: + Merged revisions 176708 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) + | 23 lines Merged revisions 176701 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) + | 17 lines Modify bridging to properly evaluate DTMF after first + warning is played The main problem is currently if the Dial flag + L is used with a warning sound, DTMF is not evaluated after the + first warning sound. To fix this, a flag has been added in + ast_generic_bridge for playing the warning which ensures that if + a scheduled warning is missed, multiple warrnings are not played + back (due to a feature evaluation or waiting for digits). + ast_channel_bridge was modified to store the nexteventts in the + ast_bridge_config structure as that information was lost every + time ast_channel_bridge was reentered, causing a hangup due to + incorrect time calculations. (closes issue #14315) Reported by: + tim_ringenbach Reviewed on reviewboard: + http://reviewboard.digium.com/r/163/ ........ ................ + +2009-02-17 21:41 +0000 [r176699] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/frame.h, /: Merged revisions 176697 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r176697 | mmichelson | 2009-02-17 15:40:09 -0600 (Tue, 17 Feb + 2009) | 3 lines Clear up documentation of AST_FRIENDLY_OFFSET in + frame.h ........ + +2009-02-17 21:24 +0000 [r176675] Russell Bryant <russell@digium.com> + + * main/timing.c, main/channel.c, /, res/res_timing_pthread.c, + res/res_timing_dahdi.c, include/asterisk/timing.h: Merged + revisions 176666 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r176666 | + russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines + Update the timing API to have better support for multiple timing + interfaces. 1) Add module use count handling so that timing + modules can be unloaded. 2) Implement unload_module() functions + for the timing interface modules. 3) Allow multiple timing + modules to be loaded, and use the one with the highest priority + value. 4) Report which timing module is being use in the "timing + test" CLI command. (closes issue #14489) Reported by: russell + Review: http://reviewboard.digium.com/r/162/ ........ + +2009-02-17 21:16 +0000 [r176644] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c, channels/chan_local.c, /: Merged revisions + 176592,176642 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r176592 | + tilghman | 2009-02-17 12:49:20 -0600 (Tue, 17 Feb 2009) | 4 lines + Add assertions in the quest to track down a refcount leak. + (closes issue #14485) Reported by: davevg ........ r176642 | + tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines + Prior to masquerade, move the group definitions to the channel + performing the masq, so that the group count lingers past the + bridge. (closes issue #14275) Reported by: kowalma Patches: + 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) + Tested by: kowalma ........ + +2009-02-17 20:57 +0000 [r176559-176637] Russell Bryant <russell@digium.com> + + * tests/test_heap.c (added), /: Merged revisions 176635 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r176635 | russell | 2009-02-17 14:56:26 -0600 (Tue, 17 + Feb 2009) | 4 lines Add a test module for the heap + implementation. Review: http://reviewboard.digium.com/r/160/ + ........ + + * include/asterisk/heap.h (added), /, main/Makefile, main/heap.c + (added): Merged revisions 176632 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r176632 | + russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines + Add an implementation of the heap data structure. A heap is a + convenient data structure for implementing a priority queue. Code + from svn/asterisk/team/russell/heap/. Review: + http://reviewboard.digium.com/r/160/ ........ + + * apps/app_queue.c, main/pbx.c, /: Merged revisions 176557 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r176557 | russell | 2009-02-17 11:33:38 -0600 (Tue, 17 + Feb 2009) | 12 lines Fix a race condition that caused device + states to become incorrect for hints. The problem here is that + the hint processing code was subscribed to the wrong event type. + So, it started processing state for a hint too soon, before the + device state cache had been updated. Also, fix a similar bug in + app_queue, as it was also subscribed to the wrong event type. + (closes issue #14461) Reported by: alecdavis ........ + +2009-02-17 14:48 +0000 [r176461-176503] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 176501 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r176501 | + tilghman | 2009-02-17 08:39:36 -0600 (Tue, 17 Feb 2009) | 3 lines + In this version, we can combine the queries, because we support + dropping nonexistent columns. ........ + + * /, channels/chan_sip.c: Merged revisions 176459 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009) + | 17 lines Merged revisions 176426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) + | 10 lines After a 'sip reload', qualifies for realtime peers + weren't immediately restarted, instead waiting until the next + registration. We're now caching the qualify across a + reload/restart and starting the qualify immediately upon loading + the peer. (closes issue #14196) Reported by: pdf Patches: + 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) + Tested by: pdf ........ ................ + +2009-02-16 23:57 +0000 [r176362] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 176355 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r176355 | dvossel | 2009-02-16 17:33:55 -0600 + (Mon, 16 Feb 2009) | 13 lines Merged revisions 176354 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) + | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not being + relayed correctly during bridging This should have been committed + with rev176247, but I missed it. srcupdate frames no longer break + out of the native bridge, but are not being sent to the other + call leg either. This fixs that. issue #13749 ........ + ................ + +2009-02-16 23:17 +0000 [r176321] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_skinny.c: Merged revisions 176320 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r176320 | tilghman | 2009-02-16 17:14:08 -0600 (Mon, 16 Feb 2009) + | 7 lines Use the correct list macros for deleting an item from + the middle of a list. (issue #13777) Reported by: pj Patches: + 20090203__bug13777.diff.txt uploaded by Corydon76 (license 14) + Tested by: pj ........ + +2009-02-16 22:00 +0000 [r176259] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/stringfields.h, /, main/utils.c: Merged + revisions 176255 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb + 2009) | 13 lines Merged revisions 176216 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb + 2009) | 3 lines fix a flaw in the ast_string_field_build() family + of API calls; these functions made no attempt to reuse the space + already allocated to a field, so every time the field was written + it would allocate new space, leading to what appeared to be a + memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 + -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the + last stringfields commit... don't mark additional space as + allocated if the string was built using already-allocated space + ........ ................ + +2009-02-16 21:50 +0000 [r176257] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_meetme.c: Merged revisions 176253 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r176253 | mmichelson | 2009-02-16 15:40:40 -0600 (Mon, 16 Feb + 2009) | 24 lines Merged revisions 176249,176252 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb + 2009) | 14 lines Open the DAHDI pseudo device and set it to be + nonblocking atomically Apparently on FreeBSD, attempting to set + the O_NONBLOCKING flag separately from opening the file was + causing an "inappropriate ioctl for device" error. While I cannot + fathom why this would be happening, I certainly am not opposed to + making the code a bit more compact/efficient if it also fixes a + bug. (closes issue #14482) Reported by: ys Patches: meetme.patch + uploaded by ys (license 281) Tested by: ys ........ r176252 | + mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 + lines Remove unused variable and make dev-mode compilation happy + ........ ................ + +2009-02-16 21:36 +0000 [r176251] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 176248 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r176248 | dvossel | 2009-02-16 15:30:17 -0600 + (Mon, 16 Feb 2009) | 11 lines Merged revisions 175597 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 + Feb 2009) | 4 lines Fixed iax2 key rotation backwards + compatibility Turns key rotation back on by default. Added bit + into encryption IE to indicate whether or not key rotation is + supported or not. If it is not supported then it is not enabled, + which insures backwards compatibility. This eliminates the need + for the keyrotate option in iax.conf, so it has been removed. + ........ ................ + +2009-02-16 18:38 +0000 [r176176] Mark Michelson <mmichelson@digium.com> + + * /, main/logger.c: Merged revisions 176174 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r176174 | + mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11 + lines Assist proper thread synchronization when stopping the + logger thread. I was finding that on my dev box, occasionally + attempting to "stop now" in trunk would cause Asterisk to hang. I + traced this to the fact that the logger thread was waiting on a + condition which had already been signalled. The logger thread + also need to be sure to check the value of the + close_logger_thread variable. The close_logger_thread variable is + only checked when the list of logmessages is empty. This allows + for the logger thread to print and free any pending messages + before exiting. ........ + +2009-02-16 17:10 +0000 [r176102] Russell Bryant <russell@digium.com> + + * /, channels/chan_features.c (removed): Merged revisions 176100 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r176100 | russell | 2009-02-16 11:09:24 -0600 (Mon, 16 + Feb 2009) | 4 lines Remove chan_features. Review: + http://reviewboard.digium.com/r/161/ ........ + +2009-02-16 17:07 +0000 [r176099] Tilghman Lesher <tlesher@digium.com> + + * configs/func_odbc.conf.sample: Eliminate mention of a variable + which exists only in trunk. (Thanks, jsmith) + +2009-02-16 15:38 +0000 [r176032] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 176030 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) | + 16 lines Merged revisions 176029 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 + lines Don't have the Via header stored as a stringfield as it can + change often during the lifetime of a dialog. This issue crept up + with subscriptions on the AA50. When an outgoing NOTIFY is sent a + new branch value is created and the Via header is changed to + reflect it. Since this was a stringfield a new spot in the pool + was used for the value while the old was left untouched/unused. + If the current pool was full a new pool was created. This would + cause memory usage to increase steadily. (issue #AA50-2332) + ........ ................ + +2009-02-16 09:42 +0000 [r176023] Michiel van Baak <michiel@vanbaak.info> + + * include/asterisk/manager.h, doc/unistim.txt, + channels/chan_unistim.c, /, channels/chan_sip.c: Merged revisions + 175952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) + | 10 lines Merged revisions 175921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) + | 3 lines fix mis-spelling of the word registered. Reported by + De_Mon on #asterisk-dev. ........ ................ + +2009-02-15 21:28 +0000 [r175831-175890] Russell Bryant <russell@digium.com> + + * main/sched.c, /, include/asterisk/sched.h: Merged revisions + 175882 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175882 | + russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines + Make ast_sched_report() and ast_sched_dump() thread safe. + ........ + + * main/sched.c, /, channels/chan_sip.c, include/asterisk/sched.h: + Merged revisions 175829 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175829 | + russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines + Fix a number of problems with ast_sched_report(). 1) It had + numerous coding guidelines violations with regards to formatting. + 2) It allocated memory using ast_calloc() that was never freed. + 3) It didn't check for failure from the allocation. 4) It used + sprintf() and strcat() to build the result, doing zero checking + to prevent writing past the end of the provided buffer. The + function also lacks API documentation, but that has not been + addressed in this commit. ........ + +2009-02-13 20:48 +0000 [r175662] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, configs/iax.conf.sample, channels/iax2.h: + Merged revisions 175597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | + dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines + Fixed iax2 key rotation backwards compatibility Turns key + rotation back on by default. Added bit into encryption IE to + indicate whether or not key rotation is supported or not. If it + is not supported then it is not enabled, which insures backwards + compatibility. This eliminates the need for the keyrotate option + in iax.conf, so it has been removed. ........ + +2009-02-13 19:52 +0000 [r175593] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 175591 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r175591 | mmichelson | 2009-02-13 13:49:38 -0600 + (Fri, 13 Feb 2009) | 22 lines Merged revisions 175590 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb + 2009) | 16 lines Fix a potential crash situation when using IMAP + voicemail If calling into VoiceMailMain when using IMAP storage, + it was possible to crash Asterisk by hanging up the phone when + prompted for a voicemail mailbox. This patch fixes the issue. + While it may appear that this patch is superficial, it allows + code execution to continue to the failure case just below the + IMAP_STORAGE code block where this patch has been applied (closes + issue #14473) Reported by: dwpaul Patches: + voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license + 689) ........ ................ + +2009-02-13 16:44 +0000 [r175551] Joshua Colp <jcolp@digium.com> + + * /, apps/app_record.c: Merged revisions 175549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | + file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add + an option to keep the recorded file upon hangup. (closes issue + #14341) Reported by: fnordian ........ + +2009-02-12 21:41 +0000 [r175370] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | + russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines + Remove useless string copy, and make sscanf safe again ........ + +2009-02-12 21:27 +0000 [r175342] Tilghman Lesher <tlesher@digium.com> + + * main/udptl.c, /: Merged revisions 175334 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) + | 16 lines Merged revisions 175311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) + | 9 lines Fix crashes when receiving certain T.38 packets. Also, + increase the maximum size of T.38 packets and warn users when + they try to set the limits above those maximums. (closes issue + #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt + uploaded by Corydon76 (license 14) Tested by: schern ........ + ................ + +2009-02-12 20:51 +0000 [r175300] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: Merged revisions 175298 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) + | 15 lines Merged revisions 175294 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) + | 9 lines Fix ParkedCall event information for From field in the + case of a blind transfer If the parker information can not be + obtained from the peer, try and see if the BLINDTRANSFER channel + variable has been set. Previously, a blind transfer to the + ParkAndAnnounce app would return nothing for the From. Closes + AST-189 ........ ................ + +2009-02-12 20:48 +0000 [r175257-175297] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | + russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines + Avoid using ast_strdupa() in a loop. ........ + + * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) + | 4 lines Don't enable something by default that has a dependency + on something _not_ enabled by default. menuselect was not happy + with this. ........ + +2009-02-12 18:50 +0000 [r175251] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 175250 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb + 2009) | 1 line correct warning message to not refer specifically + to DAHDI ........ + +2009-02-12 18:01 +0000 [r175190] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: Merged revisions 175188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) + | 12 lines Merged revisions 175187 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) + | 6 lines Fix crash in event of failed attempt to transfer to + parking The peer may not necessarily exist, such as in the case + of a transfer to ParkAndAnnounce. In this case don't try to play + a sound to it. ........ ................ + +2009-02-12 17:09 +0000 [r175130] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 175127 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r175127 | dvossel | 2009-02-12 11:07:17 -0600 (Thu, 12 Feb 2009) + | 4 lines Setting key rotation to be off by default Key rotation + breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). + As a follow up to this, I am investigating possible ways to allow + key rotation to be on by default and not affect the other + branches, but for now it must be turned off. ........ + +2009-02-12 17:08 +0000 [r175129] Russell Bryant <russell@digium.com> + + * main/rtp.c, /: Merged revisions 175125 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) + | 35 lines Merged revisions 175124 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) + | 27 lines Don't send DTMF for infinite time if we do not receive + an END event. I thought that this was going to end up being a + pretty gnarly fix, but it turns out that there was actually + already a configuration option in rtp.conf, dtmftimeout, that was + intended to handle this situation. However, in between Asterisk + 1.2 and Asterisk 1.4, the code that processed the option got + lost. So, this commit brings it back to life. The default timeout + is 3 seconds. However, it is worth noting that having this be + configurable at all is not really the recommended behavior in RFC + 2833. From Section 3.5 of RFC 2833: Limiting the time period of + extending the tone is necessary to avoid that a tone "gets + stuck". Regardless of the algorithm used, the tone SHOULD NOT be + extended by more than three packet interarrival times. A slight + extension of tone durations and shortening of pauses is generally + harmless. Three seconds will pretty much _always_ be far more + than three packet interarrival times. However, that behavior is + not required, so I'm going to leave it with our legacy behavior + for now. Code from svn/asterisk/team/russell/issue_14460 (closes + issue #14460) Reported by: moliveras ........ ................ + +2009-02-12 16:35 +0000 [r174947-175123] Mark Michelson <mmichelson@digium.com> + + * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions + 175121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | + mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 + lines Make lock information for ao2_trylock be more useful and + gnarly Core show locks information involving an ao2_trylock did + not show the function that called ao2_trylock, but would instead + show ao2_trylock as the source of the lock. This is not useful + when trying to debug locking issues. One bizarre note is that + this logic is already in 1.4 but somehow did not get merged to + trunk or the 1.6.X branches. ........ + + * apps/app_queue.c, /: Merged revisions 174951 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174951 | + mmichelson | 2009-02-11 17:12:57 -0600 (Wed, 11 Feb 2009) | 3 + lines Fix a bit of odd logic for announcing position. Sync with + 1.6.0's logic ........ + + * apps/app_queue.c, /: Merged revisions 174948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174948 | + mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 20 + lines Fix odd "thank you" sound playing behavior in app_queue.c + If someone has configured the queue to play an position or + holdtime announcement, then it is odd and potentially unexpected + to hear a "Thank you for your patience" sound when no position or + holdtime was actually announced. This fixes the announcement so + that the "thanks" sound is only played in the case that a + position or holdtime was actually announced. There is a way that + the "thank you" sound can be played without a position or + holdtime, and that is to set announce-frequency to a value but + keep announce-position and announce-holdtime both turned off. + (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch + uploaded by putnopvut (license 60) Tested by: caspy ........ + + * apps/app_dial.c, main/channel.c, main/pbx.c, /, + apps/app_dictate.c, apps/app_waitforsilence.c, + include/asterisk/channel.h: Merged revisions 174945 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb + 2009) | 29 lines Fix 'd' option for app_dial and add new option + to Answer application The 'd' option would not work for channel + types which use RTP to transport DTMF digits. The only way to + allow for this to work was to answer the channel if we saw that + this option was enabled. I realized that this may cause issues + with CDRs, specifically with giving false dispositions and answer + times. I therefore modified ast_answer to take another parameter + which would tell if the CDR should be marked answered. I also + extended this to the Answer application so that the channel may + be answered but not CDRified if desired. I also modified + app_dictate and app_waitforsilence to only answer the channel if + it is not already up, to help not allow for faulty CDR answer + times. All of these changes are going into Asterisk trunk. For + 1.6.0 and 1.6.1, however, all the changes except for the change + to the Answer application will go in since we do not introduce + new features into stable branches (closes issue #14164) Reported + by: DennisD Patches: 14164.patch uploaded by putnopvut (license + 60) Tested by: putnopvut Review: + http://reviewboard.digium.com/r/145 ........ + +2009-02-11 14:46 +0000 [r174846] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 174844 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174844 | + file | 2009-02-11 10:44:47 -0400 (Wed, 11 Feb 2009) | 10 lines + Tell the device state core a change happened when a channel is + freed but not a specific state. We need to do this because while + we know that the freeing of the channel may cause something to + become not in use we do not know this for sure. There may be + another channel that is still up which would cause it to be in + use. (closes issue #13238) Reported by: kowalma Patches: + 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) + Tested by: alecdavis ........ + +2009-02-10 23:21 +0000 [r174769-174823] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 | + mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11 + lines Fix potential for stack overflows in app_chanspy.c When + using the 'g' or 'e' options, the stack allocations that were + used could cause a stack overflow if a spyer stayed on the line + long enough without actually successfully spying on anyone. The + problem has been corrected by using static buffers and copying + the contents of the appropriate strings into them instead of + using functions like alloca or ast_strdupa ........ + + * main/manager.c, /: Merged revisions 174764 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 | + mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21 + lines Fix an fd leak that would occur in HTTP AMI sessions The + explanation behind this fix is a bit complicated, and I've + already typed it up in the code as a huge comment inside of + manager.c, so I'll give the abridged version here. We needed a + way to separate action-specific data from session-specific data. + Unfortunately, the only way to maintain API compatibility and to + not have to change every single manager action was to rename the + current mansession structure and wrap it inside a new mansession + structure which actually contains action- specific data. (closes + issue #14364) Reported by: awk Patches: 14364_better.patch + uploaded by putnopvut (license 60) Tested by: putnopvut Review: + http://reviewboard.digium.com/r/148/ ........ + +2009-02-10 20:17 +0000 [r174714] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 | + file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines + Only decrease inringing count if above zero. (issue #13238) + Reported by: kowalma ........ + +2009-02-10 18:18 +0000 [r174590] Matthew Nicholson <mnicholson@digium.com> + + * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb + 2009) | 25 lines Merged revisions 174583 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb + 2009) | 18 lines Improve behavior of jitterbuffer when + maxjitterbuffer is set. This change improves the way the + jitterbuffer handles maxjitterbuffer and dramatically reduces the + number of frames dropped when maxjitterbuffer is exceeded. In the + previous jitterbuffer, when maxjitterbuffer was exceeded, all new + frames were dropped until the jitterbuffer is empty. This change + modifies the code to only drop frames until maxjitterbuffer is no + longer exceeded. Also, previously when maxjitterbuffer was + exceeded, dropped frames were not tracked causing stats for + dropped frames to be incorrect, this change also addresses that + problem. (closes issue #14044) Patches: bug14044-1.diff uploaded + by mnicholson (license 96) Tested by: mnicholson Review: + http://reviewboard.digium.com/r/144/ ........ ................ + +2009-02-10 17:49 +0000 [r174545-174582] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 174580 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174580 | + file | 2009-02-10 13:48:29 -0400 (Tue, 10 Feb 2009) | 4 lines Set + the type for the peer structure to be a peer as the default. + (closes issue #14447) Reported by: triccyx ........ + + * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 | + file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines + Make the logic for inuse and inringing manipluation match that of + 1.4. The old broken logic would reset the values back to 0 during + certain scenarios causing the wrong state to be reported. (closes + issue #14399) Reported by: caspy (issue #13238) Reported by: + kowalma ........ + +2009-02-10 07:07 +0000 [r174471-174504] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, apps/app_voicemail.c, /: Merged revisions + 174503 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174503 | + tilghman | 2009-02-10 01:06:29 -0600 (Tue, 10 Feb 2009) | 2 lines + Fix0ring build ........ + + * apps/app_stack.c, /: Merged revisions 174470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174470 | + tilghman | 2009-02-09 23:39:33 -0600 (Mon, 09 Feb 2009) | 2 lines + Remove the usage of the KeepAlive app, as it no longer exists. + ........ + +2009-02-10 05:13 +0000 [r174428-174440] Steve Murphy <murf@digium.com> + + * apps/app_osplookup.c: This patch corrects warnings which seem to + appear only on 64-bit compilers, gcc-4.3.2. + + * apps/app_rpt.c: One final fix in the 1.6.1 release only; some + variables the compiler worries "may not be initialized". + + * apps/app_rpt.c, /: Merged revisions 174435 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174435 | + murf | 2009-02-09 21:49:02 -0700 (Mon, 09 Feb 2009) | 8 lines + This patch removes the use of AST_PBX_KEEPALIVE from app_rpt.c. + (closes issue #14435) Reported by: D_McNaul ........ + + * apps/app_rpt.c, /: Merged revisions 174432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174432 | + murf | 2009-02-09 21:36:22 -0700 (Mon, 09 Feb 2009) | 3 lines + More intptr_t work. ........ + + * apps/app_rpt.c, /: Merged revisions 174370 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174370 | murf | 2009-02-09 19:45:56 -0700 (Mon, 09 Feb 2009) | + 10 lines Merged revisions 174369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 + lines This patch solves some compiler complaints in both 32 and + 64-bit environments. ........ ................ + +2009-02-09 17:47 +0000 [r174330] David Vossel <dvossel@digium.com> + + * /, apps/app_externalivr.c: Merged revisions 174325 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r174325 | dvossel | 2009-02-09 11:26:02 -0600 (Mon, 09 Feb 2009) + | 9 lines Fixes issue with hangups not being sent and external + process never terminating. The ignore_hangup, run_dead, and + noanswer flags were never initilized to zero causing hangups to + never be issued. If the external script expects to be notified of + a hangup and never receives one, it runs indefinitely. (closes + issue #14251) Reported by: chris-mac Tested by: dvossel ........ + +2009-02-09 17:30 +0000 [r174326-174329] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 | + mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 + lines Fix something I messed up in the merge I just did ........ + + * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb + 2009) | 20 lines Merged revisions 174282 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb + 2009) | 12 lines Don't do an SRV lookup if a port is specified + RFC 3263 says to do A record lookups on a hostname if a port has + been specified, so that's what we're going to do. See section + 4.2. (closes issue #14419) Reported by: klaus3000 Patches: + patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 + (license 65) ........ ................ + +2009-02-09 14:50 +0000 [r174221] Joshua Colp <jcolp@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon, + 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 + lines Don't overwrite our pointer to the music class when music + on hold stops. We will use this if it starts again to see if we + can resume the music where it left off. (closes issue #14407) + Reported by: mostyn ........ ................ + +2009-02-07 16:18 +0000 [r174154] Russell Bryant <russell@digium.com> + + * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009) + | 10 lines Merged revisions 174148 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) + | 2 lines Fix a race condition that could cause a crash. ........ + ................ + +2009-02-07 00:09 +0000 [r174086] Dwayne M. Hubbard <dhubbard@digium.com> + + * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) + | 13 lines Merged revisions 174082 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) + | 5 lines check ast_strlen_zero() before calling ast_strdupa() in + sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter + didn't actually upload a properly-formed patch, instead a + modified chan_sip.c file was uploaded. I created a patch to + determine the changes, then modified the suggested changes to + create a proper fix. The summary above is a complete description + of the changes. (closes issue #13547) Reported by: tecnoxarxa + Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) + Tested by: tecnoxarxa ........ ................ + +2009-02-06 19:30 +0000 [r173994-174043] Joshua Colp <jcolp@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 + lines Don't subscribe to a mailbox on pseudo channels. It is + futile. This solves an issue where duplicated pseudo channels + would cause a crash because the first one would unsubscribe and + the next one would also try to unsubscribe the same subscription. + (closes issue #14322) Reported by: amessina ........ + + * /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | + 15 lines Merged revisions 173967-173968 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 + lines Some clients do not put the call-id for replaces at the + beginning, so support it being anywhere in the string. (closes + issue #14350) Reported by: fhackenberger ........ r173968 | file + | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a + debug message I put in by accident. ........ ................ + +2009-02-06 17:05 +0000 [r173964-173966] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb + 2009) | 14 lines Merged revisions 173917 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb + 2009) | 7 lines Limit the addition of the Contact header in SIP + responses according to various SIP RFCs. (closes issue #13602) + Reported by: hjourdain Tested by: mnicholson ........ + ................ + + * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h: + revert revision 173964 + + * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h: + Merged revisions 173952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb + 2009) | 14 lines Merged revisions 173917 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb + 2009) | 7 lines Limit the addition of the Contact header in SIP + responses according to various SIP RFCs. (closes issue #13602) + Reported by: hjourdain Tested by: mnicholson ........ + ................ + +2009-02-06 16:01 +0000 [r173904] Joshua Colp <jcolp@digium.com> + + * apps/app_chanspy.c, /, main/audiohook.c: Merged revisions 173902 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r173902 | file | 2009-02-06 11:59:17 -0400 (Fri, 06 Feb + 2009) | 4 lines Always detach and destroy the whisper and barge + audiohooks. Additionally also allow an audiohook to be detached + if it has not been attached. (closes issue #14414) Reported by: + bluecrow76 ........ + +2009-02-06 10:26 +0000 [r173850] Russell Bryant <russell@digium.com> + + * main/manager.c, /: Merged revisions 173848 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173848 | + russell | 2009-02-06 04:25:09 -0600 (Fri, 06 Feb 2009) | 2 lines + Resolve a memory leak that would occur on an invalid channel + given to Action: Status ........ + +2009-02-05 23:53 +0000 [r173779] Mark Michelson <mmichelson@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 173776 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, + 05 Feb 2009) | 14 lines Update extensions.conf.sample to be + correct. In trunk, the only necessary change pointed out was that + the call to ChanIsAvail uses an option that has been removed. For + the 1.6.1 branch, however, it appears that the sample file is + badly in need of updating since there are |'s used all over the + place there. My tentative plan is just to copy trunk's sample + config file to those branches since the info there is most + up-to-date and should be correct for use in 1.6.1 Thanks to macli + in #asterisk-dev for bringing this up ........ + +2009-02-05 23:51 +0000 [r173778] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_sqlite.c: Oops, merge from trunk broke 1.6.1 + +2009-02-05 23:31 +0000 [r173775] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 173773 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb + 2009) | 7 lines Properly set "seen" and "unseen" flags when + moving messages from the new to the old folder when using IMAP + for voicemail storage (closes issue #13905) Reported by: jaroth + Patches: foldermove_v2.patch uploaded by jaroth (license 50) + ........ + +2009-02-05 21:06 +0000 [r173699] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600 + (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) + | 12 lines Add new configuration option to make shared IMAP + mailboxes function as expected. The new option is "imapvmshareid" + which is an ID to tag multiple mailboxes using the same IMAP + storage location to function as one mailbox. This allows all + messages to be retrieved for any user in the group. The patch + alters the 'X-Asterisk-VM-Extension' header that is responsible + for matching voicemails for a given user. (closes issue #13673) + Reported by: howardwilkinson ........ ................ + +2009-02-05 20:35 +0000 [r173695] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 173693 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb + 2009) | 20 lines Merged revisions 173692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb + 2009) | 12 lines Fix situations where queue members could be + autopaused unexpectedly Specifically, this patch prevents us from + autopausing members when we receive a busy or congestion frame + from them. (closes issue #14376) Reported by: fiddur Patches: + 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur + ........ ................ + +2009-02-05 19:37 +0000 [r173658] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_sqlite.c, /: Merged revisions 173657 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r173657 | tilghman | 2009-02-05 13:36:29 -0600 (Thu, 05 Feb 2009) + | 2 lines Change the first field, or we don't get the necessary + field separation. ........ + +2009-02-05 18:50 +0000 [r173541-173595] Mark Michelson <mmichelson@digium.com> + + * apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600 + (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb + 2009) | 3 lines Add some missing cleanup to app_mixmonitor + ........ ................ + + * apps/app_mixmonitor.c, /: Merged revisions 173589 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600 + (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb + 2009) | 25 lines Fix a problem where a channel pointer becomes + invalid due to masquerading or hanging up. app_mixmonitor runs + its own thread to monitor the channel's activity and write the + mixed audio to a file. Since this thread runs independently of + the channel, it is possible that the mixmonitor thread's channel + pointer will point to freed memory when the channel either is + masqueraded or hangs up (technically, both cases are hangups, but + we need to handle the cases slightly differently). The solution + for this is to employ a datastore, which has the nice benefit of + allowing us to hook into channel masquerades and hangups and + update our pointer as necessary. If this looks familiar, this + same technique is employed in app_chanspy. app_chanspy is a bit + more involved since it does a lot more operations on the channel + that is being spied upon. app_mixmonitor does have an extra touch + that app_chanspy doesn't have, though. Since there is a thread + race between the channel's thread and the mixmonitor thread on a + hangup, we em- ploy a condition-and-boolean combination to ensure + that the channel thread finishes with our structure before the + mixmonitor thread attempts to free it. No crashes! (closes issue + #14374) Reported by: aragon Patches: 14374.patch uploaded by + putnopvut (license 60) Tested by: aragon, putnopvut ........ + ................ + + * apps/app_queue.c, /: Merged revisions 173507 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 | + mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 + lines Fix some areas where the incorrect interface was passed to + ast_device_state I swear it feels like I already did this once... + (closes issue #14359) Reported by: francesco_r ........ + +2009-02-04 21:32 +0000 [r173506] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions + 173502 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173502 | + dvossel | 2009-02-04 15:25:14 -0600 (Wed, 04 Feb 2009) | 9 lines + Fixes issue with IAX2 transfer not handing off calls. Reverts + changes in 116884 Fixes issue with IAX2 transfers not taking + place. As it was, a call that was being transfered would never be + handed off correctly to the call ends because of how call numbers + were stored in a hash table. The hash table, + "iax_peercallno_pvt", storing all the current call numbers did + not take into account the complications associated with + transferring a call, so a separate hash table was required. This + second hash table "iax_transfercallno_pvt" handles calls being + transfered, once the call transfer is complete the call is + removed from the transfer hash table and added to the peer hash + table resuming normal operations. Addition functions were created + to handle storing, removing, and comparing items in the + iax_transfercallno_pvt table. The changes reverted in 116884 + caused backwards compatibility issues involving iax2 transfer + with 1.6.0, 1.4, and 1.2. (closes issue #13468) Reported by: + nicox Tested by: dvossel ........ + +2009-02-04 21:28 +0000 [r173505] Jeff Peeler <jpeeler@digium.com> + + * include/asterisk/features.h, /, main/features.c: Merged revisions + 173500 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) + | 23 lines Merged revisions 173211 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) + | 17 lines Parking attempts made to one end of a bridge no longer + will hang up due to a parking failure. Parking attempts made + using either one-touch, or doing either a blind or assisted + transfer to the parking extension now keep up the bridge instead + of hanging up the attempted parked party. Normal causes for the + parking attempt to fail includes the specific specified extension + (via PARKINGEXTEN) not being available or if all the parking + spaces are currently in use. To avoid having to reverse a + masquerade park_space_reserve was made to provide foresight if a + parking attempt will succeed and if so reserve the parking space. + (closes issue #13494) Reported by: mdu113 Reviewed by Russell: + http://reviewboard.digium.com/r/133/ ........ ................ + +2009-02-04 18:52 +0000 [r173459] Tilghman Lesher <tlesher@digium.com> + + * main/tcptls.c, /: Merged revisions 173458 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 | + tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines + When using a socket as a FILE *, the stdio functions will + sometimes try to do an fseek() on the stream, which is an invalid + operation for a socket. Turning off buffering explicitly lets the + stdio functions know they cannot do this, thus avoiding a + potential error. (closes issue #14400) Reported by: fnordian + Patches: tcptls.patch uploaded by fnordian (license 110) ........ + +2009-02-04 17:46 +0000 [r173356-173399] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb + 2009) | 11 lines Merged revisions 173396 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb + 2009) | 3 lines Revert my previous change because it was stupid + ........ ................ + + * apps/app_chanspy.c, /: Merged revisions 173393 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb + 2009) | 11 lines Merged revisions 173392 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb + 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever + matter, but it's needed. ........ ................ + + * /, main/file.c: Merged revisions 173354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 | + mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 + lines Fix a problem where file playback would cause fds to remain + open forever The problem came from the fact that a frame read + from a format interpreter was not freed. Adding a call to + ast_frfree fixed this. The explanation for why this caused the + problem is a bit complex, but here goes: There was a problem in + all versions of Asterisk where the embedded frame of a filestream + structure was referenced after the filestream was freed. This was + fixed by adding reference counting to the filestream structure. + The refcount would increase every time that a filestream's frame + pointer was pointing to an actual frame of data. When the frame + was freed, the refcount would decrease. Once the refcount reached + 0, the filestream was freed, and as part of the operation, the + open files were closed as well. Thus it becomes more clear why a + missing ast_frfree would cause a reference leak and cause the + files to not be closed. You may ask then if there was a frame + leak before this patch. The answer to that is actually no! The + filestream code was "smart" enough to know that since the frame + we received came from a format interpreter, the frame had no + malloced data and thus didn't need to be freed. Now, however, + there is cleanup that needs to be done when we finish with the + frame, so we do need to call ast_frfree on the frame to be sure + that the refcount for the filestream is decremented + appropriately. (closes issue #14384) Reported by: fiddur Patches: + 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, + putnopvut ........ + +2009-02-04 00:46 +0000 [r173313] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 173311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 | + tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 + lines Ensure that commas placed in the middle of extension + character classes do not interfere with correct parsing of the + extension. Also, if an unterminated character class DOES make its + way into the pbx core (through some other method), ensure that it + does not crash Asterisk. (closes issue #14362) Reported by: + Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by + Corydon76 (license 14) Tested by: Corydon76 ........ + +2009-02-03 00:26 +0000 [r173115] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 173104 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600 + (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) + | 5 lines Add warning to standard config, that globals may be + overridden by other dialplan configuration files. (closes issue + #14388) Reported by: macli ........ ................ + +2009-02-03 00:01 +0000 [r173069] Terry Wilson <twilson@digium.com> + + * /, main/features.c: Merged revisions 173067 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) + | 9 lines Merged revisions 173066 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) + | 2 lines Fix a feature inheritance bug I added after code review + ........ ................ + +2009-02-02 18:15 +0000 [r172895] Leif Madsen <lmadsen@digium.com> + + * /, configs/res_ldap.conf.sample: Merged revisions 172894 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 + Feb 2009) | 7 lines Update the res_ldap.conf file with a better + working example. (closes issue #13861) Reported by: scramatte + Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage + (license 10) Tested by: jcovert ........ + +2009-02-01 02:45 +0000 [r172708-172743] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) + | 4 lines Blank argument crashes Asterisk (closes issue #14377) + Reported by: amorsen ........ + + * /, funcs/func_strings.c: Merged revisions 172706 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009) + | 7 lines Don't increment the loop, now that incrementing is + taken care of by the decoder function. (closes issue #14363) + Reported by: andrew53 Patches: func_strings_filter.patch uploaded + by andrew53 (license 519) ........ + +2009-01-31 00:07 +0000 [r172636-172638] Terry Wilson <twilson@digium.com> + + * configs/features.conf.sample, /: Merged revisions 172581 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30 + Jan 2009) | 2 lines Remove incorret line from sample config + ........ + + * CHANGES, configs/features.conf.sample, apps/app_dial.c, + main/global_datastores.c, /, main/features.c, + include/asterisk/global_datastores.h: Merged revisions 172580 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r172580 | twilson | 2009-01-30 15:29:12 -0600 + (Fri, 30 Jan 2009) | 44 lines Merged revisions 172517 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) + | 37 lines Fix feature inheritance with builtin features When + using builtin features like parking and transfers, the + AST_FEATURE_* flags would not be set correctly for all instances + when either performing a builtin attended transfer, or parking a + call and getting the timeout callback. Also, there was no way on + a per-call basis to specify what features someone should have on + picking up a parked call (since that doesn't involve the Dial() + command). There was a global option for setting whether or not + all users who pickup a parked call should have + AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or + PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan + variable which can be set either in the dialplan or with setvar + in channels that support it. This variable can be set to any + combination of 't', 'k', 'w', and 'h' (case insensitive matching + of the equivalent dial options), to set what features should be + activated on this channel. The patch moves the setting of the + features datastores into the bridging code instead of app_dial to + help facilitate this. 2) adds global options parkedcallparking, + parkedcallhangup, and parkedcallrecording to be similar to the + parkedcalltransfers option for globally setting features. 3) has + builtin_atxfer call builtin_parkcall if being transfered to the + parking extension since tracking everything through multiple + masquerades, etc. is difficult and error-prone 4) attempts to fix + all cases of return calls from parking and completed builtin + transfers not having the correct permissions (closes issue + #14274) Reported by: aragon Patches: + fix_feature_inheritence.diff.txt uploaded by otherwiseguy + (license 396) Tested by: aragon, otherwiseguy Review + http://reviewboard.digium.com/r/138/ ........ ................ + +2009-01-30 22:24 +0000 [r172609] Mark Michelson <mmichelson@digium.com> + + * /, include/asterisk/channel.h: Merged revisions 172598 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, + 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h + ........ + +2009-01-30 08:27 +0000 [r172509] Olle Johansson <oej@edvina.net> + + * CHANGES: Remove an extra "the" and restructure a bit + +2009-01-29 23:53 +0000 [r172504] Tilghman Lesher <tlesher@digium.com> + + * apps/app_rpt.c, main/asterisk.c, /, autoconf/ast_func_fork.m4, + configure, main/app.c: Merged revisions 172441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) + | 16 lines Merged revisions 172438 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) + | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at + startup. Otherwise, if Asterisk runs as a non-root user and the + administrator does a 'restart now', Asterisk loses the ability to + set QOS on packets. (closes issue #14004) Reported by: nemo + Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 + (license 14) Tested by: Corydon76 ........ ................ + +2009-01-29 22:05 +0000 [r172435] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged + revisions 172400 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 | + rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 + lines channels/chan_dahdi.c * Added doxygen comments to the major + dahdi structures. * Fixed PRI and SS7 using an incorrect string + value if the extension delimiter is not present in the Dial() + function. * Fixed SS7 not checking if the dialed extension is at + least as long as the stripmsd option. * Fixed PRI not handling + unknown TON/NPI prefix letters correctly. * Fixed some + uninitialized string variables on FXS ports. + configs/chan_dahdi.conf.sample * Updated some documentation. + ........ + +2009-01-29 20:54 +0000 [r172317-172402] Tilghman Lesher <tlesher@digium.com> + + * utils/muted.c, /: Merged revisions 146514 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk (closes issue + #14360) Reported by: oej ........ r146514 | russell | 2008-10-05 + 17:11:30 -0500 (Sun, 05 Oct 2008) | 2 lines Make this build on my + mac. ........ + + * configs/func_odbc.conf.sample, /: Merged revisions 172315 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29 + Jan 2009) | 2 lines Better document mode=multirow, based upon a + conversation with Jared. ........ + +2009-01-29 13:50 +0000 [r172272] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 172271 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29 + Jan 2009) | 5 lines The realtime_pgsql.sql script is missing a + couple of fields. closes issue #14339) Reported by: fiddur + Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678) + ........ + +2009-01-29 11:24 +0000 [r172218-172235] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 172234 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r172234 | + oej | 2009-01-29 12:19:29 +0100 (Tor, 29 Jan 2009) | 7 lines Make + sure register= line supports both port and expiry at the same + time. (closes issue #14185) Reported by: Nick_Lewis Patches: + chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657) + Tested by: Nick_Lewis ........ + + * /, channels/chan_sip.c: Merged revisions 172173 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 + lines Merged revisions 172169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 + lines Make sure that we always add the hangupcause headers. In + some cases, the owner was disconnected before we checked for the + cause. This patch implements a temporary storage in the pvt and + use that instead. The code is based on ideas from code from + Adomjan in issue #13385 (Add support for Reason: header) Thanks + to Klaus Darillion for testing! (closes issue #14294) related to + issue #13385 Reported by: klaus3000 and adomjan Patches: + bug14294b.diff uploaded by oej (license 306) Based on + 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan + (license 487) Tested by: oej, klaus3000 ........ ................ + + * /, configs/sip.conf.sample: Merged revisions 171880 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r171880 | oej | 2009-01-28 14:26:31 +0100 (Ons, 28 Jan 2009) | 2 + lines Add some more notes about device matching. ........ + +2009-01-28 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.1-rc1 released + +2009-01-28 22:52 +0000 [r172133] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_odbc.c, /: Merged revisions 172131 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r172131 | tilghman | 2009-01-28 16:48:01 -0600 (Wed, 28 Jan 2009) + | 7 lines Fix how we skip fields (to avoid fields which don't + exist) when doing an UPDATE. (closes issue #14205) Reported by: + maxgo Patches: 20090128__bug14205__5.diff.txt uploaded by + Corydon76 (license 14) Tested by: blitzrage ........ + +2009-01-28 20:56 +0000 [r172067] Steve Murphy <murf@digium.com> + + * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /, + main/features.c, include/asterisk/channel.h: Merged revisions + 172063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | + 52 lines Merged revisions 172030 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | + 46 lines This patch fixes h-exten running misbehavior in + manager-redirected situations. What it does: 1. A new Flag value + is defined in include/asterisk/channel.h, + AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the + bridge hangup exten code not to run the h-exten there (nor + publish the bridge cdr there). It will done at the pbx-loop level + instead. 2. In the manager Redirect code, I set this flag on the + channel if the channel has a non-null pbx pointer. I did the same + for the second (chan2) channel, which gets run if name2 is set... + and the first succeeds. 3. I restored the ending of the cdr for + the pbx loop h-exten running code. Don't know why it was removed + in the first place. 4. The first attempt at the fix for this bug + was to place code directly in the async_goto routine, which was + called from a large number of places, and could affect a large + number of cases, so I tested that fix against a fair number of + transfer scenarios, both with and without the patch. In the + process, I saw that putting the fix in async_goto seemed not to + affect any of the blind or attended scenarios, but still, I was + was highly concerned that some other scenarios I had not tested + might be negatively impacted, so I refined the patch to its + current scope, and jmls tested both. In the process, tho, I saw + that blind xfers in one situation, when the one-touch blind-xfer + feature is used by the peer, we got strange h-exten behavior. So, + I inserted code to swap CDRs and to set the HANGUP_DONT field, to + get uniform behavior. 5. I added code to the bridge to obey the + HANGUP_DONT flag, skipping both publishing the bridge CDR, and + running the h-exten; they will be done at the pbx-loop (higher) + level instead. 6. I removed all the debug logs from the patch + before committing. 7. I moved the AUTOLOOP set/reset in the + h-exten code in res_features so it's only done if the h-exten is + going to be run. A very minor performance improvement, but + technically correct. (closes issue #14241) Reported by: jmls + Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer + uploaded by murf (license 17) Tested by: murf, jmls ........ + ................ + +2009-01-28 17:29 +0000 [r171966] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600 + (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 + Jan 2009) | 2 lines Clarify log message (suggested by manxpower + on #asterisk-dev) ........ ................ + +2009-01-28 13:21 +0000 [r171857] Olle Johansson <oej@edvina.net> + + * configs/sip.conf.sample: Merged revisions 171838 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, + 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 + lines Add a better explanation of the difference between the + device namespace and the dialplan for newbies. ........ + ................ + +2009-01-27 22:01 +0000 [r171620-171693] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 171691 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600 + (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan + 2009) | 39 lines Fix devicestate problems for "always-on" agent + channels A revision to chan_agent attempted to "inherit" the + device state of the underlying channel in order to report the + device state of an agent channel more accurately. The problem + with the logic here is that it makes no sense to use this for + always-on agents. If the agent is logged in, then to the + underlying channel, the agent will always appear to be "in use," + no matter if the agent is on a call or not. The reason is that to + the underlying channel, the channel is currently in use on a call + to the AgentLogin application. The most common cause that I found + for this issue to occur was for a SIP channel to be the + underlying channel type for an Agent channel. If the SIP phone + re-registers, then the registration will cause the device state + core to query the device state of the SIP channel. Since the SIP + channel is in use, the Agent channel would also inherit this + status. Once the agent channel was set to "in use" there was no + way that the device state could change on that channel unless the + agent logged out. The solution for this problem is a bit + different in 1.4 than it is in the other branches. In 1.4, there + will be a one-line fix to make sure that only callback agents + will inherit device state from their underlying channel type. For + the other branches of Asterisk, since callback support has been + removed, there is also no need for device state inheritance in + chan_agent, so I will simply be removing it from the code. In + addition, the 1.4 source is getting a new comment to help the + next person who edits chan_agent.c. I'm adding a comment that a + agent_pvt's loginchan field may be used to determine if the agent + is a callback agent or not. (closes issue #14173) Reported by: + nathan Patches: 14173.patch uploaded by putnopvut (license 60) + Tested by: nathan, aramirez ........ ................ + + * /, main/slinfactory.c: Merged revisions 171622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan + 2009) | 26 lines Merged revisions 171621 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan + 2009) | 18 lines Prevent a crash from occurring when a jitter + buffer interpolated frame is removed from a slinfactory + slinfactory used the "samples" field of an ast_frame in order to + determine the amount of data contained within the frame. In + certain cases, such as jitter buffer interpolated frames, the + frame would have a non-zero value for "samples" but have NULL + "data" This caused a problem when a memcpy call in + ast_slinfactory_read would attempt to access invalid memory. The + solution in use here is to never feed frames into the slinfactory + if they have NULL "data" (closes issue #13116) Reported by: + aragon Patches: 13116.diff uploaded by putnopvut (license 60) + ........ ................ + + * apps/app_queue.c, /: Merged revisions 171618 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 | + mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 + lines Fix queue crashes that would occur after the calling + channel was masqueraded. The data passed to the + end_bridge_callback was assumed to be data which was still + stack'd. The problem was that with some call features, attended + transfers in particular, a new bridge thread is started once the + feature completes, meaning that when the end_bridge_callback is + called, the end_bridge_callback_data was invalid. To fix this + problem, there are two measures taken 1. Instead of pointing to + stacked data, we now used heap-allocated data for passing to the + end_bridge_callback in app_queue 2. Since bridges can end + multiple times on a single logical call, we wait until the final + bridge is broken to actually set any queue variables. This is + accomplished through reference-counting and the use of an + end_bridge_callback_data_fixup function in app_queue.c (closes + issue #14260) Reported by: ccesario Patches: 14260.patch uploaded + by putnopvut (license 60) Tested by: ccesario ........ + +2009-01-27 15:19 +0000 [r171540] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23 + lines Solving the same issue, but a bit different in trunk... + Merged revisions 171527 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 + lines Use the same branch tag in CANCEL as in INVITE Originally + putnopvut implemented some changes in revision 142079 that + according to the bug report seemed to have worked then, but + somehow fails now. I guess code, as humans, get old and forget + stuff. Anyway, this bug caused CANCEL not to work with picky + systems. Thanks Fredrik for pointing out where the bug in the SIP + messaging was. (closes issue #14346) Reported by: oej Patches: + bug14346.diff uploaded by oej (license 306) Tested by: oej + ........ ................ + +2009-01-26 14:58 +0000 [r171361] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r171326 | oej | 2009-01-26 14:44:40 +0100 (Mån, 26 Jan 2009) | + 17 lines Merged revisions 171264 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r171264 | oej | 2009-01-26 13:51:53 +0100 (Mån, 26 Jan 2009) | 9 + lines Don't retransmit 401 on REGISTER requests when + alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000 + Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by + klaus3000 (license 65) Tested by: klaus3000 ........ + ................ + +2009-01-26 00:04 +0000 [r171190] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009) + | 13 lines Merged revisions 171187 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) + | 6 lines Correctly track the hookstate (closes issue #13686) + Reported by: itiliti Patches: 20081013__bug13686.diff.txt + uploaded by Corydon76 (license 14) ........ ................ + +2009-01-25 13:40 +0000 [r170982] Sean Bright <sean.bright@gmail.com> + + * /, apps/app_page.c: Merged revisions 170980 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan + 2009) | 16 lines Merged revisions 170979 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan + 2009) | 9 lines Resolve a logic error that was causing Page() to + crash when more than one channel was specified. (closes issue + #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt + uploaded by seanbright (license 71) Tested by: kc0bvu ........ + ................ + +2009-01-25 02:52 +0000 [r170945] Russell Bryant <russell@digium.com> + + * include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) + | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second + part of this macro is written as 0[a] instead of a[0], it will + force a failure if the macro is used on a C++ object that + overloads the [] operator. ........ + +2009-01-24 13:57 +0000 [r170839] Tilghman Lesher <tlesher@digium.com> + + * configs/res_odbc.conf.sample, /: Merged revisions 170837 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600 + (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 + Jan 2009) | 2 lines Remove superfluous implementation note + (closes issue #14319) ........ ................ + +2009-01-23 23:53 +0000 [r170831] Richard Mudgett <rmudgett@digium.com> + + * /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 | + rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line + Fix asterisk.pdf generation if branch name has an underscore in + it. ........ + +2009-01-23 22:59 +0000 [r170792] Russell Bryant <russell@digium.com> + + * /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 | + russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines + Don't blow up if a branch name has an underscore in it ........ + +2009-01-23 20:57 +0000 [r170693-170722] Mark Michelson <mmichelson@digium.com> + + * configs/res_odbc.conf.sample, /: Merged revisions 170720 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600 + (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan + 2009) | 8 lines Add notes to the idlecheck explanation in + res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000 + Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by + klaus3000 (license 65) ........ ................ + + * contrib/i18n.testsuite.conf, /: Merged revisions 170677 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600 + (Fri, 23 Jan 2009) | 22 lines Merged revisions 170671 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan + 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use + deprecated syntax * Convert Wait,1 to Wait(1) * Convert + SetLanguage to Set(CHANNEL(language)) * Use 'n' for all + priorities beyond the first Also added test for Chinese numbers, + too. (closes issue #14320) Reported by: dant Patches: + i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license + 670) ........ ................ + +2009-01-23 20:20 +0000 [r170664] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 170652 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | + 11 lines Merged revisions 170648 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 + lines When a channel is answered make sure any indications + currently playing stop. Usually the phone would do this but if + the channel was already answered then they are being generated by + Asterisk and we darn well need to stop them. (closes issue + #14249) Reported by: RadicAlish ........ ................ + +2009-01-23 19:37 +0000 [r170637] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 170608 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r170608 | tilghman | 2009-01-23 13:25:10 -0600 + (Fri, 23 Jan 2009) | 9 lines Merged revisions 170588 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23 + Jan 2009) | 2 lines Additions to AST-2009-001 ........ + ................ + +2009-01-23 19:10 +0000 [r170507-170571] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c, /: Merged revisions 170569 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | + 11 lines Merged revisions 170568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 + lines When a call is forwarded stop any active indications. The + new channel will provide an indication, if need be, itself. + (closes issue #14310) Reported by: RadicAlish ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 170505 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) | + 11 lines Merged revisions 170504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 + lines Use the on hold flag to see if the call is on hold or not. + It is possible that our address for them will still be valid even + though they are on hold. (closes issue #14295) Reported by: + klaus3000 ........ ................ + +2009-01-23 17:49 +0000 [r170502] Michiel van Baak <michiel@vanbaak.info> + + * /, channels/chan_h323.c: Merged revisions 170501 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r170501 | mvanbaak | 2009-01-23 18:46:02 +0100 (Fri, 23 Jan 2009) + | 1 line let's use SENTINEL where needed ........ + +2009-01-23 16:35 +0000 [r170458] Doug Bailey <dbailey@digium.com> + + * channels/chan_dahdi.c: MWI messages included in CID spill was not + being properly handled and prevented the call from being + processed (issue #14313) Reported by: seandarcy Tested by: + dbailey + +2009-01-23 15:51 +0000 [r170395] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 170393 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170393 | mmichelson | 2009-01-23 09:44:27 -0600 (Fri, 23 Jan + 2009) | 36 lines Merged revisions 170392 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan + 2009) | 28 lines Fix broken call pickup There was a subtle change + in ast_do_masquerade which resulted in failed attempts to pickup + calls. The problem was that the value of the AST_FLAG_OUTGOING + flag was copied from the clone to the original channel. In the + case of call pickup, this meant that the AST_FLAG_OUTGOING flag + ended up being cleared on the channel that was attempting to + execute the pickup. Because this flag was not set, when ast_read + came across an answer frame, it ignored it. The result of this + was that the calling channel was never properly answered. This + fix changes the behavior in ast_do_masquerade to set the flags on + the original channel to the union of the flags on the clone + channel. This way, if the AST_FLAG_OUTGOING flag is set on either + of the two channels involved in the masquerade, the resulting + channel will have the flag set as well. (closes issue #14206) + Reported by: francesco_r Patches: 14206.patch uploaded by + putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut + ........ ................ + +2009-01-22 20:06 +0000 [r170242] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, /: Merged revisions 170240 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170240 | file | 2009-01-22 16:04:39 -0400 (Thu, 22 Jan 2009) | + 14 lines Merged revisions 170239 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 + lines Don't crash if RTCP is not enabled on an RTP structure but + statistics are output. (closes issue #14234) Reported by: jcovert + Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551) + rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........ + ................ + +2009-01-22 17:21 +0000 [r170178] Tilghman Lesher <tlesher@digium.com> + + * pbx/pbx_config.c, /: Merged revisions 170165 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170165 | tilghman | 2009-01-22 11:19:28 -0600 (Thu, 22 Jan 2009) + | 13 lines Merged revisions 170158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009) + | 6 lines Allow global variables after substitution to be as long + as other variables. (closes issue #14263) Reported by: markd + Patches: 20090120__bug14263.diff.txt uploaded by Corydon76 + (license 14) ........ ................ + +2009-01-22 16:54 +0000 [r170049-170150] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 170148 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | + 11 lines Merged revisions 170147 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 + lines If we are unable to request a DAHDI pseudo channel and we + are using the user introduction without review option make sure + it gets unset so other code does not blindly assume a DAHDI + pseudo channel exists. (closes issue #14282) Reported by: + cheesegrits ........ ................ + + * main/pbx.c, /: Merged revisions 170051 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170051 | file | 2009-01-22 11:14:50 -0400 (Thu, 22 Jan 2009) | + 13 lines Merged revisions 170050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6 + lines Do a string comparison instead of pointer comparison since + some people specify the context they are actually in as an + argument to get around some funkiness. (closes issue #14011) + Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga + (license 665) ........ ................ + + * apps/app_parkandannounce.c, /: Merged revisions 170047 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan + 2009) | 4 lines Clear the autoloop flag when parsing and setting + the context/extension/priority to go back to. When the channel + executes a PBX again we want it to start out at the point we + explicitly say and at that point it will not yet be doing + autoloop. (closes issue #14304) Reported by: jcovert ........ + +2009-01-22 00:46 +0000 [r169946] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/linkedlists.h: Merged revisions 169944 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r169944 | tilghman | 2009-01-21 18:44:52 -0600 + (Wed, 21 Jan 2009) | 16 lines Merged revisions 169943 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) + | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really + wanted to ask is whether autoconf detected a static initializer + value. This fixes rwlocks on all such platforms (mainly, Mac OS + X). (closes issue #13767) Reported by: jcovert Patches: + 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14) + Tested by: jcovert, Corydon76 ........ ................ + +2009-01-21 23:28 +0000 [r169871] Joshua Colp <jcolp@digium.com> + + * main/pbx.c, /: Merged revisions 169869 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r169869 | file | 2009-01-21 19:25:27 -0400 (Wed, 21 Jan 2009) | + 11 lines Merged revisions 169867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4 + lines Read lock the contexts to maintain the locking order when + we are notified that the state of a device has changed. (closes + issue #13839) Reported by: mcallist ........ ................ + +2009-01-21 22:23 +0000 [r169830] Michiel van Baak <michiel@vanbaak.info> + + * /, doc/tex/extensions.tex: Merged revisions 169793 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r169793 | mvanbaak | 2009-01-21 23:04:16 +0100 (Wed, 21 Jan 2009) + | 2 lines remove duplicated sentence. ........ + +2009-01-21 22:11 +0000 [r169792-169796] Mark Michelson <mmichelson@digium.com> + + * /, main/say.c: Merged revisions 169794 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169794 | + mmichelson | 2009-01-21 16:10:02 -0600 (Wed, 21 Jan 2009) | 17 + lines Fix a crash when saying certain numbers in Chinese This + commit fixes a crash that was occurring when attempting to say a + number between 10000 and 100000 due to dividing by 0. This also + removes some places where a "zero" is spoken when it should not + be. (closes issue #14291) Reported by: dant Patches: + say.c-14291.diff uploaded by dant (license 670) Tested by: dant + ........ + + * /, channels/chan_sip.c: Merged revisions 169791 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169791 | + mmichelson | 2009-01-21 15:53:55 -0600 (Wed, 21 Jan 2009) | 18 + lines Further fix some oddities in sip show users and sip show + peers logic ccesario on IRC pointed out that his sip peers were + not displayed properly when he would issue the command "sip show + peers." The problem was that the onlymatchonip field was used to + determine if the endpoint was a "peer" or "user." The tricky part + is that a "friend" is supposed to be treated as both a "user" and + a "peer" but the logic would not allow "friends" to show up as + "peers" since onlymatchonip was set to FALSE for friends. I have + modified the sip_peer structure to more explicitly keep track of + what type endpoint it is so that the various manager and CLI + commands will display the expected information Reported by + ccesario via IRC Tested by ccesario ........ + +2009-01-21 21:05 +0000 [r169725] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 169723 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r169723 | tilghman | 2009-01-21 15:03:40 -0600 (Wed, 21 Jan 2009) + | 15 lines Merged revisions 169722 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) + | 8 lines Extra NULLs in the output cause some terminal types to + abort in the middle of a color code, causing terminal weirdness. + (closes issue #14130) Reported by: coolmig Patches: + 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76, coolmig ........ ................ + +2009-01-21 17:40 +0000 [r169674] Steve Murphy <murf@digium.com> + + * utils/refcounter.c, /: Merged revisions 169673 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169673 | + murf | 2009-01-21 10:21:40 -0700 (Wed, 21 Jan 2009) | 14 lines + This patch corrects a segfault reported in 14289, due to a null + ptr being refd. Yes, seanbright is right in the bug comments, + that is the fix. Sorry for this oversight; I guess my personal + usage didn't have this happen! murf (closes issue #14289) + Reported by: jamesgolovich ........ + +2009-01-21 10:49 +0000 [r169622-169626] Russell Bryant <russell@digium.com> + + * /: Merged revisions 169625 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169625 | + russell | 2009-01-21 04:49:00 -0600 (Wed, 21 Jan 2009) | 2 lines + Remove properties that erroneously got merged into trunk ........ + + * main/tcptls.c, /: Merged revisions 169620 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169620 | + russell | 2009-01-21 04:26:07 -0600 (Wed, 21 Jan 2009) | 10 lines + Fix a regression in TCP support. This patch fixes a problem that + caused chan_sip to think that every open TCP session was to a + remote address of 0.0.0.0:0. (closes issue #14287) Reported by: + jamesgolovich Patches: bug-14287.diff.txt uploaded by + jamesgolovich (license 176) ........ + +2009-01-21 00:35 +0000 [r169559-169613] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 169611 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169611 | + mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22 + lines Fix device state parsing issues for channel names with + multiple slashes The fix being applied is a bit different for + trunk and the 1.6.X branches. For trunk, we only wish to strip + off the characters beyond the second slash if the channel is a + Local channel (i.e. we are removing the /n from the device name). + Other channel technologies with multiple slashes (e.g. DAHDI) + need the information after the second slash in order to get the + proper device state information. In addition to this fix, the + 1.6.X branches are receiving a much more important fix as well. + The problem in 1.6.X is that the member's device name was being + directly changed instead of having a copy changed. This meant + that we would strip off the second slash and trailing characters + and then leave the member's device name like that permanently + thereafter. (closes issue #14014) Reported by: kebl0155 Patches: + 14014_number2.patch uploaded by putnopvut (license 60) Tested by: + kebl0155 ........ + + * apps/app_queue.c, /: Merged revisions 169574 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169574 | + mmichelson | 2009-01-20 15:57:24 -0600 (Tue, 20 Jan 2009) | 6 + lines Use the default timeout for a queue instead of -1 (closes + issue #14272) Reported by: timking ........ + + * /, channels/chan_sip.c: Merged revisions 169557 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169557 | + mmichelson | 2009-01-20 14:10:31 -0600 (Tue, 20 Jan 2009) | 19 + lines Convert the character pointers in a sip_request to be + pointer offsets When an ast_str expands to hold more data, any + pointers that were pointing to the data prior to the expansion + will be pointing at invalid memory. This change makes such + pointers used in chan_sip.c instead be offsets from the beginning + of the string so that the same math may be applied no matter + where in memory the string resides. To help ease this transition, + a macro called REQ_OFFSET_TO_STR has been added to chan_sip.c so + that given a sip_request and an offset, the string at that offset + is returned. (closes issue #14220) Reported by: riksta Tested by: + putnopvut Review http://reviewboard.digium.com/r/126/ ........ + +2009-01-20 19:31 +0000 [r169488-169554] Terry Wilson <twilson@digium.com> + + * /, main/features.c: Merged revisions 169510 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169510 | + twilson | 2009-01-20 13:22:24 -0600 (Tue, 20 Jan 2009) | 7 lines + Make a proper builtin attended transfer to parking work This is + an ugly hack from 1.4 that allows the timeout callback from a + parked call to use the right channel name for the callback when + the park is done with a builtin attended transfer (that isn't + completed early). This hasn't ever worked in trunk and no one has + complained yet, so eh. ........ + + * /, main/features.c: Merged revisions 169486 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r169486 | twilson | 2009-01-20 12:48:14 -0600 (Tue, 20 Jan 2009) + | 13 lines Merged revisions 169485 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009) + | 6 lines Don't play audio to the channel if we've masqueraded + (closes issue #14066) Reported by: bluefox Tested by: + otherwiseguy, bluefox ........ ................ + +2009-01-19 20:10 +0000 [r169368] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /, apps/app_userevent.c: Merged revisions 169365 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r169365 | tilghman | 2009-01-19 14:05:52 -0600 + (Mon, 19 Jan 2009) | 11 lines Merged revisions 169364 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) + | 4 lines Truncate userevents at the end of a line, when the + command exceeds the buffer. (closes issue #14278) Reported by: + fnordian ........ ................ + +2009-01-19 15:55 +0000 [r169213] Mark Michelson <mmichelson@digium.com> + + * channels/chan_local.c, /: Merged revisions 169211 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r169211 | mmichelson | 2009-01-19 09:54:06 -0600 + (Mon, 19 Jan 2009) | 21 lines Merged revisions 169210 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan + 2009) | 13 lines Prevent a crash in chan_local due to a potential + NULL pointer dereference Move the check for if both channels on a + local_pvt have generators to below where p->chan is checked for + NULLity (NULLness?). This prevents a crash from occurring if + p->chan is NULL. (closes issue #14189) Reported by: sascha + Patches: 14189.patch uploaded by putnopvut (license 60) Tested + by: sascha ........ ................ + +2009-01-17 18:46 +0000 [r169154] Doug Bailey <dbailey@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add + discriminator for when ring pulse alert signal is used to preface + MWI spills This prevents the situation when MWI messages are + added to caller ID spills causing the channel to be hung up + +2009-01-17 01:59 +0000 [r168981-169082] Terry Wilson <twilson@digium.com> + + * main/tcptls.c, /, main/http.c, include/asterisk/tcptls.h: Merged + revisions 169080 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169080 | + twilson | 2009-01-16 19:56:36 -0600 (Fri, 16 Jan 2009) | 8 lines + Fix qualify for TCP peer (closes issue #14192) Reported by: + pabelanger Patches: asterisk-bug14192.diff.txt uploaded by + jamesgolovich (license 176) Tested by: jamesgolovich ........ + + * /, channels/chan_sip.c: Merged revisions 169044 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r169044 | + twilson | 2009-01-16 18:03:39 -0600 (Fri, 16 Jan 2009) | 8 lines + Fix port :0 added to SIP INVITE URI when outboundproxy used + (closes issue #14233) Reported by: chris-mac Patches: + asterisk-bug14233.diff.txt uploaded by jamesgolovich (license + 176) Tested by: jamesgolovich, chris-mac, otherwiseguy ........ + + * /, main/features.c: Merged revisions 168941 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168941 | twilson | 2009-01-16 16:16:23 -0600 (Fri, 16 Jan 2009) + | 19 lines Merged revisions 168716 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) + | 12 lines Convert call to park_call_full to + masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE + return value, we need to use masqueraded parking, otherwise we + will try to call ast_hangup() in __pbx_run() and in + do_parking_thread() and then promptly crash. (closes issue + #14215) Reported by: waverly360 Tested by: otherwiseguy (closes + issue #14228) Reported by: kobaz Tested by: otherwiseguy ........ + ................ + +2009-01-16 22:46 +0000 [r168979] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 168976 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168976 | mmichelson | 2009-01-16 16:43:09 -0600 (Fri, 16 Jan + 2009) | 26 lines Merged revisions 168975 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan + 2009) | 18 lines Account for possible NULL pointer when we + receive a 408 in response to a REGISTER It may be that by the + time we receive a reply to a REGISTER request, the attempt has + timed out and thus the registry structure pointed to by the + corresponding sip_pvt has gone away. This situation was handled + properly for a 200 OK response, but the 408 case assumed that the + sip_registry struct was non-NULL, thus potentially causing a + crash This commit fixes this assumption and prints out a message + to the console if we should receive a late 408 response to a + REGISTER (closes issue #14211) Reported by: aborghi Patches: + 14211.diff uploaded by putnopvut (license 60) Tested by: aborghi + ........ ................ + +2009-01-16 18:55 +0000 [r168836] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/say.h, apps/app_voicemail.c, /, main/say.c: + Merged revisions 168832 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168832 | tilghman | 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) + | 13 lines Merged revisions 168828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) + | 6 lines Fix the conjugation of Russian and Ukrainian languages. + (related to issue #12475) Reported by: chappell Patches: + vm_multilang.patch uploaded by chappell (license 8) ........ + ................ + +2009-01-16 00:47 +0000 [r168739-168748] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, /: Merged revisions 168746 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168746 | murf | 2009-01-15 17:34:31 -0700 (Thu, 15 Jan 2009) | + 20 lines Merged revisions 168745 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | + 14 lines This patch fixes a problem where a goto (or jump, in + this case) fails a consistency check because it can't find a + matching extension. The problem was a missing instruction to end + the range notation in the code where it converts the pattern into + a regex and uses the regex code to determine the match. I tested + using the AEL code the user supplied, and now, the consistency + check passes. (closes issue #14141) Reported by: dimas ........ + ................ + + * main/ast_expr2.c, /, main/ast_expr2.h, main/ast_expr2.y: Merged + revisions 168737 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168737 | + murf | 2009-01-15 13:54:59 -0700 (Thu, 15 Jan 2009) | 16 lines + This patch allows null args in ast_expr2 func calls, and fixes + commas being converted to pipes, which was 1.4 type stuff. If the + user says count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); + then it won't complain about the empty arg (c,,...) and fabled's + patch won't let it swap the commas for pipes. Ran it thru my + dialplan and no complaints. (closes issue #14169) Reported by: + fabled Patches: function-argument-separator-fix.diff uploaded by + fabled (license 448) ........ + +2009-01-15 19:17 +0000 [r168729] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Merged revisions 168728 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168728 | + mmichelson | 2009-01-15 13:16:29 -0600 (Thu, 15 Jan 2009) | 3 + lines Fix the compactheaders option in sip.conf ........ + +2009-01-15 19:05 +0000 [r168727] Olle Johansson <oej@edvina.net> + + * /, configs/extconfig.conf.sample: Merged revisions 168722 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r168722 | oej | 2009-01-15 19:47:14 +0100 (Tor, + 15 Jan 2009) | 10 lines Merged revisions 168721 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 + lines Meetme actually has realtime but wasn't documented ........ + ................ + +2009-01-15 19:00 +0000 [r168726] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 168725 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168725 | + mmichelson | 2009-01-15 13:00:06 -0600 (Thu, 15 Jan 2009) | 17 + lines Remove an unneeded condition for line addition to a SIP + request/response In Asterisk 1.4 and 1.6.0, the sip_request + structure had a statically allocated buffer to hold the text of + the request. There was a check in the add_line function to not + attempt to write the line into the buffer if we did not have room + for it. In trunk and Asterisk versions starting with 1.6.1, an + expandable ast_str structure is used to hold the text. Since it + may grow to fit an arbitrarily sized string, this check in + add_line is no longer valid. I found this oddity while attempting + to fix issue #14220; however, I do not believe that this is the + fix for that issue since the output supplied by the reporter did + not contain the warning message that would be printed had this + condition been satisfied. ........ + +2009-01-15 18:20 +0000 [r168714-168715] Olle Johansson <oej@edvina.net> + + * /, configs/sip.conf.sample: Merged revisions 168711 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4 + lines Clarify some misunderstandings and make it even more clear + that you can refer to a peer in the register= line. ........ + + * /, channels/chan_sip.c: Merged revisions 168712 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168712 | + oej | 2009-01-15 19:08:59 +0100 (Tor, 15 Jan 2009) | 3 lines Make + sure that we have the same terminology in sip.conf.sample and the + source code warning. Thanks Nick Lewis for pointing this out in + the bug tracker. ........ + +2009-01-15 15:37 +0000 [r168707] Sean Bright <sean.bright@gmail.com> + + * /, apps/app_meetme.c: Merged revisions 168705 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168705 | + seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11 + lines Add a missing unlock and properly handle the 'maxusers' + setting on MeetMe conferences. We were using the 'user number' + field to compare against the maximum allowed users, which works + assuming users with lower user numbers didn't leave the + conference. (closes issue #14117) Reported by: sergedevorop + Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright + (license 71) Tested by: sergedevorop ........ + +2009-01-15 00:15 +0000 [r168631] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 168629 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan + 2009) | 24 lines Merged revisions 168628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan + 2009) | 16 lines Fix some crashes from bad datastore handling in + app_queue.c * The queue_transfer_fixup function was searching for + and removing the datastore from the incorrect channel, so this + was fixed. * Most datastore operations regarding the + queue_transfer datastore were being done without the channel + locked, so proper channel locking was added, too. (closes issue + #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by + putnopvut (license 60) Tested by: ZX81, festr ........ + ................ + +2009-01-14 21:55 +0000 [r168625] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn/isdn_lib.c, /: Merged revisions 168623 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r168623 | rmudgett | 2009-01-14 15:51:06 -0600 + (Wed, 14 Jan 2009) | 11 lines Merged revisions 168622 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009) + | 4 lines * Fixed create_process() allocation of process ID + values. The allocated process IDs could overflow their respective + NT and TE fields. Affects outgoing calls. ........ + ................ + +2009-01-14 21:30 +0000 [r168621] Steve Murphy <murf@digium.com> + + * /, apps/app_page.c: Merged revisions 168613 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168613 | murf | 2009-01-14 13:51:26 -0700 (Wed, 14 Jan 2009) | 9 + lines Merged revisions 168608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 + line app_page was failing to compile in dev-mode on my gcc-4.2.4 + system. This change gets rid of the warning. ........ + ................ + +2009-01-14 21:00 +0000 [r168618] Sean Bright <sean.bright@gmail.com> + + * contrib/scripts/autosupport, /: Merged revisions 168615 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r168615 | seanbright | 2009-01-14 15:58:26 -0500 + (Wed, 14 Jan 2009) | 16 lines Merged revisions 168614 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan + 2009) | 9 lines Update autosupport script to supply info for both + Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x + and trunk instead of zttest. (closes issue #14132) Reported by: + dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by + dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded + by dsedivec (license 638) ........ ................ + +2009-01-14 20:18 +0000 [r168611] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 168610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168610 | + mmichelson | 2009-01-14 14:13:48 -0600 (Wed, 14 Jan 2009) | 9 + lines Restore the "sip show users" and "sip show user" CLI + commands (closes issue #14180) Reported by: amorsen Patches: + sip_show_users_161v3.diff uploaded by putnopvut (license 60) + Tested by: blitzrage, amorsen ........ + +2009-01-14 19:12 +0000 [r168606] Tilghman Lesher <tlesher@digium.com> + + * main/udptl.c, /: Merged revisions 168604 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168604 | tilghman | 2009-01-14 13:11:14 -0600 (Wed, 14 Jan 2009) + | 14 lines Merged revisions 168603 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) + | 7 lines Don't read into a buffer without first checking if a + value is beyond the end. (closes issue #13600) Reported by: atis + Patches: 20090106__bug13600.diff.txt uploaded by Corydon76 + (license 14) Tested by: atis ........ ................ + +2009-01-14 02:11 +0000 [r168582-168596] Terry Wilson <twilson@digium.com> + + * /, apps/app_page.c: Merged revisions 168594 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) + | 27 lines Merged revisions 168593 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) + | 20 lines Don't overflow when paging more than 128 extensions + The number of available slots for calls in app_page was hardcoded + to 128. Proper bounds checking was not in place to enforce this + limit, so if more than 128 extensions were passed to the Page() + app, Asterisk would crash. This patch instead dynamically + allocates memory for the ast_dial structures and removes the + (non-functional) arbitrary limit. This issue would have special + importance to anyone who is dynamically creating the argument + passed to the Page application and allowing more than 128 + extensions to be added by an outside user via some external + interface. The patch posted by a_villacis was slightly modified + for some coding guidelines and other cleanups. Thanks, + a_villacis! (closes issue #14217) Reported by: a_villacis + Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch + uploaded by a (license 660) Tested by: otherwiseguy ........ + ................ + + * /, res/res_http_post.c: Merged revisions 168588 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168588 | + twilson | 2009-01-13 17:05:43 -0600 (Tue, 13 Jan 2009) | 5 lines + Fully overwrite a same-named file when uploading (closes issue + #14190) Reported by: timking ........ + + * /, channels/chan_sip.c: Merged revisions 168578 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168578 | twilson | 2009-01-13 16:22:34 -0600 (Tue, 13 Jan 2009) + | 14 lines Merged revisions 168551 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009) + | 7 lines Don't pass a value with a side effect to a macro + (closes issue #14176) Reported by: paraeco Patches: + chan_sip.c.diff uploaded by paraeco (license 658) ........ + ................ + +2009-01-13 19:35 +0000 [r168565] Russell Bryant <russell@digium.com> + + * main/indications.c, main/channel.c, channels/chan_misdn.c, + channels/chan_skinny.c, funcs/func_channel.c, main/app.c, + res/snmp/agent.c, res/res_indications.c, channels/chan_unistim.c, + main/pbx.c, apps/app_read.c, /, include/asterisk/indications.h, + apps/app_readexten.c, apps/app_disa.c, + include/asterisk/channel.h: Merged revisions 168562 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r168562 | russell | 2009-01-13 13:22:13 -0600 + (Tue, 13 Jan 2009) | 10 lines Merged revisions 168561 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) + | 2 lines Revert unnecessary indications API change from rev + 122314 ........ ................ + +2009-01-13 17:52 +0000 [r168528-168549] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_logic.c: Merged revisions 168547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168547 | tilghman | 2009-01-13 11:51:12 -0600 (Tue, 13 Jan 2009) + | 13 lines Merged revisions 168546 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009) + | 6 lines If either conditional is NULL, don't try copying it. + (closes issue #14226) Reported by: caspy Patches: + 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14) + ........ ................ + + * /, channels/chan_alsa.c: Merged revisions 168526 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r168526 | tilghman | 2009-01-12 17:45:51 -0600 + (Mon, 12 Jan 2009) | 12 lines Merged revisions 167095 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 Dec 2008) + | 5 lines Repeat attempts to write when we receive -EAGAIN from + the driver, as detailed in the ALSA sample code (see + http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32) + Reported by: Jerry Geis (via the -users list) Fixed by: me + (license 14) ........ ................ + +2009-01-12 23:13 +0000 [r168524] Mark Michelson <mmichelson@digium.com> + + * main/srv.c, /: Merged revisions 168523 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168523 | + mmichelson | 2009-01-12 17:12:30 -0600 (Mon, 12 Jan 2009) | 11 + lines bump the verbosity of a message in srv.c up by one. It used + to be at this level prior to a large patch merge which converted + ast_verbose calls to ast_verb (closes issue #14221) Reported by: + jcovert Patches: srv.c.patch uploaded by jcovert (license 551) + ........ + +2009-01-12 22:00 +0000 [r168510-168519] Jeff Peeler <jpeeler@digium.com> + + * /, res/res_agi.c: Merged revisions 168517 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168517 | jpeeler | 2009-01-12 15:51:46 -0600 (Mon, 12 Jan 2009) + | 12 lines Merged revisions 168516 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) + | 5 lines (closes issue #13881) Reported by: hoowa Update the app + CDR field for AGI commands that are not executing an application + via "exec". ........ ................ + + * /, channels/chan_agent.c: Merged revisions 168508 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r168508 | jpeeler | 2009-01-12 14:53:04 -0600 + (Mon, 12 Jan 2009) | 15 lines Merged revisions 168507 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) + | 9 lines (closes issue #12269) Reported by: IgorG Tested by: + denisgalvao This gits rid of the notion of an owning_app allowing + the request and hangup to be initiated by different threads. + Originating from an active agent channel requires this. The + implementation primarily changes __login_exec to wait on a + condition variable rather than a lock. Review: + http://reviewboard.digium.com/r/35/ ........ ................ + +2009-01-12 17:26 +0000 [r168500] Olle Johansson <oej@edvina.net> + + * /, apps/app_minivm.c: Merged revisions 168497 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168497 | + oej | 2009-01-12 17:31:27 +0100 (Mån, 12 Jan 2009) | 2 lines + Better to use the proper app name ........ + +2009-01-12 15:05 +0000 [r168488] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 168486 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ ........ + +2009-01-12 14:58 +0000 [r168484] Russell Bryant <russell@digium.com> + + * /, configs/indications.conf.sample: Merged revisions 168481 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r168481 | russell | 2009-01-12 08:57:49 -0600 + (Mon, 12 Jan 2009) | 10 lines Merged revisions 168480 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) + | 2 lines s/ringdance/ringcadence/ for Bulgaria ........ + ................ + +2009-01-10 01:44 +0000 [r168336] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 168334 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168334 | + tilghman | 2009-01-09 19:42:45 -0600 (Fri, 09 Jan 2009) | 2 lines + sizeof for a stringfield is 4. Kinda low for reconstructing a + field value. ........ + +2009-01-09 23:18 +0000 [r168272] Kevin P. Fleming <kpfleming@digium.com> + + * sounds/Makefile, /: Merged revisions 168270 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168270 | kpfleming | 2009-01-09 17:16:08 -0600 (Fri, 09 Jan + 2009) | 9 lines Merged revisions 168267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan + 2009) | 1 line update to use new sound file packages that include + license files ........ ................ + +2009-01-09 23:12 +0000 [r168266] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 168192 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r168192 | rmudgett | 2009-01-09 15:43:30 -0600 + (Fri, 09 Jan 2009) | 10 lines Merged revisions 168191 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) + | 3 lines * Fix for JIRA AST-175/ABE-1757 * Miscellaneous doxygen + comments added. ........ ................ + +2009-01-09 22:23 +0000 [r168209] Russell Bryant <russell@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 168200 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r168200 | russell | 2009-01-09 16:21:05 -0600 + (Fri, 09 Jan 2009) | 10 lines Merged revisions 168198 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 Jan 2009) + | 2 lines Make this compile for mvanbaak ........ + ................ + +2009-01-09 21:57 +0000 [r168196] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 168193 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r168193 | mmichelson | 2009-01-09 15:53:26 -0600 (Fri, 09 Jan + 2009) | 21 lines Merged revisions 168128 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan + 2009) | 13 lines Add check_via calls to more request handlers + INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not + checking the topmost Via to determine where to send the response. + Adding check_via calls to those request handlers solves this. + (closes issue #13071) Reported by: baron Patches: check_via.patch + uploaded by baron (license 531) Tested by: baron ........ + ................ + +2009-01-09 20:30 +0000 [r168157] Terry Wilson <twilson@digium.com> + + * /, res/res_phoneprov.c: Merged revisions 168142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168142 | + twilson | 2009-01-09 14:25:25 -0600 (Fri, 09 Jan 2009) | 7 lines + Don't leak memory if phoneprov.conf does not exist (closes issue + #14203) Reported by: jamesgolovich Patches: + asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich + (license 176) ........ + +2009-01-09 18:42 +0000 [r168092] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 168090 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168090 | + tilghman | 2009-01-09 12:30:55 -0600 (Fri, 09 Jan 2009) | 3 lines + When using ast_str with a non-ast_str-enabled API, we need to + update the buffer or otherwise, we cannot use ast_str_strlen(). + ........ + +2009-01-09 16:41 +0000 [r168015] Matthew Nicholson <mnicholson@digium.com> + + * /, main/logger.c: Merged revisions 168014 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r168014 | + mnicholson | 2009-01-09 10:32:34 -0600 (Fri, 09 Jan 2009) | 5 + lines Use ast_safe_system() in logger.c instead of system() + (closes issue #14194) Reported by: pabelanger ........ + +2009-01-09 00:45 +0000 [r167972] Terry Wilson <twilson@digium.com> + + * apps/app_dial.c, /: Merged revisions 167935 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r167935 | + twilson | 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) | 2 lines + Set peer context and exten values so MACRO_EXTEN and + MACRO_CONTEXT will be set ........ + +2009-01-08 22:45 +0000 [r167836-167905] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 167894 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167894 | tilghman | 2009-01-08 16:37:20 -0600 (Thu, 08 Jan 2009) + | 13 lines Merged revisions 167840 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009) + | 6 lines Don't truncate database results at 255 chars. (closes + issue #14069) Reported by: evandro Patches: + 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14) + ........ ................ + + * /, apps/app_minivm.c: Merged revisions 167835 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r167835 | + tilghman | 2009-01-08 15:32:45 -0600 (Thu, 08 Jan 2009) | 6 lines + Textual changes, consistency in status variable naming, and other + minor bugs. (closes issue #13943) Reported by: Marquis Patches: + minivm_trunk_fixes3.patch uploaded by Marquis (license 32) + ........ + +2009-01-08 17:28 +0000 [r167701-167727] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c: Merged revisions 167720 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167720 | kpfleming | 2009-01-08 11:26:03 -0600 (Thu, 08 Jan + 2009) | 9 lines Merged revisions 167714 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan + 2009) | 1 line remove an unnecessary argument to queue_request() + ........ ................ + + * /, channels/chan_sip.c: Merged revisions 167700 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167700 | kpfleming | 2009-01-08 10:43:26 -0600 (Thu, 08 Jan + 2009) | 12 lines Merged revisions 167620 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan + 2009) | 5 lines When a SIP request or response arrives for a + dialog with an associated Asterisk channel, and the lock on that + channel cannot be obtained because it is held by another thread, + instead of dropping the request/response, queue it for later + processing when the channel lock becomes available. + http://reviewboard.digium.com/r/123/ ........ ................ + +2009-01-08 14:30 +0000 [r167663] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/sip-friends.sql, /: Merged revisions 167662 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r167662 | lmadsen | 2009-01-08 09:27:53 -0500 (Thu, 08 + Jan 2009) | 1 line Oops... fix the fieldname I changed yesterday + to be right. ........ + +2009-01-07 22:37 +0000 [r167544-167573] Russell Bryant <russell@digium.com> + + * /, main/file.c: Merged revisions 167569 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167569 | russell | 2009-01-07 16:36:34 -0600 (Wed, 07 Jan 2009) + | 10 lines Merged revisions 167566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) + | 2 lines Fix the last couple of places where free() was + improperly used directly. ........ ................ + + * /, main/file.c: Merged revisions 167555 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167555 | russell | 2009-01-07 16:27:23 -0600 (Wed, 07 Jan 2009) + | 10 lines Merged revisions 167554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) + | 2 lines Don't fclose() the file early, the filestream + destructor will handle it. ........ ................ + + * /, main/file.c: Merged revisions 167546 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167546 | russell | 2009-01-07 16:20:31 -0600 (Wed, 07 Jan 2009) + | 10 lines Merged revisions 167545 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) + | 2 lines Only try to close the file if one was actually opened + ........ ................ + + * /, main/file.c: Merged revisions 167542 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167542 | russell | 2009-01-07 16:05:29 -0600 (Wed, 07 Jan 2009) + | 12 lines Merged revisions 167541 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) + | 4 lines Don't use free() directly. This caused a crash since + ast_filestream is now an ao2 object. Reported by JunK-Y on IRC, + #asterisk-dev ........ ................ + +2009-01-07 18:32 +0000 [r167502] BJ Weschke <bweschke@btwtech.com> + + * apps/app_followme.c, /: Merged revisions 167478 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r167478 | + bweschke | 2009-01-07 13:20:31 -0500 (Wed, 07 Jan 2009) | 7 lines + Answer the channel if it has not already been answered and we've + already found a valid profile for followme. (closes issue #14140) + Reported by: dimas Patches: 14140.patch uploaded by dimas + ........ + +2009-01-07 18:27 +0000 [r167491] Leif Madsen <lmadsen@digium.com> + + * /, configs/queues.conf.sample: Merged revisions 167477 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r167477 | lmadsen | 2009-01-07 13:18:45 -0500 (Wed, 07 + Jan 2009) | 8 lines Update queues.conf.sample documentation. + Update the queues.conf.sample documentation to mention that you + need to preload chan_local.so as well if you plan on using Local + channels for queue members, and you're preloading pbx_config.so. + (closes issue #14179) Reported by: CrashHD Tested by: CrashHD + ........ + +2009-01-07 17:46 +0000 [r167456] Russell Bryant <russell@digium.com> + + * main/indications.c, /: Merged revisions 167442 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167442 | russell | 2009-01-07 11:35:39 -0600 (Wed, 07 Jan 2009) + | 12 lines Merged revisions 167432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009) + | 4 lines Treat an empty string the same way as a NULL country + argument. In passing, simplify the handling of returning a + default tone zone. ........ ................ + +2009-01-07 14:41 +0000 [r167376] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/sip-friends.sql, /: Merged revisions 167373 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r167373 | lmadsen | 2009-01-07 09:26:19 -0500 (Wed, 07 + Jan 2009) | 1 line Update the sip-friends.sql file to use the + non-deprecated 'defaultname' instead of 'username' and remove an + extra comma that would cause the script to fail as-is ........ + +2009-01-06 21:38 +0000 [r167306] Mark Michelson <mmichelson@digium.com> + + * main/db.c, /: Merged revisions 167301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167301 | mmichelson | 2009-01-06 15:36:44 -0600 (Tue, 06 Jan + 2009) | 16 lines Merged revisions 167299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan + 2009) | 8 lines Use the correct variable when creating the format + string (closes issue #14177) Reported by: nic_bellamy Patches: + asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic + (license 299) ........ ................ + +2009-01-06 21:10 +0000 [r167268] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 167265 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r167265 | tilghman | 2009-01-06 15:02:33 -0600 + (Tue, 06 Jan 2009) | 16 lines Merged revisions 167260 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r167260 | tilghman | 2009-01-06 14:48:05 -0600 + (Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 + Jan 2009) | 2 lines Security fix AST-2009-001. ........ + ................ ................ + +2009-01-05 17:10 +0000 [r167182] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 167180 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r167180 | mmichelson | 2009-01-05 10:59:36 -0600 (Mon, 05 Jan + 2009) | 49 lines Merged revisions 167179 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan + 2009) | 41 lines A couple of changes to T.38 SDP attribute + handling There are some boolean attributes for T.38 such as + T38FaxFillBitRemoval, T38FaxTranscodingMMR, and + T38FaxTranscodingJBIG. By simply being present, we should treat + these as a "true" value. The current code, however, was requiring + a 1 or 0 as the value of the attribute in order to parse it. This + is due to the fact that there are some T.38 endpoints and + gateways that also transmit this information incorrectly. This + patch follows the "be liberal in what you accept and strict in + what you send" philosophy by accepting both the correctly- and + incorrectly-formatted attributes, but only sending information as + it is supposed to be sent. It was also discovered that a + particular type of T.38 gateway sends some non-standard T.38 SDP + attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate, + it used T38MaxDatagram and T38FaxMaxRate respectively. We now + will properly accept these attributes as well. Note that there + are a lot of patches cited in the below commit message template. + This is because the person who submitted these patches is an + awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes + issue #13976) Reported by: linulin Patches: + chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648) + chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648) + chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648) + chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov + (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded + by arcivanov (license 648) + chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov + (license 648) Tested by: arcivanov ........ ................ + +2009-01-05 16:46 +0000 [r167178] Tilghman Lesher <tlesher@digium.com> + + * /, UPGRADE-1.6.txt: Merged revisions 167176 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r167176 | + tilghman | 2009-01-05 10:44:47 -0600 (Mon, 05 Jan 2009) | 7 lines + More clearly explain that quote marks are no longer necessary. + (closes issue #13718) Reported by: davidw Patches: + 20081020__bug13718.diff.txt uploaded by Corydon76 (license 14) + Tested by: blitzrage ........ + +2008-12-31 19:38 +0000 [r166957] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /: Merged revisions 166954 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r166954 | tilghman | 2008-12-31 13:34:28 -0600 + (Wed, 31 Dec 2008) | 12 lines Merged revisions 166953 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) + | 5 lines Also inherit the musiconhold class. (Closes #14153) + Reported by: Jerry Geis, via the users list. Patch by: me + (license 14) ........ ................ + +2008-12-30 20:57 +0000 [r166910] Terry Wilson <twilson@digium.com> + + * phoneprov/polycom_line.xml, doc/realtimetext.txt, /, + res/res_phoneprov.c, doc/sip-retransmit.txt, + doc/tex/phoneprov.tex, res/res_http_post.c: Merged revisions + 166908 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r166908 | + twilson | 2008-12-30 14:50:05 -0600 (Tue, 30 Dec 2008) | 2 lines + Fix some svn:keywords ........ + +2008-12-29 18:16 +0000 [r166863] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 166861 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, + 29 Dec 2008) | 14 lines Update app_queue to deal with the removal + of AST_PBX_KEEPALIVE When placing a call to a queue which ran a + gosub on the member's channel, Asterisk would crash every time, + stemming from the fact that the member's channel was being hung + up unexpectedly when the Gosub completed. The necessary change + was pretty much copied and pasted from app_dial's similar changes + made last week. I also took the opportunity to change a LOG_DEBUG + message in app_dial to use ast_debug. I am guessing this was due + to a direct merge from 1.4 that was not corrected to use trunk's + preferred syntax. ........ + +2008-12-29 14:52 +0000 [r166858] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Per kpfleming add a note describing why you + must never change the first element of peer_finding_info. + +2008-12-28 15:16 +0000 [r166775] Russell Bryant <russell@digium.com> + + * channels/misdn_config.c, /: Merged revisions 166773 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r166773 | russell | 2008-12-28 09:15:14 -0600 + (Sun, 28 Dec 2008) | 12 lines Merged revisions 166772 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) + | 4 lines Use strncat() instead of an sprintf() in which source + and target buffers overlap + http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html + ........ ................ + +2008-12-24 01:15 +0000 [r166730] Steve Murphy <murf@digium.com> + + * apps/app_queue.c, include/asterisk/features.h, apps/app_dial.c, + main/pbx.c, /, main/features.c, apps/app_macro.c, + include/asterisk/pbx.h: Merged revisions 166665 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk This merged from + trunk with no conflicts. I tested mostly the 'tired' cases, and + for the most part ignored the tests for reconnecting and dialing + in to fetch a parked call, after the first case. ................ + r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | + 153 lines Merged revisions 166093 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to + merge this 1.4 patch into trunk, I had to resolve some conflicts + and wait for Russell to make some changes to res_agi. I re-ran + all the tests; 39 calls in all, and made fairly careful notes and + comparisons: I don't want this to blow up some aspect of + asterisk; I completely removed the KEEPALIVE from the pbx.h + decls. The first 3 scenarios involving feature park; feature xfer + to 700; hookflash park to Park() app call all behave the same, + don't appear to leave hung channels, and no crashes. ........ + r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | + 131 lines This merges the masqpark branch into 1.4 These changes + eliminate the need for (and use of) the KEEPALIVE return code in + res_features.c; There are other places that use this result code + for similar purposes at a higher level, these appear to be left + alone in 1.4, but attacked in trunk. The reason these changes are + being made in 1.4, is that parking ends a channel's life, in some + situations, and the code in the bridge (and some other places), + was not checking the result code properly, and dereferencing the + channel pointer, which could lead to memory corruption and + crashes. Calling the masq_park function eliminates this danger in + higher levels. A series of previous commits have replaced some + parking calls with masq_park, but this patch puts them ALL to + rest, (except one, purposely left alone because a masquerade is + done anyway), and gets rid of the code that tests the KEEPALIVE + result, and the NOHANGUP_PEER result codes. While bug 13820 + inspired this work, this patch does not solve all the problems + mentioned there. I have tested this patch (again) to make sure I + have not introduced regressions. Crashes that occurred when a + parked party hung up while the parking party was listening to the + numbers of the parking stall being assigned, is eliminated. These + are the cases where parking code may be activated: 1. Feature one + touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. + Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi + hookflash xfer to 700) 4. Run Park via manager. The interesting + testing cases for parking are: I. A calls B, A parks B a. B hangs + up while A is getting the numbers announced. b. B hangs up after + A gets the announcement, but before the parking time expires c. B + waits, time expires, A is redialed, A answers, B and A are + connected, after which, B hangs up. d. C picks up B while still + in parking lot. II. A calls B, B parks A a. A hangs up while B is + getting the numbers announced. b. A hangs up after B gets the + announcement, but before the parking time expires c. A waits, + time expires, B is redialed, B answers, A and B are connected, + after which, A hangs up. d. C picks up A while still in parking + lot. Testing this throroughly involves acting all the + permutations of I and II, in situations 1,2,3, and 4. Since I + added a few more changes (ALL references to KEEPALIVE in the + bridge code eliimated (I missed one earlier), I retested most of + the above cases, and no crashes. H-extension weirdness. Current + h-extension execution is not completely correct for several of + the cases. For the case where A calls B, and A parks B, the 'h' + exten is run on A's channel as soon as the park is accomplished. + This is expected behavior. But when A calls B, and B parks A, + this will be current behavior: After B parks A, B is hung up by + the system, and the 'h' (hangup) exten gets run, but the channel + mentioned will be a derivative of A's... Thus, if A is DAHDI/1, + and B is DAHDI/2, the h-extension will be run on channel + Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be + those relating to Channel A. And, in the case where A is + reconnected to B after the park time expires, when both parties + hang up after the joyful reunion, no h-exten will be run at all. + In the case where C picks up A from the parking lot, when either + A or C hang up, the h-exten will be run for the C channel. CDR's + are a separate issue, and not addressed here. As to WHY this + strange behavior occurs, the answer lies in the procedure + followed to accomplish handing over the channel to the parking + manager thread. This procedure is called masquerading. In the + process, a duplicate copy of the channel is created, and most of + the active data is given to the new copy. The original channel + gets its name changed to XXX<ZOMBIE> and keeps the PBX + information for the sake of the original thread (preserving its + role as a call originator, if it had this role to begin with), + while the new channel is without this info and becomes a call + target (a "peer"). In this case, the parking lot manager thread + is handed the new (masqueraded) channel. It will not run an + h-exten on the channel if it hangs up while in the parking lot. + The h exten will be run on the original channel instead, in the + original thread, after the bridge completes. See bug 13820 for + our intentions as to how to clean up the h exten behavior. + Review: http://reviewboard.digium.com/r/29/ ........ + ................ + +2008-12-23 20:56 +0000 [r166698] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/app.h, /, channels/chan_sip.c, main/app.c: + Merged revisions 166696 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r166696 | + tilghman | 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) | 7 lines + Allow semicolons and extended characters in user-specified SIP + headers. (closes issue #14110) Reported by: gork Patches: + 20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14) + Tested by: gork, putnopvut ........ + +2008-12-23 15:20 +0000 [r166571] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 166569 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r166569 | mmichelson | 2008-12-23 09:17:54 -0600 (Tue, 23 Dec + 2008) | 20 lines Merged revisions 166568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec + 2008) | 12 lines Fix a crash resulting from a datastore with + inheritance but no duplicate callback The fix for this is to + simply set the newly created datastore's data pointer to NULL if + it is inherited but has no duplicate callback. (closes issue + #14113) Reported by: francesco_r Patches: 14113.patch uploaded by + putnopvut (license 60) Tested by: francesco_r ........ + ................ + +2008-12-23 04:34 +0000 [r166535] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 166533 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r166533 | tilghman | 2008-12-22 22:32:15 -0600 (Mon, 22 Dec 2008) + | 11 lines Merged revisions 166509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) + | 4 lines Use the integer form of condition for integer + comparisons. (closes issue #14127) Reported by: andrew ........ + ................ + +2008-12-22 23:27 +0000 [r166440-166472] Mark Michelson <mmichelson@digium.com> + + * /, res/res_agi.c: Merged revisions 166470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r166470 | + mmichelson | 2008-12-22 17:25:34 -0600 (Mon, 22 Dec 2008) | 11 + lines Always use the value of the AGISIGHUP when running an AGI. + Prior to this patch, the value of AGISIGUP was not always honored + when set on a channel. (closes issue #13711) Reported by: + fmueller Patches: 13711.patch uploaded by putnopvut (license 60) + ........ + + * channels/chan_dahdi.c, /: Merged revisions 166382 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r166382 | mmichelson | 2008-12-22 15:08:03 -0600 + (Mon, 22 Dec 2008) | 44 lines Merged revisions 166380 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec + 2008) | 36 lines Fix a deadlock relating to channel locks and + autoservice It has been discovered that if a channel is locked + prior to a call to ast_autoservice_stop, then it is likely that a + deadlock will occur. The reason is that the call to + ast_autoservice_stop has a check built into it to be sure that + the thread running autoservice is not currently trying to + manipulate the channel we are about to pull out of autoservice. + The autoservice thread, however, cannot advance beyond where it + currently is, though, because it is trying to acquire the lock of + the channel for which autoservice is attempting to be stopped. + The gist of all this is that a channel MUST NOT be locked when + attempting to stop autoservice on the channel. In this particular + case, the channel was locked by a call to ast_read. A call to + ast_exists_extension led to autoservice being started and stopped + due to the existence of dialplan switches. It may be that there + are future commits which handle the same symptoms but in a + different location, but based on my looks through the code, it is + very rare to see a construct such as this one. (closes issue + #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded + by putnopvut (license 60) Tested by: rtrauntvein Review: + http://reviewboard.digium.com/r/107/ ........ ................ + +2008-12-22 21:46 +0000 [r166277-166438] Russell Bryant <russell@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 166436 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r166436 | russell | 2008-12-22 15:45:28 -0600 (Mon, 22 Dec 2008) + | 2 lines Cosmetic change - don't mix struct initializer styles. + ........ + + * /, res/res_musiconhold.c: Merged revisions 166377 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r166377 | russell | 2008-12-22 14:26:48 -0600 (Mon, 22 Dec 2008) + | 2 lines Fix a bad typo. ........ + + * main/astobj2.c, /: Merged revisions 166342 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r166342 | + russell | 2008-12-22 11:44:23 -0600 (Mon, 22 Dec 2008) | 2 lines + Remove some error messages. This is the default handler that is + valid to use. ........ + + * /, main/utils.c: Merged revisions 166317 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r166317 | russell | 2008-12-22 11:29:10 -0600 (Mon, 22 Dec 2008) + | 10 lines Merged revisions 166297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) + | 2 lines Fix up timeout handling in ast_carefulwrite(). ........ + ................ + + * include/asterisk/utils.h, main/manager.c, /, main/utils.c: Merged + revisions 166282 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r166282 | + russell | 2008-12-22 11:09:36 -0600 (Mon, 22 Dec 2008) | 12 lines + Introduce ast_careful_fwrite() and use in AMI to prevent partial + writes. This patch introduces a function to do careful writes on + a file stream which will handle timeouts and partial writes. It + is currently used in AMI to address the issue that has been + reported. However, there are probably a few other places where + this could be used. (closes issue #13546) Reported by: srt Tested + by: russell http://reviewboard.digium.com/r/104/ ........ + + * /, res/res_musiconhold.c: Merged revisions 166273 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r166273 | russell | 2008-12-22 10:10:40 -0600 (Mon, 22 Dec 2008) + | 7 lines Re-work ref count handling of MoH classes using astobj2 + to resolve crashes. (closes issue #13566) Reported by: + igorcarneiro Tested by: russell Review: + http://reviewboard.digium.com/r/106/ ........ + +2008-12-22 16:17 +0000 [r166275] Mark Michelson <mmichelson@digium.com> + + * /, funcs/func_timeout.c, main/file.c: Merged revisions 166267 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r166267 | mmichelson | 2008-12-22 10:07:59 -0600 (Mon, + 22 Dec 2008) | 17 lines Fix a file playback crash and explicitly + initialize values in func_timeout.c A crash was brought up on the + bugtracker. The first run through valgrind was full of legitimate + complaints of uninitialized values in func_timeout when setting a + response timeout. These were fixed but the crash persisted. A + second run through showed the real problem. The reference + counting used for filestreams was incorrect because there were + some missing increments when a frame was read from a format + module. (closes issue #14118) Reported by: blitzrage Patches: + 14118v2.patch uploaded by putnopvut (license 60) Tested by: + blitzrage ........ + +2008-12-22 16:10 +0000 [r166272] Joshua Colp <jcolp@digium.com> + + * main/dnsmgr.c, /: Merged revisions 166268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r166268 | + file | 2008-12-22 12:08:13 -0400 (Mon, 22 Dec 2008) | 7 lines + Record the previous port in the temporary address structure so + that the comparison does not treat the host as having changed + even if it did not. This would have been uninitialized before and + would have led to a baddddd port. (closes issue #13628) Reported + by: pananix Patches: bug13628.patch uploaded by jpeeler (license + 325) Tested by: file, blitzrage ........ + +2008-12-22 14:19 +0000 [r166260] Russell Bryant <russell@digium.com> + + * /, res/res_agi.c: Merged revisions 166258 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r166258 | + russell | 2008-12-22 08:16:54 -0600 (Mon, 22 Dec 2008) | 26 lines + Remove AST_PBX_KEEPALIVE usage from res_agi. This patch removes + the usage of AST_PBX_KEEPALIVE from res_agi. The only usage was + for the AGI command, "asyncagi break". This patch removes this + feature. Normally, a feature would not be removed like this. + However, this code is broken and usage of it will result in a + memory leak. Usage of this feature will make the AGI code return + a result of AST_PBX_KEEPALIVE. The PBX handler assumes that + another thread has assumed ownership of the channel. The channel + thread will exit without destroying the channel. Unfortunately, + _no_ thread has ownership of the channel at this point. There are + a couple of serious problems here: 1) The only way to recover the + caller is to issue a channel redirect. This will work, but this + will be done with a masquerade, and the old ast_channel structure + will be lost. 2) Until the channel redirect happens, there is no + code servicing the channel. That means nothing is reading audio + or handling events coming from the channel. This is very bad. The + recommended way to get this same "break" functionality is to + issue the redirect while the channel is still being handled by + the AGI code. That way, there will be no memory leak, and there + will be no period of time that the channel is not being serviced. + ........ + +2008-12-19 23:45 +0000 [r166098-166164] Mark Michelson <mmichelson@digium.com> + + * /, main/audiohook.c: Merged revisions 166162 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r166162 | + mmichelson | 2008-12-19 17:45:00 -0600 (Fri, 19 Dec 2008) | 6 + lines Get rid of an extra space. I don't know how this crept back + in when I had already fixed it earlier ........ + + * funcs/func_audiohookinherit.c: Switch documentation formats for + func_audiohookinherit.c 1.6.1 does not have xml documentation, so + I reverted to the old way here. + + * main/channel.c, funcs/func_audiohookinherit.c (added), /, + include/asterisk/audiohook.h, main/audiohook.c: Merged revisions + 166092,166095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r166092 | + mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28 + lines Adding a new dialplan function AUDIOHOOK_INHERIT This + function is being added as a method to allow for an audiohook to + move to a new channel during a channel masquerade. The most + obvious use for such a facility is for MixMonitor when a transfer + is performed. Prior to the addition of this functionality, if a + channel running MixMonitor was transferred by another party, then + the recording would stop once the transfer had completed. By + using AUDIOHOOK_INHERIT, you can make MixMonitor continue + recording the call even after the transfer has completed. It has + also been determined that since this is seen by most as a bug fix + and is not an invasive change, this functionality will also be + backported to 1.4 and merged into the 1.6.0 branches, even though + they are feature-frozen. (closes issue #13538) Reported by: mbit + Patches: 13538.patch uploaded by putnopvut (license 60) Tested + by: putnopvut Review: http://reviewboard.digium.com/r/102/ + ........ r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri, + 19 Dec 2008) | 5 lines Remove the verbatim tag from the author + line I could have sworn I already did that before, though... + ........ + +2008-12-19 15:08 +0000 [r165892] Russell Bryant <russell@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 165890 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r165890 | russell | 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008) + | 17 lines Merged revisions 165889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) + | 9 lines Ensure that the chanspy datastore is fully initialized. + This patch resolved some random crash issues observed by a user + on a BSD system (closes issue #14111) Reported by: ys Patches: + app_chanspy.c.diff uploaded by ys (license 281) ........ + ................ + +2008-12-18 Leif Madsen <leif@digium.com> + + * Asterisk 1.6.1-beta4 released. + +2008-12-18 21:57 +0000 [r165808] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 165797 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r165797 | tilghman | 2008-12-18 15:41:02 -0600 + (Thu, 18 Dec 2008) | 15 lines Merged revisions 165767 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) + | 8 lines Add mutexes around accesses to the IMAP library + interface. This prevents certain crashes, especially when shared + mailboxes are used. (closes issue #13653) Reported by: + howardwilkinson Patches: + asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by + howardwilkinson (license 590) Tested by: jpeeler ........ + ................ + +2008-12-18 21:47 +0000 [r165804] Russell Bryant <russell@digium.com> + + * /, main/utils.c: Merged revisions 165801 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r165801 | russell | 2008-12-18 15:44:47 -0600 (Thu, 18 Dec 2008) + | 19 lines Merged revisions 165796 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) + | 11 lines Make ast_carefulwrite() be more careful. This patch + handles some additional cases that could result in partial writes + to the file description. This was done to address complaints + about partial writes on AMI. (issue #13546) (more changes needed + to address potential problems in 1.6) Reported by: srt Tested by: + russell Review: http://reviewboard.digium.com/r/99/ ........ + ................ + +2008-12-18 21:24 +0000 [r165794] Joshua Colp <jcolp@digium.com> + + * apps/app_queue.c, channels/chan_oss.c, channels/chan_dahdi.c, + channels/chan_misdn.c, /, channels/chan_sip.c, pbx/pbx_ael.c: + Merged revisions 165792 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165792 | + file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines + Numerous documentation updates. (closes issue #13970) Reported + by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt + uploaded by blitzrage (license 10) ........ + +2008-12-18 19:45 +0000 [r165728] Russell Bryant <russell@digium.com> + + * apps/app_dial.c, main/pbx.c, /, include/asterisk/pbx.h: Merged + revisions 165723 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165723 | + russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines + Remove the need for AST_PBX_KEEPALIVE with the GoSub option from + Dial. This is part of an effort to completely remove + AST_PBX_KEEPALIVE and other similar return codes from the source. + While this usage was perfectly safe, there are others that are + problematic. Since we know ahead of time that we do not want to + PBX to destroy the channel, the PBX API has been changed so that + information can be provided as an argument, instead, thus + removing the need for the KEEPALIVE return value. Further changes + to get rid of KEEPALIVE and related code is being done by murf. + There is a patch up for that on review 29. Review: + http://reviewboard.digium.com/r/98/ ........ + +2008-12-18 19:36 +0000 [r165725] Mark Michelson <mmichelson@digium.com> + + * res/res_odbc.c, /: Merged revisions 165724 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165724 | + mmichelson | 2008-12-18 13:34:33 -0600 (Thu, 18 Dec 2008) | 8 + lines Fix crashes in res_odbc. The variable "class" was being set + NULL just prior to being dereferenced in an ao2_link call. I have + moved the setting of the variable to NULL until after the + ao2_link call. ........ + +2008-12-18 18:58 +0000 [r165664] Russell Bryant <russell@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 165662 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r165662 | russell | 2008-12-18 12:54:47 -0600 + (Thu, 18 Dec 2008) | 15 lines Merged revisions 165661 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008) + | 7 lines Set the process group ID on the MOH process so that all + children will get killed (closes issue #14099) Reported by: caspy + Patches: res_musiconhold.c.patch.killpg.try2 uploaded by caspy + (license 645) ........ ................ + +2008-12-18 18:47 +0000 [r165660] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 165658 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r165658 | tilghman | 2008-12-18 12:36:48 -0600 (Thu, 18 Dec 2008) + | 2 lines Fix 2 resource leaks and fix another pipe-to-comma + conversion ........ + +2008-12-18 17:59 +0000 [r165605-165606] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merge in changes to return chan_sip to + matching based on how it was previously done and how it is done + in trunk. It will do name based for users and friends and IP + based for peers. (closes issue #14107) Reported by: jsmith + + * main/rtp.c, /: Merged revisions 165599 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) | + 11 lines Merged revisions 165591 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 + lines Only care about a compatible codec for early bridging if we + are actually bridging to another channel. If we are not we + actually want to bring the audio back to us. (closes issue + #13545) Reported by: davidw ........ ................ + +2008-12-18 16:48 +0000 [r165543] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c, /: Merged revisions 165541 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165541 | + tilghman | 2008-12-18 10:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines + Fix reference counts of the class and add an assertion to the + end. ........ + +2008-12-17 21:48 +0000 [r165332] Mark Michelson <mmichelson@digium.com> + + * res/res_odbc.c, /: Merged revisions 165330 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165330 | + mmichelson | 2008-12-17 15:46:19 -0600 (Wed, 17 Dec 2008) | 3 + lines Fix a refcount leak in res_odbc ........ + +2008-12-17 21:31 +0000 [r165329] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_macro.c: Merged revisions 165325 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165325 | + tilghman | 2008-12-17 15:28:51 -0600 (Wed, 17 Dec 2008) | 2 lines + Oops, broke trunk ........ + +2008-12-17 21:25 +0000 [r165324] Mark Michelson <mmichelson@digium.com> + + * apps/app_directory.c, apps/app_queue.c, apps/app_voicemail.c, /, + res/res_realtime.c: Merged revisions 165318 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec + 2008) | 15 lines Merged revisions 165255 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec + 2008) | 7 lines Fix some memory leaks found while looking at how + realtime configs are handled. Also cleaned up some coding + guidelines violations in app_realtime.c, mostly related to + spacing ........ ................ + +2008-12-17 21:22 +0000 [r165323] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_macro.c: Merged revisions 165319 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r165319 | tilghman | 2008-12-17 15:18:57 -0600 (Wed, 17 Dec 2008) + | 11 lines Merged revisions 165317 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) + | 4 lines Reverse the fix from issue #6176 and add proper + handling for that issue. (Closes issue #13962, closes issue + #13363) Fixed by myself (license 14) ........ ................ + +2008-12-17 21:02 +0000 [r165279] Steve Murphy <murf@digium.com> + + * /, utils/extconf.c: Merged revisions 165254 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165254 | + murf | 2008-12-17 13:50:19 -0700 (Wed, 17 Dec 2008) | 5 lines + This patch is here committed to satisfy the buildbot, who has a + problem with the const. ........ + +2008-12-17 20:02 +0000 [r165242] Terry Wilson <twilson@digium.com> + + * /, res/res_phoneprov.c: Merged revisions 165219 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165219 | + twilson | 2008-12-17 13:55:10 -0600 (Wed, 17 Dec 2008) | 2 lines + Polycom phones close the connection after reading a little bit of + the firmware files, we should stop sending in that case. Also, + make that case print out a debug statement instead of a scary + WARNING. ........ + +2008-12-17 19:54 +0000 [r165218] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 165216 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165216 | + file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines + Call proxy_update so that the IP address gets populated. Sending + stuff to 0.0.0.0 is silly! (closes issue #14055) Reported by: + chris-mac ........ + +2008-12-17 17:56 +0000 [r165146] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 165142-165143 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r165142 | mmichelson | 2008-12-17 11:52:50 -0600 (Wed, + 17 Dec 2008) | 10 lines Use the create_vm_state_from_user + function in a place where it was not being used before. Also, + I've moved the urgent folder check in messagecount() up a bit so + that the flow is a bit better. This was something I noticed while + taking a look at issue #13973, although I don't think this is the + underlying cause of the issue. ........ r165143 | mmichelson | + 2008-12-17 11:53:37 -0600 (Wed, 17 Dec 2008) | 3 lines And + actually assign the function to a pointer... ........ + +2008-12-17 05:53 +0000 [r165093] Steve Murphy <murf@digium.com> + + * utils/conf2ael.c, pbx/ael/ael-test/ref.ael-vtest13, + utils/check_expr.c, utils/Makefile, + pbx/ael/ael-test/ref.ael-vtest17, /, pbx/pbx_ael.c, + utils/ael_main.c, utils/extconf.c: Merged revisions 165071 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk I + might add here that in I tested the merged fixes from trunk in + both 1.6.0 and 1.6.1 via both 'make' and ./runtests in the ael + regression tests for all but DEBUG_CHANNEL_LOCKS, + DEBUG_SCHEDULER, and CHANNEL_TRACE options. ........ r165071 | + murf | 2008-12-16 22:04:56 -0700 (Tue, 16 Dec 2008) | 31 lines A + possibly "horrible fix" for a "horribly broken" situation. As + stuff shifts around in the asterisk code, the miscellaneous + inclusions from the standalone stuff gets broken. There's no easy + fix for this situation. I made sure that everything in utils + builds without problem ***AND*** that aelparse runs the + regressions correctly with the following make menuselect options + both on and off: DONT_OPTIMIZE DEBUG_THREADS DEBUG_CHANNEL_LOCKS + MALLOC_DEBUG MTX_PROFILE DEBUG_SCHEDULER DEBUG_THREADLOCALS + DETECT_DEADLOCKS CHANNEL_TRACE I think from now on, I'm going to + #undef all these features in the various utils native files; I + guess I could do the same for the copied-in files, surrounded by + STANDALONE ifdef. A standalone isn't going to care about threads, + mutexes, etc. ........ + +2008-12-16 23:07 +0000 [r164980] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 164978 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r164978 | mmichelson | 2008-12-16 17:06:04 -0600 (Tue, 16 Dec + 2008) | 15 lines Merged revisions 164977 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec + 2008) | 7 lines After looking through SIP registration code most + of the day, this is one of the few things I could find that was + just plain wrong. Even though it probably isn't possible for it + to happen, it seems weird to have code that checks if a pointer + is NULL and then immediately dereferences that pointer if it was + NULL. ........ ................ + +2008-12-16 22:52 +0000 [r164960] Jeff Peeler <jpeeler@digium.com> + + * /, apps/app_record.c: Merged revisions 164942 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164942 | + jpeeler | 2008-12-16 16:45:39 -0600 (Tue, 16 Dec 2008) | 6 lines + (closes issue #13669) Reported by: pj Delete file recording if + recording terminated from a hangup. ........ + +2008-12-16 21:40 +0000 [r164813-164884] Russell Bryant <russell@digium.com> + + * /, main/utils.c: Merged revisions 164882 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r164882 | russell | 2008-12-16 15:39:15 -0600 (Tue, 16 Dec 2008) + | 17 lines Merged revisions 164881 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008) + | 9 lines Fix an issue where DEBUG_THREADS may erroneously report + that a thread is exiting while holding a lock. If the last lock + attempt was a trylock, and it failed, it will still be in the + list of locks so that it can be reported. (closes issue #13219) + Reported by: pj ........ ................ + + * /, apps/app_macro.c: Merged revisions 164877 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008) + | 14 lines Merged revisions 164876 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) + | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has + been returned. This is a bug I noticed while looking at the code + for app_macro. This return code means that another thread has + assumed ownership of the channel and it can no longer be touched. + (I hate this return code with a passion, by the way.) ........ + ................ + + * main/manager.c, /: Merged revisions 164807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r164807 | russell | 2008-12-16 14:41:51 -0600 (Tue, 16 Dec 2008) + | 17 lines Merged revisions 164806 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) + | 9 lines Add "restart gracefully" to the AMI blacklist of CLI + commands. "module unload" was already identified as a command + that can not be used from the AMI. "restart gracefully" + effectively unloads all modules, and will run in to the same + problems. (closes issue #13894) Reported by: kernelsensei + ........ ................ + +2008-12-16 20:18 +0000 [r164805] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 164801 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164801 | + murf | 2008-12-16 13:04:46 -0700 (Tue, 16 Dec 2008) | 36 lines + (closes issue #14076) Reported by: toc Tested by: murf OK, Well + this issue has had its share of flip-flopping. I found the + following: 1. the code in question, in ext_cmp1 in pbx.c, would + not allow two extensions that vary only by any dashes contained + within them, to be defined in the same context. 2. for input + dialstrings, dashes are NOT ignored. So, skipping them when + sorting patterns seemed a bit silly. Thus, you might declare ext + 891 in a context, but if you try dialing 8-9-1, it will NOT match + 891. So, I proposed to remove the code from ext_cmp1 to skip the + spaces and dashes. Just kept us from declaring 891 and 8-9-1 in + the same context, forcing users to generate otherwise uselessly + obfuscated dialplan code to get the same effect. Then, I tried + out 1.4, and found that: 1. you can declare 891 and 8-9-1 in the + same context! 2. You can't define 891, and have 8-9-1 match it! + Nor can you define 8-9-1, and have 891 match it! So, it appears + that my proposal simply restores the pbx to behaving as it did in + 1.4. ........ + +2008-12-16 19:54 +0000 [r164799] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/safe_asterisk, /: Merged revisions 164798 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r164798 | tilghman | 2008-12-16 13:54:11 -0600 (Tue, 16 + Dec 2008) | 4 lines Set up umask as a possible configuration + option. (closes issue #13753) Reported by: irroot ........ + +2008-12-16 17:18 +0000 [r164677-164739] Russell Bryant <russell@digium.com> + + * include/asterisk/threadstorage.h, /, main/threadstorage.c: Merged + revisions 164737 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r164737 | russell | 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008) + | 22 lines Merged revisions 164736 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) + | 14 lines Fix memory leak and invalid reporting issues with + DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was + being used within the context of the thread local data + destructors. We would go off and allocate more thread local data + while the pthread lib was in the middle of destroying it all. + This led to a memory leak. Another issue was an invalid argument + being provided to the the object_add API call. (closes issue + #13678) Reported by: ys Tested by: Russell ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 164675 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008) + | 19 lines Merged revisions 164672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) + | 11 lines Fix a memory leak related to the use of the "setvar" + configuration option. The problem was that these variables were + being appended to the list of vars on the sip_pvt every time a + re-registration or re-subscription came in. Since it's just a + waste of memory to put them there unless the request was an + INVITE, then the fix is to check the request type before copying + the vars. (closes issue #14037) Reported by: marvinek Tested by: + russell ........ ................ + +2008-12-16 15:47 +0000 [r164662] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 164659 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164659 | + file | 2008-12-16 11:44:28 -0400 (Tue, 16 Dec 2008) | 4 lines + When using externhost make sure the port gets set to the bindaddr + port if one was not specified in the externhost value itself. + (closes issue #13634) Reported by: performer ........ + +2008-12-16 15:42 +0000 [r164658] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 164648 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r164648 | murf | 2008-12-16 08:31:54 -0700 (Tue, 16 Dec 2008) | + 13 lines Merged revisions 164634 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5 + lines I added a sentence to clarify why - and ' ' are ignored in + patterns as per bug 14076. Leif says he'll put some stuff about + it in the extensions.conf sample, etc. ........ ................ + +2008-12-16 15:02 +0000 [r164521-164625] Russell Bryant <russell@digium.com> + + * /, apps/app_minivm.c: Merged revisions 164623 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164623 | + russell | 2008-12-16 09:00:27 -0600 (Tue, 16 Dec 2008) | 5 lines + Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable + that was not needed. (closes issue #14081) Reported by: pkempgen + ........ + + * /, res/res_musiconhold.c: Merged revisions 164606 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r164606 | russell | 2008-12-16 08:31:02 -0600 + (Tue, 16 Dec 2008) | 13 lines Merged revisions 164605 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16 Dec 2008) + | 5 lines Don't try to change working directory if a directory + was not configured. (closes issue #14089) Reported by: caspy + ........ ................ + + * channels/chan_dahdi.c, /: Merged revisions 164602 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r164602 | russell | 2008-12-16 08:17:45 -0600 (Tue, 16 Dec 2008) + | 7 lines Fix usage of the DAHDI_VMWI ioctl. (closes issue + #14090) Reported by: alecdavis Patches: + chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license + 585) ........ + + * channels/chan_iax2.c, /: Merged revisions 164525 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r164525 | russell | 2008-12-15 16:25:46 -0600 (Mon, 15 Dec 2008) + | 6 lines Open a timer before loading configuration so that the + trunking configuration option will take effect. (closes issue + #14082) Reported by: seandarcy ........ + + * channels/chan_iax2.c, /: Merged revisions 164522 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r164522 | russell | 2008-12-15 16:22:43 -0600 (Mon, 15 Dec 2008) + | 4 lines Fix log message to refer to the generic timing + interface, not DAHDI specifically (inspired by issue #14082) + ........ + + * main/frame.c, /: Merged revisions 164519 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164519 | + russell | 2008-12-15 15:53:30 -0600 (Mon, 15 Dec 2008) | 7 lines + Make sure we handle a uint32_t payload in ast_frdup() (closes + issue #14080) Reported by: fnordian Patches: frame.patch uploaded + by fnordian (license 110) ........ + +2008-12-15 19:54 +0000 [r164421-164425] Mark Michelson <mmichelson@digium.com> + + * /, include/asterisk/pbx.h: Merged revisions 164423 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r164423 | mmichelson | 2008-12-15 13:53:29 -0600 + (Mon, 15 Dec 2008) | 11 lines Merged revisions 164422 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec + 2008) | 3 lines Add the deadlock note to ast_spawn_extension as + well ........ ................ + + * /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged + revisions 164419 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r164419 | mmichelson | 2008-12-15 13:51:24 -0600 (Mon, 15 Dec + 2008) | 12 lines Merged revisions 164416 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec + 2008) | 4 lines Add notes to autoservice and pbx doxygen + regarding a potential deadlock scenario so that it is avoided in + the future ........ ................ + +2008-12-15 18:27 +0000 [r164355] Tilghman Lesher <tlesher@digium.com> + + * /, cdr/cdr_pgsql.c: Merged revisions 164349 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164349 | + tilghman | 2008-12-15 12:09:58 -0600 (Mon, 15 Dec 2008) | 4 lines + When querying for the structure of the CDR table, remove the + schema, if it exists. (Closes issue #14058) ........ + +2008-12-15 18:14 +0000 [r164314-164353] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 164351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r164351 | file | 2008-12-15 14:12:24 -0400 (Mon, 15 Dec 2008) | + 13 lines Merged revisions 164350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 + lines Do not try to unlock a non-existant channel if the transfer + fails. (closes issue #13800) Reported by: dwagner Patches: + asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license + 608) ........ ................ + + * /, main/file.c: Merged revisions 164312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164312 | + file | 2008-12-15 13:24:28 -0400 (Mon, 15 Dec 2008) | 4 lines Use + ast_seekstream to return the file stream back to the beginning + instead of directly seeking to zero. This is because some audio + formats have headers at the front that need to be skipped, which + will be done by the format module. (closes issue #14079) Reported + by: elguero ........ + +2008-12-15 16:32 +0000 [r164276-164300] Russell Bryant <russell@digium.com> + + * main/channel.c, /, main/features.c: Merged revisions 164203 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r164203 | russell | 2008-12-15 08:40:24 -0600 + (Mon, 15 Dec 2008) | 39 lines Merged revisions 164201 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) + | 31 lines Handle a case where a call can be bridged to a channel + that is still ringing. The issue that was reported was about a + case where a RINGING channel got redirected to an extension to + pick up a call from parking. Once the parked call got taken out + of parking, it heard silence until the other side answered. + Ideally, the caller that was parked would get a ringing + indication. This patch fixes this case so that the caller + receives ringback once it comes out of parking until the other + side answers. The fixes are: - Make sure we remember that a + channel was an outgoing channel when doing a masquerade. This + prevents an erroneous ast_answer() call on the channel, which + causes a bogus 200 OK to be sent in the case of SIP. - Add some + additional comments to explain related parts of code. - Update + the handling of the ast_channel visible_indication field. Storing + values that are not stateful is pointless. Control frames that + are events or commands should be ignored. - When a bridge first + starts, check to see if the peer channel needs to be given + ringing indication because the calling side is still ringing. - + Rework ast_indicate_data() a bit for the sake of readability. + (closes issue #13747) Reported by: davidw Tested by: russell + Review: http://reviewboard.digium.com/r/90/ ........ + ................ + + * /, pbx/pbx_dundi.c: Merged revisions 164272 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164272 | + russell | 2008-12-15 10:17:55 -0600 (Mon, 15 Dec 2008) | 8 lines + When a reload is issued, always process the configuration for + dundi.conf. The reason is that a reload can be used to refresh + DNS lookups for defined peers. Even if the config file hasn't + changed, we want to process it for that purpose. (closes issue + #13776) Reported by: kombjuder ........ + +2008-12-15 16:18 +0000 [r164273-164274] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 164270 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164270 | + mmichelson | 2008-12-15 10:16:47 -0600 (Mon, 15 Dec 2008) | 4 + lines Fix a compile warning and a logic error that could have + been bad for non-realtime queues ........ + + * apps/app_queue.c, /: Merged revisions 164268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164268 | + mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17 + lines Fix up a few issues with regards to queues * Fix reference + counting used in the __queues_show function * Add code to be sure + that the "queue show" command does not print information for a + realtime queue which has been deleted from the backend * Add a + missing unref to the realtime queue loading function for the case + where a queue is in the module's container but has been deleted + from the realtime backend (closes issue #14033) Reported by: + cristiandimache Patches: 14033.patch uploaded by putnopvut + (license 60) Tested by: cristiandimache ........ + +2008-12-15 15:50 +0000 [r164266] Joshua Colp <jcolp@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, apps/app_fax.c, + configure.ac: Merged revisions 164257 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r164257 | + file | 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) | 4 lines + Make app_fax compatible with newer versions of spandsp. This + remains backwards compatible with earlier versions though so do + not fret. (closes issue #14073) Reported by: seandarcy ........ + +2008-12-13 01:01 +0000 [r163914] Joshua Colp <jcolp@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 163912 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r163912 | + file | 2008-12-12 20:59:24 -0400 (Fri, 12 Dec 2008) | 2 lines + Only detach and destroy the whisper audiohooks if they are + actually in use. ........ + +2008-12-13 00:08 +0000 [r163875] Terry Wilson <twilson@digium.com> + + * apps/app_queue.c, /: Merged revisions 163873 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r163873 | + twilson | 2008-12-12 17:48:26 -0600 (Fri, 12 Dec 2008) | 6 lines + When using realtime queues, app_queue wasn't updating the + strategy if it was changed in the realtime backend. This patch + resolves the issue for almost all situations. It is currently not + supported to switch to the linear strategy via realtime since the + ao2_container for members will have been set to have multiple + buckets and therefore the members would be unordered. (closes + issue #14034) Reported by: cristiandimache Tested by: + otherwiseguy, cristiandimache ........ + +2008-12-12 23:08 +0000 [r163830] Russell Bryant <russell@digium.com> + + * /: Merged revisions 163829 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ ........ + +2008-12-12 22:05 +0000 [r163764] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, main/editline/read.c, /: Merged revisions 163762 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r163762 | tilghman | 2008-12-12 16:04:26 -0600 + (Fri, 12 Dec 2008) | 14 lines Merged revisions 163761 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) + | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk, + but also add a pointer inside editline to look back to + asterisk.c, so others don't spend as much time as I did looking + (in the wrong place) for the appropriate function. Reported by: + ZX81, via the #asterisk-users channel Fixed by: me (license 14) + ........ ................ + +2008-12-12 19:58 +0000 [r163715] Steve Murphy <murf@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 163675 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r163675 | murf | 2008-12-12 12:16:32 -0700 (Fri, 12 Dec 2008) | 1 + line demote always-appearing debug message (for certain boards) + to ast_debug lev 3 msg instead ........ + +2008-12-12 18:53 +0000 [r163656-163672] Russell Bryant <russell@digium.com> + + * main/tcptls.c, /, channels/chan_sip.c: Merged revisions 163670 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12 + Dec 2008) | 6 lines Rename a number of tcptls_session variables. + There are no functional changes here. The name "ser" was used in + a lot of places. However, it is a relic from when the struct was + a server_instance, not a session_instance. It was renamed since + it represents both a server or client connection. ........ + + * /, channels/chan_sip.c: Merged revisions 163667 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r163667 | + russell | 2008-12-12 12:33:27 -0600 (Fri, 12 Dec 2008) | 5 lines + Fix a small race condition in sip_tcp_locate(). We must increase + the reference count on the tcptls_session _before_ unlocking the + thread list. ........ + + * /, channels/chan_sip.c: Merged revisions 163642 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r163642 | + russell | 2008-12-12 12:19:47 -0600 (Fri, 12 Dec 2008) | 7 lines + Resolve crashes when using SIP TCP/TLS with qualify. The problem + was a reference count error on the tcptls_session structure. + (closes issue #13989) Reported by: Nugget ........ + +2008-12-12 18:19 +0000 [r163640] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 163629 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r163629 | + file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines + When a device registers we need to unlink them (if linked) from + the peers_by_ip container and link them back in since their IP + address has changed. This would have manifested itself if you + configured a new device (as type=peer), registered, and then + tried to place a call from the device. Since the peer was not + linked into the peers_by_ip container it would have never been + found. (closes issue #13811) Reported by: pj ........ + +2008-12-12 17:26 +0000 [r163624] Michiel van Baak <michiel@vanbaak.info> + + * res/res_monitor.c, /: Merged revisions 163612 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r163612 | + mvanbaak | 2008-12-12 18:22:47 +0100 (Fri, 12 Dec 2008) | 7 lines + Document default Monitor file location. (closes issue #14065) + Reported by: kshumard Patches: + res_monitor.documentation.patch.txt uploaded by kshumard (license + 92) ........ + +2008-12-12 16:57 +0000 [r163581] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /, channels/chan_sip.c: Merged revisions 163579 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec + 2008) | 4 lines Since chan_sip is callback devicestate driven do + not pass in actual states, pass in unknown so we get asked. + Additionally do not pass in an actual device state value in + ast_setstate since the channel may be callback driven. (closes + issue #13525) Reported by: pj ........ + +2008-12-12 14:48 +0000 [r163514-163515] Russell Bryant <russell@digium.com> + + * main/channel.c, main/autoservice.c, /, + include/asterisk/channel.h: Merged revisions 163449 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r163449 | russell | 2008-12-12 07:55:30 -0600 + (Fri, 12 Dec 2008) | 34 lines Merged revisions 163448 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) + | 26 lines Resolve issues that could cause DTMF to be processed + out of order. These changes come from team/russell/issue_12658 1) + Change autoservice to put digits on the head of the channel's + frame readq instead of the tail. If there were frames on the + readq that autoservice had not yet read, the previous code would + have resulted in out of order processing. This required a new API + call to queue a frame to the head of the queue instead of the + tail. 2) Change up the processing of DTMF in ast_read(). Some of + the problems were the result of having two sources of pending + DTMF frames. There was the dtmfq and the more generic readq. Both + were used for pending DTMF in various scenarios. Simplifying + things to only use the frame readq avoids some of the problems. + 3) Fix a bug where a DTMF END frame could get passed through when + it shouldn't have. If code set END_DTMF_ONLY in the middle of + digit emulation, and a digit arrived before emulation was + complete, digits would get processed out of order. (closes issue + #12658) Reported by: dimas Tested by: russell, file Review: + http://reviewboard.digium.com/r/85/ ........ ................ + + * /, pbx/pbx_dundi.c: Merged revisions 163512 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r163512 | russell | 2008-12-12 08:44:06 -0600 (Fri, 12 Dec 2008) + | 13 lines Merged revisions 163511 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008) + | 5 lines Specify uint32_t for variables storing a CRC32 so that + it is actually 32 bits on 64-bit machines, as well. (inspired by + issue #13879) ........ ................ + +2008-12-11 23:48 +0000 [r163386] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 163384 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r163384 | tilghman | 2008-12-11 17:38:56 -0600 (Thu, 11 Dec 2008) + | 16 lines Merged revisions 163383 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) + | 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on + certain shells, the terminal is messed up. By intercepting those + events with a signal handler in the remote console, we can avoid + those issues. (closes issue #13464) Reported by: tzafrir Patches: + 20081110__bug13464.diff.txt uploaded by Corydon76 (license 14) + Tested by: blitzrage ........ ................ + +2008-12-11 22:52 +0000 [r163319] Matt Nicholson <mnicholson@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 163317 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r163317 | mnicholson | 2008-12-11 16:49:59 -0600 (Thu, 11 Dec + 2008) | 16 lines Merged revisions 163316 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec + 2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes + issue #13819) Reported by: adomjan Patches: + pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487) + dundi_clearecache3.diff uploaded by mnicholson (license 96) + Tested by: adomjan ........ ................ + +2008-12-11 21:50 +0000 [r163252-163256] Russell Bryant <russell@digium.com> + + * /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions + 163254 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r163254 | russell | 2008-12-11 15:48:08 -0600 (Thu, 11 Dec 2008) + | 16 lines Merged revisions 163253 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008) + | 8 lines Fix some observed slowdowns in dialplan processing. The + change is to remove autoservice usage from dialplan functions + that do not need it because they do not perform operations that + potentially block. (closes issue #13940) Reported by: tbelder + ........ ................ + + * /, res/res_timing_pthread.c: Merged revisions 163241 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r163241 | russell | 2008-12-11 15:21:31 -0600 (Thu, 11 Dec 2008) + | 8 lines Fix a problem where continuous mode will get + inadvertently get turned off if set_rate() is used while + continuous mode was already turned on. (closes issue #13738) + Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix + (license 547) ........ + +2008-12-11 21:00 +0000 [r163214] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged + revisions 163213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r163213 | + mmichelson | 2008-12-11 14:57:44 -0600 (Thu, 11 Dec 2008) | 9 + lines Add an option to voicemail.conf to allow urgent messages to + be forwarded as not urgent. (closes issue #14063) Reported by: + jaroth Patches: urgfwd_v2.patch uploaded by jaroth (license 50) + ........ + +2008-12-11 20:10 +0000 [r163173] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 163171 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r163171 | + russell | 2008-12-11 14:07:47 -0600 (Thu, 11 Dec 2008) | 16 lines + Fix the "failed" extension for outgoing calls. The conversion to + use ast_check_hangup() everywhere instead of checking the + softhangup flag directly introduced this problem. The issue is + that ast_check_hangup() checked for tech_pvt to be NULL. + Unfortunately, this will be NULL is some valid circumstances, + such as with a dummy channel. The fix is simple. Don't check + tech_pvt. It's pointless, because the code path that sets this to + NULL is when the channel hangup callback gets called. This + happens inside of ast_hangup(), which is the same function + responsible for freeing the channel. Any code calling + ast_check_hangup() better not be calling it after that point, and + if so, we have a bigger problem at hand. (closes issue #14035) + Reported by: erogoza ........ + +2008-12-11 20:05 +0000 [r163170] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 163168 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r163168 | tilghman | 2008-12-11 14:02:35 -0600 (Thu, 11 Dec 2008) + | 5 lines Sometimes even Linux needs -lm to link libtonezone, + such as when libtonezone is compiled statically. (closes issue + #13887) Reported by: tzafrir ........ + +2008-12-11 17:16 +0000 [r163100] Russell Bryant <russell@digium.com> + + * /, main/features.c: Merged revisions 163094 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r163094 | russell | 2008-12-11 11:06:16 -0600 (Thu, 11 Dec 2008) + | 19 lines Merged revisions 163092 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) + | 11 lines Fix an issue that made it so you could only have a + single caller executing a custom feature at a time. This was + especially problematic when custom features ran for any + appreciable amount of time. The fix turned out to be quite + simple. The dynamic features are now stored in a read/write list + instead of a list using a mutex. (closes issue #13478) Reported + by: neutrino88 Fix suggested by file ........ ................ + +2008-12-11 16:54 +0000 [r163091] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 163089 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r163089 | tilghman | 2008-12-11 10:52:24 -0600 (Thu, 11 Dec 2008) + | 13 lines Merged revisions 163088 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008) + | 6 lines Don't wait forever, if there's a specified recording + timeout. (closes issue #13885) Reported by: bamby Patches: + res_agi.c.patch uploaded by bamby (license 430) ........ + ................ + +2008-12-11 16:49 +0000 [r163083-163087] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 163085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r163085 | mmichelson | 2008-12-11 10:47:34 -0600 (Thu, 11 Dec + 2008) | 12 lines Merged revisions 163084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec + 2008) | 4 lines Revert this cast to long. Using time_t here + causes build failures on a FreeBSD 32-bit build. ........ + ................ + + * apps/app_queue.c, /: Merged revisions 163081 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r163081 | mmichelson | 2008-12-11 10:33:16 -0600 (Thu, 11 Dec + 2008) | 22 lines Merged revisions 163080 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec + 2008) | 14 lines Fix a potential crash due to unsafe datastore + handling. This patch also contains a conversion from using long + to time_t for representing times for a queue, as well as some + whitespace fixes. (closes issue #14060) Reported by: nivek + Patches: datastore_fixup.patch.corrected uploaded by nivek + (license 636) with slight modification from me Tested by: nivek + ........ ................ + +2008-12-11 15:07 +0000 [r163006] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 162997 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r162997 | + file | 2008-12-11 11:05:49 -0400 (Thu, 11 Dec 2008) | 4 lines + When a device registers to use it is entirely possible that they + may be in use, so tell the core that we don't know the devstate + and have it ask us for it. (closes issue #13525) Reported by: pj + ........ + +2008-12-10 23:13 +0000 [r162949] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 162922,162930 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r162922 | + tilghman | 2008-12-10 16:48:09 -0600 (Wed, 10 Dec 2008) | 7 lines + Checking global variables here actually overwrote the previous + substitution by channel variables, and in any case, was + redundant; pbx_substitute_variables_helper ALREADY does + substitution for global variables. (closes issue #13327) Reported + by: pj ........ r162930 | tilghman | 2008-12-10 17:01:14 -0600 + (Wed, 10 Dec 2008) | 2 lines Previously missing line, now the + substitution works correctly ........ + +2008-12-10 22:54 +0000 [r162896-162929] Jeff Peeler <jpeeler@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 162927 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r162927 | jpeeler | 2008-12-10 16:53:34 -0600 + (Wed, 10 Dec 2008) | 11 lines Merged revisions 162926 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10 Dec 2008) + | 3 lines Oops, inverted logic for a strcasecmp check. Pointed + out by mmichelson, thanks! ........ ................ + + * /, res/res_musiconhold.c: Merged revisions 162891 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r162891 | jpeeler | 2008-12-10 16:11:46 -0600 + (Wed, 10 Dec 2008) | 13 lines Merged revisions 162874 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 Dec 2008) + | 5 lines (closes issue #13229) Reported by: + clegall_proformatique Ensure that moh_generate does not return + prematurely before local_ast_moh_stop is called. Also, the sleep + in mp3_spawn now only occurs for http locations since it seems to + have been added originally only for failing media streams. + ........ ................ + +2008-12-10 19:05 +0000 [r162741-162807] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 162805 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162805 | file | 2008-12-10 15:02:57 -0400 (Wed, 10 Dec 2008) | + 13 lines Merged revisions 162804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 + lines Fix subscription based MWI up a bit. We only want to put + sip: at the beginning of the URI if it is not already there and + revert code to ignore destination check if subscribing for MWI. + (closes issue #12560) Reported by: vsauer Patches: patch001.diff + uploaded by ramonpeek (license 266) ........ ................ + + * /, channels/chan_sip.c: Merged revisions 162739 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162739 | file | 2008-12-10 13:53:09 -0400 (Wed, 10 Dec 2008) | + 13 lines Merged revisions 162738 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 + lines When a SIP peer unregisters set the expiry time back to 0 + so that the 200 OK contains an expires of 0. (closes issue + #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded + by hjourdain (license 583) ........ ................ + +2008-12-10 16:39 +0000 [r162666-162669] Mark Michelson <mmichelson@digium.com> + + * doc/tex/misdn.tex, /: Merged revisions 162667 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162667 | mmichelson | 2008-12-10 10:39:10 -0600 (Wed, 10 Dec + 2008) | 16 lines Merged revisions 162659 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec + 2008) | 8 lines Add missing documentation to misdn.txt (closes + issue #14052) Reported by: festr Patches: misdn.txt.patch + uploaded by festr (license 443) ........ ................ + + * /, channels/chan_sip.c: Merged revisions 162664 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162664 | mmichelson | 2008-12-10 10:34:35 -0600 (Wed, 10 Dec + 2008) | 19 lines Merged revisions 162663 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec + 2008) | 11 lines Revert fix for issue 13570. It has caused more + problems than it helped to fix. (closes issue #13783) Reported + by: navkumar (closes issue #14025) Reported by: ffs ........ + ................ + +2008-12-10 16:08 +0000 [r162622-162658] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, /: Merged revisions 162656 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162656 | file | 2008-12-10 12:06:59 -0400 (Wed, 10 Dec 2008) | + 13 lines Merged revisions 162653 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 + lines Increment the sequence number on the end packets for + RFC2833. After reading the RFC some more and doing some testing I + agree with this change. (closes issue #12983) Reported by: vt + Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license + 520) ........ ................ + + * /, channels/chan_sip.c: Merged revisions 162619 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r162619 | + file | 2008-12-10 11:22:26 -0400 (Wed, 10 Dec 2008) | 4 lines + When transmitting a register set the socket port to the local one + for the transport being used, not the port for the remote server. + (closes issue #13633) Reported by: performer ........ + +2008-12-10 11:37 +0000 [r162585] Michiel van Baak <michiel@vanbaak.info> + + * /, res/snmp/agent.c: Merged revisions 162583 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r162583 | + mvanbaak | 2008-12-10 12:34:09 +0100 (Wed, 10 Dec 2008) | 5 lines + Make res_snmp.so compile on OpenBSD. OpenBSD uses an old version + of gcc which throws an error if you use a macro that's not + #defined ........ + +2008-12-09 23:45 +0000 [r162490] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/stringfields.h, /: Merged revisions 162488 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r162488 | kpfleming | 2008-12-09 17:41:02 -0600 (Tue, 09 + Dec 2008) | 1 line it does help if the compiler attribute syntax + is correct ........ + +2008-12-09 23:12 +0000 [r162472] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 162466 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r162466 | tilghman | 2008-12-09 17:10:34 -0600 + (Tue, 09 Dec 2008) | 9 lines Merged revisions 162463 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 + Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........ + ................ + +2008-12-09 22:34 +0000 [r162416] Russell Bryant <russell@digium.com> + + * main/asterisk.c, include/asterisk/utils.h, /, main/utils.c: + Merged revisions 162414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162414 | russell | 2008-12-09 16:25:06 -0600 (Tue, 09 Dec 2008) + | 16 lines Merged revisions 162413 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) + | 8 lines Remove the test_for_thread_safety() function + completely. The test is not valid. Besides, if we actually + suspected that recursive mutexes were not working, we would get a + ton of LOG_ERROR messages when DEBUG_THREADS is turned on. + (inspired by a discussion on the asterisk-dev list) ........ + ................ + +2008-12-09 22:02 +0000 [r162372] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 162355 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r162355 | tilghman | 2008-12-09 15:57:09 -0600 + (Tue, 09 Dec 2008) | 11 lines Merged revisions 162348 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) + | 4 lines We appear to have documented tz= in the [general] + section of voicemail.conf, without actually having implemented + it. Oops. (Reported by Olivier on the -users list) ........ + ................ + +2008-12-09 21:18 +0000 [r162344] Joshua Colp <jcolp@digium.com> + + * /, apps/app_directed_pickup.c: Merged revisions 162342 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r162342 | file | 2008-12-09 17:16:37 -0400 (Tue, + 09 Dec 2008) | 11 lines Merged revisions 162341 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 + lines Add 'down' as a valid state for directed call pickup. This + creeps up when we receive session progress when dialing a device + and not ringing. (closes issue #14005) Reported by: ddl ........ + ................ + +2008-12-09 21:03 +0000 [r162302] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 162291 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162291 | russell | 2008-12-09 14:59:54 -0600 (Tue, 09 Dec 2008) + | 17 lines Merged revisions 162286 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) + | 9 lines Fix an issue where callers on an incoming call on an + SLA trunk would not hear ringback. We need to make sure that we + don't start writing audio to the trunk channel until we're + actually ready to answer it. Otherwise, the channel driver will + treat it as inband progress, even though all they are getting is + silence. (closes issue #12471) Reported by: mthomasslo ........ + ................ + +2008-12-09 20:48 +0000 [r162278] Joshua Colp <jcolp@digium.com> + + * /, apps/app_festival.c: Merged revisions 162275 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162275 | file | 2008-12-09 16:46:11 -0400 (Tue, 09 Dec 2008) | + 11 lines Merged revisions 162273 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 + lines Fix double declaration of 'x' on the PPC platform. (closes + issue #14038) Reported by: ffloimair ........ ................ + +2008-12-09 20:47 +0000 [r162277] Steve Murphy <murf@digium.com> + + * res/ael/ael.flex, /, res/ael/ael_lex.c: Merged revisions 162271 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r162271 | murf | 2008-12-09 13:40:31 -0700 (Tue, + 09 Dec 2008) | 9 lines Merged revisions 162264 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 + line In discussion with seanbright on #asterisk-dev, I have added + a default rule, and an option to suppress the default rule from + being generated in the flex output, for the sake of those OS's + where they didn't tweak flex's ECHO macro, and the compiler + doesn't like it. The regressions are OK with this. ........ + ................ + +2008-12-09 20:31 +0000 [r162269] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c, /: Merged revisions 162266 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162266 | mmichelson | 2008-12-09 14:30:07 -0600 (Tue, 09 Dec + 2008) | 14 lines Merged revisions 162265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec + 2008) | 6 lines If we fail to start a thread for the pbx to run + in, we need to be sure to decrease the number of active calls on + the system. This fix may relate to ABE-1713, but it is not + certain yet. ........ ................ + +2008-12-09 19:52 +0000 [r162202-162207] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, /: Merged revisions 162205 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162205 | file | 2008-12-09 15:48:35 -0400 (Tue, 09 Dec 2008) | + 14 lines Merged revisions 162204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 + lines Make sure that the timestamp for DTMF is not the same as + the previous voice frame and do not send audio when transmitting + DTMF as this confuses some equipment. (closes issue #13209) + Reported by: ip-rob Patches: 13209.diff uploaded by file (license + 11) Tested by: ip-rob, bujones ........ ................ + + * main/rtp.c, /: Merged revisions 162197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162197 | file | 2008-12-09 15:08:39 -0400 (Tue, 09 Dec 2008) | + 11 lines Merged revisions 162188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 + lines Take video into account when early bridging RTP. (closes + issue #13535) Reported by: davidw ........ ................ + +2008-12-09 18:49 +0000 [r162082-162142] Steve Murphy <murf@digium.com> + + * res/ael/ael.flex, /, res/ael/ael_lex.c: Merged revisions 162140 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r162140 | murf | 2008-12-09 11:35:35 -0700 (Tue, + 09 Dec 2008) | 9 lines Merged revisions 162136 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1 + line Previous fix used ast_malloc and ast_copy_string and messed + up the standalone stuff. Fixed. ........ ................ + + * res/ael/ael.flex, res/ael/pval.c, /, include/asterisk/pval.h, + res/ael/ael_lex.c: Merged revisions 162079 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162079 | murf | 2008-12-09 10:18:03 -0700 (Tue, 09 Dec 2008) | + 53 lines Merged revisions 162013 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | + 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches: + 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme, + murf This crash was the result of a few small errors that would + combine in 64-bit land to result in a crash. 32-bit land might + have seen these combine to mysteriously drop the args to an + application call, in certain circumstances. Also, in trying to + find this bug, I spotted a situation in the flex input, where, in + passing back a 'word' to the parser, it would allocate a buffer + larger than necessary. I changed the usage in such situations, so + that strdup was not used, but rather, an ast_malloc, followed by + ast_copy_string. I removed a field from the pval struct, in u2, + that was never getting used, and set in one spot in the code. I + believe it was an artifact of a previous fix to make switch cases + work invisibly with extens. And, for goto's I removed a '!' from + before a strcmp, that has been there since the initial merging of + AEL2, that might prevent the proper target of a goto from being + found. This was pretty harmless on its own, as it would just + louse up a consistency check for users. Many thanks to + ckjohnsonme for providing a simplified and complete set of + information about the bug, that helped considerably in finding + and fixing the problem. Now, to get aelparse up and running again + in trunk, and out of its "horribly broken" state, so I can run + the regression suite! ........ ................ + +2008-12-09 16:50 +0000 [r161963-162018] Russell Bryant <russell@digium.com> + + * /, apps/app_disa.c: Merged revisions 162016 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r162016 | russell | 2008-12-09 10:47:39 -0600 (Tue, 09 Dec 2008) + | 13 lines Merged revisions 162014 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) + | 5 lines Allow DISA to handle extensions that start with #. + (closes issue #13330) Reported by: jcovert ........ + ................ + + * /, main/app.c: Merged revisions 161951 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r161951 | russell | 2008-12-09 08:57:39 -0600 (Tue, 09 Dec 2008) + | 23 lines Merged revisions 161948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) + | 15 lines Fix a problem with GROUP() settings on a masquerade. + The previous code carried over group settings from the old + channel to the new one. However, it did nothing with the group + settings that were already on the new channel. This patch removes + all group settings that already existed on the new channel. I + have a more complicated version of this patch which addresses + only the most blatant problem with this, which is that a channel + can end up with multiple group settings in the same category. + However, I could not think of a use case for keeping any of the + group settings from the old channel, so I went this route for + now. (closes AST-152) ........ ................ + +2008-12-08 20:55 +0000 [r161835] Joshua Colp <jcolp@digium.com> + + * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8: + Merged revisions 161830 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r161830 | + file | 2008-12-08 16:53:50 -0400 (Mon, 08 Dec 2008) | 2 lines + Update autosupport script with a few changes. ........ + +2008-12-08 18:52 +0000 [r161792] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /: Merged revisions 161790 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r161790 | + tilghman | 2008-12-08 12:49:50 -0600 (Mon, 08 Dec 2008) | 6 lines + Allocate enough space initially for the message. (closes issue + #14027) Reported by: junky Patches: M14027.diff uploaded by junky + (license 177) ........ + +2008-12-08 18:49 +0000 [r161729-161789] Joshua Colp <jcolp@digium.com> + + * main/pbx.c, /: Merged revisions 161787 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r161787 | + file | 2008-12-08 14:47:32 -0400 (Mon, 08 Dec 2008) | 6 lines Fix + a regression introduced when the PBX timeouts were converted to + milliseconds. collect_digits now gets milliseconds fed to it, not + seconds. (closes issue #14012) Reported by: dveiga Patches: + 14012.patch uploaded by bkruse (license 132) ........ + + * /, channels/chan_sip.c: Merged revisions 161726 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r161726 | file | 2008-12-08 13:53:32 -0400 (Mon, 08 Dec 2008) | + 13 lines Merged revisions 161725 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 + lines Make the usereqphone option work again. (closes issue + #13474) Reported by: mmaguire Patches: 20080912_bug13474.diff + uploaded by mmaguire (license 571) ........ ................ + +2008-12-08 17:24 +0000 [r161722] Matt Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 161721 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r161721 | + mnicholson | 2008-12-08 11:23:41 -0600 (Mon, 08 Dec 2008) | 7 + lines Fix a crash that can occur on a transfer in chan_sip when + attempting to collect rtp stats. (closes issue #13956) Reported + by: chris-mac Tested by: chris-mac ........ + +2008-12-05 23:29 +0000 [r161496] Mark Michelson <mmichelson@digium.com> + + * apps/app_stack.c, /: Merged revisions 161493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r161493 | + mmichelson | 2008-12-05 17:24:38 -0600 (Fri, 05 Dec 2008) | 8 + lines If the autoloop flag is set on a channel, then we need to + add 1 to the priority when checking if the extension exists. + Otherwise, gosubs will fail. This was discovered when + investigating an asterisk-users mailing list post made by Gary + Hawkins. ........ + +2008-12-05 21:16 +0000 [r161352-161429] Sean Bright <sean.bright@gmail.com> + + * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions + 161427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r161427 | seanbright | 2008-12-05 16:08:43 -0500 (Fri, 05 Dec + 2008) | 22 lines Merged revisions 161426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r161426 | seanbright | 2008-12-05 16:02:20 -0500 + (Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec + 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned + int). (closes issue #14006) Reported by: alphaque Patches: + astobj2.h-patch uploaded by alphaque (license 259) (Slightly + modified by seanbright) ........ ................ + ................ + + * apps/app_voicemail.c, /: Merged revisions 161349-161350 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r161349 | seanbright | 2008-12-05 10:56:15 -0500 (Fri, + 05 Dec 2008) | 5 lines When using IMAP_STORAGE, it's important to + convert bare newlines (\n) in emailbody and pagerbody to CR-LF so + that the IMAP server doesn't spit out an error. This was + informally reported on #asterisk-dev a few weeks ago. Reviewed by + Mark M. on IRC. ........ r161350 | seanbright | 2008-12-05 + 11:04:36 -0500 (Fri, 05 Dec 2008) | 2 lines Use ast_free() + instead of free(), pointed out by eliel on IRC. ........ + +2008-12-05 14:18 +0000 [r161285-161290] Russell Bryant <russell@digium.com> + + * main/pbx.c, /: Merged revisions 161288 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r161288 | russell | 2008-12-05 08:16:24 -0600 (Fri, 05 Dec 2008) + | 10 lines Merged revisions 161287 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008) + | 2 lines Fix a NULL format string warning found by buildbot. + ........ ................ + + * /, apps/app_minivm.c: Merged revisions 161252 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r161252 | + russell | 2008-12-05 07:46:01 -0600 (Fri, 05 Dec 2008) | 2 lines + Resolve a compiler warning from buildbot about a NULL format + string. ........ + +2008-12-05 05:42 +0000 [r161182] Tilghman Lesher <tlesher@digium.com> + + * main/config.c, /: Merged revisions 161181 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r161181 | + tilghman | 2008-12-04 23:41:41 -0600 (Thu, 04 Dec 2008) | 11 + lines The first file should have a blank config filename in the + structure, so that when a save occurs to a different filename, + everything goes to the alternate filename, instead of appending + to the original. This is important for the AMI command + UpdateConfig. (closes issue #13301) Reported by: trevo Patches: + 20081113__bug13301.diff.txt uploaded by Corydon76 (license 14) + 20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license + 14) Tested by: Corydon76, blitzrage ........ + +2008-12-05 02:52 +0000 [r161149] Sean Bright <sean.bright@gmail.com> + + * apps/app_voicemail.c, /: Merged revisions 161147 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r161147 | seanbright | 2008-12-04 21:47:54 -0500 (Thu, 04 Dec + 2008) | 3 lines Check the return value of fread/fwrite so the + compiler doesn't complain. Only a problem when IMAP_STORAGE is + enabled. ........ + +2008-12-04 18:37 +0000 [r161016] Jeff Peeler <jpeeler@digium.com> + + * main/rtp.c, /: Merged revisions 161014 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r161014 | jpeeler | 2008-12-04 12:32:20 -0600 (Thu, 04 Dec 2008) + | 17 lines Merged revisions 161013 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) + | 9 lines (closes issue #13835) Reported by: matt_b Tested by: + jpeeler This mirrors a check that was present in ast_rtp_read to + also be in ast_rtp_raw_write to not schedule sending the receiver + report if the remote RTCP endpoint address isn't present in the + RTCP structure. Closes AST-142. ........ ................ + +2008-12-04 16:48 +0000 [r160947] Mark Michelson <mmichelson@digium.com> + + * /, main/callerid.c: Merged revisions 160945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r160945 | mmichelson | 2008-12-04 10:45:06 -0600 (Thu, 04 Dec + 2008) | 23 lines Merged revisions 160943 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec + 2008) | 15 lines Fix a callerid parsing issue. If someone + formatted callerid like the following: "name <number>" (including + the quotation marks), then the parts would be parsed as name: + "name number: number This is because the closing quotation mark + was not discovered since the number and everything after was + parsed out of the string earlier. Now, there is a check to see if + the closing quote occurs after the number, so that we can know if + we should strip off the opening quote on the name. Closes AST-158 + ........ ................ + +2008-12-04 01:41 +0000 [r160858-160859] Richard Mudgett <rmudgett@digium.com> + + * funcs/func_callerid.c, /: Merged revisions 160856 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r160856 | rmudgett | 2008-12-03 19:36:39 -0600 (Wed, 03 Dec 2008) + | 1 line Jcolp pointed out that num will also match number + ........ + + * funcs/func_callerid.c, /: Merged revisions 160854 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r160854 | rmudgett | 2008-12-03 19:14:22 -0600 (Wed, 03 Dec 2008) + | 4 lines * Found a couple more places where num/number needed to + be done so 1.4 upgraders will not have problems. * Added curly + braces and minor tweaks. ........ + +2008-12-03 22:02 +0000 [r160811] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 160791 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r160791 | tilghman | 2008-12-03 15:58:21 -0600 + (Wed, 03 Dec 2008) | 9 lines Merged revisions 160770 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 + Dec 2008) | 2 lines Some compilers warn on null format strings; + some don't (caught by buildbot) ........ ................ + +2008-12-03 21:40 +0000 [r160766] Steve Murphy <murf@digium.com> + + * funcs/func_callerid.c, /: Merged revisions 160760 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r160760 | murf | 2008-12-03 14:09:15 -0700 (Wed, + 03 Dec 2008) | 23 lines Merged revisions 160703 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | + 11 lines (closes issue #13597) Reported by: john8675309 Patches: + patch.13597 uploaded by murf (license 17) Tested by: murf, + john8675309 This patch causes the setcid func to update the CDR + clid after setting the channel field. I also notice that in + trunk, the num/number of 1.4 is left out; I decided to include + the option to use either in trunk, so as not to have 1.4 + upgraders not to have problems. ........ ................ + +2008-12-03 20:36 +0000 [r160702] Jason Parker <jparker@digium.com> + + * main/manager.c, /: Merged revisions 160699-160700 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r160699 | qwell | 2008-12-03 14:32:20 -0600 (Wed, 03 Dec 2008) | + 7 lines Fix typo when ListCategories returns none. (closes issue + #13994) Reported by: mika Patches: ListCategoriesActionPatch.diff + uploaded by mika (license 624) ........ r160700 | qwell | + 2008-12-03 14:35:36 -0600 (Wed, 03 Dec 2008) | 1 line Another + place this is missing ........ + +2008-12-03 19:49 +0000 [r160665] Eliel C. Sardanons <eliels@gmail.com> + + * /, channels/iax2-provision.c: Merged revisions 160663 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r160663 | eliel | 2008-12-03 17:25:30 -0200 (Wed, 03 Dec + 2008) | 13 lines - iax2-provision was not freeing iax_templates + structure when unloading the chan_iax2.so module. - Move the code + to start using the LIST macros. Review: + http://reviewboard.digium.com/r/72 (closes issue #13232) Reported + by: eliel Patches: iax2-provision.patch.txt uploaded by eliel + (license 64) (with minor changes pointed by Mark Michelson on + review board) Tested by: eliel ........ + +2008-12-03 18:42 +0000 [r160628] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, apps/app_stack.c, apps/app_dial.c, /: Merged + revisions 160626 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r160626 | + mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16 + lines Add some safety measures when using gosub, especially when + using the options for app_dial and app_queue to run a gosub when + the call is answered. * Check for the existence of the gosub + target in gosub_exec. If it is nonexistent, then this will cause + errors when we attempt to actually run the gosub, including a + definite memory leak and potential crashes. Return an error in + this situation * Check the return value of pbx_exec in app_dial + and app_queue before attempting to actually run the gosub + routine. If there was an error, we should not attempt to run the + gosub. * Change a '|' to a ',' in app_queue. * Add some extra + curly braces where they had been missing previously. (closes + issue #13548) Reported by: fiddur ........ + +2008-12-03 17:41 +0000 [r160561] Tilghman Lesher <tlesher@digium.com> + + * pbx/pbx_spool.c, /: Merged revisions 160559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r160559 | tilghman | 2008-12-03 11:38:59 -0600 (Wed, 03 Dec 2008) + | 14 lines Merged revisions 160558 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008) + | 7 lines If an entry is added to the directory during a scan + when another entry expires, then that new entry will not be + processed promptly, but must wait for either a future entry to + start or a current entry's retry to occur. If no other entries + exist in the directory (other than the new entries) when a bunch + expire, then the new entries must wait until another new entry is + added to be processed. This was a rather weird race condition, + really. Fixes AST-147. ........ ................ + +2008-12-03 17:10 +0000 [r160557] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 160555 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r160555 | + mmichelson | 2008-12-03 11:07:09 -0600 (Wed, 03 Dec 2008) | 11 + lines When investigating issue #13548, I found that gosub + handling in app_queue was just completely wrong, mostly because + the channel operations being performed were being done on the + incorrect channel. With this set of changes, a gosub will + correctly run on the answering queue member's channel. There are + still crash issues which occur if there are dialplan syntax + errors, so I cannot yet close the referenced issue. ........ + +2008-12-03 17:02 +0000 [r160483-160554] Tilghman Lesher <tlesher@digium.com> + + * pbx/pbx_spool.c, /: Merged revisions 160552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r160552 | tilghman | 2008-12-03 11:01:03 -0600 (Wed, 03 Dec 2008) + | 11 lines Merged revisions 160551 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008) + | 4 lines Don't start scanning the directory until all modules + are loaded, because some required modules (channels, apps, + functions) may not yet be in memory yet. Fixes AST-149. ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 160481 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) + | 14 lines Merged revisions 160480 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) + | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I + guess that having only ip-phones in mind is not a good approach. + Since it is possible to have a sip proxy connected to asterisk we + could receive a 407 (unauthorized) or 483 (too many hops) as + response and dialog ending would not be a good behavior." So + modified. ........ ................ + +2008-12-02 18:05 +0000 [r160329-160339] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 160333 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008) + | 1 line remove duplicate comment that I accidentally merged + ........ + + * channels/chan_dahdi.c, /: Merged revisions 160319 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008) + | 7 lines (closes issue #13786) Reported by: tzafrir Readding + DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which + fixes not being able to make outgoing calls on some FXO adapters: + http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553 + ........ + +2008-12-02 18:03 +0000 [r160234-160325] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 160308 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) + | 17 lines Merged revisions 160297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) + | 10 lines When the text does not match exactly (e.g. RTP/SAVP), + then the %n conversion fails, and the resulting integer is + garbage. Thus, we must initialize the integer and check it + afterwards for success. (closes issue #14000) Reported by: folke + Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke + (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by + folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff + uploaded by folke (license 626) ........ ................ + + * include/asterisk/stringfields.h, apps/app_voicemail.c, + main/cli.c, main/pbx.c, main/frame.c, /, + channels/chan_features.c: Merged revisions 160208 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600 + (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) + | 3 lines Ensure that Asterisk builds with --enable-dev-mode, + even on the latest gcc and glibc. ........ ................ + +2008-12-01 23:53 +0000 [r160175] Sean Bright <sean.bright@gmail.com> + + * channels/chan_phone.c, main/manager.c, /, utils/smsq.c: Merged + revisions 160170-160172 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec + 2008) | 1 line Pay attention to the return value of system(), + even if we basically ignore it. ................ r160171 | + seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1 + line Silence a build warning. (chan_phone.c:810: warning: value + computed is not used) ................ r160172 | seanbright | + 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines Merged + revisions 159976 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) + | 3 lines Get rid of the useless format string and argument in + the Bogus/ manager channelname. Noted by kpfleming and name + Bogus/manager suggested by eliel ........ ................ + +2008-12-01 Tilghman Lesher <tlesher@digium.com> + + * Released 1.6.1-beta3 + +2008-12-01 21:46 +0000 [r160101] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 160097 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008) + | 2 lines Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or + bad things happen. ........ + +2008-12-01 17:45 +0000 [r160006] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 160004 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r160004 | russell | 2008-12-01 11:34:31 -0600 + (Mon, 01 Dec 2008) | 14 lines Merged revisions 160003 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) + | 6 lines Apply some logic used in iax2_indicate() to + iax2_setoption(), as well, since they both have the potential to + send control frames in the middle of call setup. We have to wait + until we have received a message back from the remote end before + we try to send any more frames. Otherwise, the remote end will + consider it invalid, and we'll get stuck in an INVAL/VNAK storm. + ........ ................ + +2008-12-01 16:06 +0000 [r159975] Michiel van Baak <michiel@vanbaak.info> + + * main/manager.c, /: Merged revisions 159898 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008) + | 11 lines Merged revisions 159897 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008) + | 4 lines make manager compile on OpenBSD. The last (10th) + argument to ast_channel_alloc here should be a pointer and NULL + is not really a pointer. ........ ................ + +2008-12-01 14:57 +0000 [r159920] Russell Bryant <russell@digium.com> + + * .cleancount, /: Merged revisions 159911 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008) + | 10 lines Merged revisions 159900 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008) + | 2 lines Force a "make clean" to avoid a bizarre build issue ... + ........ ................ + +2008-11-29 18:34 +0000 [r159854] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_readexten.c: Merged revisions 159853 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r159853 | tilghman | 2008-11-29 12:33:18 -0600 (Sat, 29 Nov 2008) + | 2 lines Allow the '#' sign to exist within an extension + (inspired by issue #13330) ........ + +2008-11-29 18:16 +0000 [r159851] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_iax2.c, cdr/cdr_tds.c, include/asterisk/logger.h, + include/asterisk/res_odbc.h, channels/chan_misdn.c, + include/asterisk/astmm.h, include/asterisk/lock.h, + utils/extconf.c, makeopts.in, main/dns.c, funcs/Makefile, + include/asterisk/stringfields.h, include/asterisk/utils.h, + include/asterisk/devicestate.h, /, include/asterisk/dundi.h, + configure.ac, utils/astman.c, include/asterisk/cli.h, + include/asterisk/channel.h, include/asterisk/manager.h, + res/res_config_sqlite.c, utils/conf2ael.c, utils/frame.c, + channels/misdn_config.c, main/ast_expr2.c, Makefile, main/srv.c, + include/asterisk/compat.h, configure, channels/misdn/ie.c, + include/asterisk/module.h, main/features.c, + include/asterisk/linkedlists.h, main/logger.c, main/event.c, + include/asterisk/dlinkedlists.h, include/asterisk/strings.h, + utils/check_expr.c, channels/chan_vpb.cc, channels/chan_sip.c, + main/Makefile, include/asterisk/enum.h, channels/chan_agent.c, + main/utils.c, include/jitterbuf.h: Merged revisions 159818 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 + Nov 2008) | 18 lines incorporates r159808 from branches/1.4: + ------------------------------------------------------------------------ + r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov + 2008) | 7 lines update dev-mode compiler flags to match the ones + used by default on Ubuntu Intrepid, so all developers will see + the same warnings and errors since this branch already had some + printf format attributes, enable checking for them and tag + functions that didn't have them format attributes in a consistent + way + ------------------------------------------------------------------------ + in addition: move some format attributes from main/utils.c to the + header files they belong in, and fix up references to the + relevant functions based on new compiler warnings ........ + +2008-11-26 19:58 +0000 [r159561] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, /: Merged revisions 159554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 | + mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 + lines Add some necessary hangup commands in the case that + forwarding a call fails 1) Hang up the original destination if + the local channel cannot be requested. 2) Hang up the local + channel (in addition to the original destination) if ast_call + fails when calling the newly created local channel. This prevents + channels from sticking around forever in the case of a botched + call forward (e.g. to an extension which does not exist). (closes + issue #13764) Reported by: davidw Patches: 13764_v2.patch + uploaded by putnopvut (license 60) Tested by: putnopvut, davidw + ........ + +2008-11-26 19:17 +0000 [r159535] Kevin P. Fleming <kpfleming@digium.com> + + * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules, + Makefile.rules: Merged revisions 159534 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov + 2008) | 11 lines Merged revisions 159476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov + 2008) | 7 lines simplify (and slightly bug-fix) the recent + developer-oriented COMPILE_DOUBLE mode ensure that 'make clean' + removes dependency files for .i files that are created in + COMPILE_DOUBLE mode ........ ................ + +2008-11-26 18:38 +0000 [r159477] Tilghman Lesher <tlesher@digium.com> + + * main/udptl.c, /: Merged revisions 159475 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r159475 | + tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines + If the config file does not exist, then the first use crashes + Asterisk. (closes issue #13848) Reported by: klaus3000 Patches: + udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage + ........ + +2008-11-26 14:59 +0000 [r159438] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 159437 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r159437 | mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov + 2008) | 10 lines Don't allow for configuration options to + overwrite options set via channel variables on a reload. (closes + issue #13921) Reported by: davidw Patches: 13921.patch uploaded + by putnopvut (license 60) Tested by: davidw ........ + +2008-11-26 03:19 +0000 [r159403] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: Merged revisions 159402 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r159402 | + jpeeler | 2008-11-25 21:18:01 -0600 (Tue, 25 Nov 2008) | 3 lines + Always parse arguments in park_call_exec so that app_args is + valid. This prevents a crash when executing Park from the + dialplan with no arguments. ........ + +2008-11-25 23:27 +0000 [r159375] Steve Murphy <murf@digium.com> + + * channels/chan_iax2.c, main/cdr.c, /: Merged revisions 159360 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue, + 25 Nov 2008) | 23 lines Merged revisions 159316 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | + 15 lines (closes issue #12694) Reported by: yraber Patches: + 12694.2nd.diff uploaded by murf (license 17) Tested by: murf, + laurav Thanks to file (Joshua Colp) for his IAX fix. the change + to cdr.c allows no-answer to percolate up into CDR's, and feels + like the right place to locate this fix; if BUSY is done here, + no-answer should be, too. ........ ................ + +2008-11-25 21:58 +0000 [r159249-159280] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 159276 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r159276 | tilghman | 2008-11-25 15:57:59 -0600 + (Tue, 25 Nov 2008) | 14 lines Merged revisions 159269 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) + | 7 lines Don't try to send a response on a NULL pvt. (closes + issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch + uploaded by eliel (license 64) Tested by: barthpbx ........ + ................ + + * channels/chan_iax2.c, /: Merged revisions 159247 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r159247 | tilghman | 2008-11-25 15:42:42 -0600 + (Tue, 25 Nov 2008) | 21 lines Merged revisions 159246 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600 + (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) + | 7 lines Regression fix for last security fix. Set the iseqno + correctly. (closes issue #13918) Reported by: ffloimair Patches: + 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) + Tested by: ffloimair ........ ................ ................ + +2008-11-25 16:21 +0000 [r159095] Terry Wilson <twilson@digium.com> + + * /, apps/app_festival.c: Merged revisions 159093 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 | + twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines + Add missing variable declaration for PPC code ........ + +2008-11-25 05:05 +0000 [r159053] Tilghman Lesher <tlesher@digium.com> + + * channels/xpmr/xpmr.c, apps/app_rpt.c, channels/chan_usbradio.c, + /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 159050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r159050 | tilghman | 2008-11-24 23:02:11 -0600 (Mon, 24 Nov 2008) + | 10 lines Merged revisions 159025 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) + | 3 lines System call ioperm is non-portable, so check for its + existence in autoconf. (Closes issue #13863) ........ + ................ + +2008-11-25 03:51 +0000 [r158993] Terry Wilson <twilson@digium.com> + + * channels/chan_usbradio.c, /: Merged revisions 158992 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008) + | 2 lines Make chan_usbradio compile under dev mode ........ + +2008-11-25 00:41 +0000 [r158894-158927] Matt Nicholson <mnicholson@digium.com> + + * apps/app_queue.c, /, UPGRADE.txt: Merged revisions 158924 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r158924 | mnicholson | 2008-11-24 18:05:41 -0600 (Mon, + 24 Nov 2008) | 6 lines Make the Join event from app_queue use + CallerIDNum insead of CallerID for indicating the callerid number + just like the rest of asterisk. (closes issue #13883) Reported + by: davidw ........ + + * /, main/file.c: Merged revisions 158925 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158925 | + mnicholson | 2008-11-24 18:19:55 -0600 (Mon, 24 Nov 2008) | 2 + lines Fix compiling in dev mode. ........ + + * include/asterisk/manager.h, main/manager.c, /, res/res_agi.c: + Merged revisions 158876 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158876 | + mnicholson | 2008-11-24 15:56:22 -0600 (Mon, 24 Nov 2008) | 7 + lines Added EVENT_FLAG_AGI and used it for manager calls in + res_agi.c (closes issue #13873) Reported by: fnordian Patches: + ami_agievent.patch uploaded by fnordian (license 110) ........ + +2008-11-24 21:53 +0000 [r158861] Tilghman Lesher <tlesher@digium.com> + + * main/dsp.c, /: Merged revisions 158857 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158857 | + tilghman | 2008-11-24 15:52:34 -0600 (Mon, 24 Nov 2008) | 3 lines + Add a bit of documentation (thanks, I-MOD) on what the silence + threshold constant actually does and what values are valid for + it. ........ + +2008-11-24 21:44 +0000 [r158855] Matt Nicholson <mnicholson@digium.com> + + * /, main/file.c: Merged revisions 158851 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158851 | + mnicholson | 2008-11-24 15:27:26 -0600 (Mon, 24 Nov 2008) | 6 + lines Make ast_streamfile() check the result of ast_openstream() + before doing anything with it. (closes issue #13955) Reported by: + chris-mac ........ + +2008-11-22 17:00 +0000 [r158689-158701] Michiel van Baak <michiel@vanbaak.info> + + * /, channels/chan_skinny.c: Merged revisions 158694 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r158694 | mvanbaak | 2008-11-22 17:57:11 +0100 (Sat, 22 Nov 2008) + | 8 lines dont send reorder tone after a device is hungup if a + dialout is abandoned or failed. Without this reorder tone will + play after hangup and both wedhorn's and my wife have threatened + to use an axe on our asterisk box (closes issue #13948) Reported + by: wedhorn Patches: switch.diff uploaded by wedhorn (license 30) + ........ + + * /, channels/chan_skinny.c: Merged revisions 158688 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r158688 | mvanbaak | 2008-11-22 17:06:38 +0100 (Sat, 22 Nov 2008) + | 4 lines fix a very occasional core dump in chan_skinny found by + wedhorn. (issue #13948) ........ + +2008-11-21 23:45 +0000 [r158607] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 158606 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r158606 | murf | 2008-11-21 16:40:46 -0700 (Fri, 21 Nov 2008) | + 19 lines Merged revisions 158603 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) | + 11 lines In reference to the fix made for 13871, I was merging + the fix into 1.6.0 and realized I missed the code in the h-exten + block, and didn't catch it because my test case had the h-exten + commented out. So, this corrects the code I missed, as a + preventative against another crash report. Tested with the + h-exten defined, all is well. ........ ................ + +2008-11-21 23:15 +0000 [r158604] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 158602 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) + | 12 lines Merged revisions 158600 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) + | 5 lines The passed extension may not be the same in the list as + the current entry, because we strip spaces when copying the + extension into the structure. Therefore, use the copied item to + place the item into the list. (found by lmadsen on -dev, fixed by + me) ........ ................ + +2008-11-21 22:57 +0000 [r158572] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 158484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) | + 19 lines Merged revisions 158483 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | + 11 lines (closes issue #13871) Reported by: mdu113 This one is + totally my fault. The code doesn't even create a bridge CDR if + the channel CDR has POST_DISABLED. I didn't check for that at the + end of the bridge. Fixed with a few small insertions. Tested. + Looks good. No cdr generated, no crash, no unnecc. data objects + created either. ........ ................ + +2008-11-21 22:13 +0000 [r158541] Russell Bryant <russell@digium.com> + + * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions + 158540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) + | 10 lines Merged revisions 158539 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) + | 2 lines When compiling with DEBUG_THREADS, report the real + file/func/line for ao2_lock/ao2_unlock ........ ................ + +2008-11-21 20:43 +0000 [r158450] Kevin P. Fleming <kpfleming@digium.com> + + * CHANGES, /, UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, + UPGRADE-1.6.txt: Merged revisions 158449 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158449 | + kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov 2008) | 3 + lines as suggested by jtodd, document the purposes of the CHANGES + and UPGRADE files ........ + +2008-11-21 19:42 +0000 [r158415] Jason Parker <jparker@digium.com> + + * main/manager.c, /: Merged revisions 158414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158414 | + qwell | 2008-11-21 13:40:57 -0600 (Fri, 21 Nov 2008) | 7 lines + Make sure we add the Event header for CoreShowChannels. (closes + issue #13334) Reported by: srt Patches: + 13334_missing_event_header_in_core_show_channel.diff uploaded by + srt (license 378) ........ + +2008-11-21 17:17 +0000 [r158377] Terry Wilson <twilson@digium.com> + + * cdr/cdr_csv.c, /: Merged revisions 158374 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158374 | + twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines + Reloading the config and having no changes still initialized some + settings to 0. Initialize settings after doing all of the cfg + checks. (closes issue #13942) Reported by: davidw Patches: + cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by: + davidw ........ + +2008-11-21 01:23 +0000 [r158223-158268] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 158265-158266 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, + 20 Nov 2008) | 4 lines Use some magic constants to get the right + size for this sscanf statement. Thanks Richard! ........ r158266 + | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 + lines Use a more expressive constant for a 64-bit scanned int + ........ + + * /, channels/chan_sip.c: Merged revisions 158262 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 | + mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 + lines Fix the build for 32-bit systems. %lu is only 32-bits on + 32-bit systems, so we need to use %llu instead. Of course %llu is + 128-bits on 64-bit systems, so we have to cast to unsigned long + long. No harm, but it's sure annoying. ........ + + * /, channels/chan_sip.c: Merged revisions 158230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 | + mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 + lines Change the remote user agent session version variable from + an int to a uint64_t. This prevents potential comparison problems + from happening if the version string exceeds INT_MAX. This was an + apparent problem for one user who could not properly place a call + on hold since the version in the SDP of the re-INVITE to place + the call on hold greatly exceeded INT_MAX. This also aligns with + RFC 2327 better since it recommends using an NTP timestamp for + the version (which is a 64-bit number). (closes issue #13531) + Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut + (license 60) Tested by: sgofferj ........ + + * channels/chan_sip.c: Change this so it actually compiles. Thanks, + Terry! + +2008-11-20 19:43 +0000 [r158191] Sean Bright <sean.bright@gmail.com> + + * res/ael/pval.c, /: Merged revisions 158188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158188 | + seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10 + lines Fix one case where the application argument was not + converted from a pipe to a comma. This was causing problems with + switch statements with empty expressions. (closes issue #13901) + Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by + seanbright (license 71) Tested by: seanbright Reviewed by: murf + ........ + +2008-11-20 18:23 +0000 [r158135] Terry Wilson <twilson@digium.com> + + * cdr/cdr_odbc.c, cdr/cdr_radius.c, cdr/cdr_custom.c, + cdr/cdr_manager.c, cdr/cdr_csv.c, cdr/cdr_sqlite3_custom.c, /, + cdr/cdr_sqlite.c, cdr/Makefile, cdr/cdr_adaptive_odbc.c, + cdr/cdr_pgsql.c: Merged revisions 158072 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | + twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines + Begin on a crusade to end trailing whitespace! ........ + +2008-11-20 18:20 +0000 [r158084-158134] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c, + /, channels/chan_sip.c, main/file.c: Merged revisions 158133 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r158133 | mmichelson | 2008-11-20 12:20:00 -0600 + (Thu, 20 Nov 2008) | 10 lines Merged revisions 158072 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 + Nov 2008) | 2 lines Begin on a crusade to end trailing + whitespace! ........ ................ + + * /, channels/chan_sip.c: Merged revisions 158082 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov + 2008) | 24 lines Merged revisions 158071 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov + 2008) | 16 lines We don't handle 4XX responses to BYE well. + According to section 15 of RFC 3261, we should terminate a dialog + if we receive a 481 or 408 in response to our BYE. Since I am + aware of at least one phone manufacturer who may sometimes send a + 404 as well, I am being liberal and saying that any 4XX response + to a BYE should result in a terminated dialog. (closes issue + #12994) Reported by: pabelanger Patches: 12994.patch uploaded by + putnopvut (license 60) Closes AST-129 ........ ................ + +2008-11-20 17:42 +0000 [r158069] Jeff Peeler <jpeeler@digium.com> + + * /, main/file.c: Merged revisions 158062 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158062 | + jpeeler | 2008-11-20 11:37:31 -0600 (Thu, 20 Nov 2008) | 6 lines + (closes issue #12929) Reported by: snyfer This handles the case + for a zero length file to attempt to be streamed. Instead of + failing from not playing any data, go ahead and return success as + ast_streamfile should consider playing nothing a success when + there is nothing to play. ........ + +2008-11-20 17:40 +0000 [r158067] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 158066 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r158066 | mmichelson | 2008-11-20 11:39:06 -0600 + (Thu, 20 Nov 2008) | 20 lines Merged revisions 158053 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov + 2008) | 12 lines Make sure to set the hangup cause on the calling + channel in the case that ast_call() fails. For incoming SIP + channels, this was causing us to send a 603 instead of a 486 when + the call-limit was reached on the destination channel. (closes + issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded + by putnopvut (license 60) Tested by: blitzrage ........ + ................ + +2008-11-20 00:10 +0000 [r157975] Kevin P. Fleming <kpfleming@digium.com> + + * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree, + channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, res/ael, + channels, main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash, + codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules, + main/db1-ast/mpool, res/ais, channels/misdn, res/snmp, + Makefile.rules, pbx/Makefile, res/Makefile: Merged revisions + 157974 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r157974 | kpfleming | 2008-11-19 18:08:12 -0600 (Wed, 19 Nov + 2008) | 13 lines Merged revisions 157859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov + 2008) | 7 lines the gcc optimizer frequently finds broken code + (use of uninitalized variables, unreachable code, etc.), which is + good. however, developers usually compile with the optimizer + turned off, because if they need to debug the resulting code, + optimized code makes that process very difficult. this means that + we get code changes committed that weren't adequately checked + over for these sorts of problems. with this build system change, + if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is + turned on, when a source file is compiled it will actually be + preprocessed (into a .i or .ii file), then compiled once with + optimization (with the result sent to /dev/null) and again + without optimization (but only if the first compile succeeded, of + course). while making these changes, i did some cleanup work in + Makefile.rules to move commonly-used combinations of flag + variables into their own variables, to make the file easier to + read and maintain ........ ................ + +2008-11-19 18:29 +0000 [r157785] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 157784 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r157784 | tilghman | 2008-11-19 12:28:14 -0600 (Wed, 19 Nov 2008) + | 6 lines Add check for t38_terminal_init in spandsp (not found + in 0.0.6, so it should fail reasonably) (closes issue #13473) + Reported by: genie Patches: 20080916__bug13473.diff.txt uploaded + by Corydon76 (license 14) ........ + +2008-11-19 13:47 +0000 [r157719-157744] Kevin P. Fleming <kpfleming@digium.com> + + * /, res/res_agi.c: Merged revisions 157743 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157743 | + kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line + correct small bug introduced during API conversion ........ + + * CHANGES, apps/app_stack.c, include/asterisk/agi.h, /, + res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added): Merged + revisions 157706 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | + kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 + lines make some corrections to the ast_agi_register_multiple(), + ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to + be consistent with API guidelines also, move UPGRADE.txt to + UPGRADE-1.6.txt and make the new UPGRADE.txt contain information + about upgrading between Asterisk 1.6 releases ........ + +2008-11-19 01:08 +0000 [r157641] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/logger.h, /, main/logger.c, main/utils.c, + include/asterisk/strings.h: Merged revisions 157639 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r157639 | tilghman | 2008-11-18 19:02:45 -0600 (Tue, 18 Nov 2008) + | 7 lines Starting with a change to ensure that ast_verbose() + preserves ABI compatibility in 1.6.1 (as compared to 1.6.0 and + versions of 1.4), this change also deprecates the use of Asterisk + with FreeBSD 4, given the central use of va_copy in core + functions. va_copy() is C99, anyway, and we already require C99 + for other purposes, so this isn't really a big change anyway. + This change also simplifies some of the core ast_str_* functions. + ........ + +2008-11-19 01:00 +0000 [r157636] Mark Michelson <mmichelson@digium.com> + + * /, main/astmm.c: Merged revisions 157632 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157632 | + mmichelson | 2008-11-18 18:59:48 -0600 (Tue, 18 Nov 2008) | 10 + lines If malloc returns NULL, we need to return NULL immediately + or else Asterisk will crash when attempting to dereference the + NULL pointer (closes issue #13858) Reported by: eliel Patches: + astmm.c.patch uploaded by eliel (license 64) ........ + +2008-11-19 00:38 +0000 [r157602] Sean Bright <sean.bright@gmail.com> + + * build_tools/make_buildopts_h, makeopts.in, Makefile, /, + build_tools/make_version, configure, configure.ac: Merged + revisions 157600 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157600 | + seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10 + lines Fix a few build problems on Solaris (and check for an md5 + utility in configure instead of the icky loop I was doing + before). (closes issue #13842) Reported by: snuffy Patches: + bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff + uploaded by seanbright (license 71) Tested by: snuffy ........ + +2008-11-18 23:59 +0000 [r157429-157596] Mark Michelson <mmichelson@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 157592 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r157592 | mmichelson | 2008-11-18 17:59:02 -0600 (Tue, 18 Nov + 2008) | 10 lines This change prevents a crash from occurring if + res_musiconhold.so is unloaded and then Asterisk is stopped. The + problem was that we are not unregistering the ast_moh_destroy + function at exit. (closes issue #13761) Reported by: eliel + Patches: res_musiconhold.c.patch uploaded by eliel (license 64) + ........ + + * apps/app_voicemail.c, /: Merged revisions 157562 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r157562 | mmichelson | 2008-11-18 17:28:23 -0600 (Tue, 18 Nov + 2008) | 11 lines Fix the logic for when delete=yes when IMAP + storage is in use so that the message is deleted from both local + and IMAP storage. (closes issue #13642) Reported by: jaroth + Patches: deleteyes.patch uploaded by jaroth (license 50) ........ + + * /, channels/chan_sip.c: Merged revisions 157512 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov + 2008) | 21 lines Merged revisions 157503 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov + 2008) | 13 lines Add some missing invite state changes necessary + in the sip_write function. Not setting the invite state correctly + on the call was resulting in the Record application leaving empty + files. I also have updated the doxygen comment next to the + declaration of the INV_EARLY_MEDIA constant to reflect that we + also use this state when we *send* a 18X response to an INVITE. + (closes issue #13878) Reported by: nahuelgreco Patches: + sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco + (license 162) Tested by: putnopvut ........ ................ + + * /, channels/chan_sip.c: Merged revisions 157496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 | + mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 + lines Based on Russell's advice on the asterisk-dev list, I have + changed from using a global lock in update_call_counter to using + the locks within the sip_pvt and sip_peer structures instead. + ........ + + * /, channels/chan_sip.c: Merged revisions 157427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 | + mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 + lines * Add a lock to be used in the update_call_counter + function. * Revert logic to mirror 1.4's in the sense that it + will not allow the call counter to dip below 0. These two + measures prevent potential races that could cause a SIP peer to + appear to be busy forever. (closes issue #13668) Reported by: mjc + Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic + (license 586) ........ + +2008-11-18 19:18 +0000 [r157367] Jeff Peeler <jpeeler@digium.com> + + * /, apps/app_meetme.c: Merged revisions 157366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r157366 | jpeeler | 2008-11-18 13:16:00 -0600 (Tue, 18 Nov 2008) + | 14 lines Merged revisions 157365 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) + | 6 lines (closes issue #13899) Reported by: akkornel This fix is + the result of a bug fix in ast_app_separate_args r124395. If an + argument does not exist it should always be set to a null string + rather than a null pointer. ........ ................ + +2008-11-18 18:32 +0000 [r157308] Mark Michelson <mmichelson@digium.com> + + * apps/app_followme.c, apps/app_dial.c, channels/chan_local.c, /, + main/features.c, include/asterisk/channel.h: Merged revisions + 157306 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov + 2008) | 20 lines Merged revisions 157305 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov + 2008) | 12 lines Fix a crash in the end_bridge_callback of + app_dial and app_followme which would occur at the end of an + attended transfer. The error occurred because we initially stored + a pointer to an ast_channel which then was hung up due to a + masquerade. This commit adds a "fixup" callback to the + bridge_config structure to allow for end_bridge_callback_data to + be changed in the case that a new channel pointer is needed for + the end_bridge_callback. ........ ................ + +2008-11-18 18:20 +0000 [r157304] Steve Murphy <murf@digium.com> + + * main/config.c, /: Merged revisions 157302 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157302 | + murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines + (closes issue #13420) Reported by: alex70 Patches: + 13420.13539.patch uploaded by murf (license 17) Tested by: murf, + awk This fixes two problems: a spurious linefeed insertion + probably left over from pre-precomment times. Only generated when + category had no previous comments. The other problem: Insertions + could get the line-numbering out of whack and generate negative + line numbers, causing chunks of line numbers to be emitted, on + the scale of the number of lines up to that point in the file. In + such cases, abort the looping, and all is well. ........ + +2008-11-17 22:39 +0000 [r157255] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, /: Merged revisions 157253 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157253 | + tilghman | 2008-11-17 16:25:06 -0600 (Mon, 17 Nov 2008) | 8 lines + Can't use items duplicated off the stack frame in an element + returned from a function: in these cases, we have to use the + heap, or garbage will result. (closes issue #13898) Reported by: + alecdavis Patches: 20081114__bug13898__2.diff.txt uploaded by + Corydon76 (license 14) Tested by: alecdavis ........ + +2008-11-15 19:49 +0000 [r157108-157166] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged + revisions 157164 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov + 2008) | 13 lines Merged revisions 157162-157163 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov + 2008) | 1 line dist-clean should remove dependency information + files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03 + +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory + dist-clean is run, run clean in that directory first, and when + running top-level dist-clean, do not run subdirectory clean + operations twice ........ ................ + + * /, contrib/asterisk-ng-doxygen: Merged revisions 157105 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 + Nov 2008) | 13 lines major update to doxygen configuration file: + 1) update to doxygen 1.5.x style file, as used in trunk 2) tell + doxygen where are header files are, so include-file processing + can be done 3) make all macros that are used to define + variables/functions be expanded, so that doxygen will properly + document the resulting variable/function 4) make all macros that + are used to provide the contents of a variable (structure) be + expanded, so that doxygen will be able to document the resulting + fields 5) suppress compiler attributes (__attribute__(xxx)) from + being seen by doxygen, so it will properly match up function + definition and usage (for an example of th effect of this, look + at the doxygen docs for ast_log() from before and afte this + commit) ........ + +2008-11-15 04:30 +0000 [r157040-157042] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c, main/features.c, main/taskprocessor.c: + Merged revisions 157041 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157041 | + russell | 2008-11-14 22:25:57 -0600 (Fri, 14 Nov 2008) | 3 lines + Fix a few more places where the case insensitive hash should be + used since the comparison is case insensitive. ........ + + * /, channels/chan_console.c: Merged revisions 157039 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r157039 | russell | 2008-11-14 22:08:42 -0600 (Fri, 14 Nov 2008) + | 3 lines Use the new case insensitive hash function for console + interfaces. The comparison function is case insensitive. ........ + +2008-11-14 21:21 +0000 [r156963] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 156962 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r156962 | + mmichelson | 2008-11-14 15:19:58 -0600 (Fri, 14 Nov 2008) | 7 + lines Revision 155513 of chan_sip.c in trunk inadvertently + removed a very important line to set the "len" field for incoming + SIP requests. The result was that all incoming SIP messages + appeared to be 0-length, meaning Asterisk could do no meaningful + processing of anything SIP-related ........ + +2008-11-14 17:04 +0000 [r156913] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /: Merged revisions 156911 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r156911 | + tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines + Ping is missing the standard double-newline after the event. + (closes issue #13903) Reported by: kebl0155 ........ + +2008-11-14 16:57 +0000 [r156819-156894] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, include/asterisk/strings.h: This is the 1.6.1 + version of trunk commit 156883. It is functionally equivalent to + the 1.6.0 commit + + * apps/app_voicemail.c, /: Merged revisions 156817 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r156817 | mmichelson | 2008-11-14 09:20:03 -0600 + (Fri, 14 Nov 2008) | 18 lines Merged revisions 156816 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov + 2008) | 10 lines If the prompt to reenter a voicemail password + timed out, it resulted in the password not being saved, even if + the input matched what you gave when first prompted to enter a + new password. This is because the return value of ast_readstring + was checked, but not checked properly. This bug was discovered by + Jared Smith during an Asterisk training course. Thanks for + reporting it! ........ ................ + +2008-11-14 00:44 +0000 [r156691-156757] Tilghman Lesher <tlesher@digium.com> + + * apps/app_while.c, /: Merged revisions 156756 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) + | 13 lines Merged revisions 156755 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) + | 6 lines ast_waitfordigit() requires that the channel be up, for + no good logical reason. This prevents While/EndWhile from working + within the "h" extension. Reported by: jgalarneau (for ABE C.2) + Fixed by: me ........ ................ + + * main/manager.c, /: Merged revisions 156690 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) + | 14 lines Merged revisions 156688 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) + | 7 lines Provide more space for all the data which can appear in + an originating channel name. (closes issue #13398) Reported by: + bamby Patches: manager.c.diff uploaded by bamby (license 430) + ........ ................ + +2008-11-13 19:29 +0000 [r156654] Brandon Kruse <bkruse@digium.com> + + * main/manager.c: Merged revisions 156017 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r156017 | + pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines + Patch by Ryan Brindley -- Make sure that manager refuses any + duplicate 'new category' requests in updateconfig (closes issue + #13539) ........ + +2008-11-13 19:18 +0000 [r156650] Jeff Peeler <jpeeler@digium.com> + + * main/pbx.c, /: Merged revisions 156649 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r156649 | + jpeeler | 2008-11-13 13:17:50 -0600 (Thu, 13 Nov 2008) | 6 lines + (closes issue #13891) Reported by: smurfix This reverts a change + I made in 116297. At the time it seemed the change was required + to solve an issue with attempting a transfer but then letting it + timeout without dialing any digits. However, I didn't realize + that having an empty extension was possible. I'm removing the + immediate return that was added in pbx_find_extension if the + extension is null. ........ + +2008-11-13 17:12 +0000 [r156614] Mark Michelson <mmichelson@digium.com> + + * autoconf/ast_c_compile_check.m4, /, configure: Merged revisions + 156612 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r156612 | + mmichelson | 2008-11-13 11:07:56 -0600 (Thu, 13 Nov 2008) | 4 + lines Kevin sent a note indicating that this change is not + necessary, so I am reverting it ........ + +2008-11-12 21:36 +0000 [r156389] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, /: Merged revisions 156388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) + | 12 lines Merged revisions 156386 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) + | 5 lines When using call limits under 1 second, infinite call + lengths are allowed, instead. (closes issue #13851) Reported by: + ruddy ........ ................ + +2008-11-12 20:11 +0000 [r156354] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 156299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) | + 26 lines Merged revisions 156297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | + 18 lines It turns out that the 0x0XX00 codes being returned for + N, X, and Z are off by one, as per conversation with jsmith on + #asterisk-dev; he was teaching a class and disconcerted that this + published rule was not being followed, with patterns _NXX, + _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should + have been. This change, tested on these 3 patterns now picks the + proper one. However, this change may surprise users who set up + dialplans based on previous behavior, which has been there for + what, 2 and half years or so now. ........ ................ + +2008-11-12 19:29 +0000 [r156296] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_meetme.c: Merged revisions 156295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) + | 13 lines Merged revisions 156294 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) + | 6 lines If the SLA thread is not started, then reload causes a + memory leak. (closes issue #13889) Reported by: eliel Patches: + app_meetme.c.patch uploaded by eliel (license 64) ........ + ................ + +2008-11-12 19:11 +0000 [r156291] Jeff Peeler <jpeeler@digium.com> + + * /, apps/app_meetme.c: Merged revisions 156290 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008) + | 11 lines Merged revisions 156289 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) + | 3 lines For whatever reason, gcc only warned me about the + possible use of an uninitialized variable when compiling 1.6.1. + ........ ................ + +2008-11-12 19:05 +0000 [r156284-156288] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 156243 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r156243 | tilghman | 2008-11-12 12:55:18 -0600 + (Wed, 12 Nov 2008) | 18 lines Merged revisions 156229 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) + | 11 lines Revert revision 132506, since it occasionally caused + IAX2 HANGUP packets not to be sent, and instead, schedule a task + to destroy the iax2 pvt structure 10 seconds later. This allows + the IAX2 HANGUP packet to be queued, transmitted, and ACKed + before the pvt is destroyed. (closes issue #13645) Reported by: + dzajro Patches: 20081111__bug13645__3.diff.txt uploaded by + Corydon76 (license 14) Tested by: vazir Reviewed: + http://reviewboard.digium.com/r/51/ ........ ................ + + * apps/app_meetme.c: Fix build (res possibly unused in this + function, says gcc) + +2008-11-12 18:55 +0000 [r156247] Jeff Peeler <jpeeler@digium.com> + + * /, apps/app_meetme.c: Merged revisions 156228 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008) + | 16 lines Merged revisions 156178 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) + | 8 lines (closes issue #13173) Reported by: pep This change adds + an announce_thread responsible for playing announcements to an + existing conference. This allows all announcing to be immediately + stopped if necessary but more importantly allows other threads + that need to play something to not block. There are multiple + benefits to this, but the actual bug is for solving the scenario + for a channel to be unusable after hang up for the entire + duration of the parting announcement. The parting announcement + can be extremely long depending on what the user recorded upon + joining the conference. Reviewed by Russell on Review Board: + http://reviewboard.digium.com/r/25/ ........ ................ + +2008-11-12 17:48 +0000 [r156171] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, /: Merged revisions 156169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov + 2008) | 15 lines Merged revisions 156167 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov + 2008) | 7 lines When doing some tests, I was having a crash at + the end of every call if an attended transfer occurred during the + call. I traced the cause to the CDR on one of the channels being + NULL. murf suggested a check in the end bridge callback to be + sure the CDR is non-NULL before proceeding, so that's what I'm + adding. ........ ................ + +2008-11-12 17:38 +0000 [r156168] Russell Bryant <russell@digium.com> + + * main/asterisk.c, /: Merged revisions 156166 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) + | 15 lines Merged revisions 156164 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) + | 7 lines Move the sanity check that makes sure "always fork" is + not set along with the console option to be after the code that + reads options from asterisk.conf. This resolves a situation where + Asterisk can start taking up 100% when misconfigured. (Thanks to + Bryce Porter (x86 on IRC) for letting me log in to his system to + figure out what was causing the 100% CPU problem.) ........ + ................ + +2008-11-12 15:34 +0000 [r156128] Mark Michelson <mmichelson@digium.com> + + * autoconf/ast_c_compile_check.m4, /, configure: Merged revisions + 156127 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r156127 | + mmichelson | 2008-11-12 09:33:11 -0600 (Wed, 12 Nov 2008) | 5 + lines Add a couple of AC_SUBST calls to the AST_C_COMPILE_CHECK + macro. These missing calls were discovered when working on + timerfd support in a separate branch. ........ + +2008-11-11 19:52 +0000 [r156005] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_realtime.c: Merged revisions 155862 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155862 | + tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines + Make documentation of update method match documentation and + update update2 method to match. Reported by: atis, via -dev + mailing list. Fixed by: me ........ + +2008-11-10 21:15 +0000 [r155864] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 155863 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r155863 | mmichelson | 2008-11-10 15:14:44 -0600 + (Mon, 10 Nov 2008) | 22 lines Merged revisions 155861 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov + 2008) | 14 lines Channel drivers assume that when their indicate + callback is invoked, that the channel on which the callback was + called is locked. This patch corrects an instance in chan_agent + where a channel's indicate callback is called directly without + first locking the channel. This was leading to some observed + locking issues in chan_local, but considering that all channel + drivers operate under the same expectations, the generic fix in + chan_agent is the right way to go. AST-126 ........ + ................ + +2008-11-10 20:56 +0000 [r155764-155826] Tilghman Lesher <tlesher@digium.com> + + * doc/valgrind.txt, /: Merged revisions 155804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155803 | + tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line + I got tired of saying this in every single bugnote referring to + this file. ........ + + * /, main/editline/readline.c: Merged revisions 155763 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r155763 | tilghman | 2008-11-10 12:04:30 -0600 (Mon, 10 Nov 2008) + | 6 lines Fix memory leak when MALLOC_DEBUG is enabled. (closes + issue #13864) Reported by: eliel Patches: readline.c.patch + uploaded by eliel (license 64) ........ + +2008-11-09 16:32 +0000 [r155556-155672] Sean Bright <sean.bright@gmail.com> + + * configs/chan_dahdi.conf.sample, /: Merged revisions 155671 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r155671 | seanbright | 2008-11-09 11:30:29 -0500 (Sun, + 09 Nov 2008) | 1 line Fix this as well. Pointed out by tzafrir. + ........ + + * apps/app_followme.c, apps/app_queue.c, apps/app_dial.c, /, + main/features.c, include/asterisk/channel.h: Merged revisions + 155554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov + 2008) | 14 lines Merged revisions 155553 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov + 2008) | 6 lines Use static functions here instead of nested ones. + This requires a small change to the ast_bridge_config struct as + well. To understand the reason for this change, see the following + post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html + ........ ................ + +2008-11-08 21:48 +0000 [r155515-155517] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c, include/asterisk/strings.h: Merged + revisions 155516 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155516 | + russell | 2008-11-08 15:46:43 -0600 (Sat, 08 Nov 2008) | 3 lines + - Check for failure when putting the packet in the ast_str - fix + a spelling error in a header file ........ + + * /, channels/chan_sip.c: Merged revisions 155513 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155513 | + russell | 2008-11-08 15:34:36 -0600 (Sat, 08 Nov 2008) | 3 lines + Remove some code that is basically a no-op. Code above this + already ensures that the buffer is terminated. ........ + +2008-11-07 23:42 +0000 [r155469] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 155467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 | + mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 + lines Set the invite state to INV_CANCELLED in a place that makes + more sense. Where it was set before, it was impossible to + actually delay sending a CANCEL if we had not yet received a + provisional response to an INVITE. (closes issue #13626) Reported + by: atis Patches: 13626.patch uploaded by putnopvut (license 60) + Tested by: atis ........ + +2008-11-07 22:29 +0000 [r155396-155400] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 155399 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) + | 14 lines Merged revisions 155398 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) + | 7 lines Clarify error message. (closes issue #13809) Reported + by: denke Patches: 20081104__bug13809.diff.txt uploaded by + Corydon76 (license 14) Tested by: denke ........ ................ + + * /, funcs/func_odbc.c: Merged revisions 155395 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155395 | + tilghman | 2008-11-07 16:03:50 -0600 (Fri, 07 Nov 2008) | 2 lines + Two bugs relating to colnames found by Marquis42 on #asterisk-dev + ........ + +2008-11-07 21:16 +0000 [r155362] Mark Michelson <mmichelson@digium.com> + + * /, configs/voicemail.conf.sample: Merged revisions 155360 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri, + 07 Nov 2008) | 8 lines Remove one more instance of the sample + configuration lying about what's possible. The tz cannot be set + in a context like this. It can only be set in the general section + or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing + this out ........ + +2008-11-07 20:19 +0000 [r155325] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 155324 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r155324 | tilghman | 2008-11-07 14:13:32 -0600 (Fri, 07 Nov 2008) + | 7 lines Send call release with unallocated cause instead of + normal call clearing, when invalid extension is called. (closes + issue #13408) Reported by: adomjan Patches: + chan_dahdi.c-ss7-unallocated-2 uploaded by adomjan (license 487) + ........ + +2008-11-07 15:43 +0000 [r155242-155272] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 155264 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155264 | + russell | 2008-11-07 09:42:04 -0600 (Fri, 07 Nov 2008) | 3 lines + Remove a bogus ast_free() that Kevin noticed. This was probably + just left over from pre-astobj2ified chan_sip. ........ + + * /, include/asterisk/astobj2.h: Merged revisions 155244 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r155244 | russell | 2008-11-07 09:01:02 -0600 (Fri, 07 + Nov 2008) | 4 lines Clarify which part of OBJ_MULTIPLE is not + implemented, and under what case it is perfectly fine to use. + (Inspired by a question I received about my last commit.) + ........ + + * /, channels/chan_sip.c: Merged revisions 155241 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155241 | + russell | 2008-11-07 08:50:30 -0600 (Fri, 07 Nov 2008) | 4 lines + Fix some code in chan_sip that was intended to unlink multiple + objects from a container. The OBJ_MULTIPLE flag must be provided + here. Otherwise, this would only remove a single object. ........ + +2008-11-06 22:49 +0000 [r155117-155122] Kevin P. Fleming <kpfleming@digium.com> + + * res/ael/ael.flex, /, res/ael/ael_lex.c, utils/extconf.c: Merged + revisions 155121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155121 | + kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3 + lines don't blindly assume that Darwin and Cygwin need + GLOB_ABORTED defined; only define it if it is not already defined + ........ + + * configure, configure.ac: ensure that an adequately new version of + libpri is in place so that chan_dahdi will compile with PRI + support + +2008-11-06 19:48 +0000 [r155014] Mark Michelson <mmichelson@digium.com> + + * /, configs/voicemail.conf.sample: Merged revisions 155012 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600 + (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov + 2008) | 8 lines The documentation listed the ability to set + 'maxmsg' per context. The truth is that you can only set this in + the general section or per mailbox. Thus I am updating the sample + config file to be more accurate. Thanks to sasargen on IRC for + bringing up this issue. ........ ................ + +2008-11-05 22:02 +0000 [r154920] Sean Bright <sean.bright@gmail.com> + + * include/asterisk.h, /: Merged revisions 154919 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 | + seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2 + lines Fix a problem found while building res_snmp. ........ + +2008-11-05 22:00 +0000 [r154917] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 154428 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r154428 | tilghman | 2008-11-04 17:03:00 -0600 (Tue, 04 Nov 2008) + | 7 lines Switch to using a thread condition to signal that a + child thread is ready for work, rather than a busy wait. (closes + issue #13011) Reported by: jpgrayson Patches: + chan_iax2_find_idle.patch uploaded by jpgrayson (license 492) + ........ + +2008-11-05 16:14 +0000 [r154690] Steve Murphy <murf@digium.com> + + * main/channel.c, /: Merged revisions 154687 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r154687 | murf | 2008-11-05 09:11:11 -0700 (Wed, 05 Nov 2008) | 9 + lines Merged revisions 154685 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1 + line This fix was prompted by communication from user, who was + seeing thousands of error logs... looks like EAGAIN. Made such + uninteresting. ........ ................ + +2008-11-04 20:52 +0000 [r154367] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 154366 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600 + (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) + | 9 lines On busy systems, it's possible for the values checked + within a single line of code to change, unless the structure is + locked to ensure a consistent state. (closes issue #13717) + Reported by: kowalma Patches: 20081102__bug13717.diff.txt + uploaded by Corydon76 (license 14) Tested by: kowalma ........ + ................ + +2008-11-04 19:09 +0000 [r154269] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 154268 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600 + (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) + | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state + when it receives the indication AST_CONTROL_RINGING. ........ + ................ + +2008-11-04 19:02 +0000 [r154024-154267] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_h323.c: Merged revisions 154264 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r154264 | tilghman | 2008-11-04 12:59:48 -0600 + (Tue, 04 Nov 2008) | 10 lines Recorded merge of revisions 154263 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) + | 3 lines Make the monitor thread non-detached, so it can be + joined (suggested by Russell on -dev list). ........ + ................ + + * apps/app_voicemail.c, /: Recorded merge of revisions 154072 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r154072 | tilghman | 2008-11-03 16:28:12 -0600 + (Mon, 03 Nov 2008) | 12 lines Merged revisions 154066 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008) + | 5 lines Attempting to expunge a mailbox when the mailstream is + NULL will crash Asterisk. (Closes issue #13829) Reported by: + jaroth Patch by: me (modified jaroth's patch) ........ + ................ + + * main/rtp.c, /: Merged revisions 154060 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) + | 3 lines Remove the potential for a division by zero error. + (Closes issue #13810) ........ + + * /, funcs/func_odbc.c: Recorded merge of revisions 154023 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r154023 | tilghman | 2008-11-03 15:01:30 -0600 (Mon, 03 + Nov 2008) | 4 lines Should have passed the string pointer, not + the ast_str structure. (closes issue #13830) Reported by: Marquis + ........ + +2008-11-03 00:21 +0000 [r153710-153711] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/compiler.h, apps/app_stack.c, + include/asterisk/agi.h, configure, + include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4, + configure.ac: Merged revision 153709 from trunk + ------------------------------------------------------------------------ + r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov + 2008) | 3 lines instead of trying to forcibly load res_agi when + app_stack is loaded (even if the administrator didn't want it + loaded), use GCC weak symbols to determine whether it was loaded + already or not; if it was loaded, then use it. + ------------------------------------------------------------------------ + + * channels/chan_iax2.c, res/res_jabber.c, channels/chan_oss.c, + utils/stereorize.c, main/channel.c, main/manager.c, + res/ael/ael_lex.c, main/file.c, pbx/pbx_dundi.c, + formats/format_gsm.c, main/asterisk.c, utils/muted.c, /, + formats/format_wav.c, apps/app_authenticate.c, + res/res_phoneprov.c, res/res_crypto.c, utils/astman.c, + res/res_musiconhold.c, res/res_http_post.c, apps/app_queue.c, + res/res_config_sqlite.c, agi/eagi-sphinx-test.c, utils/frame.c, + channels/chan_dahdi.c, res/ael/ael.tab.c, funcs/func_odbc.c, + main/ast_expr2f.c, res/res_agi.c, main/http.c, main/logger.c, + channels/chan_h323.c, apps/app_sms.c, res/ael/ael.flex, + pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c, + utils/streamplayer.c, apps/app_adsiprog.c, apps/app_voicemail.c, + apps/app_dial.c, channels/chan_sip.c, apps/app_festival.c, + main/db1-ast/hash/hash_page.c, res/ael/ael.y, agi/eagi-test.c, + pbx/pbx_lua.c, formats/format_ogg_vorbis.c, main/utils.c, + utils/astcanary.c, formats/format_wav_gsm.c: import gcc 4.3.2 + warning fixes from trunk, with a few changes specific to this + branch + +2008-11-02 20:07 +0000 [r153363-153653] Russell Bryant <russell@digium.com> + + * include/asterisk/features.h, /: Merged revisions 153652 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r153652 | russell | 2008-11-02 14:06:03 -0600 + (Sun, 02 Nov 2008) | 10 lines Merged revisions 153651 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008) + | 2 lines features.h depends on linkedlists.h, so include it + ........ ................ + + * /, channels/chan_sip.c: Merged revisions 153362 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r153362 | + russell | 2008-11-01 15:41:38 -0500 (Sat, 01 Nov 2008) | 3 lines + Ensure that the sip_pvt properly has its refcount incremented + when the scheduler holds a reference to it for session timer + processing. ........ + +2008-10-31 22:11 +0000 [r153266] Terry Wilson <twilson@digium.com> + + * apps/app_followme.c, apps/app_queue.c, apps/app_dial.c, /, + main/features.c, include/asterisk/channel.h: Merged revisions + 153181 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 | + twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines + Recent CDR fixes moved execution of the 'h' exten into the + bridging code, so variables that were set after ast_bridge_call + was called would not show up in the 'h' exten. Added a callback + function to handle setting variables, etc. from w/in the bridging + code. Calls back into a nested function within the function + calling ast_bridge_call (closes issue #13793) Reported by: + greenfieldtech ........ + +2008-10-31 20:10 +0000 [r153225] Mark Michelson <mmichelson@digium.com> + + * main/dial.c, include/asterisk/dial.h: This commit contains the + bug fixes and documentation updates which were committed to trunk + in revision 153223. I blocked that commit from 1.6.1 since it + also contained a new feature. Note to self: Separate commits so + that you don't end up with a situation where part of a commit + should be merged but part should be blocked from stable branches. + +2008-10-31 16:36 +0000 [r153123] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Turn off qualify on uncached realtime + peers. (Closes issue #13383) + +2008-10-30 21:01 +0000 [r152995] Sean Bright <sean.bright@gmail.com> + + * bootstrap.sh: The -I argument to aclocal needs a space before + the include directory name. + +2008-10-30 20:36 +0000 [r152924-152974] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_h323.c: Cannot join detached threads. See + http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html + (Closes issue #13400) + + * channels/chan_local.c: Unlock before returning, when extension + doesn't exist. (closes issue #13807) Reported by: eliel Patches: + chan_local.c.patch uploaded by eliel (license 64) + +2008-10-30 19:41 +0000 [r152878-152921] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Fix the sip_peer reference count with + respect to scheduler entries for scheduling peer pokes, and + scheduling peer poke expirations. + + * channels/chan_sip.c: Fix the sip_peer reference count with + respect to scheduler entries for registration expirations. + + * include/asterisk/sched.h: Fix a bug in AST_SCHED_REPLACE_UNREF(). + The reference count of the object _must_ be increased before + creating the scheduler entry. Otherwise, you create a race + condition where the reference count may hit zero and the object + can disappear out from under you. This could also would have + incorrectly decreased the reference count in the case that the + scheduler add failed. + + * channels/chan_sip.c: Modify the documentation of the sip_registry + struct - Remove a comment that says that the monitor thread is the + only one that ever touches these objects. This is no longer the + case with TCP. Also, I would eventually like to get the scheduler + in its own thread, so this is just a poor assumption to make. - + Note that reference counting of these objects with respect to + scheduler entries is not complete. There are some leaked + references when deleting scheduler entries. + +2008-10-30 16:55 +0000 [r152814] Kevin P. Fleming <kpfleming@digium.com> + + * main/cdr.c: instead of comparing the string pointer to 0, + let's compare the value that was actually parsed out of the + string (found by sparse) + +2008-10-30 04:29 +0000 [r152690-152777] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample: Set up an example stdexten that + preserves the original context and extension in the CDR. (Related + to issue #13799) Reported by: davidw + + * main/pbx.c: Track down and fix annoying lock errors. These would + occur when merging hints that resulted from a pattern matched hint + during a 'dialplan reload'. + +2008-10-29 20:55 +0000 [r152648] Mark Michelson <mmichelson@digium.com> + + * apps/app_directory.c: If there was no named defined in a + voicemail.conf mailbox entry, then app_directory would crash when + attempting to read that entry from the file. We now check for the + NULL or empty string properly so that there will be no crash. + (closes issue #13804) Reported by: bluecrow76 + +2008-10-29 20:16 +0000 [r152645] Terry Wilson <twilson@digium.com> + + * apps/app_queue.c: Small modification to putnopvut's patch to fix + this issue. Thanks for all the help, putnopvut! (closes issue + #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch + uploaded by otherwiseguy (license 396) Tested by: otherwiseguy + +2008-10-29 05:52 +0000 [r152606] Steve Murphy <murf@digium.com> + + * apps/app_queue.c, configs/features.conf.sample, apps/app_dial.c: + A little documentation cross-ref between features and dial and + queue... I wasted some time (stupidly) trying to get the + one-touch parking stuff working, because it didn't occur to me + that I had to also have the corresponding options in the dial + command! Duh! (In all this time, I never set this up before!) So, + to keep some poor fool from suffering the same fate, I made the + features.conf.sample file mention the corresponding opts in + dial/queue; and the docs for dial/app specifically mention the + corresponding decls in the feature.conf file. I hope this doesn't + spoil some vast, eternal plan... + +2008-10-29 05:35 +0000 [r152573] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes + issue #13795) Reported by: andrew53 Patches: + chan_sip_sizeof.patch uploaded by andrew53 (license 519) + +2008-10-29 05:09 +0000 [r152537] Steve Murphy <murf@digium.com> + + * apps/app_queue.c, include/asterisk/features.h, apps/app_dial.c, + main/features.c, include/asterisk/pbx.h: The magic trick to avoid + this crash is not to try to find the channel by name in the list, + which is slow and resource consuming, but rather to pay attention + to the result codes from the ast_bridge_call, to which I added the + AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when + a channel is parked. Why? because CDR's aren't generated via + parking, so nothing is needed, but if a transfer occurred, there + are critical things I need. If you get AST_PBX_KEEPALIVE, then + don't touch the channel pointer. If you get + AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then + don't touch the peer pointer. Updated the several places where + the results from a bridge were not being properly obeyed, and + fixed some code I had introduced so that the results of the + bridge were not overridden (in trunk). All the places that + previously tested for AST_PBX_NO_HANGUP_PEER now have to check + for both AST_PBX_NO_HANGUP_PEER and + AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common + parking scenarios: 1. A calls B; B answers; A parks B; B hangs up + while A is getting the parking slot announcement, immediately + after being put on hold. 2. A calls B; B answers; A parks B; B + hangs up after A has been hung up, but before the park times out. + 3. A calls B; B answers; B parks A; A hangs up while B is getting + the parking slot announcement, immediately after being put on + hold. 4. A calls B; B answers; B parks A; A hangs up after B has + been hung up, but before the park times out. No crash. I also ran + the scenarios above against valgrind, and accesses looked good. + +2008-10-28 22:35 +0000 [r152370-152471] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Quoting in the wrong direction (Fixes + AST-107) + + * channels/chan_mgcp.c: Only re-add the io port if it was closed, + otherwise reload causes a memory leak. (closes issue #13785) + Reported by: eliel Patches: chan_mgcp.c.patch uploaded by eliel + (license 64) + + * apps/app_dial.c: Reset all DIAL variables back to blank, in case + Dial is called multiple times per call (which could otherwise + lead to inconsistent status reports). (closes issue #13216) + Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded + by Corydon76 (license 14) Tested by: ruddy + +2008-10-27 23:32 +0000 [r152288] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: Buffer policy setting for half is not + needed. + +2008-10-27 21:53 +0000 [r152173-152217] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c: Inherit ALL elements of CallerID across a + local channel. (closes issue #13368) Reported by: Peter Schlaile + Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 + (license 14) + + * apps/app_stack.c: Oops, only delete the ARG variables once upon + release. The following section would have removed them again + (removing variables from 2 stack frames, instead of just one). + +2008-10-27 16:06 +0000 [r152133] Jason Parker <jparker@digium.com> + + * apps/app_transfer.c: Remove options argument parsing/syntax (it + isn't used any longer) (closes issue #13789) Reported by: IgorG + Patches: app_transfer.c.diff uploaded by IgorG (license 20) + +2008-10-26 20:27 +0000 [r152068] Sean Bright <sean.bright@gmail.com> + + * funcs/func_strings.c: Since passing \0 as the second argument to + strchr is valid (and will match the trailing \0 of a string) we + need to check that first, otherwise we end up with incorrect + results. Fix suggested by reporter. (closes issue #13787) + Reported by: meitinger + +2008-10-25 11:11 +0000 [r151907] Russell Bryant <russell@digium.com> + + * main/asterisk.c: Move AMI initialization to occur after loading + modules. This prevents a deadlock when someone tries to + initiate a module reload from the AMI just as Asterisk is + starting. (closes issue #13778) Reported by: hotsblanc Fix + suggested by hotsblanc + +2008-10-22 20:08 +0000 [r151603] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/live_ast: Add a contributed script for running + Asterisk without installing it, first. (closes issue #11680) + Reported by: tzafrir Patches: live_ast_6 uploaded by tzafrir + (license 46) + +2008-10-22 20:05 +0000 [r151421-151602] Mark Michelson <mmichelson@digium.com> + + * channels/chan_dahdi.c: Change some logical ands to bitwise ands + and add messages alerting that a channel is being ignored if the + PROC_DAHDI_NOCHAN option is set in process_dahdi. (closes issue + #13759) Reported by: smurfix Patches: dahdi.patch uploaded by + smurfix (license 547) + + * channels/chan_sip.c: The logic of a strncasecmp call was reversed. + (closes issue #13706) Reported by: andrew53 Patches: + sip_notify_from_rfc3265.patch uploaded by andrew53 (license 519) + + * channels/chan_sip.c: Make the sip_standard_port function more + granular by allowing separate type and port arguments. This is + necessary because when building our From and Contact headers, we + need to be absolutely sure that we are placing our source port + there and not the peer's source port. (closes issue #12761) + Reported by: asbestoshead Patches: patch-chan-sip-contact-port.txt + uploaded by asbestoshead (license 455) + + * channels/chan_sip.c: Get this compiling in dev-mode + + * channels/chan_sip.c: If a peer uses any transport other than UDP, + then MWI will fail for that peer since sip_alloc will allocate a + sip_pvt with a default transport of UDP. This change resets the + socket type immediately after allocating the sip_pvt in + sip_send_mwi_from_peer, so that the proceeding call to + create_addr_from_peer does not fail right away. The socket data + from the peer is properly copied to the sip_pvt in + create_addr_from_peer. (closes issue #13710) Reported by: + andrew53 Patches: sip_notify_use_tcp.patch uploaded by andrew53 + (license 519) + + * channels/chan_sip.c: When attempting to resolve hostnames, we + need to be sure to remove any parameters from the string so that + name resolution succeeds. (closes issue #13727) Reported by: + fnordian Patches: resolvewithouturiparameter.patch uploaded by + fnordian (license 110) + +2008-10-21 15:21 +0000 [r151372] Tilghman Lesher <tlesher@digium.com> + + * apps/app_mixmonitor.c: Default file modes should always be full + read and write, to allow the system administrator to make the + decision of what permissions will actually be given, through the + use of the process umask. (Closes issue# 13751) + +2008-10-21 11:03 +0000 [r151328] BJ Weschke <bweschke@btwtech.com> + + * channels/chan_sip.c: Fix configuration parsing so type=friend + still identifies "friend" as a peer even though it is now a + legacy configuration verb. (closes issue #13705) reported by: + blitzrage patched by: bweschke + +2008-10-20 05:06 +0000 [r151135-151245] Kevin P. Fleming <kpfleming@digium.com> + + * autoconf (added), autoconf/ast_check_pwlib.m4, + autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure, + autoconf/ast_gcc_attribute.m4, bootstrap.sh, + autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4, + autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4, + autoconf/ast_c_define_check.m4, autoconf/ast_prog_egrep.m4, + autoconf/ast_ext_tool_check.m4, autoconf/ast_check_mandatory.m4, + autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4, + configure.ac, acinclude.m4 (removed), autoconf/ast_prog_sed.m4: + break up acinclude.m4 into individual files, which will make it + easier to maintain, easier to add new macros (less patching) and + will ease maintenance of these macros across Asterisk branches. + Rename this macro to properly reflect what it does + + * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c, + apps/app_externalivr.c, include/asterisk/tcptls.h: cleaup of the + TCP/TLS socket API: 1) rename 'struct server_args' to 'struct + ast_tcptls_session_args', to follow coding guidelines 2) make + ast_make_file_from_fd() static and rename it to something that + indicates what it really is for (again coding guidelines) 3) + rename address variables inside 'struct ast_tcptls_session_args' + to be more descriptive (dare i say it... coding guidelines) 4) + change ast_tcptls_client_start() to use the new 'remote_address' + field of the session args for the destination of the connection, + and use the 'local_address' field to bind() the socket to the + proper source address, if one is supplied 5) in chan_sip, ensure + that we pass in the PP address we are bound to when creating + outbound (client) connections, so that our connections will + appear from the correct address + +2008-10-18 02:29 +0000 [r150829] BJ Weschke <bweschke@btwtech.com> + + * main/manager.c: Using the GetVar handler in AMI is potentially + dangerous (insta-crash [tm]) when you use a dialplan function + that requires a channel and then you don't provide one or provide + an invalid one in the Channel: parameter. We'll handle this + situation exactly the same way it was handled in pbx.c back on + r61766. We'll create a bogus channel for the function call and + destroy it when we're done. If we have trouble allocating the + bogus channel then we're not going to try executing the function + call at all and run the risk of crashing. (closes issue #13715) + reported by: makoto patch by: bweschke + +2008-10-17 17:10 +0000 [r150606-150636] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c: Make helper call a little safer (suggested + by Russell on IRC) + + * channels/chan_iax2.c, include/asterisk/sched.h: Fix the FRACK! + warnings in chan_iax2 when POKE/LAGRQ packets are not answered. + +2008-10-16 23:41 +0000 [r150208-150306] Mark Michelson <mmichelson@digium.com> + + * main/manager.c: Reverting changes from commits 150298 and 150301 + since I was mistakenly under the assumption that dialplan + functions *always* required that a channel be present. I need to + go home earlier, I think :) + + * main/manager.c: Don't try to call a dialplan function's read + callback from the manager's GetVar handler if an invalid channel + has been specified. Several dialplan functions, including CHANNEL + and SIP_HEADER, do not check for NULL-ness of the channel being + passed in. (closes issue #13715) Reported by: makoto + And don't forget to return on the error condition + + * apps/app_sms.c: Answer the channel prior to checking for the 'a' + option in app_sms. (closes issue #13675) Reported by: alecdavis + Patches: app_sms.bug13675.148985.diff.txt uploaded by alecdavis + (license 585) + + * configure, configure.ac: Change configure script to search for + openais in both /usr/lib and /usr/lib64 since some distros place + 64-bit libraries only in the /usr/lib64 directory. (closes issue + #13721) Reported by: jcollie Patches: + 0007-Look-in-64bit-dirs-for-openais.patch uploaded by jcollie + (license 412) + + * channels/chan_sip.c: INVITES with proxy auth were sent with a + different branch than what was in the invite_branch of a sip_pvt, + meaning that if a CANCEL were sent later, the branch in the + CANCEL would not match the branch in the latest INVITE sent out, + leading to some endpoints responding to the CANCEL with a 481. + (closes issue #13714) Reported by: fnordian Patches: + invite_branch.patch uploaded by fnordian (license 110) + +2008-10-16 16:17 +0000 [r150127] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c: Fix memory leak found by customer + +2008-10-16 13:32 +0000 [r149919-149995] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: return this logic to where it used to be, + *after* the dialog->needdestroy flag has been determined to be set; + otherwise, we generate these debug messages every time we inspect + every active dialog + + * apps/app_stack.c: building this module depends on res_agi being + built as well + + * res/res_phoneprov.c: inter-module dependencies should be included + in the source code, not just in sample config files + + * res/res_phoneprov.c: correct file name in message + +2008-10-15 21:00 +0000 [r149803] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Make the sip_proxy struct reference counted. + This is necessary to allow for a sip_pvt to maintain a reference + to a sip_peer's outboundproxy even after the peer has been freed. + (closes issue #13700) Reported by: fnordian Patches: 13700.patch + uploaded by putnopvut (license 60) Tested by: fnordian + +2008-10-15 20:22 +0000 [r149758] BJ Weschke <bweschke@btwtech.com> + + * configs/agents.conf.sample: An update to the documentation/example + of agents.conf.sample with the correct parameter for this feature + as defined in chan_agent.c (closes issue #13709) + +2008-10-15 19:09 +0000 [r149589-149688] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_odbc.c: Permit data fields to contain more than 255 + characters. (closes issue #13631) Reported by: seanbright Patches: + 20081015__bug13631.diff.txt uploaded by Corydon76 (license 14) + Tested by: blitzrage + + * funcs/func_odbc.c: Only set buf to blank before the goto. + + * codecs/lpc10/lpcini.c: When using MALLOC_DEBUG, codec_lpc10 leaks + memory, because it matches a library malloc() with an ast_free + (which, of course, doesn't match up with known allocated memory, + so the free fails). (closes issue #13702) Reported by: eliel + Patches: codec_lpc10_lpcini.c uploaded by eliel (license 64) + + * apps/app_echo.c: Minor spacing change (closes issue #13697) + Reported by: alecdavis Patches: app_echo.bug13697.103249.diff.txt + uploaded by alecdavis (license 585) + +2008-10-15 11:32 +0000 [r149512] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: fix some problems when parsing SIP messages + that have the maximum number of headers or body lines that we + support + +2008-10-14 23:58 +0000 [r149203-149280] Mark Michelson <mmichelson@digium.com> + + * CHANGES, apps/app_dial.c: When specifying an invalid timeout to + Dial, take it to mean that no timeout is desired. (closes issue + #13625) Reported by: atis + + * channels/chan_sip.c: Change this warning to an error message. + Suggestion comes from Sean Bright. Thanks Sean! + + * channels/chan_sip.c: Call register_peer_exten even in the case + that the peer's IP/port does not change. (closes issue #13309) + Reported by: dimas Patches: v2-13309.patch uploaded by dimas + (license 88) + + * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance + period for sync-triggered audiohooks so that if packetization of + audio is close (but not equal) we don't end up flushing the + audiohooks over small inconsistencies in synchronization. Related + to issue #13005, and solves the issue for most people who were + experiencing the problem. However, a small number of people are + still experiencing the problem on long calls, so I am not + closing the issue yet + + * apps/app_queue.c: Update the queue with the correct number of + calls and whether the call was completed within the service level + when a transfer takes place. This way, we do not "break" the + leastrecent and fewestcalls strategies by not logging a call + until after the transferred call has ended. (closes issue #13395) + Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded + by Marquis (license 32) + +2008-10-14 22:42 +0000 [r149202] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/hashtab.h, main/chanvars.c, main/config.c, + main/hashtab.c, pbx/pbx_spool.c, channels/chan_sip.c, + include/asterisk/chanvars.h, include/asterisk/config.h, + include/asterisk/strings.h, res/res_indications.c: Add additional + memory debugging to several core APIs, and fix several memory + leaks found with these changes. (Closes issue #13505, closes + issue #13543) Reported by: mav3rick, triccyx Patches: + 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) + Tested by: mav3rick, triccyx + +2008-10-14 21:09 +0000 [r149132] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Don't allow reserved characters to be used in + register lines in sip.conf. (closes issue #13570) Reported by: + putnopvut + +2008-10-14 20:17 +0000 [r149063] Tilghman Lesher <tlesher@digium.com> + + * apps/app_waitforsilence.c: Check correct values in the return of + ast_waitfor(); also, get rid of a possible memory leak. (closes + issue #13658) Reported by: explidous Patch by: me + +2008-10-14 19:42 +0000 [r149060] Leif Madsen <lmadsen@digium.com> + + * doc/manager_1_1.txt: Add missing documentation for SipShowRegistry + action and RegistryEntry event. (closes issue #13342) Reported and + patch by: Laureano + +2008-10-14 18:59 +0000 [r148918-148986] Tilghman Lesher <tlesher@digium.com> + + * apps/app_sms.c: App is ignoring 'p' parameter -- initial pause. + (closes issue #13617) Reported by: alecdavis Patches: + app_sms.13oct.diff.txt uploaded by alecdavis (license 585) + + * apps/app_voicemail.c: Ensure that mail headers are 7-bit clean, + even when UTF-8 characters are used in headers like 'Subject' and + 'To'. Closes AST-107. + +2008-10-14 17:39 +0000 [r148915] Mark Michelson <mmichelson@digium.com> + + * channels/chan_local.c: Deadlock prevention in chan_local. (closes + issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded by + putnopvut (license 60) Tested by: tacvbo + +2008-10-14 15:18 +0000 [r148869] Tilghman Lesher <tlesher@digium.com> + + * apps/app_fax.c: API differences in spandsp 0.0.6pre1 and higher + (closes issue #13688) Reported by: irroot Patches: + app_fax-span6.patch uploaded by irroot (license 52) with minor + modifications by me + +2008-10-14 11:35 +0000 [r148614-148763] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: fix some references to the owner of a + private structure that may not be present + + * Makefile: on Ubuntu (at least), recent versions of ld in binutils + delete all debugging symbols when -x is supplied; since the + reasons why -x is being passed are lost in the mists of time, + remove it so debugging will work properly + + * channels/chan_sip.c: ensure that *all* fields in the req + structure are cleared out before reusing it; has_to_tag was not + cleared, which caused the second incoming call over a TCP socket + to fail if pedantic checking was enabled + + * main/translate.c: it would be nice if this message printing code + had actually been tested before it was committed... + +2008-10-13 17:56 +0000 [r148562] Steve Murphy <murf@digium.com> + + * main/pbx.c: Hmmm. Nobody (but me) is interested in seeing the + trie info when they do 'dialplan show ...' (even with debug set + to non-zero); so I set up a 'dialplan debug [context]' cli + command instead, to explicitly show just the trie info. I even + added an extension_exists() call to make sure the trie info is + built. I moved the explanatory header to above the extension loop + to ensure it only prints once. And it will do this now, whether + debug is set or not. I removed the trie printing from the + 'dialplan show' command entirely. + +2008-10-13 15:36 +0000 [r148472] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Sending a 403 after a 200 is considered very + bad. (found at SIPit) + +2008-10-10 21:22 +0000 [r148375-148377] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: The logic used when checking a peer got + changed subtly in the "kill the user" commit and caused calls + relying on the insecure setting to not work properly. I changed + for finding a peer back to how it was prior to that commit. + (closes issue #13644) Reported by: pj Patches: 13644_trunkv2.patch + uploaded by putnopvut (license 60) Tested by: pj + + * channels/chan_sip.c: Make sure that the inUse and inRinging + fields for a sip peer cannot go below zero. This is a + regression from 1.4 and so it will be applied to 1.6.0 as + well. (closes issue #13668) Reported by: mjc + +2008-10-10 16:37 +0000 [r148269] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: User not notified of temporary greeting, if + ODBC storage is in use. (closes issue #13659) Reported by: + moliveras Patches: 20081009__bug13659.diff.txt uploaded by + Corydon76 (license 14) Tested by: moliveras + +2008-10-10 01:33 +0000 [r148240] Sean Bright <sean.bright@gmail.com> + + * res/res_config_sqlite.c, apps/app_voicemail.c, + include/asterisk.h, main/tdd.c, main/cryptostub.c: Don't include + logger.h in asterisk.h by default as it is causing problems + building app_voicemail. Instead, include it where it is needed. + This turned out to be a relatively minor issue because other + headers include logger.h as well. Need to test -addons before + merging this back to 1.6.0. (closes issue #13605) Reported by: + tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright + (license 71) Tested by: mmichelson + +2008-10-09 23:55 +0000 [r148151-148161] Mark Michelson <mmichelson@digium.com> + + * main/manager.c: The priority was unnecessary for the manager + atxfer, so it has been removed. Furthermore, now we actually use + the Context argument passed to set the transfer context and don't + error out if no context is specified. This addresses the actual + problems outlined in issue 12158. Regarding the other points + brought up, regarding the inability to not transfer to extensions + which cannot be represented by DTMF, it is not enough of a + constraint that it is worth attempting to rework the feature. + (closes issue #12158) Reported by: davidw + + * apps/app_voicemail.c: Read the callerid in the correct order and + make sure to read the Urgent flag value from the IMAP headers. + (closes issue #13652) Reported by: jaroth Patches: + imapheaders.patch uploaded by jaroth (license 50) + +2008-10-09 23:27 +0000 [r148128] Tilghman Lesher <tlesher@digium.com> + + * configs/res_ldap.conf.sample: Fix example schema (closes issue + #12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded + by flyn (license 503) + +2008-10-09 23:20 +0000 [r148115] Mark Michelson <mmichelson@digium.com> + + * main/features.c: (closes issue #13579) Reported by: dwagner + (closes issue #13584) Reported by: dwagner Tested by: murf, + putnopvut The thought occurred to me that the res= from the + extension spawn was ending up being returned from the bridge. + "Thou shalt not poison the return value". Made the change and it + appears to allow blind xfers to work as normal. If I'm wrong, + reopen the bugs. But it looks good to me! Many thanks to + putnopvut for helping me reproduce this! + +2008-10-09 20:01 +0000 [r148006-148011] Tilghman Lesher <tlesher@digium.com> + + * sounds/Makefile, sounds/sounds.xml: Publish MOH files in sln16 + format + + * apps/app_voicemail.c: When blank, callerid name and number + should display "unknown caller" in voicemail emails. (Closes + issue #13643) + +2008-10-09 19:28 +0000 [r147957] Jeff Peeler <jpeeler@digium.com> + + * main/features.c: (closes issue #13139) Reported by: krisk84 + Tested by: krisk84 This change prevents a call that is placed in + the parkinglot to be picked up before the PBX is finished. If + another extension dials the parking extension before the PBX + thread has completed at minimum warnings will occur about the PBX + not properly being terminated. At worst, a crash could occur. + +2008-10-09 17:54 +0000 [r147901] Michiel van Baak <michiel@vanbaak.info> + + * include/asterisk/endian.h: only include this for OpenBSD. At least + FreeBSD is borked when including it (closes issue #13649) + Reported by: ys + +2008-10-09 17:47 +0000 [r147898] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample: Remove "second form" of + extensions, as it no longer applies. Also, cleanup the grammar, + formatting, and introduce several clarifications to the text. + (Closes issue #13654) + +2008-10-09 15:06 +0000 [r147811] Steve Murphy <murf@digium.com> + + * channels/chan_iax2.c, main/astobj2.c, channels/chan_oss.c, + main/config.c, main/rtp.c, main/cli.c, channels/chan_usbradio.c, + configure, channels/console_gui.c, utils/extconf.c, main/pbx.c, + include/asterisk.h, doc/CODING-GUIDELINES, + include/asterisk/autoconfig.h.in, main/translate.c, + channels/vcodecs.c, configure.ac, channels/console_video.c: + (closes issue #13557) Reported by: nickpeirson Patches: + pbx.c.patch uploaded by nickpeirson (license 579) + replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) + Tested by: nickpeirson, murf 1. replaced all refs to bzero and + bcopy to memset and memmove instead. 2. added a note to the + CODING-GUIDELINES 3. add two macros to asterisk.h to prevent + bzero, bcopy from creeping back into the source 4. removed bzero + from configure, configure.ac, autoconfig.h.in + +2008-10-08 22:33 +0000 [r147719] Mark Michelson <mmichelson@digium.com> + + * apps/app_meetme.c: Some small tweaks regarding realtime conference + announcements. (closes issue #13522) Reported by: DEA Patches: + meetme-rt-fixes.txt uploaded by DEA (license 3) + +2008-10-08 22:27 +0000 [r147692] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c: when parsing a text configuration option, + ensure that the buffer on the stack is actually large enough to + hold the legal values of that option, and also ensure that + sscanf() knows to stop parsing if it would overrun the buffer + (without these changes, specifying "buffers=...,immediate" would + overflow the buffer on the stack, and could not have worked as + expected) + +2008-10-08 19:09 +0000 [r147593] Tilghman Lesher <tlesher@digium.com> + + * apps/app_sms.c: Correct a typo in the help; also, ensure that the + date and time are correctly set, if not specified in the message. + (Closes issue #13594, closes issue #13595) Reported by: alecdavis + Patches: 20081001__bug13595.diff.txt uploaded by Corydon76 + (license 14) Tested by: alecdavis + +2008-10-08 15:10 +0000 [r147519] Mark Michelson <mmichelson@digium.com> + + * apps/app_speech_utils.c: If we receive DTMF make sure that the + state of the speech structure goes back to being not ready. + (issue #LUMENVOX-8) + +2008-10-07 16:54 +0000 [r147196] Sean Bright <sean.bright@gmail.com> + + * apps/app_voicemail.c: Make 'imapsecret' an alias to + 'imappassword' in voicemail.conf. + +2008-10-07 16:05 +0000 [r147147] Jeff Peeler <jpeeler@digium.com> + + * main/features.c: Explicitly setting these fields to NULL was done + because I wasn't sure if they would be NULL otherwise. Since they + will be set automatically, removing. + +2008-10-07 15:06 +0000 [r147100] Richard Mudgett <rmudgett@digium.com> + + * funcs/func_callerid.c: Independent change from branch issue8824 + that is not part of COLP. (-r142574 rmudgett) + +2008-10-07 12:03 +0000 [r147052] Sean Bright <sean.bright@gmail.com> + + * apps/app_dial.c: Make sure to compare the correct number of + characters when special-casing our DAHDI operator mode stuff. + Technically, it would work fine, as 'DAH' is currently unique + amongst our channel technologies, but as Jared points out: + <@jsmith> Sure... as long as the technology starts whith DAH.... + but it could be DAHDOO! + +2008-10-07 00:13 +0000 [r146972] Terry Wilson <twilson@digium.com> + + * channels/chan_sip.c: A blind transfer to the parking thread would + cause a segfault because copy_request accesses dst->data w/o + being able to tell whether it is proerly initialized + +2008-10-06 23:22 +0000 [r146930] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/threadstorage.h: Update documentation; + AST_THREADSTORAGE() in trunk only takes a single argument. + +2008-10-06 23:08 +0000 [r146876-146924] Jeff Peeler <jpeeler@digium.com> + + * include/asterisk/features.h, main/features.c, res/res_agi.c: + Similar to r143204, masquerade the channel in the case of Park + being called from AGI. ........ + + * include/asterisk/endian.h: Mvanbaak said this was needed to + compile on OpenBSD, so put it in the OpenBSD section. + + * main/features.c: This commit squashes together three commits + because the wrong approach was originally used. (One of the + commits was only one line.) 1) r143204: The main change here was + to masquerade the channel if the channel that was to be parked + was running a PBX on it. The PBX thread can then maintain full + control of the channel (the zombie) as it expects to while + allowing the parking thread full control of the real (parked) + channel. 2) r143270: Changed park_call_full to hold the + parkinglot lock a little longer, which protects the parkeduser + struct from being freed out from underneath. Made sure that the + parking extension is added to the parking context while holding + the lock thereby ensuring that there are no spurious warnings + from removal attempts when a hangup occurs while the parking lot + is being announced. 3) r143475: (the one liner) compare peer and + chan instead of looking at the parked user (pu), which could have + possibly already have been freed by the parking thread + + * main/features.c: fix some comment placement + + * main/features.c: Explicitly set args in park_call_exec NULL so in + the case of no options being passed in, there is no garbage + attempted to be used. Also, do not set args to unknown value + again if there are no options passed in. + +2008-10-06 21:53 +0000 [r146874] Michiel van Baak <michiel@vanbaak.info> + + * include/asterisk/endian.h: make aescrypt.c compile on OpenBSD again + +2008-10-06 21:32 +0000 [r146715-146838] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c, funcs/func_callerid.c, + apps/app_speech_utils.c, funcs/func_curl.c, + funcs/func_groupcount.c, res/res_smdi.c, channels/chan_sip.c, + funcs/func_timeout.c, funcs/func_odbc.c, funcs/func_cdr.c, + funcs/func_math.c: Dialplan functions should not actually return + 0, unless they have modified the workspace. To signal an error + (and no change to the workspace), -1 should be returned instead. + (closes issue #13340) Reported by: kryptolus Patches: + 20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14) + + * channels/chan_local.c: Check whether an extension exists in the + _call method, rather than the _alloc method, because we need to + evaluate the callerid (since that data affects whether an + extension exists). (closes issue #13343) Reported by: efutch + Patches: 20080915__bug13343.diff.txt uploaded by Corydon76 + (license 14) Tested by: efutch + +2008-10-06 16:39 +0000 [r146698] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c: ensure that the private structure for + pseudo channels is created without 'leaking' configuration data + from other configured channels (closes issue #13555) Reported by: + jeffg Patches: issue_13555.patch uploaded by kpfleming (license + 421) Tested by: jeffg + +2008-10-06 00:23 +0000 [r146557] Sean Bright <sean.bright@gmail.com> + + * utils/Makefile: Quote arguments to cp so we can handle spaces in + our paths. + +2008-10-05 21:24 +0000 [r146451] Jason Parker <jparker@digium.com> + + * channels/chan_sip.c: Fix silly formatting. + +2008-10-04 01:57 +0000 [r146314] Sean Bright <sean.bright@gmail.com> + + * configs/sip_notify.conf.sample: Add ability to remotely reboot + snom phones. Also cleaned up and reorganized + sip_notify.conf.sample a bit as well. Tested snom reboot on snom + 360 and verified snom-check-cfg worked as well. (closes issue + #13601) Reported by: mjc Tested by: seanbright + +2008-10-03 22:42 +0000 [r146243] Jeff Peeler <jpeeler@digium.com> + + * main/features.c: remove superfluous reference counting operations + in manage_parkinglot since ao2_interator_next increments the ref + count automatically + +2008-10-03 22:13 +0000 [r146200] Sean Bright <sean.bright@gmail.com> + + * main/cli.c: Resolve a subtle bug where we would never + successfully be able to get the first item in the CLI entry list. + This was preventing '!' from showing up in either 'help' or in tab + completion. (closes issue #13578) Reported by: mvanbaak + +2008-10-02 19:31 +0000 [r145960-145964] Russell Bryant <russell@digium.com> + + * CHANGES: The 'P' command for ExternalIVR was also added in 1.6.0 + + * CHANGES: 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: This is much cleaner, methinks. + +2008-10-02 15:19 +0000 [r145754] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c: 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: 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: 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: if (!(x) == 0) is the same as if (x). + + * main/features.c: 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 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0 released. + +2008-09-09 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-rc6 released. + +2008-09-09 15:44 +0000 [r142065] 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 [r141996-141999] Mark Michelson <mmichelson@digium.com> + + * 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 01:49 +0000 [r141950] Russell Bryant <russell@digium.com> + + * 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 21:07 +0000 [r141808] 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 ........ ................ + +2008-09-08 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-rc5 released. + +2008-09-08 20:19 +0000 [r141746] 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 17:14 +0000 [r141683] 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 20:21 +0000 [r141567] 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 15:40 +0000 [r141505-141508] 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 22:06 +0000 [r141368-141426] 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 14:24 +0000 [r141116-141158] 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 ........ + ................ + + * 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, res/ael/ael.flex: 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 18:35 +0000 [r141086] 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 [r140976] 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. ........ + +2008-09-03 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-rc4 released. + +2008-09-03 14:17 +0000 [r140825-140827] 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. ........ ................ + +2008-09-03 13:27 +0000 [r140818] 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 18:17 +0000 [r140607] 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 15:12 +0000 [r140564-140567] 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 ........ + +2008-08-27 20:15 +0000 [r140302-140304] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Revert commit 140302. Should not be merging + changes like that into a release-candidate branch + + * 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 18:12 +0000 [r140170] 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. + ........ + +2008-08-25 21:33 +0000 [r139918] 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 21:05 +0000 [r139872] 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 16:00 +0000 [r139774] 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 16:30 +0000 [r139705-139708] 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 22:35 +0000 [r139628-139671] 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 20:21 +0000 [r139458-139564] Mark Michelson <mmichelson@digium.com> + + * 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 23:46 +0000 [r139400] 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) ........ ................ + +2008-08-21 10:02 +0000 [r139282] Philippe Sultan <philippe.sultan@gmail.com> + + * /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008) + | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel! + (closes issue #13310) Reported by: eliel Patches: + chan_gtalk.c.patch uploaded by eliel (license 64) ........ + +2008-08-20 Kevin P. Fleming <kpfleming@digium.com> + + * Asterisk 1.6.0-rc3 released. + +2008-08-20 22:17 +0000 [r139216] 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 20:12 +0000 [r139155] Shaun Ruffell <sruffell@digium.com> + + * codecs/codec_dahdi.c: Fix bug where the samples were not accurate + when in G723 mode, which would cause the timestamp field of the + RTP header to be invalid. + +2008-08-20 17:30 +0000 [r139104] 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 [r138889-139017] 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 + ........ ................ + + * apps/app_chanspy.c: Manually add revision 138887 from trunk to + the 1.6.0 branch. I had misunderstood the policy for when to + merge to 1.6.0 since it moved to rc status. + +2008-08-19 16:38 +0000 [r138846-138847] 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 00:15 +0000 [r138776-138781] 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 20:23 +0000 [r138688-138695] Mark Michelson <mmichelson@digium.com> + + * /, 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 15:54 +0000 [r138632] 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 [r138519] 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 14:14 +0000 [r138443-138483] 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 13:07 +0000 [r138410-138413] 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) ........ + +2008-08-16 01:14 +0000 [r138333-138362] 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 23:03 +0000 [r138207-138262] 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 ........ + ................ + + * configs/extensions.conf.sample, main/pbx.c, /: Merged revisions + 138206 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 | + tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines + Remove deprecated syntax from sample config file (closes issue + #13314) Reported by: kue ........ + +2008-08-15 20:20 +0000 [r138156-138157] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to + dfd to match 1.4 (left over from DAHDI transition) + +2008-08-15 15:12 +0000 [r138029] 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 15:04 +0000 [r138025] Tilghman Lesher <tlesher@digium.com> + + * /, 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 [r137988] 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 18:48 +0000 [r137934] 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 17:01 +0000 [r137849-137852] 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 Jason Parker <jparker@digium.com> + + * Asterisk 1.6.0-rc2 released. + +2008-08-14 15:37 +0000 [r137814] 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 15:20 +0000 [r137783] 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 15:05 +0000 [r137781] 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 02:04 +0000 [r137681] 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 ........ + ................ + +2008-08-13 Kevin P. Fleming <kpfleming@digium.com> + + * Asterisk 1.6.0-rc1 released. + +2008-08-13 23:00 +0000 [r137631-137641] Kevin P. Fleming <kpfleming@digium.com> + + * /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug + 2008) | 1 line make this script actually work ........ + + * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions + 137627 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug + 2008) | 9 lines 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 21:09 +0000 [r137497-137533] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 | + qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines + 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: Merged revisions 137496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 | + qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines + Add FAXMODE variable with what fax transport was used. (closes + issue #13252) Patches: v1-13252.patch uploaded by dimas (license + 88) ........ + +2008-08-13 14:47 +0000 [r137350-137407] Sean Bright <sean.bright@gmail.com> + + * /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400 + (Wed, 13 Aug 2008) | 9 lines 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, /: Merged revisions 137403 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 | + seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1 + line Use the ast_vasprintf macro instead of vasprintf directly. + ........ + +2008-08-12 19:48 +0000 [r137300-137302] Russell Bryant <russell@digium.com> + + * doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008) + | 2 lines Grammar hax from Qwell ........ + + * doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008) + | 3 lines Note that developer documentation belongs in doxygen, + and not integrated with the user manual stuff in doc/tex/. + ........ + +2008-08-11 16:15 +0000 [r137240] Russell Bryant <russell@digium.com> + + * Makefile, /: Merged revisions 137239 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 | + russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines + Make PRINT_DIR work as advertised. ........ + +2008-08-11 14:31 +0000 [r137217] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon, + 11 Aug 2008) | 7 lines 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 00:27 +0000 [r137160] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c, /: Merged revisions 137150 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008) + | 13 lines 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 ........ ................ + +2008-08-09 15:27 +0000 [r136948] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged + revisions 136947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008) + | 18 lines 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 01:16 +0000 [r136860] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 136859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 | + tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines + 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 15:33 +0000 [r136785] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug + 2008) | 3 lines Fix compilation for ODBC voicemail ........ + +2008-08-08 06:45 +0000 [r136778] Steve Murphy <murf@digium.com> + + * 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, res/ael/pval.c, /, + pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h, + utils/ael_main.c: Merged revisions 136746 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) | + 40 lines 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. ........ ................ + +2008-08-08 02:36 +0000 [r136753] Tilghman Lesher <tlesher@digium.com> + + * /: Merged revisions 136751 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 | + tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines + Removing bad properties ........ + +2008-08-07 23:42 +0000 [r136719-136724] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a + bunch of functions over one level during a merge. + + * apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug + 2008) | 3 lines Remove one last batch of debug messages ........ + + * apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug + 2008) | 18 lines 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 20:41 +0000 [r136672-136674] Shaun Ruffell <sruffell@digium.com> + + * codecs/codec_dahdi.c: Removing code that was commented out. + + * codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder + interface in the DAHDI. (Issue: DAHDI-42) + +2008-08-07 20:26 +0000 [r136632-136663] Mark Michelson <mmichelson@digium.com> + + * /, main/features.c: Merged revisions 136660 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 | + mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4 + lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears + once for every bridged call ........ + + * main/pbx.c, /: Merged revisions 136635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 | + mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5 + lines Don't allow Answer() to accept a negative argument. + Negative argument means an infinite delay and we don't want that. + ........ + + * main/channel.c, /: Merged revisions 136633 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 | + mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7 + lines 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, /: Merged revisions 136631 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 | + mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13 + lines 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 19:19 +0000 [r136598] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn_config.c, channels/chan_misdn.c, /, + configs/misdn.conf.sample: Merged revisions 136594 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500 + (Thu, 07 Aug 2008) | 13 lines 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 [r136506-136543] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/doxyref.h, /: Merged revisions 136542 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500 + (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + ........ ................ + +2008-08-07 16:57 +0000 [r136490] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_queue.c: Merged revisions 136489 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008) + | 15 lines 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 + ........ ................ + +2008-08-06 20:16 +0000 [r136113-136192] Tilghman Lesher <tlesher@digium.com> + + * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500 + (Wed, 06 Aug 2008) | 12 lines 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 ........ + ................ + + * /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008) + | 7 lines 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 16:00 +0000 [r136064] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500 + (Wed, 06 Aug 2008) | 24 lines 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 13:59 +0000 [r136006] Olle Johansson <oej@edvina.net> + + * /, res/res_jabber.c: Merged revisions 136005 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 | + oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines - + 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 03:56 +0000 [r135901-135951] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 135950 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008) + | 12 lines 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 135938 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008) + | 12 lines 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/threadstorage.h, include/asterisk/utils.h, /: + Merged revisions 135900 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008) + | 12 lines 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 00:31 +0000 [r135852] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/abstract_jb.h, main/channel.c, /, + main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions + 135851 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug + 2008) | 48 lines 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 23:52 +0000 [r135822] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c, + include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | + 42 lines 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 21:38 +0000 [r135749] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500 + (Tue, 05 Aug 2008) | 17 lines 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). ........ ................ + +2008-08-05 13:27 +0000 [r135599] Sean Bright <sean.bright@gmail.com> + + * main/cli.c, /: Merged revisions 135598 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug + 2008) | 9 lines 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 [r135538] Russell Bryant <russell@digium.com> + + * configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135537 | russell | 2008-08-04 15:15:27 -0500 + (Mon, 04 Aug 2008) | 10 lines 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 [r135478-135486] 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): Merged revisions + 135485 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 | + tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines + Rename Mandrake scripts to Mandriva (Closes issue #13221) + ........ + + * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500 + (Mon, 04 Aug 2008) | 11 lines 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 135480 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500 + (Mon, 04 Aug 2008) | 14 lines 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) ........ ................ + +2008-08-04 16:28 +0000 [r135440-135475] Russell Bryant <russell@digium.com> + + * configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135474 | russell | 2008-08-04 11:28:07 -0500 + (Mon, 04 Aug 2008) | 10 lines 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: Merged revisions 135439 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008) + | 4 lines 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. ........ + +2008-08-02 05:15 +0000 [r135266] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 135265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 | + murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines + (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. in 1.6.0, the changes to only + main/pbx.c were applicable, as apparently the code added to + main/features by jpeeler were not included in 1.6.0. ........ + +2008-08-01 19:30 +0000 [r135198] Sean Bright <sean.bright@gmail.com> + + * channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug + 2008) | 6 lines 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 17:10 +0000 [r135127-135129] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 | + tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines + Picky, picky, buildbot ........ + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 135126 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 | + tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines + 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 14:43 +0000 [r135070] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged + revisions 135067-135068 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 | + mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13 + lines 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) ........ + r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug + 2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE + defines... ........ + +2008-08-01 12:18 +0000 [r135057-135062] Michiel van Baak <michiel@vanbaak.info> + + * /, apps/app_ices.c: Merged revisions 135059 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008) + | 10 lines 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 135056 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200 + (Fri, 01 Aug 2008) | 16 lines 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 22:34 +0000 [r135034] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/http.c: Merged revisions 135016 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul + 2008) | 11 lines 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 21:58 +0000 [r134926-134981] Tilghman Lesher <tlesher@digium.com> + + * sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions + 134980 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008) + | 16 lines Blocked revisions 134976 via svnmerge ........ r134976 + | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9 + lines Specify codecs in callfiles and manager, to allow video + calls to be set up from callfiles and AMI. (closes issue #9531) + Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt + uploaded by Corydon76 (license 14) + 20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license + 14) Tested by: Corydon76 ........ ................ + + * res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008) + | 2 lines Switch command order, to meet with current specs + ........ + +2008-07-31 19:54 +0000 [r134923] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 134922 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) | + 63 lines 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. + ........ ................ + +2008-07-31 19:41 +0000 [r134918] Russell Bryant <russell@digium.com> + + * /, apps/app_ices.c: Merged revisions 134917 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008) + | 17 lines 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 ........ ................ + +2008-07-31 16:53 +0000 [r134816] Russell Bryant <russell@digium.com> + + * channels/iax2-parser.c: Merged revisions 134815 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008) + | 15 lines 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 ........ ................ + +2008-07-31 16:07 +0000 [r134760] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 134759 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul + 2008) | 24 lines 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 22:41 +0000 [r134651-134707] Tilghman Lesher <tlesher@digium.com> + + * main/sched.c, /, include/asterisk/sched.h: Merged revisions + 134703 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 | + tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines + Oops, wrong define ........ + + * /, configure, configure.ac: Merged revisions 134650 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500 + (Wed, 30 Jul 2008) | 12 lines 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 21:06 +0000 [r134599] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 134598 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul + 2008) | 15 lines 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 20:39 +0000 [r134597] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008) + | 14 lines 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 20:25 +0000 [r134561] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.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 19:56 +0000 [r134542] Russell Bryant <russell@digium.com> + + * funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008) + | 12 lines 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 19:49 +0000 [r134482-134539] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 134538 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500 + (Wed, 30 Jul 2008) | 12 lines 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: Merged revisions 134483 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 | + tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines + 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 134481 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008) + | 13 lines 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 [r134477] Mark Michelson <mmichelson@digium.com> + + * /, main/app.c: Merged revisions 134476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul + 2008) | 12 lines 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 15:34 +0000 [r134356] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /: Merged revisions 134355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul + 2008) | 10 lines 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 + ........ ................ + +2008-07-29 22:29 +0000 [r134283] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /, + apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c: + Merged revisions 134260 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 | + kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2 + lines build against the now-typedef-free dahdi/user.h, and remove + some #ifdefs for features that will always be present in DAHDI + ........ + +2008-07-28 22:16 +0000 [r134164] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500 + (Mon, 28 Jul 2008) | 15 lines 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 ........ + ................ + +2008-07-28 19:55 +0000 [r134126] Mark Michelson <mmichelson@digium.com> + + * /, configure, main/Makefile, configure.ac, CHANGES: Merged + revisions 134125 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 | + mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27 + lines 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 [r134087] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_parkandannounce.c, main/loader.c, sample.call, + contrib/scripts/autosupport, build_tools/cflags.xml, + main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c, + doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions + 134086 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 | + kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3 + lines remove remaining Zaptel references in various places + ........ + +2008-07-28 16:13 +0000 [r134052] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, + /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions + 134050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 | + mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 + lines merging the zap_and_dahdi_trunk branch up to trunk ........ + +2008-07-26 15:34 +0000 [r133942-133982] Russell Bryant <russell@digium.com> + + * main/asterisk.c, include/asterisk/doxyref.h, /: Include the + licensing page in 1.6.0 as well. Now, this page exists in 1.4, + trunk, and 1.6.0. + + * /: unblock 133575 + + * /, main/devicestate.c: Merged revisions 133945-133946 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26 + Jul 2008) | 6 lines 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. ........ r133946 | russell | 2008-07-26 10:16:20 + -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache + argument ........ + +2008-07-25 22:09 +0000 [r133863-133905] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25 + Jul 2008) | 6 lines Update version (closes issue #13163) Reported + by: suretec Patches: asterisk.ldif uploaded by suretec (license + 70) ........ + +2008-07-25 19:37 +0000 [r133804-133806] Brandon Kruse <bkruse@digium.com> + + * /: Blocking revert of code changes in r133770 + + * main/http.c: Include the http_decode function from trunk to + replace the + with a space. + +2008-07-25 17:33 +0000 [r133694] Brandon Kruse <bkruse@digium.com> + + * /: Blocking a fix from trunk for the function http_decode. 1.6.0 + does not have this function. + +2008-07-25 17:26 +0000 [r133671] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /, channels/chan_agent.c, main/devicestate.c: + Merged revisions 133665 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008) + | 16 lines 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 15:01 +0000 [r133576-133580] Russell Bryant <russell@digium.com> + + * /, LICENSE: Merged revisions 133579 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008) + | 18 lines 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 ........ + ................ ................ + +2008-07-25 14:41 +0000 [r133571-133574] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul + 2008) | 15 lines 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. ........ + ................ + +2008-07-25 13:25 +0000 [r133567-133569] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 | + russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines + Minor coding guidelines tweaks ... - Use ast_strlen_zero in one + place - check for successful string comparison the way most of + Asterisk code does it ........ + +2008-07-24 21:31 +0000 [r133524] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008) + | 11 lines 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 20:41 +0000 [r133487] Russell Bryant <russell@digium.com> + + * /, channels/chan_agent.c: Merged revisions 133486 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008) + | 3 lines 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 19:55 +0000 [r133449] Mark Michelson <mmichelson@digium.com> + + * /, main/logger.c: Merged revisions 133448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 | + mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12 + lines 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 [r133392-133405] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/asterisk.logrotate, Makefile, /: Merged revisions + 133400 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133400 | + tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines + Build the logrotate script according to paths (Closes issue + #13147) ........ + + * Makefile, /: Merged revisions 133391 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133391 | + tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines + Optionally install logrotate file (Closes issue #13148) ........ + +2008-07-23 22:07 +0000 [r133300] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 133299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 | + murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines + (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 [r133297] Jason Parker <jparker@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 133296 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r133296 | qwell | 2008-07-23 16:50:20 -0500 + (Wed, 23 Jul 2008) | 9 lines 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 20:39 +0000 [r133218] Brett Bryant <bbryant@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 | + bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines + 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 19:50 +0000 [r133042-133172] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, + /: Merged revisions 133171 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul + 2008) | 20 lines 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 133106 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul + 2008) | 13 lines 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 133102 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul + 2008) | 14 lines 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 133041 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r133041 | mmichelson | 2008-07-23 12:54:03 -0500 + (Wed, 23 Jul 2008) | 15 lines 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 17:21 +0000 [r132978-132983] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 132981 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008) + | 6 lines 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), /: Merged revisions + 132977 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132977 | + tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines + Add logrotate script for Asterisk (closes issue #13085) Reported + by: pabelanger Patches: logrotate uploaded by pabelanger (license + 224) ........ + +2008-07-23 16:42 +0000 [r132965-132967] Kevin P. Fleming <kpfleming@digium.com> + + * channels/misdn/isdn_lib.c, /: Merged revisions 132883,132966 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r132883 | crichter | 2008-07-23 07:07:15 -0500 + (Wed, 23 Jul 2008) | 9 lines 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. ........ ................ + r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul + 2008) | 2 lines use correct function name... please compile with + --enable-dev-mode ................ + + * include/asterisk/stringfields.h, /, main/utils.c: Merged + revisions 132964 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul + 2008) | 10 lines 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 08:18 +0000 [r132824] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 132823 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 | + oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines + 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 22:20 +0000 [r132797] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 132795 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul + 2008) | 11 lines 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 22:15 +0000 [r132793] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132791 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul + 2008) | 2 lines correct fix made in r132777... the code *did* + compile in dev-mode, as long as libpri was installed and enabled + ........ + +2008-07-22 21:59 +0000 [r132782] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged + revisions 132703 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 + lines 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 21:55 +0000 [r132780] Tilghman Lesher <tlesher@digium.com> + + * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged + revisions 132778 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008) + | 18 lines 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 21:54 +0000 [r132779] Mark Michelson <mmichelson@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132777 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul + 2008) | 3 lines Get chan_dahdi to compile in devmode ........ + +2008-07-22 21:23 +0000 [r132574-132729] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132721 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r132721 | kpfleming | 2008-07-22 16:21:56 -0500 + (Tue, 22 Jul 2008) | 14 lines 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 132705 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r132705 | kpfleming | 2008-07-22 15:54:07 -0500 + (Tue, 22 Jul 2008) | 10 lines 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 ........ ................ + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, + configure, include/asterisk/autoconfig.h.in, configure.ac: Merged + revisions 132643 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul + 2008) | 10 lines 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 132573 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r132573 | kpfleming | 2008-07-21 17:51:16 -0500 + (Mon, 21 Jul 2008) | 10 lines 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 21:13 +0000 [r132515] Brett Bryant <bbryant@digium.com> + + * configs/features.conf.sample, configs/gtalk.conf.sample, /, + configs/jingle.conf.sample, configs/manager.conf.sample: Merged + revisions 132514 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132514 | + bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines + 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 21:02 +0000 [r132512-132513] Tilghman Lesher <tlesher@digium.com> + + * main/fskmodem.c (added), /, include/asterisk/fskmodem.h (added): + Merged revisions 132511 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 | + tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines + (Step 2 of 2) ........ + + * main/fskmodem.c (removed), 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): Merged revisions + 132510 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 | + tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines + 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 20:55 +0000 [r132467-132509] Brett Bryant <bbryant@digium.com> + + * /, apps/app_sendtext.c: Merged revisions 132508 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132508 | + bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines + 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_sip.c: Merged revisions 132468 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 | + bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines + 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: Merged revisions 132466 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008) + | 3 lines 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 15:34 +0000 [r132426] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132425 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008) + | 2 lines make buffers config option (chan_dahdi.conf) parsing + safer and added logging in case of failure ........ + +2008-07-21 14:48 +0000 [r132389-132391] Russell Bryant <russell@digium.com> + + * apps/app_jack.c, 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: + Merged revisions 132390 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 | + russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines + 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. ........ + + * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions + 132388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 | + russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines + Enable higher quality resampling, as it doesn't have a noticeable + performance impact on my machine .. ........ + +2008-07-19 16:47 +0000 [r132313] Kevin P. Fleming <kpfleming@digium.com> + + * /, LICENSE: Merged revisions 132312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul + 2008) | 10 lines 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 10:47 +0000 [r132278] Michiel van Baak <michiel@vanbaak.info> + + * res/res_config_sqlite.c, /: Merged revisions 132277 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008) + | 7 lines 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 22:20 +0000 [r132245] Brett Bryant <bbryant@digium.com> + + * main/manager.c, /: Merged revisions 132242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 | + bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines + 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 20:58 +0000 [r132114-132207] Tilghman Lesher <tlesher@digium.com> + + * /, main/say.c: Merged revisions 132113 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008) + | 14 lines 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) ........ ................ + +2008-07-18 18:53 +0000 [r132111] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132108 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) | + 1 line 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 18:51 +0000 [r132110] Tilghman Lesher <tlesher@digium.com> + + * main/config.c, /: Merged revisions 132109 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008) + | 14 lines 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 17:56 +0000 [r132051] Brett Bryant <bbryant@digium.com> + + * main/hashtab.c, /, cdr/cdr_radius.c: Merged revisions 132050 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18 + Jul 2008) | 8 lines 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 17:40 +0000 [r131984-132047] Tilghman Lesher <tlesher@digium.com> + + * main/sched.c, /: Merged revisions 131989 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008) + | 10 lines 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 ........ ................ + + * main/sched.c, /, include/asterisk/sched.h: Merged revisions + 131986 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008) + | 10 lines 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 ........ + ................ + + * main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c: + Merged revisions 131982 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008) + | 10 lines 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 [r131924] Kevin P. Fleming <kpfleming@digium.com> + + * main/dlfcn.c (removed), main/loader.c, /, main/Makefile, + include/asterisk/dlfcn-compat.h (removed): Merged revisions + 131923 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul + 2008) | 10 lines 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 15:39 +0000 [r131917] Brett Bryant <bbryant@digium.com> + + * /, main/features.c: Merged revisions 131916 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131916 | bbryant | 2008-07-18 10:38:22 -0500 (Fri, 18 Jul 2008) + | 12 lines 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 22:45 +0000 [r131869] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 131868 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008) + | 6 lines 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 21:27 +0000 [r131830] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_senddtmf.c: Merged revisions 131824 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r131824 | + mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10 + lines 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 20:38 +0000 [r131754-131792] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 131791 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r131791 | tilghman | 2008-07-17 15:37:14 -0500 + (Thu, 17 Jul 2008) | 15 lines 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, /: Merged revisions 131753 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r131753 | tilghman | 2008-07-17 13:36:34 -0500 (Thu, 17 Jul 2008) + | 6 lines 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 18:15 +0000 [r131718] Brett Bryant <bbryant@digium.com> + + * /, main/features.c: Merged revisions 131717 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r131717 | + bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines + 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 15:46 +0000 [r131682] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_sqlite.c, /: Merged revisions 131681 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r131681 | tilghman | 2008-07-17 10:45:25 -0500 (Thu, 17 Jul 2008) + | 4 lines Fix memory leak. (Closes issue #13096) Reported by + gknispel_proformatique ........ + +2008-07-16 23:56 +0000 [r131571] Steve Murphy <murf@digium.com> + + * /: The commit from 131570 should not be applied to 1.6.0, as it + is not as necessary, because log_show_lock in trunk is not + available in 1.6.0, and is estimated to be the only function that + might care about the lock_addr values. + +2008-07-16 22:18 +0000 [r131493] Brett Bryant <bbryant@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 131492 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r131492 | bbryant | 2008-07-16 17:17:36 -0500 + (Wed, 16 Jul 2008) | 14 lines 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) ........ + ................ + +2008-07-16 21:54 +0000 [r131455-131486] Brett Bryant <bbryant@digium.com> + + * /, funcs/func_sysinfo.c: Merged revisions 131484 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r131484 | bbryant | 2008-07-16 16:54:08 -0500 (Wed, 16 Jul 2008) + | 4 lines Fixes sysinfo operator issue also fixed elsewhere in + r131445. (issue #13057) ........ + + * main/asterisk.c, /: Merged revisions 131445 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r131445 | + bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines + 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 [r131423] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 131422 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r131422 | russell | 2008-07-16 15:48:27 -0500 + (Wed, 16 Jul 2008) | 15 lines 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 [r131301-131378] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 131375 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul + 2008) | 22 lines 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, /: Merged revisions 131361 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r131361 | mmichelson | 2008-07-16 14:57:02 -0500 (Wed, 16 Jul + 2008) | 9 lines 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 131358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131358 | mmichelson | 2008-07-16 14:37:42 -0500 (Wed, 16 Jul + 2008) | 14 lines 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 131300 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul + 2008) | 21 lines 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 18:20 +0000 [r131248] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, /: Merged revisions 131243 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) | + 27 lines 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 15:04 +0000 [r131206] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_agent.c: add missing terminator argument for + ast_event_subscribe(). Without it the function will randomly walk + on the stack possibly causing a panic + +2008-07-16 00:54 +0000 [r131168] Tilghman Lesher <tlesher@digium.com> + + * /, main/logger.c: Merged revisions 131166 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r131166 | + tilghman | 2008-07-15 19:52:48 -0500 (Tue, 15 Jul 2008) | 3 lines + Fix rotate strategy (Closes issue #13086) ........ + +2008-07-15 23:41 +0000 [r131131] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 131129 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r131129 | + murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines + (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 18:47 +0000 [r131073] Russell Bryant <russell@digium.com> + + * /, res/res_agi.c: Merged revisions 131072 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r131072 | + russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines + 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 18:29 +0000 [r131060] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, main/manager.c, /, channels/chan_sip.c: Merged + revisions 131044 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008) + | 16 lines 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 18:00 +0000 [r131014] Michiel van Baak <michiel@vanbaak.info> + + * main/cdr.c, /: Merged revisions 131013 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r131013 | mvanbaak | 2008-07-15 19:49:48 +0200 (Tue, 15 Jul 2008) + | 15 lines 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 13:14 +0000 [r130946] Steve Murphy <murf@digium.com> + + * 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, res/ael/ael.tab.h, + utils/ael_main.c, include/asterisk/pbx.h, utils/extconf.c, + res/ael/ael.flex, pbx/pbx_config.c, apps/app_stack.c, + apps/app_dial.c, main/pbx.c, include/asterisk/pval.h, /, + channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y, + channels/chan_iax2.c, apps/app_queue.c: Merged revisions 130145 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + Merging this rev from trunk to 1.6.0 was not simple. Why? Because + we've enhanced trunk to do a [fast] merge-and-delete operation + which also solved problems with contexts having entries from + different registrars. Fast as in the amount of time the contexts + are locked down. That *is* fast, but traversing the entire + dialplan looking for priorities to delete takes more time + overall. This particular fix involved pulling in those + enhancements from trunk, along with all the various fixes and + refinements made along the way. Merging all this from trunk into + 1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert + all but the prop changes c. catalog all revisions to pbx.c since + 1.6.0 was forked (at rev 105596). d. catalog all revisions to + pbx.c in trunk since 1.6.0 was forked, making special note of all + revs that were not merged into 1.6.0. e. study each rev in trunk + not applied to 1.6.0, and determine if it was involved in the + merge_and_delete enhancements in trunk. 25 commits were done in + 1.6.0, all but one (106306) was a merge from trunk. Trunk had 22 + additional changes, of which 7 were involved in the + merge_and_delete enhancements: 106757 108894 109169 116461 123358 + 130145 130297 f. Go to trunk and collect patches, one by one, of + the changes made by each rev across the entire source tree, using + svn diff -c <num> > pfile g. Apply each patch in order to 1.6.0, + and resolve all failures and compilation problems before + proceding to the next patch. h. test the stuff. i. profit! + ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul + 2008) | 40 lines (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. ........ + +2008-07-15 00:00 +0000 [r130891] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 130890 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r130890 | tilghman | 2008-07-14 18:59:54 -0500 + (Mon, 14 Jul 2008) | 16 lines 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 22:24 +0000 [r130795-130855] Mark Michelson <mmichelson@digium.com> + + * main/asterisk.c, /: Merged revisions 130854 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r130854 | + mmichelson | 2008-07-14 17:22:57 -0500 (Mon, 14 Jul 2008) | 9 + lines 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 130794 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul + 2008) | 16 lines 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 17:22 +0000 [r130752] Michiel van Baak <michiel@vanbaak.info> + + * main/dnsmgr.c, /: Merged revisions 130744 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r130744 | mvanbaak | 2008-07-14 19:21:18 +0200 (Mon, 14 Jul 2008) + | 18 lines 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 10:40 +0000 [r130636-130637] Russell Bryant <russell@digium.com> + + * /, include/asterisk/astobj.h: Merged revisions 129987 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r129987 | russell | 2008-07-11 09:22:44 -0500 + (Fri, 11 Jul 2008) | 10 lines 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 ... ........ + ................ + + * /, main/audiohook.c: Merged revisions 130635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008) + | 10 lines 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 23:20 +0000 [r130575-130582] Michiel van Baak <michiel@vanbaak.info> + + * /, doc/tex/Makefile, build_tools/prep_tarball, res/Makefile: + Merged revisions 130578 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r130578 | + mvanbaak | 2008-07-14 01:14:03 +0200 (Mon, 14 Jul 2008) | 15 + lines 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. ........ + + * main/manager.c, /: Merged revisions 130574 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r130574 | mvanbaak | 2008-07-14 00:50:31 +0200 (Mon, 14 Jul 2008) + | 16 lines 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 17:59 +0000 [r130516] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 130515 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r130515 | tilghman | 2008-07-13 12:58:47 -0500 + (Sun, 13 Jul 2008) | 12 lines 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 15:00 +0000 [r130480] Michiel van Baak <michiel@vanbaak.info> + + * doc/tex/asterisk.tex, /: Merged revisions 130479 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r130479 | mvanbaak | 2008-07-13 16:58:40 +0200 (Sun, 13 Jul 2008) + | 3 lines restore ASTERISKVERSION marker to asterisk.tex. This + got lost in commit 97634 ........ + +2008-07-13 02:35 +0000 [r130445] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_agent.c: Merged revisions 130444 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r130444 | tilghman | 2008-07-12 21:34:32 -0500 (Sat, 12 Jul 2008) + | 2 lines Unlock list before returning ........ + +2008-07-11 21:39 +0000 [r130294] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 130293 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r130293 | mattf | 2008-07-11 16:36:26 -0500 (Fri, 11 Jul 2008) | + 1 line Support new TRANSPORT definitions in libss7 ........ + +2008-07-11 20:04 +0000 [r130238] Mark Michelson <mmichelson@digium.com> + + * /, main/audiohook.c: Merged revisions 130237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul + 2008) | 11 lines 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 19:57 +0000 [r130231-130235] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /, channels/chan_agent.c, utils/astman.c: + Merged revisions 130230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r130230 | + tilghman | 2008-07-11 14:40:55 -0500 (Fri, 11 Jul 2008) | 2 lines + Fix trunk breakage ........ + +2008-07-11 19:14 +0000 [r130175] Mark Michelson <mmichelson@digium.com> + + * /, main/audiohook.c: Merged revisions 130174 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul + 2008) | 15 lines 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 18:53 +0000 [r130171] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 130170 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r130170 | tilghman | 2008-07-11 13:52:42 -0500 + (Fri, 11 Jul 2008) | 15 lines 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 18:33 +0000 [r130168] Sean Bright <sean.bright@gmail.com> + + * /, channels/chan_sip.c: Merged revisions 130167 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r130167 | + seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1 + line Missed one. Formatting only. ........ + +2008-07-11 18:14 +0000 [r130130] Brett Bryant <bbryant@digium.com> + + * 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, main/asterisk.c, apps/app_milliwatt.c, + main/dsp.c, 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: Merged revisions + 130129 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 | + bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines + 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 17:30 +0000 [r130127] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_agent.c: Merged revisions 130126 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r130126 | tilghman | 2008-07-11 12:29:24 -0500 + (Fri, 11 Jul 2008) | 17 lines 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 [r129936-130045] Kevin P. Fleming <kpfleming@digium.com> + + * doc/ss7.txt, /, contrib/utils/zones2indications.c, CHANGES: + Merged revisions 130044 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r130044 | + kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2 + lines 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 130040 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul + 2008) | 12 lines 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) ........ ................ + + * /, main/astmm.c: Merged revisions 129968 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r129968 | kpfleming | 2008-07-11 09:16:15 -0500 (Fri, 11 Jul + 2008) | 18 lines 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 129916 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r129916 | kpfleming | 2008-07-11 07:21:29 -0500 (Fri, 11 Jul + 2008) | 10 lines 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 01:01 +0000 [r129865] Sean Bright <sean.bright@gmail.com> + + * res/res_config_pgsql.c, /, res/res_config_ldap.c: Merged + revisions 129864 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r129864 | + seanbright | 2008-07-10 20:55:06 -0400 (Thu, 10 Jul 2008) | 1 + line Fix some usages of snprintf, and clarify a couple variable + names. ........ + +2008-07-10 22:07 +0000 [r129764-129805] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 129804 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r129804 | tilghman | 2008-07-10 17:06:07 -0500 + (Thu, 10 Jul 2008) | 16 lines 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 129758 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r129758 | tilghman | 2008-07-10 16:23:23 -0500 + (Thu, 10 Jul 2008) | 10 lines 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 21:05 +0000 [r129739] Terry Wilson <twilson@digium.com> + + * Makefile, /: Merged revisions 129738 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r129738 | + twilson | 2008-07-10 15:56:20 -0500 (Thu, 10 Jul 2008) | 2 lines + Move phoneprov config files to be installed with 'make samples' + so changes aren't inadvertently lost on a 'make install' ........ + +2008-07-10 19:14 +0000 [r129685] Brett Bryant <bbryant@digium.com> + + * /, apps/app_queue.c: Merged revisions 129684 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r129684 | + bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines + 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 18:20 +0000 [r129640-129647] Sean Bright <sean.bright@gmail.com> + + * /, channels/chan_sip.c: Merged revisions 129642 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r129642 | + seanbright | 2008-07-10 14:19:17 -0400 (Thu, 10 Jul 2008) | 1 + line A couple more minor text changes ........ + + * /, channels/chan_sip.c: Merged revisions 129638 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r129638 | + seanbright | 2008-07-10 14:16:21 -0400 (Thu, 10 Jul 2008) | 1 + line Remove extraneous \n. Pointed out by eliel on #asterisk-dev. + ........ + +2008-07-10 16:13 +0000 [r129570] Russell Bryant <russell@digium.com> + + * sample.call, /: Merged revisions 129569 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r129569 | russell | 2008-07-10 11:12:51 -0500 (Thu, 10 Jul 2008) + | 11 lines 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 14:09 +0000 [r129504-129507] Sean Bright <sean.bright@gmail.com> + + * /, main/editline: Merged revisions 129503 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r129503 | + seanbright | 2008-07-10 09:54:29 -0400 (Thu, 10 Jul 2008) | 2 + lines Update svn:ignore ........ + +2008-07-09 19:41 +0000 [r129438] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /: Merged revisions 129437 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul + 2008) | 21 lines 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 16:01 +0000 [r129400] Matthew Fredrickson <creslin@digium.com> + + * main/pbx.c, /: Merged revisions 129399 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r129399 | + mattf | 2008-07-09 10:57:06 -0500 (Wed, 09 Jul 2008) | 1 line Add + Proceeding() application (#13025) ........ + +2008-07-09 13:46 +0000 [r129345] Sean Bright <sean.bright@gmail.com> + + * main/editline/makelist (removed), main/editline/makelist.in + (added), /, main/editline/configure, main/editline/Makefile.in, + main/editline/configure.in: Merged revisions 129344 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r129344 | seanbright | 2008-07-09 09:44:43 -0400 + (Wed, 09 Jul 2008) | 12 lines 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. ........ ................ + +2008-07-08 22:56 +0000 [r129160-129271] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 129270 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r129270 | mmichelson | 2008-07-08 17:56:12 -0500 (Tue, 08 Jul + 2008) | 3 lines Fix compilation error when IMAP storage is + enabled ........ + +2008-07-08 21:04 +0000 [r129157] Brett Bryant <bbryant@digium.com> + + * main/dns.c, main/srv.c, /: Merged revisions 129156 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r129156 | bbryant | 2008-07-08 16:00:01 -0500 (Tue, 08 Jul 2008) + | 6 lines 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 20:31 +0000 [r129049-129153] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, /, channels/chan_sip.c, + include/asterisk/causes.h: Merged revisions 129152 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r129152 | tilghman | 2008-07-08 15:30:29 -0500 + (Tue, 08 Jul 2008) | 16 lines 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 ........ + ................ + + * /, channels/chan_iax2.c: Merged revisions 129048 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r129048 | tilghman | 2008-07-08 11:49:01 -0500 + (Tue, 08 Jul 2008) | 15 lines 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 16:41 +0000 [r129041-129046] Brett Bryant <bbryant@digium.com> + + * main/rtp.c, main/channel.c, channels/chan_dahdi.c, + main/manager.c, formats/format_pcm.c, main/logger.c, + main/callerid.c, apps/app_parkandannounce.c, apps/app_adsiprog.c, + 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: Merged revisions 129045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r129045 | + bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines + Janitor project to convert sizeof to ARRAY_LEN macro. (closes + issue #13002) Reported by: caio1982 Patches: + janitor_arraylen5.diff uploaded by caio1982 (license 22) ........ + + * /, channels/chan_sip.c: Merged revisions 127621 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r127621 | + bbryant | 2008-07-02 17:16:29 -0500 (Wed, 02 Jul 2008) | 1 line + Update transport= in sip so that the option is not broken from a + recent commit. ........ + + * /, channels/chan_sip.c: Merged revisions 127434 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r127434 | + bbryant | 2008-07-02 12:27:36 -0500 (Wed, 02 Jul 2008) | 1 line + Fix to sip_parse_host so that it passes the correct information + to sip_registry. ........ + +2008-07-08 14:18 +0000 [r129007] Russell Bryant <russell@digium.com> + + * /, apps/app_fax.c: Merged revisions 129006 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r129006 | + russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines + 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 10:06 +0000 [r128913-128952] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 128951 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19 + lines 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 128927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r128927 | oej | 2008-07-08 11:26:37 +0200 (Tis, 08 Jul 2008) | 15 + lines 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 00:03 +0000 [r128855-128858] Tilghman Lesher <tlesher@digium.com> + + * /: Merged revisions 128857 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r128857 | tilghman | 2008-07-07 19:02:11 -0500 (Mon, 07 Jul 2008) + | 15 lines 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 128830 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r128830 | tilghman | 2008-07-07 18:25:39 -0500 + (Mon, 07 Jul 2008) | 10 lines 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 22:44 +0000 [r128797] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 128796 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r128796 | russell | 2008-07-07 17:42:30 -0500 + (Mon, 07 Jul 2008) | 16 lines 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 20:51 +0000 [r128739] Sean Bright <sean.bright@gmail.com> + + * /, channels/chan_iax2.c: Merged revisions 128738 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r128738 | seanbright | 2008-07-07 16:50:29 -0400 + (Mon, 07 Jul 2008) | 17 lines 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 20:31 +0000 [r128601-128735] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 128733 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r128733 | mmichelson | 2008-07-07 15:30:46 -0500 (Mon, 07 Jul + 2008) | 3 lines Crap ........ + + * apps/app_voicemail.c, /: Merged revisions 128731 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul + 2008) | 7 lines 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 ........ + + * /, channels/chan_iax2.c: Merged revisions 128640 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r128640 | mmichelson | 2008-07-07 12:09:11 -0500 + (Mon, 07 Jul 2008) | 18 lines 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) ........ ................ + + * configs/extensions.conf.sample, /, doc/tex/extensions.tex: Merged + revisions 128599 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r128599 | + mmichelson | 2008-07-07 09:35:27 -0500 (Mon, 07 Jul 2008) | 6 + lines Update a few instances of "extensions reload" to "dialplan + reload" in the documentation. Patch provided by caio1982 (license + 22) ........ + +2008-07-06 20:22 +0000 [r128288-128543] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 128524 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r128524 | + oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines - + 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: Merged revisions 128491 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r128491 | + oej | 2008-07-06 21:14:06 +0200 (Sön, 06 Jul 2008) | 3 lines - + Remove unused variable "expiry" - Set global_outboundproxy.force + at reload. ........ + + * doc/realtimetext.txt (added), /: The following patch with + references to t140red removed, since it only exists in trunk. + Merged revisions 128417 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r128417 | + oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines + Adding documentation on the T.140 support in Asterisk. This is a + function that we're the reference implementation on now. :-) + ........ + + * /: Merged revisions 128343 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r128343 | + oej | 2008-07-06 10:10:27 +0200 (Sön, 06 Jul 2008) | 2 lines + Removing the CLI dumpdb command (see asterisk-dev discussion and + decision) ........ + + * /, channels/chan_sip.c: Merged revisions 128290 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r128290 | + oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines + 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: Merged revisions 128287 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r128287 | + oej | 2008-07-05 23:37:57 +0200 (Lör, 05 Jul 2008) | 3 lines + Adding TCP and TLS to "sip show settings". TLS needs to have one + configuration per configured domain at some point. ........ + + * /: Blocking changes in trunk. + +2008-07-05 21:02 +0000 [r128238-128243] Olle Johansson <oej@edvina.net> + + * /: Keep the "sip-user" structure in 1.6.0, while testing new + funky stuff in trunk. + + * /: Blocking the AGi changes from 1.6.0. Let's test them for a + while in trunk before a release. + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 128237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r128237 | + oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines + Make TCP disabled by default (it's considered experimental) + ........ + + * /, configs/sip.conf.sample: Merged revisions 128236 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2 + lines Reformatting the config sample ........ + +2008-07-05 15:19 +0000 [r128161] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif, /: Merged revisions 128160 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r128160 | tilghman | 2008-07-05 10:17:51 -0500 (Sat, 05 + Jul 2008) | 7 lines 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 03:40 +0000 [r128124-128127] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 128125 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r128125 | mattf | 2008-07-04 22:39:07 -0500 (Fri, 04 Jul 2008) | + 1 line It would help if we actually parsed the ss7_explicitacm + option in the config file... ........ + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged + revisions 128122 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r128122 | + mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line 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 16:12 +0000 [r128028-128031] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: Merged + revisions 128027 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r128027 | tilghman | 2008-07-04 11:06:34 -0500 (Fri, 04 Jul 2008) + | 16 lines 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 ........ + ................ + +2008-07-03 22:23 +0000 [r127905] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /, apps/Makefile, main/editline/np/vis.c: Merged + revisions 127903 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r127903 | kpfleming | 2008-07-03 17:23:04 -0500 (Thu, 03 Jul + 2008) | 20 lines 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 19:12 +0000 [r127830] 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 127793 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) | + 38 lines 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 16:50 +0000 [r127790-127792] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 127791 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r127791 | + oej | 2008-07-03 18:48:23 +0200 (Tor, 03 Jul 2008) | 5 lines Make + sure we stop session timers as soon as we start hanging up an + active call. May fix issue 12919. ........ + + * /, channels/chan_sip.c: Merged revisions 127779 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r127779 | + oej | 2008-07-03 18:25:59 +0200 (Tor, 03 Jul 2008) | 4 lines + 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 16:24 +0000 [r127778] Kevin P. Fleming <kpfleming@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + acinclude.m4: Merged revisions 127767 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r127767 | + kpfleming | 2008-07-03 11:22:02 -0500 (Thu, 03 Jul 2008) | 2 + lines some minor fixes found while working on issue #12911 (and + block the rev from 1.4 since the equivalent is already here) + ........ + +2008-07-02 21:10 +0000 [r127567] Mark Michelson <mmichelson@digium.com> + + * /, doc/janitor-projects.txt: Merged revisions 127566 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r127566 | mmichelson | 2008-07-02 16:09:18 -0500 (Wed, 02 Jul + 2008) | 4 lines Add a janitor project to use ARRAY_LEN instead of + in-line sizeof() and division. ........ + +2008-07-02 20:49 +0000 [r127559-127563] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 127562 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r127562 | mmichelson | 2008-07-02 15:49:08 -0500 + (Wed, 02 Jul 2008) | 11 lines 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 ........ ................ + +2008-07-02 19:48 +0000 [r127467-127503] Tilghman Lesher <tlesher@digium.com> + + * /, main/acl.c: Merged revisions 127466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 | + tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines + Solaris fix (closes issue #12949) Reported by: snuffy Patches: + bug_12949.diff uploaded by snuffy (license 35) ........ + +2008-07-02 14:30 +0000 [r127396-127399] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_tds.c, /: Merged revisions 127398 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r127398 | + seanbright | 2008-07-02 10:30:09 -0400 (Wed, 02 Jul 2008) | 1 + line Fix a bug I noticed while doing the previous merge ........ + + * cdr/cdr_tds.c, /, doc/tex/freetds.tex, configure, + include/asterisk/autoconfig.h.in, configure.ac, UPGRADE.txt: + Merged revisions 126226,126513 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r126226 | + seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8 + lines 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 ........ r126513 | seanbright + | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines 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) ........ + + * /: Unblock some revisions so I can merge the cdr_tds changes from + trunk + +2008-07-02 12:09 +0000 [r127364] Russell Bryant <russell@digium.com> + + * doc/CODING-GUIDELINES, /: Merged revisions 127363 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r127363 | russell | 2008-07-02 07:08:33 -0500 (Wed, 02 Jul 2008) + | 13 lines 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 02:49 +0000 [r127298] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 127297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r127297 | + tilghman | 2008-07-01 21:48:43 -0500 (Tue, 01 Jul 2008) | 12 + lines 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 23:39 +0000 [r127246] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 127245 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r127245 | mmichelson | 2008-07-01 18:38:12 -0500 + (Tue, 01 Jul 2008) | 13 lines 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. ........ |