From fbf6d95b4b70e678fbf45f43d86a24709dbb3528 Mon Sep 17 00:00:00 2001 From: Leif Madsen Date: Fri, 22 Jul 2011 19:46:50 +0000 Subject: [PATCH] Importing files for 10.0.0-beta1 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.0.0-beta1@329324 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- .lastclean | 3 + .version | 1 + ChangeLog | 15582 +++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 15586 insertions(+) create mode 100644 .lastclean create mode 100644 .version create mode 100644 ChangeLog diff --git a/.lastclean b/.lastclean new file mode 100644 index 0000000000..c7f5afc52a --- /dev/null +++ b/.lastclean @@ -0,0 +1,3 @@ +39 + + diff --git a/.version b/.version new file mode 100644 index 0000000000..9ebbe8a0c3 --- /dev/null +++ b/.version @@ -0,0 +1 @@ +10.0.0-beta1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 0000000000..be452258af --- /dev/null +++ b/ChangeLog @@ -0,0 +1,15582 @@ +2011-07-22 Leif Madsen + + * Asterisk 10.0.0-beta1 Released. + +2011-07-21 20:22 +0000 [r329257] Russell Bryant + + * channels/chan_dahdi.c, main/features.c, + include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c, + include/asterisk/rtp_engine.h: s/1.10/10.0/ + +2011-07-21 18:05 +0000 [r329200-329204] Richard Mudgett + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged + revisions 329203 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) + | 6 lines Document parkinglot in chan_dahdi.conf.sample. * + Document existing feature in chan_dahdi.conf.sample. * Remove + some dead code related to the parkinglot option. ........ + + * /, apps/app_directed_pickup.c: Merged revisions 329199 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) + | 17 lines Update PickupChan documentation. The PickupChan uses + the ampersand as the argument separator. Was documented as: + PickupChan(channel[,channel2[,...][,options]]) Fixed + documentation to: + PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) + This is a continuation of ASTERISK-17494 for v1.8 and later. + (closes issue ASTERISK-18144) Reported by: Erik Smith Patches: + pickupchan_ducumentation-v2.patch (License #6263) patch uploaded + by Erik Smith Tested by: Erik Smith ........ + +2011-07-21 17:27 +0000 [r329188] Jason Parker + + * UPGRADE.txt: Fix version number in UPGRADE.txt. + +2011-07-21 16:52 +0000 [r329145] Richard Mudgett + + * /, main/features.c: Merged revisions 329144 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) + | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed + more times than we've locked! This appears to be a leftover from + when ast_channel was converted to ao2 objects. Simply removed the + extraneous unlock. (closes issue ASTERISK-17772) ........ + +2011-07-21 16:04 +0000 [r329104] Russell Bryant + + * / (added): Change Asterisk 2.0 to 2.0 in binary + +2011-07-20 21:31 +0000 [r329056] Paul Belanger + + * /, UPGRADE-1.8.txt: Merged revisions 329055 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/2.0 + ................ r329055 | pabelanger | 2011-07-20 17:27:50 -0400 + (Wed, 20 Jul 2011) | 9 lines Merged revisions 329027 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, + 20 Jul 2011) | 2 lines Asterisk now requires libpri 1.4.11+ for + PRI support. ........ ................ + +2011-07-20 20:19 +0000 [r328996] Terry Wilson + + * /, tests/test_netsock2.c: Merged revisions 328992 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/2.0 + ................ r328992 | twilson | 2011-07-20 15:18:25 -0500 + (Wed, 20 Jul 2011) | 12 lines Merged revisions 328987 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) + | 5 lines We can't guarantee an eth0 is present FreeBSD test + fails on this case presumably because there is no eth0 on the + test machine. Better to just remove this test for now. ........ + ................ + +2011-07-20 19:03 +0000 [r328937] Kinsey Moore + + * /, channels/chan_sip.c: Merged revisions 328936 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/2.0 + ................ r328936 | kmoore | 2011-07-20 14:01:37 -0500 + (Wed, 20 Jul 2011) | 15 lines Merged revisions 328935 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | + 8 lines Inband DTMF regression The functionality of inband DTMF + in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not + working properly to avoid calling ast_rtp_instance_dtmf_begin/end + on RTP streams with inband DTMF. According to documentation, + ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF, + never inband. This fixes the regression introduced in revision + 328823. ........ ................ + +2011-07-19 21:32 +0000 [r328880-328881] Kevin P. Fleming + + * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged + revisions 328879 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/2.0 + ................ r328879 | kpfleming | 2011-07-19 16:31:16 -0500 + (Tue, 19 Jul 2011) | 23 lines Merged revisions 328878 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul + 2011) | 17 lines Revert partial attempt at handling pathnames + with spaces. Revision 299794 attempted to improve the build + system to be able to handle pathnames (primarily DESTDIR) with + spaces in them, since this is common on some platforms (including + Mac OSX). Unfortunately, the changes were incomplete and did not + actually provide the desired behavior, and as a side effect the + functionality that ensured that stale headers in the Asterisk + 'include' directory were removed got broken. In addition, the + check for stale (and possibly incompatible) modules in the + Asterisk 'modules' directory also got broken, and would never + report any stale modules. Users upgrading to this version or + later versions would then see unexpected module load errors. + Since there are few users who actually want to install Asterisk + into paths that contain spaces, and a proper fix for the build + system would take many hours, the best solution for now is to + just revert the partial solution. ........ ................ + + * /: Edit the merge properties to match their names. + +2011-07-19 21:21 +0000 [r328877] Russell Bryant + + * /: Fix properties after twilson's change so merges still work + +2011-07-19 18:07 +0000 [r328772-328825] Kinsey Moore + + * res/res_rtp_asterisk.c, main/rtp_engine.c, /, + channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged + revisions 328824 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328824 | kmoore | 2011-07-19 13:05:21 -0500 + (Tue, 19 Jul 2011) | 18 lines Merged revisions 328823 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | + 11 lines RTP bridge away with inband DTMF and feature detection + When deciding whether Asterisk was allowed to bridge the call + away from the core, chan_sip did not take into account the usage + of features on dialed channels that require monitoring of DTMF on + channels utilizing inband DTMF. This would cause Asterisk to + allow the call to be locally or remotely bridged, preventing + access to the data required to detect activations of such + features. (closes 17237) Review: + https://reviewboard.asterisk.org/r/1302/ ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 328771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328771 | kmoore | 2011-07-19 10:46:54 -0500 + (Tue, 19 Jul 2011) | 18 lines Merged revisions 328770 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | + 11 lines MeetMe requests a PIN twice in some circumstances If a + call to MeetMe includes both the dynamic(D) and always request + PIN(P) options, MeetMe will ask for the PIN two times: once for + creating the conference and once for entering the conference. + This behavior was introduced in rev 311616 when adding the + CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN + entry for joining a conference. (closes AST-601) Review: + https://reviewboard.asterisk.org/r/1305/ ........ + ................ + +2011-07-19 02:00 +0000 [r328718] Terry Wilson + + * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c + (added): Merged revisions 328717 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328717 | twilson | 2011-07-18 20:55:32 -0500 + (Mon, 18 Jul 2011) | 14 lines Merged revisions 328716 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) + | 7 lines Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't + modify the element passed in if it isn't found. This commit also + adds linked list unit tests. Review: + https://reviewboard.asterisk.org/r/1321/ ........ + ................ + +2011-07-18 20:51 +0000 [r328610-328665] Mark Murawki + + * apps/app_dial.c, /: Merged revisions 328664 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328664 | markm | 2011-07-18 16:50:13 -0400 + (Mon, 18 Jul 2011) | 15 lines Merged revisions 328663 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | + 9 lines app_dial may double free a channel datastore When + starting a call with originate, and having the callee channel run + Bridge() on pickup, we will double free the dialed_interface_info + datastore, causing a crash. Make sure to check if the datastore + still exists before trying to free it. (closes issue + ASTERISK-17917) Reported by: Mark Murawski Tested by: Mark + Murawski ........ ................ + + * /, channels/chan_sip.c: Merged revisions 328611 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328611 | markm | 2011-07-18 08:56:49 -0400 + (Mon, 18 Jul 2011) | 15 lines Merged revisions 328608 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | + 9 lines If the sip private structure is null, sip_setoption() + will defref the null pointer and crash. Ideally, sip_setoption + shouldn't be called if there is a lack of a sip private + structure. But this will fix a crash. (closes issue + ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark + Murawski ........ ................ + + * /, main/asterisk.c: Merged revisions 328609 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328609 | markm | 2011-07-18 08:37:53 -0400 + (Mon, 18 Jul 2011) | 15 lines Merged revisions 328593 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | + 8 lines Fixed invalid read and null pointer deref on asterisk + shutdown. In some cases when starting asterisk with -c and + hitting control-c to shutdown, there will be an invalid read and + null pointer deref causing a crash. (closes issue ASTERISK-17927) + Reported by: Mark Murawski Tested by: Mark Murawski, Kinsey + Moore, Tilghman Lesher ........ ................ + +2011-07-18 07:12 +0000 [r328542] Tilghman Lesher + + * /, funcs/func_odbc.c: Merged revisions 328541 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328541 | tilghman | 2011-07-18 02:11:26 -0500 + (Mon, 18 Jul 2011) | 9 lines Merged revisions 328540 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 + Jul 2011) | 2 lines Typo ........ ................ + +2011-07-15 21:41 +0000 [r328502] Alexandr Anikin + + * addons/ooh323c/src/ooGkClient.c, /: Merged revisions + 328428-328429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328428 | may | 2011-07-15 23:31:09 +0400 (Fri, + 15 Jul 2011) | 13 lines Merged revisions 328427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 + lines small gk processing fixes: - decrease for 1 second + registration ttl for very low expirations (some providers expire + few earlier than TTL) - delete rrq and registration expire timers + on URQ received as we make new registration. ........ + ................ r328429 | may | 2011-07-15 23:35:50 +0400 (Fri, + 15 Jul 2011) | 2 lines delete unproperly changed svn props + ................ + +2011-07-15 21:19 +0000 [r328449-328459] Leif Madsen + + * /, apps/app_macro.c: Merged revisions 328451 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ + r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) + | 1 line Build app_macro by default because things depend on it. + ........ + + * /, UPGRADE-1.10.txt, UPGRADE.txt, CHANGES: Merged revisions + 328448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ + r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) + | 2 lines Update UPGRADE.txt and CHANGES files. Update + documentation files stating that deprecated modules are no longer + built by default. ........ + +2011-07-15 08:19 +0000 [r328381] Damien Wedhorn + + * channels/chan_skinny.c: Add SLA to skinny. Adds sublines to + skinny lines. Each subline can be attached to an SLA + station/trunk combo. Includes the following functionality: Callid + is persistent for both in/out calls on all skinny devices. Can + join, hold, resume. All sublines appear under a single line + button. See: + https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for + doc. (closes issue ASTERISK-17947) Review: + https://reviewboard.asterisk.org/r/1239/ + +2011-07-15 00:23 +0000 [r328318-328344] Richard Mudgett + + * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c, + include/asterisk/extconf.h, include/asterisk/pbx.h, + apps/app_queue.c: Merged revisions 328329 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ + r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) + | 2 lines Make hint watcher callback take const strings for + context and exten parameters. ........ + + * /, channels/chan_sip.c: Merged revisions 328317 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328317 | rmudgett | 2011-07-14 18:28:49 -0500 + (Thu, 14 Jul 2011) | 13 lines Merged revisions 328302 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) + | 6 lines Missing SIP pvt and channel unlock in + sip_set_rtp_peer(). Regression introduced by -r326144. Add + missing SIP pvt and channel unlock in sip_set_rtp_peer(). + ........ ................ + +2011-07-14 20:28 +0000 [r328259] Leif Madsen + + * funcs/func_speex.c, apps/app_playtones.c, + bridges/bridge_softmix.c, apps/app_alarmreceiver.c, + res/res_calendar_caldav.c, apps/app_ices.c, apps/app_exec.c, + channels/chan_iax2.c, res/res_pktccops.c, channels/chan_skinny.c, + pbx/pbx_ael.c, formats/format_h263.c, cdr/cdr_odbc.c, + cdr/cdr_manager.c, utils/refcounter.c, funcs/func_timeout.c, + formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c, + bridges/bridge_simple.c, funcs/func_cut.c, apps/app_talkdetect.c, + apps/app_db.c, funcs/func_callcompletion.c, funcs/func_channel.c, + funcs/func_iconv.c, pbx/pbx_config.c, res/res_odbc.c, + apps/app_voicemail.c, formats/format_sln.c, + apps/app_authenticate.c, apps/app_readexten.c, + res/res_phoneprov.c, apps/app_userevent.c, codecs/codec_gsm.c, + tests/test_func_file.c, apps/app_setcallerid.c, + res/res_config_odbc.c, funcs/func_audiohookinherit.c, + apps/app_osplookup.c, funcs/func_odbc.c, cel/cel_custom.c, + tests/test_utils.c, apps/app_mp3.c, res/res_timing_timerfd.c, + codecs/codec_resample.c, formats/format_h264.c, + apps/app_directory.c, formats/format_siren14.c, + tests/test_amihooks.c, res/res_config_pgsql.c, + funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c, + res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c, + apps/app_forkcdr.c, funcs/func_blacklist.c, apps/app_sms.c, + formats/format_g723.c, utils/extconf.c, tests/test_poll.c, + apps/app_stack.c, apps/app_verbose.c, utils/check_expr.c, + funcs/func_module.c, codecs/codec_adpcm.c, tests/test_event.c, + cdr/cdr_adaptive_odbc.c, apps/app_image.c, + formats/format_wav_gsm.c, utils/stereorize.c, pbx/pbx_loopback.c, + tests/test_time.c, funcs/func_shell.c, apps/app_skel.c, + channels/chan_alsa.c, apps/app_externalivr.c, + apps/app_milliwatt.c, formats/format_gsm.c, res/res_speech.c, + apps/app_dial.c, apps/app_page.c, apps/app_fax.c, utils/astman.c, + apps/app_disa.c, res/res_monitor.c, apps/app_waitforring.c, + addons/cdr_mysql.c, res/res_fax_spandsp.c, + res/res_timing_kqueue.c, apps/app_chanspy.c, apps/app_cdr.c, + channels/chan_unistim.c, funcs/func_base64.c, + channels/chan_multicast_rtp.c, funcs/func_md5.c, + apps/app_meetme.c, tests/test_gosub.c, funcs/func_sysinfo.c, + funcs/func_vmcount.c, res/res_musiconhold.c, cdr/cdr_radius.c, + apps/app_followme.c, res/res_config_sqlite.c, + apps/app_controlplayback.c, cdr/cdr_csv.c, formats/format_ilbc.c, + channels/chan_phone.c, funcs/func_enum.c, main/manager.c, + funcs/func_groupcount.c, tests/test_stringfields.c, + tests/test_locale.c, tests/test_devicestate.c, + funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c, + tests/test_astobj2.c, apps/app_ivrdemo.c, res/res_clioriginate.c, + apps/app_jack.c, apps/app_nbscat.c, res/res_calendar_icalendar.c, + codecs/codec_a_mu.c, tests/test_ast_format_str_reduce.c, + tests/test_dlinklists.c, res/res_convert.c, apps/app_waituntil.c, + pbx/pbx_lua.c, utils/astcanary.c, apps/app_queue.c, + channels/chan_oss.c, cdr/cdr_tds.c, channels/chan_usbradio.c, + apps/app_flash.c, apps/app_senddtmf.c, funcs/func_callerid.c, + addons/app_saycountpl.c, cel/cel_pgsql.c, apps/app_dahdibarge.c, + channels/chan_local.c, funcs/func_dialgroup.c, + tests/test_logger.c, apps/app_record.c, funcs/func_env.c, + funcs/func_strings.c, res/res_timing_dahdi.c, + apps/app_chanisavail.c, bridges/bridge_multiplexed.c, + res/res_rtp_multicast.c, cel/cel_odbc.c, channels/chan_dahdi.c, + pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_pcm.c, + apps/app_dumpchan.c, main/http.c, res/res_clialiases.c, + res/res_calendar_exchange.c, res/res_ais.c, funcs/func_sprintf.c, + codecs/codec_g722.c, tests/test_expr.c, cel/cel_tds.c, + tests/test_app.c, utils/smsq.c, apps/app_morsecode.c, + formats/format_ogg_vorbis.c, tests/test_sched.c, + res/res_calendar_ews.c, apps/app_speech_utils.c, + tests/test_acl.c, apps/app_sendtext.c, funcs/func_cdr.c, + utils/hashtest2.c, utils/ael_main.c, apps/app_mixmonitor.c, + formats/format_g726.c, utils/streamplayer.c, res/res_calendar.c, + cel/cel_radius.c, channels/chan_vpb.cc, tests/test_heap.c, + addons/format_mp3.c, res/res_snmp.c, apps/app_dictate.c, + channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c, + res/res_jabber.c, funcs/func_uri.c, cel/cel_manager.c, + apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c, + apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, + codecs/codec_lpc10.c, apps/app_read.c, cdr/cdr_syslog.c, + codecs/codec_alaw.c, res/res_adsi.c, agi/eagi-test.c, + utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c, + formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c, + pbx/pbx_dundi.c, funcs/func_devstate.c, + addons/res_config_mysql.c, funcs/func_rand.c, + apps/app_readfile.c, addons/chan_ooh323.c, + cdr/cdr_sqlite3_custom.c, /, apps/app_sayunixtime.c, + apps/app_test.c, res/res_http_post.c, res/res_smdi.c, + main/features.c, funcs/func_srv.c, apps/app_amd.c, + pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c, + formats/format_g719.c, channels/chan_bridge.c, + apps/app_privacy.c, apps/app_echo.c, codecs/codec_speex.c, + apps/app_saycounted.c, apps/app_dahdiras.c, + channels/chan_agent.c, funcs/func_math.c, res/res_ael_share.c, + apps/app_transfer.c, res/res_mutestream.c, apps/app_playback.c, + res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c, + tests/test_skel.c, apps/app_macro.c, apps/app_zapateller.c, + codecs/codec_ilbc.c, addons/app_mysql.c, + tests/test_substitution.c, utils/muted.c, utils/hashtest.c, + funcs/func_sha1.c, formats/format_siren7.c, + tests/test_security_events.c, funcs/func_config.c, + bridges/bridge_builtin_features.c, funcs/func_volume.c, + res/res_agi.c, apps/app_confbridge.c, addons/chan_mobile.c, + apps/app_parkandannounce.c, res/res_security_log.c, + cdr/cdr_custom.c, apps/app_while.c, res/res_rtp_asterisk.c, + funcs/func_dialplan.c, funcs/func_db.c, apps/app_festival.c, + res/res_limit.c, res/res_fax.c, apps/app_waitforsilence.c, + channels/chan_console.c, apps/app_getcpeid.c, + funcs/func_global.c, res/res_srtp.c, funcs/func_extstate.c, + tests/test_strings.c, res/res_timing_pthread.c, + apps/app_directed_pickup.c, channels/chan_h323.c, + cel/cel_sqlite3_custom.c, codecs/codec_ulaw.c, + channels/chan_nbs.c, formats/format_g729.c: Merged revisions + 328247 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 + (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) + | 6 lines Introduce tags in MODULEINFO. This + change introduces MODULEINFO into many modules in Asterisk in + order to show the community support level for those modules. This + is used by changes committed to menuselect by Russell Bryant + recently (r917 in menuselect). More information about the support + level types and what they mean is available on the wiki at + https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States + ........ ................ + +2011-07-14 19:56 +0000 [r328208] Jonathan Rose + + * /, res/res_monitor.c: Merged revisions 328207 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328207 | jrose | 2011-07-14 14:45:18 -0500 + (Thu, 14 Jul 2011) | 13 lines Merged revisions 328205 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | + 6 lines Monitor application arguments requirements fixed. Monitor + was requiring options in spite of no individual option on Monitor + being required. Review: https://reviewboard.asterisk.org/r/1320/ + ........ ................ + +2011-07-14 17:47 +0000 [r328163] Matthew Nicholson + + * /, main/dsp.c: Merged revisions 328162 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ + r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul + 2011) | 3 lines tune the v21 preamble detector to properly detect + the desired sequence of hits and misses ........ + +2011-07-13 22:10 +0000 [r328121] David Vossel + + * /, apps/app_mixmonitor.c: Merged revisions 328120 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.10 + ........ r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 + Jul 2011) | 15 lines Preserve sample rate quality of wideband + mixmonitor recordings. MixMonitor has the ability to record in + any file format Asterisk supports, but the quality of wideband + audio is not preserved. This is because regardless of the sample + rate the call is being recorded in, the audio is always + downsampled to 8khz and then upsampled to whatever wideband + format it is being written as. This patch resolves this by + requesting the audio from the audiohook in the signed linear + format closest to the sample rate of the format we are writing. + This fix is only possible for Asterisk 1.10 because audio hooks + in 1.8 are not capable of wideband audio. Review: + https://reviewboard.asterisk.org/r/1314/ ........ + +2011-07-13 21:06 +0000 [r328079] Leif Madsen + + * BUGS, UPGRADE-1.10.txt (added), UPGRADE.txt: Add UPGRADE-1.10.txt + file from UPGRADE.txt. + +2011-07-13 20:40 +0000 [r328075-328076] Russell Bryant + + * /: set 1.10 merge properties + + * /: remove 1.8 merge properties + +2011-07-13 18:47 +0000 [r328016] Richard Mudgett + + * /, configs/features.conf.sample: Merged revisions 328014 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011) + | 1 line Add ATXFER_NULL_TECH note in features.conf.sample. + ........ + +2011-07-12 23:02 +0000 [r327953] Kevin P. Fleming + + * main/manager.c, /: Merged revisions 327950 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul + 2011) | 14 lines Correct double-free situation in manager output + processing. The process_output() function calls ast_str_append() + and xml_translate() on its 'out' parameter, which is a pointer to + an ast_str buffer. If either of these functions need to + reallocate the ast_str so it will have more space, they will free + the existing buffer and allocate a new one, returning the address + of the new one. However, because process_output only receives a + pointer to the ast_str, not a pointer to its caller's variable + holding the pointer, if the original ast_str is freed, the caller + will not know, and will continue to use it (and later attempt to + free it). (reported by jkroon on #asterisk-dev) ........ + +2011-07-12 20:08 +0000 [r327891] Matthew Nicholson + + * /, apps/app_directory.c: Merged revisions 327890 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue, + 12 Jul 2011) | 2 lines search in the current context for 'a' and + 'o' instead of 'default' ........ + +2011-07-12 19:39 +0000 [r327889] Jason Parker + + * Makefile, /: Merged revisions 327888 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327888 | qwell | 2011-07-12 14:38:44 -0500 (Tue, 12 Jul 2011) | + 1 line Fix uninstall target, so that modules dir gets cleared + again. ........ + +2011-07-12 19:18 +0000 [r327856] Matthew Jordan + + * /, apps/app_voicemail.c: Merged revisions 327852 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 + Jul 2011) | 12 lines Added additional checks for mailbox / + password beginning with '*' character A bug existed such that if + a user entered a password with '*', and the extension 'a' did not + exist, an invalid mailbox would be created and the user + authenticated. The code was changed to prevent this from + occurring, and to prevent users from having mailboxes or + passwords defined that begin with the '*' character. (closes + issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/ + ........ + +2011-07-12 15:38 +0000 [r327794] Tilghman Lesher + + * tests/test_substitution.c, /: Merged revisions 327793 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011) + | 14 lines Use 'printf' (POSIX issue 4) instead of 'echo -n', for + portability. The problem with using 'echo -n' is that it is not + portable. While BSD systems required that the '-n' option be + removed and interpreted, System V required that all strings + should be echoed with no interpretation of options. This + fundamental difference of behavior means that it is never + possible to use the '-n' flag to echo in tests which are meant to + be portable. In this case, on Mac OS X 10.6, the /bin/sh shell + builtin 'echo' uses the System V semantics of the command, and + thus the SHELL test failed on that platform. + http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16 + ........ + +2011-07-12 15:23 +0000 [r327769] Matthew Nicholson + + * res/res_fax.c, include/asterisk/dsp.h, main/dsp.c: do v21 + detection instead of CED detection for the fax gateway + +2011-07-12 14:55 +0000 [r327749] David Vossel + + * main/bridging.c: Send video update frame to new video source in + follow_talker correctly. + +2011-07-12 14:40 +0000 [r327748] Kinsey Moore + + * apps/app_confbridge.c: Segfault on shutdown when confbridge is + active When undergoing a shutdown and channels are kicked out of + a bridge, a segfault occurs because ConfBridge tries to play + sounds on the bridge after the underlying channels have been + blown away due to the shutdown. (closes ASTERISK-18040) Review: + https://reviewboard.asterisk.org/r/1283/ + +2011-07-11 20:06 +0000 [r327684] Matthew Nicholson + + * tests/test_substitution.c: use printf instead of echo -n in + substitution test + +2011-07-11 19:49 +0000 [r327683] Terry Wilson + + * /, include/asterisk/jingle.h, channels/chan_gtalk.c: Merged + revisions 327682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) + | 9 lines Update chan_gtalk to work with changed GMail-based + calls The messages sent by the GMail client have changed, but + include the old-style messages as well. This patch checks for + this case and uses the old-style offer. (closes issue + ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/ + ........ + +2011-07-11 18:44 +0000 [r327640] David Vossel + + * include/asterisk/bridging.h, bridges/bridge_softmix.c, + main/bridging.c: Updates follow_talker video_mode in confbridge + application. follow_talker mode originally echoed the same video + stream to all participants. As the primary talker switched + around, the video stream would result in the talker seeing + themselves. Now the primary talker sees the last person who was + talking rather than themselves. + +2011-07-11 17:23 +0000 [r327469-327598] Matthew Nicholson + + * res/res_fax.c: renamed fax_gateway_send_ced() to + fax_gateway_request_t38() + + * res/res_fax.c: actually do something with the ced timeout, also + added more debug output + + * res/res_fax.c: write silence on the channel during t.38 + negotiation + + * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327512 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul + 2011) | 2 lines reset our buffer each iteration when doing + variable substitution ........ + + * res/res_fax.c: Delay sending an CED tone generated T.38 reinvite + to give the CED tone generating party time to send its own T.38 + reinvite. Also don't forward frames through the gateway if we are + negotiating T.38. + + * res/res_fax.c: fixed wording in a comment + +2011-07-11 10:57 +0000 [r327413] Tzafrir Cohen + + * /, main/Makefile: Merged revisions 327411 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327411 | tzafrir | 2011-07-11 13:46:34 +0300 (ב', 11 יול 2011) | + 5 lines fix building the Debian armhf (HardFloat) port Fixes + http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385 + (Missing pthreads) ........ + +2011-07-10 01:37 +0000 [r327359] Alexandr Anikin + + * addons/chan_ooh323.c, configs/chan_ooh323.conf.sample: Full T.38 + handshaking and fax detection Add full t.38 handshaking for + OOH323 that are required for newest T.38 gateway codes. Add fax + detection (cng tone, t38) and dialplan redirection to fax ext on + fax event detected. Add OOH323() function to set/get t38support + and faxdetect parameters. (closes issue ASTERISK-17754) Reported + by: irroot Patches: ooh323_faxdetect.patch uploaded by irroot + (license 52) issue19183-final.patch uploaded by may213 (license + 454) Tested by: may213, irroot Review: + https://reviewboard.asterisk.org/r/1174/ + +2011-07-08 22:25 +0000 [r327246] Jason Parker + + * main/stdtime, utils, codecs, utils/db1-ast/recno, apps, cel, + apps/confbridge, cdr, formats, codecs/gsm/src, + utils/db1-ast/hash, funcs, bridges, codecs/lpc10, + utils/db1-ast/db, codecs/g722, utils/db1-ast/mpool, main, + codecs/speex, channels/sip, pbx, res, res/ael, channels, + utils/db1-ast/btree: Add .o files to svn:ignore property, since + it's only ignored if locally configured to do so. + +2011-07-08 21:43 +0000 [r327212] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 327211 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) + | 9 lines INVITE 403 Forbidden response always retransmits the + maximum times. Asterisk sends a 403 Forbidden response if + authentication fails for an INVITE as required. However, it + ignores the ACK and keeps retransmitting the response. * Made not + delete the to-tag in the dialog so the expected ACK can be + matched with the dialog and stop the retransmissions. ........ + +2011-07-08 20:33 +0000 [r327116-327168] David Vossel + + * UPGRADE.txt, CHANGES: Adds entry in UPDATES.txt for removal of + formats/format_sln16.c. Fixes typo in CHANGES as well. + + * CHANGES: Updates CHANGES log to reflect new slinear read/write + file interpreters. + + * formats/format_sln.c, formats/format_sln16.c (removed): Support + for writing and reading raw slin files 8khz-192khz. + + * formats/format_attr_silk.c (removed), formats/format_attr_celt.c + (removed), res/res_format_attr_silk.c (added), + res/res_format_attr_celt.c (added): Moves celt and silk format + attribute files into res folder. It was inconsistent to have the + silk and celt format attribute modules in the format file + interpreter folder. + +2011-07-08 19:54 +0000 [r327107] Matthew Nicholson + + * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327106 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul + 2011) | 11 lines Reset our ast_str before passing it on to + dialplan function backends. It is possible for a dialplan backend + to not modify the given buffer or ast_str and still return + success. This causes any previous value stored in the buffer to + be used as if the new function call provided it. Some functions + also append to the given buffer assuming it is empty. The + test_substitution unit test has also been modified to detect this + problem. (closes issue ASTERISK-17878) ........ + +2011-07-08 16:00 +0000 [r327045-327047] Russell Bryant + + * /, tests/test_netsock2.c: Merged revisions 327046 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r327046 | russell | 2011-07-08 11:00:05 -0500 (Fri, 08 + Jul 2011) | 2 lines Fix an error and add more log message info to + help see why this fails on FreeBSD. ........ + + * channels/chan_dahdi.c, /: Merged revisions 327044 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08 + Jul 2011) | 2 lines Resolve some set-but-unused-variable + warnings. ........ + +2011-07-08 01:26 +0000 [r327000] Richard Mudgett + + * main/pbx.c, /: Merged revisions 326985 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) + | 12 lines Some code cleanup in pbx.c * Mostly comment and format + changes. * ast_context_remove_extension_callerid() and + ast_add_extension_nolock() will write lock the found specific + context. * ast_context_find() will now tolerate a NULL name. * + Eliminated some inlined versions of find_context() and + find_context_locked(). ........ + +2011-07-07 22:39 +0000 [r326943] Jason Parker + + * include/asterisk/celt.h: I think reviewboard broke this. The + whole file was doubled. + +2011-07-07 22:17 +0000 [r326855-326904] David Vossel + + * formats/format_attr_celt.c (added): Adds the format_attr_celt + file which was also missing from the CELT pass through patch. + + * include/asterisk/celt.h (added): Adds missing celt.h file from + celt pass-through support patch. + + * CHANGES: Fixes spelling errors in CHANGES as well as adding a few + entries for CELT and confbridge. + + * main/channel.c, main/format.c, res/res_rtp_asterisk.c, + main/frame.c, main/rtp_engine.c, channels/chan_sip.c, + include/asterisk/format.h, configs/codecs.conf.sample: Adds + pass-through support for codec CELT. This patch adds pass-through + support for CELT. CELT formats are defined in codecs.conf and can + be configured to any sample rate a CELT endpoint supports. This + patch also addresses a crash in channel.c resulting from a frame + list being freed incorrectly. This crash was discovered while + testing a CELT translator which had to split encoded audio into + multiple frames. The codec translator is not a part of this + patch, but may be contributed in the future. Review: + https://reviewboard.asterisk.org/r/1294/ + +2011-07-07 19:20 +0000 [r326842] Tilghman Lesher + + * /, res/res_http_post.c: Merged revisions 326830 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326830 | tilghman | 2011-07-07 14:17:19 -0500 (Thu, 07 Jul 2011) + | 1 line libgen.h is also needed on Darwin for basename(3) + ........ + +2011-07-07 17:24 +0000 [r326782] David Vossel + + * configs/confbridge.conf.sample, + apps/confbridge/include/confbridge.h, apps/app_confbridge.c, + apps/confbridge/conf_config_parser.c: Updates confbridge.conf + video documentation and adds dtmf action for releasing video src. + +2011-07-07 16:50 +0000 [r326750] Terry Wilson + + * utils/astdb2sqlite3.c, main/db.c: Use older functions out of + deference to older distros + +2011-07-07 16:18 +0000 [r326694] Jonathan Rose + + * res/res_config_odbc.c, /: Merged revisions 326689 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul + 2011) | 10 lines res_odbc patch by tilghman to fix integers with + null values Addresses some improper sql statements in res_odbc + that would cause an update to fail on realtime peers due to + trying to set as "(NULL)" rather than an actual NULL. (closes + issue #1922STERISK-17791) Reported by: marcelloceschia Patches: + 20110505__issue19223.diff.txt uploaded by tilghman (license 14) + ........ + +2011-07-07 15:28 +0000 [r326682-326684] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 326683 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326683 | mnicholson | 2011-07-07 10:28:25 -0500 (Thu, 07 Jul + 2011) | 3 lines use sips: or sip: depending on the transport in + use when building reply digest URIs ........ + + * /, channels/chan_sip.c: Merged revisions 326681 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326681 | mnicholson | 2011-07-07 10:25:49 -0500 (Thu, 07 Jul + 2011) | 3 lines make the uri parameter used in reply digests more + standards compliant in certain cases by prepending "sip:" or + "sips:" to it ........ + +2011-07-07 09:49 +0000 [r326636] Tzafrir Cohen + + * contrib/scripts/live_ast: live_ast: valgrind: run asterisk under + valgrind Adds a new sub-command, "valgrind" to live_ast. It runs + asterisk under valgrind. The extra command-line parameters are + passed to Asterisk as usual, and parameters to valgrind are + passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review: + https://reviewboard.asterisk.org/r/1109/ + +2011-07-06 20:58 +0000 [r326589] Terry Wilson + + * utils/db1-ast/btree/bt_open.c, utils/db1-ast/hash/hash_log2.c, + utils/db1-ast/hash/hsearch.c, utils/db1-ast/btree/bt_page.c, + utils/db1-ast/hash/page.h, utils/db1-ast/mpool, configure, + utils/db1-ast/btree/extern.h, utils/db1-ast/include/db.h, + main/db.c, utils/db1-ast/btree/bt_seq.c, + utils/db1-ast/recno/recno.h, main/Makefile, + utils/db1-ast/btree/bt_utils.c, utils/db1-ast/recno/rec_seq.c, + configure.ac, utils/db1-ast/btree/bt_close.c, CHANGES, + utils/db1-ast/hash/search.h, utils/db1-ast/hash/README, + utils/db1-ast/recno/rec_open.c, utils/db1-ast/hash/hash_bigkey.c, + utils/db1-ast/recno/rec_delete.c, Makefile, + utils/db1-ast/include, utils/db1-ast/hash/hash_buf.c, + utils/db1-ast/db, utils/db1-ast/libdb.map, + utils/db1-ast/include/ndbm.h, utils/db1-ast/include/compat.h, + utils/db1-ast/mpool/mpool.c, utils/db1-ast/btree/bt_debug.c, + main/asterisk.c, utils/db1-ast (added), + utils/db1-ast/btree/bt_split.c, utils, utils/db1-ast/recno, + utils/db1-ast/btree/bt_delete.c, + utils/db1-ast/include/circ-queue.h, tests/test_db.c, + utils/db1-ast/Makefile, utils/db1-ast/hash/extern.h, + utils/db1-ast/recno/rec_search.c, utils/db1-ast/btree/bt_get.c, + utils/db1-ast/hash/hash.c, utils/db1-ast/btree/btree.h, + utils/db1-ast/db/db.c, utils/db1-ast/hash/hash.h, + utils/db1-ast/include/mpool.h, utils/db1-ast/recno/rec_get.c, + utils/db1-ast/hash/hash_func.c, utils/utils.xml, + utils/astdb2sqlite3.c (added), utils/db1-ast/btree/bt_overflow.c, + UPGRADE.txt, utils/db1-ast/btree/bt_conv.c, + utils/db1-ast/btree/bt_search.c, utils/db1-ast/btree/bt_put.c, + utils/db1-ast/recno/rec_utils.c, utils/Makefile, + utils/db1-ast/hash/hash_page.c, utils/db1-ast/hash, + utils/db1-ast/mpool/README, utils/db1-ast/hash/ndbm.c, + main/db1-ast (removed), utils/db1-ast/recno/rec_close.c, + utils/db1-ast/recno/rec_put.c, utils/db1-ast/recno/extern.h, + utils/db1-ast/btree: Replace Berkeley DB with SQLite 3 There were + some bugs in the very ancient version of Berkeley DB that + Asterisk used. Instead of spending the time tracking down the + bugs in the Berkeley code we move to the much better documented + SQLite 3. Conversion of the old astdb happens at runtime by + running the included astdb2sqlite3 utility. The ast_db API with + SQLite 3 backend should behave identically to the old Berkeley + backend, but in the future we could offer a much more robust + interface. We do not include the SQLite 3 library in the source + tree, but instead rely upon the distribution-provided libraries. + SQLite is so ubiquitous that this should not place undue burden + on administrators. + +2011-07-06 17:39 +0000 [r326485-326544] David Vossel + + * channels/chan_sip.c: Fixes newlines from being stripped from out + of dialog sip MESSAGES. + + * /, res/res_timing_timerfd.c: Merged revisions 326484 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 + Jul 2011) | 10 lines Reverts fix for timerfd locking issue. jrose + discovered a performance issue with this fix that prevents his + analog phones from working when using timerfd as a timing source. + Until it is understood what is causing this performance problem, + this patch is being reverted. ........ + +2011-07-05 22:11 +0000 [r326412] Tilghman Lesher + + * channels/chan_jingle.c, channels/chan_dahdi.c, + funcs/func_speex.c, /, channels/chan_sip.c, codecs/codec_speex.c, + funcs/func_aes.c, pbx/pbx_dundi.c, channels/chan_gtalk.c, + apps/app_queue.c, channels/chan_iax2.c, res/res_jabber.c, + apps/app_stack.c, channels/chan_mgcp.c, apps/app_voicemail.c: + Merged revisions 326411 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) + | 14 lines Add the attribute "type" to each "" for + menuselect. This matters only when autoconf fails to detect that + weak linking is supported. External optional dependencies will + become optional in both cases, as they are removed at compile + time when not detected. However, runtime-optional modules are + made mandatory when weak linking is not found. This change + affects only the external optional dependencies; previously, they + were incorrectly required when weak linking support was not + detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt + by tilghman (License #5003) Tested by: iasgoscouk ........ + +2011-07-05 20:25 +0000 [r326368] Kinsey Moore + + * contrib/scripts/file.convert.sh (added): Prompt conversion script + Several variables in the script control which files are converted + and the source and destination formats. Patch-by: Trey Blancher + (closes AST-560) + +2011-07-05 17:35 +0000 [r326321] Richard Mudgett + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged + revisions 326291 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) + | 23 lines Used auth= parameter freed during "sip reload" causes + crash. If you use the auth= parameter and do a "sip reload" while + there is an ongoing call. The peer->auth data points to free'd + memory. The patch does several things: 1) Puts the authentication + list into an ao2 object for reference counting to fix the + reported crash during a SIP reload. 2) Converts the + authentication list from open coding to AST list macros. 3) Adds + display of the global authentication list in "sip show settings". + (closes issue ASTERISK-17939) Reported by: wdoekes Patches: + jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by + rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA + SWP-3526 ........ + +2011-07-05 16:46 +0000 [r326267] Mark Murawki + + * main/manager.c, CHANGES: New feature: AMI Action FilterAdd This + adds a new action, FilterAdd to the manager interface that allows + control over event filters for the current session (closes issue + ASTERISK-16795) Reported by: kobaz Tested by: kobaz,loloski + +2011-07-05 13:38 +0000 [r326210] Matthew Jordan + + * /, main/file.c: Merged revisions 326209 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) + | 7 lines Updated filestream destructor to block until move is + complete when cache is used When a cache directory is used, the + process is forked and a mv command is executed to move the + temporary file to the permanent location. This caused issues with + voicemail, where a race condition occurred when the parent + expected the file to be in the permanent location prior to the mv + command completing. The parent process is now blocked until the + mv command completes. (closes issue ASTERISK-17724) Reported by: + Adiren P. Tested by: mjordan ........ + +2011-07-01 21:11 +0000 [r326145] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 326144 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) + | 16 lines Better way to get chan and pvt lock for issue + ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock + fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the + channels in the defined order and avoid the need for a deadlock + avoidance loop. * Lock the channel before getting the pointer to + the private structure to be sure that the pointer will not change + due to a masquerade or channel hangup. * To preserve sanity, + check that chan and p->owner are the same. (Pointer rearangements + should not happen without the protection of locks because bad + things tend to happen otherwise.) ........ + +2011-07-01 16:36 +0000 [r326056-326101] Gregory Nietsky + + * CHANGES: Change CHANGES move the commits to the right place + r296249 r318141 Application changes + + * CHANGES: Change CHANGES move the commits to the right place in + the file missed in review + +2011-07-01 12:45 +0000 [r326006] Matthew Nicholson + + * res/res_fax.c, res/res_fax_spandsp.c: updated irroots info for + the authors section + +2011-06-30 21:05 +0000 [r325937] David Vossel + + * channels/chan_bridge.c: Fixes warning message caused by + confbridge playback chan not being answered. + +2011-06-30 20:47 +0000 [r325936] Richard Mudgett + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 325935 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) + | 11 lines Misc minor changes in chan_sip. * Add load failure + exit if primary SIP container(s) could not get created in + chan_sip.c:load_module(). * Removed a redundant static prototype. + * Some typos. * Some whitespace. ........ + +2011-06-30 20:33 +0000 [r325931] David Vossel + + * configs/confbridge.conf.sample, + apps/confbridge/include/confbridge.h, + include/asterisk/bridging.h, include/asterisk/dsp.h, + bridges/bridge_softmix.c, apps/app_confbridge.c, CHANGES, + main/bridging.c, main/dsp.c, apps/app_voicemail.c, + apps/confbridge/conf_config_parser.c: Video support for + ConfBridge. Review: https://reviewboard.asterisk.org/r/1288/ + +2011-06-30 20:24 +0000 [r325900] Matthew Jordan + + * /, apps/app_voicemail.c: Merged revisions 325877 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30 + Jun 2011) | 9 lines Patched voicemail user option for emailbody / + emailsubject Incorporated changes per ASTERISK-16795; updated + unit tests to check for vmu->emailbody / vmu->emailsubject + (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: + mjordan ........ + +2011-06-30 19:31 +0000 [r325864] Jonathan Rose + + * /, res/res_musiconhold.c: Merged revisions 325821 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun + 2011) | 10 lines Fixes an issue with Music on Hold classes losing + files in playlist when realtime is used. The bug occurs rather + intermittently and I relied on the reporters to test the patch. + After a sanity check and some testing, I'm giving it an OK. + (closes issue ASTERISK-17875) Reported by: David Cunningham + Patches: res_musiconhold.c.mohrt17875_v1 uploaded by Igor + Goncharovsky (license #5009) ........ + +2011-06-30 18:22 +0000 [r325815-325816] Matthew Nicholson + + * res/res_fax.c, include/asterisk/res_fax.h, CHANGES, + res/res_fax_spandsp.c: Fax gateway functionality (i.e. + translating between a T.30 terminal and a T.38 terminal). Can be + enabled on a channel by setting FAXOPT(gateway)=yes in the + dialplan. Big thanks to irroot for porting this code to use the + framehooks api. + + * main/frame.c: copy all flags on asterisk frames instead of just + the timing flag + +2011-06-29 21:50 +0000 [r325741] Kinsey Moore + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged + revisions 325740 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) | + 7 lines chan_sip: cleanup from the introduction of ast_str Remove + the length field from sip_req and sip_pkt in chan_sip since they + are redundant (ast_str holds its own length) and refactor the + necessary functions. Review: + https://reviewboard.asterisk.org/r/1281/ ........ + +2011-06-29 19:02 +0000 [r325674] David Vossel + + * /, res/res_timing_timerfd.c: Merged revisions 325673 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325673 | dvossel | 2011-06-29 13:59:33 -0500 (Wed, 29 + Jun 2011) | 6 lines Fixes timerfd locking issue. (closes + ASTERISK-17867, ASTERISK-17415) Patches: fix uploaded by kobaz + Review: https://reviewboard.asterisk.org/r/1255/ ........ + +2011-06-29 18:18 +0000 [r325611-325616] Richard Mudgett + + * /, apps/app_queue.c: Merged revisions 325614 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325614 | rmudgett | 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011) + | 5 lines Fixed some error exit cleanup in app_queue.c. * Fixed + error exit cleanup in app_queue.c copy_rules() and + reload_queue_rules(). ........ + + * /, apps/app_queue.c: Merged revisions 325610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) + | 18 lines Response to QueueRule manager command does not contain + ActionID if it was specified. * Add ActionID support as + documented for the QueueRule AMI action. * Remove documentation + for ActionID with the Queues AMI action. The output does not + follow normal AMI response output and there is no place to put an + ActionID header. (closes issue AST-602) Reported by: Vlad + Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621) + patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett + Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575 + ........ + +2011-06-29 16:19 +0000 [r325538-325547] Matthew Nicholson + + * main/channel.c, /: Merged revisions 325545 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun + 2011) | 2 lines make framehooks prevent native bridging (for real + this time) ........ + + * apps/app_dial.c, main/rtp_engine.c, /: Merged revisions 325537 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun + 2011) | 2 lines don't do native/remote bridging if a framehook is + active on the channel ........ + +2011-06-29 06:39 +0000 [r325483] Gregory Nietsky + + * configs/queues.conf.sample, UPGRADE.txt, CHANGES, + apps/app_queue.c: Commit "distrotech" app_queue changes to Trunk + * Added general option negative_penalty_invalid default off. when + set members are seen as invalid/logged out when there penalty is + negative. for realtime members when set remove from queue will + set penalty to -1. * Added queue option autopausedelay when + autopause is enabled it will be delayed for this number of + seconds since last successful call if there was no prior call the + agent will be autopaused immediately. * Added member option + ignorebusy this when set and ringinuse is not will allow per + member control of multiple calls as ringinuse does for the Queue. + - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for + realtime members - QUEUE_MEMBER is now R/W supporting setting + paused, ignorebusy and penalty. (closes issue ASTERISK-17421) + (closes issue ASTERISK-17391) Reported by: irroot Tested by: + irroot, jrose Review: https://reviewboard.asterisk.org/r/1119/ + +2011-06-28 21:51 +0000 [r325417] Kevin P. Fleming + + * /, channels/chan_sip.c: Merged revisions 325416 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun + 2011) | 3 lines Fix random misspelling noticed on asterisk-users. + ........ + +2011-06-28 20:32 +0000 [r325345] David Vossel + + * /, channels/chan_sip.c: Merged revisions 325339 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) + | 4 lines Fixes locking inversion caused by holding sip pvt lock + during async_goto. (closes ASTERISK-17352) ........ + +2011-06-28 17:38 +0000 [r325213] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 325212 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 + Jun 2011) | 7 lines Use the device name and not the channel name + to initialize the device state. Correct ASTERISK-11323 + implementation as I don't see how it ever worked as claimed when + it used the channel name and not the device name. (issue + ASTERISK-11323) ........ + +2011-06-28 16:04 +0000 [r325153] Jonathan Rose + + * /, res/res_musiconhold.c: Merged revisions 325152 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325152 | jrose | 2011-06-28 10:46:29 -0500 (Tue, 28 Jun + 2011) | 5 lines Fixes moh reload breaking custom mode moh classes + when the config file is untouched (closes issue ASTERISK-17730) + Reported by: sdolloff ........ + +2011-06-28 15:34 +0000 [r325151] David Vossel + + * channels/chan_sip.c: Fixes issue with video and text not being + reinvited correctly with directmedia If a SDP does not modify the + session, we ignore it. However, we were defaulting no text and + video support to true before checking to see if the sdp modified + anything or not. This would result in process_sdp ignoring an sdp + but removing video and text from the call during direct media + reinvites. + +2011-06-28 15:12 +0000 [r325092] Leif Madsen + + * /, build_tools/prep_tarball: Merged revisions 325091 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325091 | lmadsen | 2011-06-28 10:12:00 -0500 (Tue, 28 + Jun 2011) | 1 line Remove line from prep_tarball that kills + mkrelease. ........ + +2011-06-28 00:07 +0000 [r325046] Terry Wilson + + * channels/chan_sip.c: Don't forget to build the Via when sending + MESSAGE + +2011-06-27 16:32 +0000 [r324961] Tilghman Lesher + + * /, main/asterisk.c: Merged revisions 324955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) + | 5 lines Save and restore errno from within signal handlers. + This is recommended by the POSIX standard, as well as by the + sigaction(2) manpage for various platforms that we support (e.g. + Mac OS X). ........ + +2011-06-27 15:38 +0000 [r324915] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 324914 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) + | 21 lines When subscribing MWI to an unsolicited mailbox the + first notification is incorrect. A remote peer subscribed to MWI + with the unsolicited option and a local phone subscribed to the + remote mailbox. The notify message-summary events are sent + correctly except for the first one when subscribing, which will + always be 0. This means the phone MWI indicator will be wrong + until the mailbox read/unread count changes and the event is + fired. Looks like this is a regression from ASTERISK-16149. * Fix + the logic to check the cache and if allowed then fallback to + manually counting mailbox messages. (closes issue ASTERISK-17997) + Reported by: rsw686 Patches: jira_asterisk_17997_v1.8.patch + (license #5621) uploaded by rmudgett Tested by: rsw686 JIRA + SWP-3551 ........ + +2011-06-24 20:50 +0000 [r324850] Richard Mudgett + + * /, pbx/pbx_config.c: Merged revisions 324849 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011) + | 15 lines Syntax errors in dialplan do not display the file + name. When issuing the CLI command "dialplan reload" syntax + errors and warnings are displayed on the console. The offending + line number is displayed on the console, but the file name is not + displayed. Errors caught in main/config.c do display the file + name. (closes issue ASTERISK-17985) Reported by: ulogic Patches: + pbx_config.patch uploaded by ulogic (License #5685) modified + format Tested by: rmudgett JIRA SWP-3554 ........ + +2011-06-24 16:50 +0000 [r324769] Jonathan Rose + + * include/asterisk/logger.h, /: Merged revisions 324768 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | + 11 lines DTMF wasn't being logged on connected consoles when + enabled in logger.conf Previously in order for DTMF to be logged + in a connected console session, the user would have to do logger + set channel DTMF on. This corrects that so that it is on by + default. This issue was caused by an off by one error incurred by + a logger level count of 6 in logger.h where it should have been + 7. (closes issue: ASTERISK-17974) Reported by: Luke H ........ + +2011-06-23 18:56 +0000 [r324708-324709] Kinsey Moore + + * apps/app_confbridge.c: ConfBridge: redundant code cleanup There + is no reason to clean up features twice. Review: + https://reviewboard.asterisk.org/r/1279/ + + * /, channels/chan_sip.c: Merged revisions 324678 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r324678 | kmoore | 2011-06-23 13:29:17 -0500 + (Thu, 23 Jun 2011) | 11 lines Merged revisions 324643 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | + 4 lines Addresses AST-2011-008, memory corruption and remote + crash in SIP driver. AST-2011-008 ........ ................ + +2011-06-23 18:31 +0000 [r324664-324689] David Vossel + + * /, channels/sip/reqresp_parser.c: Merged revisions 324685 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) + | 8 lines Fixes sip crash when calling remove_uri_parameters with + NULL AST-2011-009 (closes issue ASTERISK-18017) Reported by: + jaredmauch ........ + + * /, main/features.c, channels/chan_iax2.c, + include/asterisk/frame.h: Merged revisions 324652 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500 + (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 + (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) + | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver + Thanks to twilson for identifying the issue and providing the + patches. AST-2011-010 ........ ................ ................ + +2011-06-23 03:16 +0000 [r324558] Terry Wilson + + * /, tests/test_netsock2.c: Merged revisions 324557 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 + Jun 2011) | 5 lines Remove tests for parsing address with invalid + port getaddrinfo on OS X returns with EAI_NONAME error when + passed a port greater than 65535. Linux throws no error, so + remove the tests for now. ........ + +2011-06-22 19:17 +0000 [r324495] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 324491 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) + | 1 line Use correct variable for text SRTP media. ........ + +2011-06-22 19:12 +0000 [r324487] Terry Wilson + + * main/netsock2.c, /, channels/chan_sip.c, + include/asterisk/netsock2.h, tests/test_netsock2.c (added): + Merged revisions 324484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) + | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages + The idea behind the patch listed below was used, but in a more + targeted manner. There are now address stringification functions + for addresses that are meant to be sent to a remote party. + Link-local scope-ids only make sense on the machine from which + they originate and so are stripped in the new functions. There is + also a host sanitization function added to chan_sip which is used + for when peer and dialog tohost fields or sip_registry hostnames + are used to craft a SIP message. Also added are some basic unit + tests for netsock2 address parsing. (closes issue ASTERISK-17711) + Reported by: ch_djalel Patches: + asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel + (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ + ........ + +2011-06-22 18:45 +0000 [r324480-324482] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 324481 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 Also fixed a + reference leak in an error path in sip_msg_send(). ........ + r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) + | 19 lines Timout or error on INFO or MESSAGE transaction causes + call to be lost. When exchanging INFO messages within a call, 4xx + error causes the call to be disconnected although RFC 2976 + explicitly states that such transactions do not modify the state + of the dialog. When exchanging MESSAGE messages within a call, + 4xx error causes the call to be disconnected. To provide least + surprise, we should not disconnect the call since a MESSAGE is + like INFO in this case. (Implied by RFC 3428 Section 2) (closes + issue ASTERISK-17901) Reported by: neutrino88 Review: + https://reviewboard.asterisk.org/r/1257/ Review: + https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486 ........ + + * /, channels/chan_sip.c: Merged revisions 324479 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) + | 1 line Comments and whitespace in chan_sip.c ........ + +2011-06-21 21:55 +0000 [r324365-324422] David Vossel + + * apps/app_confbridge.c: Fixes issue with channel write format + being incorrectly restored when MOH is used in confbridge. + + * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 324364 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) + | 10 lines Fixes locking inversion issue in ast_async_goto() + During this function we can not hold the "chan" lock while doing + the masquerade, the explicit goto on the tmp chan, or the channel + alloc. Instead we need to get the channel lock, store off + information about the channel that we need, and then let the + channel lock go for the remainder of the function. Review: + https://reviewboard.asterisk.org/r/1275/ ........ + +2011-06-21 16:06 +0000 [r324304] Kinsey Moore + + * apps/app_confbridge.c: ConfBridge does not handle hangup properly + When playing back a prompt to a channel, confbridge neglects to + check for hangup events causing lockup condititions for hangups + that occur before actually joining the conference. This change + ensures that the user is removed from the conference in the event + of a premature hangup. Review: + https://reviewboard.asterisk.org/r/1277/ + +2011-06-21 15:49 +0000 [r324302] David Vossel + + * channels/chan_sip.c: Fixes issue with finding correct extension + when message context is used. + +2011-06-20 18:13 +0000 [r324242] Leif Madsen + + * /, configs/queuerules.conf.sample: Merged revisions 324241 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) + | 2 lines Remove extra 'the'. Reported by Vlad Povorozniuc + ........ + +2011-06-20 17:34 +0000 [r324238] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 324237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) + | 12 lines Ignore media offers with a port of 0 Section 5.1 of + RFC3264 states: A port number of zero in the offer indicates that + the stream is offered but MUST NOT be used. (closes issue + ASTERISK-17845) Reported by: jacco Patches: issue19281_2.patch + uploaded by jacco (license 1277) Tested by: jacco, twilson + ........ + +2011-06-17 18:52 +0000 [r324177-324179] Leif Madsen + + * main/manager.c, /: Merged revisions 324178 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324178 | lmadsen | 2011-06-17 14:51:16 -0400 (Fri, 17 Jun 2011) + | 2 lines Add Username and Secret fields to manager Login action. + Pointed out by Vlad Povorozniuc ........ + + * /, apps/app_meetme.c: Merged revisions 324176 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) + | 2 lines Fix typo in documentation. Pointed out by Vlad + Povorozniuc ........ + +2011-06-17 18:23 +0000 [r324175] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 324174 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 + Jun 2011) | 5 lines Add header string to libpri debug output. Add + header string to libpri debug output so the libpri output can be + found/extracted easier from huge debug trace files. ........ + +2011-06-17 15:32 +0000 [r324131] Leif Madsen + + * main/pbx.c, /: Merged revisions 324115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324115 | lmadsen | 2011-06-17 11:14:54 -0400 (Fri, 17 Jun 2011) + | 3 lines Fix grammar in documentation for Goto() and GotoIf() + (closes issue ASTERISK-18023) Reported by: Tim Osman ........ + +2011-06-16 22:49 +0000 [r324050] Terry Wilson + + * main/channel.c, channels/chan_local.c, /, channels/chan_sip.c, + include/asterisk/channel.h: Merged revisions 324048 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 + Jun 2011) | 8 lines Lock the channel before calling the setoption + callback The channel needs to be locked before calling these + callback functions. Also, sip_setoption needs to lock the pvt and + a check p->rtp is non-null before using it. Review: + https://reviewboard.asterisk.org/r/1220/ ........ + +2011-06-16 18:13 +0000 [r323991] Richard Mudgett + + * /, tests/test_event.c: Merged revisions 323990 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323990 | rmudgett | 2011-06-16 13:12:32 -0500 (Thu, 16 Jun 2011) + | 5 lines The test_event unit test is occasionally failing. Wait + for the special posted event to process before adding a new + subscription. ........ + +2011-06-16 15:59 +0000 [r323673-323933] Terry Wilson + + * Makefile, /: Merged revisions 323932 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323932 | twilson | 2011-06-16 10:58:22 -0500 (Thu, 16 Jun 2011) + | 4 lines Don't assume ASTDBDIR exists It most likely doesn't on + FreeBSD ........ + + * /, tests/test_db.c: Merged revisions 323866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323866 | twilson | 2011-06-15 15:03:58 -0500 (Wed, 15 Jun 2011) + | 2 lines Remove now-useless cast of ARRAY_LEN ........ + + * include/asterisk/utils.h, /: Merged revisions 323863 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r323863 | twilson | 2011-06-15 14:58:18 -0500 (Wed, 15 + Jun 2011) | 2 lines Make ARRAY_LEN() return the same type on x86 + and x86_64 systems ........ + + * /, tests/test_db.c: Merged revisions 323859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323859 | twilson | 2011-06-15 14:45:20 -0500 (Wed, 15 Jun 2011) + | 2 lines Fix more ARRAY_LEN format string issues ........ + + * /, main/features.c: Merged revisions 323754 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r323754 | twilson | 2011-06-15 13:21:52 -0500 + (Wed, 15 Jun 2011) | 23 lines Merged revisions 323733 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r323733 | twilson | 2011-06-15 13:13:00 -0500 + (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) + | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a + recent DTMF change. This patch makes sure that dynamic features + are also checked when deciding whether or not to pass DTMF + through or store it for interpreting. (closes issue + ASTERISK-17914) Reported by: vrban ........ ................ + ................ + + * /, tests/test_db.c: Merged revisions 323672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323672 | twilson | 2011-06-15 10:09:51 -0700 (Wed, 15 Jun 2011) + | 5 lines Cast ARRAY_LEN to size_t for ast_logging 32-bit and + 64-bit machines return different types for ARRAY_LEN(), so cast + it before using in a format string. ........ + +2011-06-15 16:49 +0000 [r323671] Richard Mudgett + + * /, tests/test_event.c, main/event.c: Merged revisions + 323669-323670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) + | 21 lines [regression] Voicemail MWI is no longer sent. When + leaving a voicemail, the MWI message is never sent. The same + thing happens when checking a voicemail and marking it as read. + If you restart Asterisk, everything comes up at that state + correctly, but changes to the messages in voicemail causes the + light to not be set appropriately. Very easy to reproduce. * Made + ast_event_check_subscriber() return TRUE if there are ANY + subscribers to an event type when there are no restricting ie + values passed. This allows an event being queued to be queued. + (closes issue ASTERISK-18002) Reported by: lmadsen Tested by: + lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded + by rmudgett (License #5621) (closes issue ASTERISK-18019) + ........ r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 + Jun 2011) | 7 lines Add a test to the event unit tests to catch + ASTERISK-18002. The new tests check to see if there are ANY + subscribers to the event type when ast_event_check_subscriber() + is not passed any specific ie values. (issue ASTERISK-18002) + ........ + +2011-06-15 16:19 +0000 [r323621] Jonathan Rose + + * res/res_config_pgsql.c, /: Merged revisions 323610 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r323610 | jrose | 2011-06-15 11:09:24 -0500 (Wed, 15 Jun + 2011) | 7 lines Adds PQclear calls on result to various parts of + res_conf_pgsql (closes issue ASTERISK-17812) Reported by: + byronclark Patches: pgsql_pqclear.patch uploaded by byronclark + (license 1200) ........ + +2011-06-15 15:33 +0000 [r323609] Sean Bright + + * main/manager.c, /: Merged revisions 323608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 + (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 + (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun + 2011) | 25 lines Resolve a segfault/bus error when we try to map + memory that falls on a page boundary. The fix for ASTERISK-15359 + was incorrect in that it added 1 to the length of the mmap'd + region. The problem with this is that reading/writing to that + extra byte outside of the bounds of the underlying fd causes a + bus error. The real issue is that we are working with both a FILE + * and the raw fd underneath it and not synchronizing between + them. The code that was removed in ASTERISK-15359 was correct, + but we weren't flushing the FILE * before mapping the fd. Looking + at the manager code in 1.4 reveals that the FILE * in 'struct + mansession' is never used except to create a temporary file that + we immediately fdopen. This means we just need to write a 0 byte + to the fd and everything will just work. The other branches + require a call to fflush() which, while not a guaranteed fix, + should reduce the likelihood of a crash. This all makes sense in + my head. (closes issue ASTERISK-16460) Reported by: + Ravelomanantsoa Hoby (hoby) Patches: + issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license + #5060) ........ ................ ................ + +2011-06-15 13:45 +0000 [r323517] Kinsey Moore + + * apps/app_confbridge.c, CHANGES: CONFBRIDGE_INFO function to get + conference data Added the CONFBRIDGE_INFO dialplan function to + get information about a conference bridge including locked status + and number of parties, admins, and marked users. Review: + https://reviewboard.asterisk.org/r/1271/ + +2011-06-15 00:51 +0000 [r323397-323457] Richard Mudgett + + * /, main/event.c: Merged revisions 323456 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323456 | rmudgett | 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011) + | 1 line Add missing break in ast_event_get_cached(). ........ + + * main/netsock2.c, main/dnsmgr.c, /: Merged revisions 323392,323394 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323392 | rmudgett | 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011) + | 6 lines Add more strict hostname checking to + ast_dnsmgr_lookup(). Change suggested in review. Review: + https://reviewboard.asterisk.org/r/1240/ ........ r323394 | + rmudgett | 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) | 2 lines + Made ast_sockaddr_split_hostport() port warning msgs more + meaningful. ........ + +2011-06-14 17:03 +0000 [r323374] Terry Wilson + + * res/res_rtp_asterisk.c, main/rtp_engine.c, /, + channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged + revisions 323370 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) + | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine + conversion left out support for handling rtpkeepalives. This + patch adds them back. (closes issue ASTERISK-17304) Reported by: + lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ + +2011-06-14 16:47 +0000 [r323372] Jonathan Rose + + * /, channels/chan_sip.c: Merged revisions 323371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | + 12 lines Changes contact use in build_peer to use the FORCE_RPORT + flag instead of RPORT_PRESENT It turned out that this was causing + NAT=Yes to always use rport when present which was against 1.6.2 + behavior and the check itself was redundant since the only way + this segment of code could be reached was if RPORT_PRESENT was + already evaluated as true earlier. (closes issue ASTERISK-17789) + Reported by: byronclark Patches: use_sip_nat_force_rport.patch + uploaded by byronclark (license 1200) ........ + +2011-06-14 14:37 +0000 [r323325] David Vossel + + * channels/chan_sip.c: Store sip peer name as var data on a + outofcall msg. + +2011-06-13 20:44 +0000 [r323272] Kinsey Moore + + * apps/confbridge/conf_config_parser.c: Config inheritance doesn't + work with ConfBridge() menu definitions Current behavior in + ConfBridge menu definitions is that first definition takes + precedence, even in templated situations. This change allows + inheritance and overriding to work as expected so that the last + definition takes precedence. (closes ASTERISK-17986) Review: + https://reviewboard.asterisk.org/r/1267/ + +2011-06-13 19:54 +0000 [r323214] Leif Madsen + + * main/channel.c, /: Merged revisions 323213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323213 | lmadsen | 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011) + | 6 lines Avoid dividing by zero with L() option to Dial() + Reported by: nicolasom Patches: issue-17995.patch - nicolasom + (License #5994) ........ + +2011-06-13 19:43 +0000 [r323212] David Vossel + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: Addition of + "outofcall_message_context" sip.conf option. Review: + https://reviewboard.asterisk.org/r/1265/ + +2011-06-13 19:03 +0000 [r323155] Leif Madsen + + * /, res/res_agi.c: Merged revisions 323154 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323154 | lmadsen | 2011-06-13 15:00:41 -0400 (Mon, 13 Jun 2011) + | 6 lines Tweak documentation for AGI Hangup command. (closes + issue ASTERISK-17999) Reported by: Ben Klang Patches: + hangup-doc.diff - uploaded by Ben Klang (License #5876) ........ + +2011-06-13 14:38 +0000 [r323106-323107] Kinsey Moore + + * apps/confbridge/include/confbridge.h, apps/app_confbridge.c: MOH + for only user not working with ConfBridge This adds the + playing_moh flag to the conference_bridge_user struct that + signifies when MOH should be playing so code doesn't have to + guess whether MOH is playing. This change also adds the necessary + checking to ensure that MOH continues playing for a single user + in a conference after the join sound is played when configured to + do so. (closes ASTERISK-17988) Review: + https://reviewboard.asterisk.org/r/1263/ + + * apps/app_confbridge.c: ConfBridge: Use of bridge or user profiles + that don't exist Bridge and user profiles are not checked for + existence before use. The lack of a fully formed bridge profile + can cause a segfault when sounds are accessed. This change + ensures that bridge and user profiles exist prior to usage + attempts. Review: https://reviewboard.asterisk.org/r/1264/ + +2011-06-10 19:22 +0000 [r323041] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 323040 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun + 2011) | 5 lines Unlock the sip channel during fax detection like + chan_dahdi does to prevent a deadlock with ast_autoservice_stop. + (closes issue ASTERISK-17798) tested by mnicholson ........ + +2011-06-10 15:30 +0000 [r322866-322982] Terry Wilson + + * /, main/db.c: Merged revisions 322981 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) + | 11 lines Avoid a DB1 infinite loop bug Explicity check the last + entry in the DB and make sure that we don't iterate past it. + Since there can be no duplicates, this just makes sure that we + stop after matching the last key. This patch also refactors the + code to get away from some code duplication. A previous patch + added many astdb tests and this patch passed them. Review: + https://reviewboard.asterisk.org/r/1259/ ........ + + * /, tests/test_db.c (added): Merged revisions 322923 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r322923 | twilson | 2011-06-09 19:33:23 -0700 (Thu, 09 + Jun 2011) | 2 lines Add some astdb unit tests ........ + + * /, include/asterisk/astdb.h: Merged revisions 322865 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r322865 | twilson | 2011-06-09 15:29:20 -0700 (Thu, 09 + Jun 2011) | 4 lines Correct ast_db_deltree documentation + ast_db_deltree returns -1 on error, otherwise the number of + deletions ........ + +2011-06-09 17:43 +0000 [r322808] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 322807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322807 | mnicholson | 2011-06-09 12:37:07 -0500 (Thu, 09 Jun + 2011) | 5 lines don't drop any voice frames when checking for + T.38 during early media (closes issue ASTERISK-17705) Review: + https://reviewboard.asterisk.org/r/1186/ patch by oej reported by + oej ........ + +2011-06-09 16:47 +0000 [r322750] Richard Mudgett + + * /, apps/app_directed_pickup.c, main/features.c, + include/asterisk/features.h: Merged revisions 322749 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 + Jun 2011) | 15 lines Remove potential deadlock in call pickup + race. Deadlock is possible in ast_do_pickup() when holding the + target channel lock and trying to get the chan channel lock. + Also, holding the target lock when calling + ast_channel_masquerade() is not a good idea because that routine + does deadlock avoidance. * Removed the need to hold the target + lock after marking the target with a datastore and getting the + connected line data off of the target channel. * Moved + can_pickup() to ast_can_pickup() in features.c. Now all the call + pickup methods use the same basic call pickup availability check. + Review: https://reviewboard.asterisk.org/r/1234/ ........ + +2011-06-09 11:05 +0000 [r322544] Damien Wedhorn + + * channels/chan_skinny.c: Add autoanswer to skinny. Autoanswer + added to skinny based on incoming chan var SKINNY_AUTOANSWER. + Initial value must be the time to autoanswer in ms, then + optionally :BEEP to play a tone when answered and :MUTE to mute + the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs, + then answer, mute the mic, and play a beep. just 3000 would + answer afer 3 secs of ringing with no beep and full two way + audio. + +2011-06-08 20:48 +0000 [r322426-322485] Richard Mudgett + + * /, apps/app_queue.c: Merged revisions 322484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) + | 15 lines Ring all queue with more than 255 agents will cause + crash. 1. Create a ring-all queue with 500 permanent agents. 2. + Call it. 3. Asterisk will crash. The watchers array in + app_queue.c has a hard limit of 255. Bounds checking is not done + on this array. No sane person should put 255 people in a ring-all + queue, but we should not crash anyway. * Added bounds checking to + the watchers array. JIRA AST-464 JIRA SWP-2903 ........ + + * main/dnsmgr.c, /: Merged revisions 322425 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) + | 16 lines SRV lookup attempted for SIP peers listed as an IP + address. Asterisk attempts to SRV lookup a host name even if the + host name is an IP address. Regression introduced when IPv6 + support was added. * Restored the check in ast_dnsmgr_lookup() to + see if the given host name is an IP address. The IP address could + be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815) + Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett + Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett + (License #5621) Review: https://reviewboard.asterisk.org/r/1240/ + ........ + +2011-06-08 11:38 +0000 [r322381] Damien Wedhorn + + * channels/chan_skinny.c: Remove skinny do_monitor and use + ast_sched_start instead The do_monitor seemed to be there for + task scheduling and network monitoring. However, the network + monitoring has a dedicated thread so the ast_io_wait was + basically just a usleep as it didn't actually seem to be + monitoring anything. Review: + https://reviewboard.asterisk.org/r/1256/ + +2011-06-08 06:45 +0000 [r322323] Gregory Nietsky + + * /, channels/chan_sip.c: Merged revisions 322322 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | + 18 lines Make handle_request_publish do dialog expiration and + destruction. This patch fixes handle_request_publish so that it + does dialog expiration and destruction. Without this patch the + incoming PUBLISH requests will get stuck in the dialog list. + Restarting asterisk is the only way to remove them. Personal + observation on one system the server hung up while looping + through the channels rendering asterisk unusable and all sip + phones unregisterd when they try reregister more requests are + added. (closes issue #18898) Reported by: gareth Tested by: + loloski, Chainsaw, wimpy, se, kuj, irroot Jira: + https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review: + https://reviewboard.asterisk.org/r/1253 ........ + +2011-06-07 23:14 +0000 [r322284] Richard Mudgett + + * channels/chan_sip.c, include/asterisk/message.h: Correct some + whitespace and a reference debug message. + +2011-06-07 19:17 +0000 [r322244] Russell Bryant + + * res/res_jabber.c: Actually check the "sendtodialplan" option + setting for xmpp. (closes issue ASTERISK-17978) Reported by: + elguero Patches: stop_messages_going_to_dialplan.patch (license + #5026) + +2011-06-07 18:01 +0000 [r322190] Paul Belanger + + * configs/sip_notify.conf.sample, /: Merged revisions 322189 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun + 2011) | 4 lines Use correct syntax for 'sip notify snom-reboot' + (closes issue ASTERISK-17915) ........ + +2011-06-06 19:39 +0000 [r322111-322128] Gregory Nietsky + + * apps/app_queue.c: Remove Unused Var Warning + rt_handle_member_record + + * apps/app_queue.c: Refactor rt_handle_member_record Review: + https://reviewboard.asterisk.org/r/1172 + +2011-06-06 19:15 +0000 [r322070] Jonathan Rose + + * include/asterisk/logger.h, /, main/asterisk.c: Merged revisions + 322069 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) | + 8 lines Fixes level toggling for logger set levels since it was + reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested + by: jrose, Luke H Review: + https://reviewboard.asterisk.org/r/1244/ ........ + +2011-06-03 22:15 +0000 [r321814-321927] Richard Mudgett + + * cel/cel_radius.c, /, cdr/cdr_radius.c: Merged revisions 321926 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011) + | 18 lines Asterisk crash when unloading cdr_radius/cel_radius. + The rc_openlog() API call is passed a string that is used by + openlog() to format log messages. The openlog() does not copy the + string it just keeps a pointer to it. When the module is + unloaded, the string is gone from memory. Depending upon module + load order and if the other module then has an error, a crash + happens. * Pass rc_openlog() a strdup'd string with the + understanding that there will be a small memory leak if the + cdr_radius/cel_radius modules are unloaded. * Call rc_destroy() + to free the rc handle memory when the module is unloaded. JIRA + AST-483 JIRA SWP-3062 ........ + + * /, main/ccss.c: Merged revisions 321924 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) + | 5 lines Be more explicit for CCSS generic device state event + subscription. Make CCSS generic device state event subscription + specify the AST_EVENT_IE_STATE ie exists to be safe. ........ + + * /, tests/test_event.c, main/event.c: Merged revisions 321871 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) + | 27 lines Event subscription fixes. Must commit the subscription + fixes together with the integration subscription tests. The + subscription fixes cause an erroneously passing test to fail. The + new subscription tests detect errors without the subscription + fixes. * Added missing event_names[] table entry. * Reworked + ast_event_check_subscriber()/match_sub_ie_val_to_event() to + correctly detect if a subscriber exists for the proposed event. * + Made match_ie_val() and match_sub_ie_val_to_event() check the + buffer length for RAW payload types. * Fixed error handling + memory leak in ast_event_sub_activate(), ast_event_unsubscribe(), + and ast_event_queue(). * Made ast_event_new() and + ast_event_check_subscriber() better protect themselves from an + invalid payload type. * Added container lock protection between + removing old cache events and adding the new cached event in + ast_event_queue_and_cache()/event_update_cache(). * Added new + event subscription tests. ........ + + * include/asterisk/event.h, /, channels/chan_sip.c, main/event.c, + channels/chan_iax2.c: Merged revisions 321812-321813 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 + Jun 2011) | 1 line Correct IAX2 and SIP event subscription + description string. ........ r321813 | rmudgett | 2011-06-03 + 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription + description parameter string. ........ + +2011-06-03 18:25 +0000 [r321752] Russell Bryant + + * tests/test_astobj2.c, main/astobj2.c: Fix some astobj2 iterator + breakage, add another unit test. Review: + https://reviewboard.asterisk.org/r/1254/ + +2011-06-03 13:18 +0000 [r321689] Leif Madsen + + * /, configs/queues.conf.sample: Merged revisions 321685 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) + | 5 lines Also document the 'queue-minute' option. (closes issue + #19386) Reported by: juanmol ........ + +2011-06-02 22:09 +0000 [r321617] Russell Bryant + + * channels/chan_sip.c: Fix message destination extension. Don't + send all messages to 's'. Get the destination from the request + URI. (Found using automated test cases). + +2011-06-01 23:12 +0000 [r321548] Richard Mudgett + + * main/cdr.c, /: Merged revisions 321547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321547 | rmudgett | 2011-06-01 18:11:55 -0500 (Wed, 01 Jun 2011) + | 1 line CDR comment tweaks. ........ + +2011-06-01 21:31 +0000 [r321546] Russell Bryant + + * main/channel.c, channels/chan_sip.c, configs/jabber.conf.sample, + include/asterisk/message.h (added), include/asterisk/jabber.h, + include/asterisk/channel.h, configs/sip.conf.sample, + include/asterisk/_private.h, CHANGES, res/res_jabber.c, + main/message.c (added), channels/sip/include/sip.h, + main/asterisk.c: Support routing text messages outside of a call. + Asterisk now has protocol independent support for processing text + messages outside of a call. Messages are routed through the + Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. + There are options in sip.conf and jabber.conf that enable these + features. There is a new application, MessageSend(). There are + two new functions, MESSAGE() and MESSAGE_DATA(). Documentation + will be available on the project wiki, wiki.asterisk.org. Thanks + to Terry Wilson for the assistance with development and to David + Vossel for helping with some additional testing. Review: + https://reviewboard.asterisk.org/r/1042/ + +2011-06-01 20:11 +0000 [r321538] Brett Bryant + + * /, apps/app_voicemail.c: Merged revisions 321537 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01 + Jun 2011) | 8 lines This patch fixes an issue with using the + wrong voicemail folders with greetings. (closes issue #17871) + Reported by: edhorton Patches: digium_bug_17871_2 uploaded by + fhackenberger (license 592) Tested by: edhorton, fhackenberger + ........ + +2011-06-01 10:45 +0000 [r321529] Alexandr Anikin + + * addons/chan_ooh323.c, /, addons/ooh323c/src/ooh245.c, + addons/ooh323c/src/oochannels.c: Merged revisions 321528 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 + lines Fix double alerting, add forced alerting before answer Fix + double alerting (it wasn't fixed here by issue #18542) Add forced + alerting before connect (if it wasn't before) Try to send all + packets from outgoing queue rather than one only Call goes into + clearing state when disconnect command is received (closes issue + #19361) Reported by: vmikhelson Patches: issue19361-3.patch + uploaded by may213 (license 454) Tested by: vmikhelson ........ + +2011-05-31 20:55 +0000 [r321518] Richard Mudgett + + * include/asterisk/acl.h, /, include/asterisk/dnsmgr.h: Merged + revisions 321517 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321517 | rmudgett | 2011-05-31 15:54:35 -0500 (Tue, 31 May 2011) + | 1 line Update some comments. ........ + +2011-05-31 19:01 +0000 [r321516] David Vossel + + * channels/chan_local.c, /: Merged revisions 321515 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 + May 2011) | 12 lines Chan_local locking cleanup. This patch + removes all of the unnecessary deadlock avoidance loops that + occur in chan_local. It also resolves an issue with a deadlock + triggered by local channel optimizations. (issue #18028) Review: + https://reviewboard.asterisk.org/r/1231/ ........ + +2011-05-31 16:06 +0000 [r321512] Leif Madsen + + * /, channels/chan_sip.c: Merged revisions 321511 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) + | 8 lines Enhance NOTICE message to know who couldn't access the + dialplan. (closes issue #19390) Reported by: lmadsen Patches: + __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10) + Tested by: russell ........ + +2011-05-28 00:29 +0000 [r321338-321445] Richard Mudgett + + * /, res/res_agi.c: Merged revisions 321436 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321436 | rmudgett | 2011-05-27 19:27:52 -0500 (Fri, 27 May 2011) + | 4 lines Some hagi launch cleanup. Inspired by issue 19256. This + patch would also fix the crash. ........ + + * main/srv.c, /: Merged revisions 321392 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) + | 12 lines Crash when using hagi and no servers are available. + When none of the servers returned by the SRV querey respond, + asterisk crashes. The problem is that if the loop over all the + SRV entries finishes then the srv_context has already been + cleaned up. * Make ast_srv_cleanup() check to see if the context + is already cleaned up. (closes issue #19256) Reported by: + byronclark ........ + + * /, apps/app_privacy.c, UPGRADE.txt, CHANGES: Merged revisions + 321337 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert + -r321331 and -r321332. ........ r321337 | rmudgett | 2011-05-27 + 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines The app_privacy args + have undocumented "options" position, interferes with "context" + position. * Add documention for unused "options" position to + match existing code. (closes issue #19273) Reported by: + mdavenport ........ + +2011-05-27 21:40 +0000 [r321334] Leif Madsen + + * /, main/features.c: Merged revisions 321333 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321333 | lmadsen | 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011) + | 7 lines Allow parking lot hints and musicclass to be set. + (closes issue #19378) Reported by: sboily_proformatique Patches: + pf_parkinghint_music_fix uploaded by sboily proformatique + (license 206) Tested by: russell ........ + +2011-05-27 21:37 +0000 [r321331-321332] Richard Mudgett + + * UPGRADE.txt: Add note about PrivacyManager to UPGRADE.txt + + * /, apps/app_privacy.c, CHANGES: Merged revisions 321330 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) + | 8 lines The app_privacy args have undocumented "options" + position, interferes with "context" position. * Add documention + for unused "options" position to match existing code. The + trunk(v1.10) version will remove the unused options position. + (closes issue #19273) Reported by: mdavenport ........ + +2011-05-27 16:35 +0000 [r321289] Jonathan Rose + + * /, channels/sip/reqresp_parser.c: Merged revisions 321273 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | + 3 lines markm committed a patch I was working on yesterday, this + fixes it to mesh up with suggestions by mnicholson. ........ + +2011-05-27 08:37 +0000 [r321212] Alec L Davis + + * /, main/features.c: Merged revisions 321211 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May + 2011) | 16 lines Fix *8 directed pickup locks system during + pickupsound play out move playout from sip_pickup_thread to + bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2 + threads trying to write audio to same channel. In addition fixes + choppy audio beep in issue 19177. (issue #18654) (issue #19177) + Reported by: Docent Patches: review1232-1.8.diff.txt alecdavis + (license 585) Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/1232/ ........ + +2011-05-26 21:50 +0000 [r321101-321156] Mark Murawki + + * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged + revisions 321155 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | + 10 lines Fixed build problem with dev mode enabled, which was + caused by commit 321100. Reformulated patch to be more generic. + Moved the sip uri parse variable initalization to parse_uri_full + in reqresp_parser.c. This will ensure that any use of parse uri + will have null output variables if the parse fails. (closes issue + #19346) Reported by: kobaz Tested by: kobaz,JonathanRose Review: + [full review board URL with trailing slash] ........ + + * main/netsock2.c, /, channels/chan_sip.c: Merged revisions 321100 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | + 11 lines ast_sockaddr_resolve() in netsock2.c may deref a null + pointer Added a null check in netsock2 ast_sockaddr_resolve() as + well as added default initalizers in chan_sip + parse_uri_legacy_check() to make sure that invalid uris will make + null (and not undefined) user,pass,domain,transport variables + (closes issue #19346) Reported by: kobaz Patches: netsock2.patch + uploaded by kobaz (license 834) Tested by: kobaz, Marquis + ........ + +2011-05-26 18:10 +0000 [r321045] Richard Mudgett + + * /, include/asterisk/netsock2.h: Merged revisions 321044 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321044 | rmudgett | 2011-05-26 13:10:17 -0500 (Thu, 26 May 2011) + | 1 line Update ast_sockaddr comment with an important note. + ........ + +2011-05-26 17:35 +0000 [r321043] Terry Wilson + + * main/rtp_engine.c, /: Merged revisions 321042 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321042 | twilson | 2011-05-26 10:29:54 -0700 (Thu, 26 May 2011) + | 6 lines Initialize stack-allocated ast_sockaddrs before use It + is important to always initialize ast_sockaddrs before use--even + if they are passed to ast_sockaddr_copy as the underlying storage + could be bigger than what ends up being copied--leaving part of + the data unitialized. ........ + +2011-05-26 16:54 +0000 [r321003] Russell Bryant + + * /, channels/chan_alsa.c: Merged revisions 320947 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320947 | russell | 2011-05-26 10:57:13 -0500 (Thu, 26 + May 2011) | 2 lines Remove some variables that were set but + unused. ........ + +2011-05-26 15:55 +0000 [r320946] Terry Wilson + + * main/channel.c, main/utils.c, include/asterisk/stringfields.h: + Use va_copy for stringfields The ast_string_field_build_va + functions were written to take to separate va_lists to work + around FreeBSD 4 not having va_copy defined. In the end, we don't + support anything using gcc < 3 anyway because we use va_copy all + over the place anyway. This patch just simplifies things by + removing the second va_list function arguments in favor of + va_copy. Review: https://reviewboard.asterisk.org/r/1233/ --This + line, and those below, will be ignored-- M + include/asterisk/stringfields.h M main/utils.c M main/channel.c + +2011-05-25 22:28 +0000 [r320820-320884] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 320883 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) + | 17 lines Native SIP CCSS sends bad CC cancel SUBSCRIBE message. + The SUBSCRIBE message used to cancel a CC request has incorrect + To/From SIP headers. They are reversed and the dialog tags are + the same when they should not be. If pedantic mode was disabled, + then the cancel would have succeeded despite the incorrect + message. * The SIP_OUTGOING flag was not set correctly for the + dialog and I had to move some CC subscribe handling code as a + result. * Initialized the dialog subscribed type to + CALL_COMPLETION earlier. If a CC request SUBSCRIBE message comes + in and the CC instance is not found, the 404 response was + duplicated. JIRA AST-568 JIRA SWP-3493 ........ + + * apps/app_dial.c, main/channel.c, main/manager.c, /, + apps/app_meetme.c, apps/app_fax.c, main/features.c, CHANGES, + apps/app_queue.c, UPGRADE-1.8.txt: Merged revisions 320823 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) + | 18 lines The AMI Newstate event contains different information + between v1.4 and v1.8. The addition of connected line support in + v1.8 changes the behavior of the channel caller ID somewhat. The + channel caller ID value no longer time shares with the connected + line ID on outgoing call legs. The timing of some AMI + events/responses output the connected line ID as caller ID. These + party ID's are now separate. * The ConnectedLineNum and + ConnectedLineName headers were added to many AMI events/responses + if the CallerIDNum/CallerIDName headers were also present. + (closes issue #18252) Reported by: gje Tested by: rmudgett + Review: https://reviewboard.asterisk.org/r/1227/ ........ + + * main/channel.c, /, main/format_cap.c, main/features.c, + include/asterisk/channel.h: Merged revisions 320796 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25 + May 2011) | 17 lines Give zombies a safe channel driver to use. + Recent crashes from zombie channels suggests that they need a + safe home to goto. When a masquerade happens, the physical part + of the zombie channel is hungup. The hangup normally sets the + channel private pointer to NULL. If someone then blindly does a + callback to the channel driver, a crash is likely because the + private pointer is NULL. The masquerade now sets the channel + technology of zombie channels to the kill channel driver. Related + to the following issues: (issue #19116) (issue #19310) Review: + https://reviewboard.asterisk.org/r/1224/ ........ + +2011-05-25 15:43 +0000 [r320772] Gregory Nietsky + + * funcs/func_channel.c, CHANGES: CHANNEL(pickupgroup) Allow Setting + / Reading the pickupgroup of a channel with func_channel.c + (closes issue #19045) Reported by: irroot Review: + https://reviewboard.asterisk.org/r/1148/ + +2011-05-25 00:52 +0000 [r320717] Terry Wilson + + * /, addons/chan_mobile.c: Merged revisions 320716 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320716 | twilson | 2011-05-24 17:49:10 -0700 (Tue, 24 + May 2011) | 4 lines Cast data as char * before using S_OR This is + required for compiling successfully under dev mode ........ + +2011-05-23 18:00 +0000 [r320651] Richard Mudgett + + * main/manager.c, /, CHANGES: Merged revisions 320650 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 + May 2011) | 16 lines Add ConnectedLineNum/Name headers to output + of AMI action Status. * Add ConnectedLineNum and + ConnectedLineName headers to the output of the AMI action Status. + This makes it easier to find out who the channel is connected to + without having to lookup BridgedChannel or when they are + connected to an application (e.g.: VoiceMail) which has no + bridged channel. * Bridged channels with no CallerID had "" + instead of "" output, that might be a bug as "" + was what older versions used. (closes issue #18158) Reported by: + gareth Patches: svn-292308.diff uploaded by gareth (license 208) + ........ + +2011-05-23 16:28 +0000 [r320606] David Vossel + + * main/tcptls.c, /: Merged revisions 320568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r320568 | dvossel | 2011-05-23 11:18:33 -0500 + (Mon, 23 May 2011) | 14 lines Merged revisions 320562 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011) + | 9 lines Adds missing part to the ast_tcptls_server_start fails + second attempt to bind patch. (closes issue #19289) Reported by: + wdoekes Patches: + issue19289_delay_old_address_setting_tcptls_2.patch uploaded by + wdoekes (license 717) ........ ................ + +2011-05-23 16:20 +0000 [r320579] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 320573 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320573 | tilghman | 2011-05-23 11:19:32 -0500 (Mon, 23 + May 2011) | 7 lines GNU libiconv uses symbol "libiconv_open" + instead of "iconv_open". (closes issue #19344) Reported by: + rohanl Patches: iconv-check.patch uploaded by rohanl (license + 1284) ........ + +2011-05-23 15:48 +0000 [r320561] Kevin P. Fleming + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 320560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320560 | kpfleming | 2011-05-23 10:47:14 -0500 (Mon, 23 May + 2011) | 4 lines Don't generate spurious "No: command not found" + messages when running the configure script on a system that has + neither gmime-config nor pkg-config. ........ + +2011-05-23 14:40 +0000 [r320505] Jonathan Rose + + * /, channels/chan_sip.c: Merged revisions 320504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320504 | jrose | 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) | + 10 lines Fixes segfault occuring in chan_sip.c at + __set_address_from_contact Checks to see if domain contains + anything before sending it off to ast_sockaddr_resolve which is + where the segfault was occuring due to null str. (closes issue + #18857) Reported by: sybasesql Review: + https://reviewboard.asterisk.org/r/1225/ ........ + +2011-05-22 23:36 +0000 [r320446] Tilghman Lesher + + * /, res/res_odbc.c: Merged revisions 320445 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r320445 | tilghman | 2011-05-22 18:34:57 -0500 + (Sun, 22 May 2011) | 15 lines Merged revisions 320444 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011) + | 8 lines Don't crash when the connection fails. (closes issue + #19250) Reported by: seadweller Patches: + 20110514__issue19250.diff.txt uploaded by tilghman (license 14) + Tested by: seadweller, sum ........ ................ + +2011-05-20 21:40 +0000 [r320340] David Vossel + + * main/tcptls.c, /: Merged revisions 320338 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r320338 | dvossel | 2011-05-20 16:39:36 -0500 + (Fri, 20 May 2011) | 14 lines Merged revisions 320271 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011) + | 8 lines Fixes issue with ast_tcptls_server_start failing on + second attempt to bind. (closes issue #19289) Reported by: + wdoekes Patches: + issue19289_delay_old_address_setting_tcptls.patch uploaded by + wdoekes (license 717) ........ ................ + +2011-05-20 20:53 +0000 [r320238] Richard Mudgett + + * /, apps/app_meetme.c: Merged revisions 320237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r320237 | rmudgett | 2011-05-20 15:49:03 -0500 + (Fri, 20 May 2011) | 27 lines Merged revisions 320236 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500 + (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) + | 13 lines The meetme CLI command completion leaves conferences + mutex locked. When issuing a meetme kick CLI command and an + invalid (non-existent) conference number is specified, pressing + Tab leaves the conferences mutex locked and, therefore, all + conferences deadlock. Add missing unlock. (closes issue #19336) + Reported by: zvision Patches: app_meetme.diff uploaded by zvision + (license 798) ........ ................ ................ + +2011-05-20 18:49 +0000 [r320181] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 320180 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May + 2011) | 16 lines This commit modifies the way polling is done on + TLS sockets. Because of the buffering the TLS layer does, polling + is unreliable. If poll is called while there is data waiting to + be read in the TLS layer but not at the network layer, the + messaging processing engine will not proceed until something else + writes data to the socket, which may not occur. This change + modifies the logic around TLS sockets to only poll after a failed + read on a non-blocking socket. This way we know that there is no + data waiting to be read from the buffering layer. (closes issue + #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by + mnicholson (license 96) Tested by: mnicholson ........ + +2011-05-20 18:29 +0000 [r320178] Jonathan Rose + + * /, apps/app_voicemail.c: Merged revisions 320162 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May + 2011) | 15 lines Fixes an imapfolder related crash imapfolders + being set in the general section of voicemail would cause the + inbox folder name to change. Since sound file names are made + based on the names of the folders, this would cause the audio + related to that folder name to change and if Asterisk attempted + to play it, the channel would instantly hang up when the audio + file couldn't be found. This patch searches for the name of the + folder first to leave existing behavior in tact and if that + fails, it uses the normal inbox name to get the sound file + instead. (closes issue #16104) Reported by: blkline Review: + https://reviewboard.asterisk.org/r/1215/ ........ + +2011-05-20 17:04 +0000 [r320058-320060] Richard Mudgett + + * /, main/features.c: Merged revisions 320059 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320059 | rmudgett | 2011-05-20 12:03:49 -0500 (Fri, 20 May 2011) + | 1 line Misc comment cleanup in features.c. ........ + + * main/channel.c, /, main/features.c: Merged revisions 320057 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320057 | rmudgett | 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011) + | 19 lines Crash while transferring a call during DTMF feature + timeout. When a call is being attended transferred during the + time between AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the + transferred channel becomes a zombie (so tech data is not + available), making ast_dtmf_stream() segfault when it tries to + send the DTMF digit (at least with SIP channels). Patch based on + feature-end-zombie.patch uploaded by Irontec (license 1256) * + Check for zombies when ast_channel_bridge() returns. * Guarantee + that the fo parameter value is initialized in + ast_channel_bridge() before any returns. (closes issue #19116) + Reported by: Irontec Tested by: rmudgett ........ + +2011-05-20 16:27 +0000 [r320040] Jonathan Rose + + * funcs/func_strings.c, CHANGES: Adds STRREPLACE function Adds a + new STRREPLACe function to func_strings.c that allows users to + search and replace against a variable in the dialplan. (closes + issue #18023) Reported by: wdoekes Review: + https://reviewboard.asterisk.org/r/1219/ + +2011-05-20 16:20 +0000 [r319998-320013] Richard Mudgett + + * /, apps/app_directed_pickup.c, main/features.c: Merged revisions + 320007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) + | 2 lines Change some variable names to make pickup code easier + to understand. ........ + + * /, apps/app_directed_pickup.c, main/features.c: Merged revisions + 319997 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) + | 25 lines Crash when using directed pickup applications. The + directed pickup applications can cause a crash if the pickup was + successful because the dialplan keeps executing. This patch does + the following: * Completes the channel masquerade on a successful + pickup before the application returns. The channel is now + guaranteed a zombie and must not continue executing the dialplan. + * Changes the return value of the directed pickup applications to + return zero if the pickup failed and nonzero(-1) if the pickup + succeeded. * Made some code optimizations that no longer require + re-checking the pickup channel to see if it is still available to + pickup. (closes issue #19310) Reported by: remiq Patches: + issue19310_v1.8_v2.patch uploaded by rmudgett (license 664) + Tested by: alecdavis, remiq, rmudgett Review: + https://reviewboard.asterisk.org/r/1221/ ........ + +2011-05-20 13:42 +0000 [r319867-319939] Jonathan Rose + + * /, channels/chan_sip.c, configs/sip.conf.sample, + channels/sip/include/sip.h: Merged revisions 319938 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May + 2011) | 12 lines Adds legacy_useroption_parsing to address + interoperability concerns. With the new option engaged, Asterisk + should interpret user fields with useroptions contained within + the userfield of the uri by stripping them out of the original + message whenever a semicolon is encountered in the userfield + string. (closes issue #18344) Reported by: danimal Tested by: + jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ + + * /, main/features.c: Merged revisions 319866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319866 | jrose | 2011-05-19 13:32:38 -0500 (Thu, 19 May 2011) | + 11 lines Fix Randomize option on Park() The randomize option was + generally not working like it should have at all on Park(). This + patch restores intended functionality. (closes issue #18862) + Reported by: davidw Tested by: jrose Review: + https://reviewboard.asterisk.org/r/1222/ ........ + +2011-05-19 18:12 +0000 [r319813] Mark Murawki + + * cel/cel_odbc.c, /: Merged revisions 319812 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319812 | markm | 2011-05-19 13:59:01 -0400 (Thu, 19 May 2011) | + 9 lines In cel_odbc, an uninitialized RWLIST is attempted to be + locked. Added INIT and DESTROY for the RWLIST odbc_tables (closes + issue #19331) Reported by: kobaz Patches: odbc_cel.patch uploaded + by kobaz (license 834) ........ + +2011-05-19 16:52 +0000 [r319759] Richard Mudgett + + * /, main/ccss.c: Merged revisions 319758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) + | 21 lines CCSS generic agent with POTS and ISDN phones fail + caller busy call-back test. If the following is true after a CCSS + activation: * The generic agent is for an analog phone or ISDN + phone. (Caller party) * The called party becomes available. * The + caller party is not available. When the caller party becomes + available, the caller is not alerted to the called party being + available. The generic agent still thinks the caller is busy. * + Fixed the generic agent device state event subscription to look + for all device states that are considered available. * + Encapsulated the device state test for CCSS generic device + available in cc_generic_is_device_available(). Made the generic + agent and monitor use the new function instead of the manually + coded inline equivalent. JIRA AST-559 JIRA SWP-3462 ........ + +2011-05-18 23:18 +0000 [r319530-319661] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 319654 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319654 | twilson | 2011-05-18 16:15:58 -0700 + (Wed, 18 May 2011) | 22 lines Merged revisions 319653 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r319653 | twilson | 2011-05-18 16:11:57 -0700 + (Wed, 18 May 2011) | 15 lines Merged revisions 319652 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) + | 8 lines Make sure everyone gets an unhold when a transfer + succeeds Some phones, like the Snom phones, send a hold to the + transfer target after before sending the REFER. We need to make + sure that we unhold the parties that are being connected after + the masquerade. If Local channels with the /nm option are used + when dialing the parties, hold music would still be playing on + the transfer target, even after being connected with the + transferee. ........ ................ ................ + + * /, channels/chan_sip.c: Merged revisions 319552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) + | 11 lines Unbreak the storing of registrations for restart The + fix for issue 18882 broke retrieving non-realtime peers from the + ast_db on restart/reload. This patch tries to unbreak things + while leaving the intent of the original fix intact. (closes + issue #19318) Reported by: remiq Patches: diff.txt uploaded by + twilson (license 396) Tested by: lmadsen, remiq ........ + + * apps/app_dial.c, /: Merged revisions 319529 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319529 | twilson | 2011-05-18 13:05:34 -0700 + (Wed, 18 May 2011) | 24 lines Merged revisions 319528 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r319528 | twilson | 2011-05-18 13:02:06 -0700 + (Wed, 18 May 2011) | 17 lines Merged revisions 319527 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) + | 10 lines Fix app_dial ring groups Revert part of r315643. We + need to remove the datastore here as well. The code in bridging + code will catch anything that app_dial might miss. (closes issue + #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff + uploaded by elguero (license 37) ........ ................ + ................ + +2011-05-17 22:04 +0000 [r319471] Richard Mudgett + + * /, channels/misdn/isdn_lib.c: Merged revisions 319469 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319469 | rmudgett | 2011-05-17 16:57:56 -0500 + (Tue, 17 May 2011) | 22 lines Merged revision 319468 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, + 17 May 2011) | 15 lines The mISDN HDLC mode is prevented on + dialed channels. The use of mISDN HDLC mode is prevented if the + mISDN dial technology option 'h1' is used when config option + astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which + prevents the use of HDLC mode. Instead of setting the channel to + HDLC mode it is set to transparent(no dsp, no hdlc), although + hdlc is not "no hdlc". I.e the logging message is correct, but + the if condition is not. Make check the nodsp and hdlc flags. + JIRA ABE-2787 JIRA SWP-3437 .......... ................ + +2011-05-17 21:59 +0000 [r319470] Damien Wedhorn + + * channels/chan_skinny.c: Remove extraneous line variables. The + vars were either explicitly or implicitly not used. + +2011-05-17 20:13 +0000 [r319427] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c: Option needed for Q931_IE_TIME_DATE to be + optional in CONNECT message. The NEC SV8300 rejects the + Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how + much of the current time is put in Q931_IE_TIME_DATE. * Send + date/time ie never. * Send date/time ie date only. * Send + date/time ie date and hour. * Send date/time ie date, hour, and + minute. * Send date/time ie date, hour, minute, and second. * + Send date/time ie default: Libpri will send date and hhmm only + when in NT PTMP mode to support ISDN phones. (closes issue + #19221) Reported by: kenner JIRA SWP-3396 + +2011-05-17 12:54 +0000 [r319366-319368] Leif Madsen + + * /, apps/app_voicemail.c: Merged revisions 319367 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r319367 | lmadsen | 2011-05-17 07:53:50 -0500 (Tue, 17 + May 2011) | 10 lines Don't create [general] voicemail context + when using users.conf Prior to this patch, app_voicemail would + create a [general] context when parsing users.conf. (closes issue + #18891) Reported by: pdugas Patches: + app_voicemail-ignore-general.patch uploaded by pdugas (license + 1222) app_voicemail-ignore-general-style-guidelines.patch + uploaded by seanbright (license 71) Tested by: pdugas ........ + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 319365 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319365 | lmadsen | 2011-05-17 07:39:37 -0500 (Tue, 17 May 2011) + | 6 lines Make Debian init script lsb compliant (closes issue + #18896) Reported by: manwe Patches: debian_init_lsb.patch + uploaded by manwe (license 1223) ........ + +2011-05-16 21:39 +0000 [r319316] Damien Wedhorn + + * channels/chan_skinny.c: Fix up skinny hints. Probably haven't + been working for a couple of years. May still need some more + love, but they are now working, both as a hint device and + monitoring a hint. Changes centre around the long ago change to + remove the requirement for a device name in a skinny line, and + changes to the transmit_* functions. + +2011-05-16 21:08 +0000 [r319262] Jonathan Rose + + * main/dsp.c: Merged revisions 319261 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | + 2 lines Makes busy detection in dsp.c always allow for at least + one frame (20ms) of error so that 200ms tone lengths don't get + ignored by single frame error lengths. ........ + +2011-05-16 20:41 +0000 [r319260] Richard Mudgett + + * /, main/ccss.c: Merged revisions 319259 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) + | 13 lines Deadlock between generic CCSS agent and native ISDN + CCSS. Deadlock can occur when the generic CCSS agent is deleting + duplicate CC offers and the native ISDN CC driver is processing + an incoming CC message. The cc_core_instances container lock + cannot be held when an agent or monitor callback is invoked + without the possibility of a deadlock. * Make + kill_duplicate_offers() remove the reference in cc_core_instances + outside of the container lock. JIRA AST-566 JIRA SWP-3469 + ........ + +2011-05-16 18:21 +0000 [r319212] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 319204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319204 | twilson | 2011-05-16 13:17:43 -0500 + (Mon, 16 May 2011) | 11 lines Merged revisions 319202 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) + | 4 lines Unlink a peer from peers_by_ip when expiring a + registration Review: https://reviewboard.asterisk.org/r/1218/ + ........ ................ + +2011-05-16 15:58 +0000 [r319146] David Vossel + + * /, channels/chan_sip.c: Merged revisions 319145 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319145 | dvossel | 2011-05-16 10:57:26 -0500 + (Mon, 16 May 2011) | 9 lines Merged revisions 319144 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 + May 2011) | 2 lines Fixes issue with peer ref-counting during + handle_request_subscribe. (closes issue #19293) Reported by: + irroot ........ ................ + +2011-05-16 15:54 +0000 [r319143] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 319142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319142 | mnicholson | 2011-05-16 10:53:26 -0500 (Mon, 16 May + 2011) | 8 lines Make sure tcptls_session exists before + dereferencing it. (closes issue #19192) Reported by: stknob + Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by + Chainsaw (license 723) Tested by: vois, Chainsaw ........ + +2011-05-16 14:56 +0000 [r319087] Gregory Nietsky + + * channels/chan_sip.c, res/res_fax.c, CHANGES, + channels/sip/include/sip.h: When a error in T.38 negotiation + happens or its rejected on a channel the state of the channel + reverts to unknown this should be rejected. this is important for + negotiating T.38 gateway see #13405 This patch adds a option + T38_REJECTED that behaves as T38_DISABLED except it reports state + rejected. Trivial Change to res_fax to honnor UNAVAILABLE and + REJECTED states. (closes issue #18889) Reported by: irroot Tested + by: irroot, darkbasic, mnicholson Review: + https://reviewboard.asterisk.org/r/1115 + +2011-05-16 14:38 +0000 [r319086] Paul Belanger + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + res/res_http_post.c: Merged revisions 319085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319085 | pabelanger | 2011-05-16 10:35:21 -0400 (Mon, 16 May + 2011) | 10 lines Support gmime-2.4 (closes issue #18863) Reported + by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir + (license 46) Tested by: tzafrir Review: + https://reviewboard.asterisk.org/r/1213/ ........ + +2011-05-16 14:29 +0000 [r319084] David Vossel + + * /, formats/format_wav.c: Merged revisions 319083 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r319083 | dvossel | 2011-05-16 09:26:33 -0500 (Mon, 16 + May 2011) | 5 lines Fixes Big Endian build issue. (closes issue + #19298) Reported by: tzafrir ........ + +2011-05-15 23:17 +0000 [r319024] Damien Wedhorn + + * channels/chan_skinny.c: Add activatesub and dialandactivate sub. + When called, activatesub first cleans up the active sub and then + handles the sub passed. dialandactivatesub first sets sub->exten + and then calls activatesub. Revise handle_offhook to utilise the + callid sent to chan_skinny. Some other minor fixes especially + around d->hookstate (which still needs some more work). + +2011-05-13 18:10 +0000 [r318918-318922] Brett Bryant + + * main/channel.c, /: Merged revisions 318921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318921 | bbryant | 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) + | 8 lines Fixes a segmentation fault in dynamic hints when a + channel technology isn't loaded for a hint. (closes issue #18495) + Reported by: bertrand Tested by: bertrand ........ + + * /, res/res_srtp.c: Merged revisions 318919 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) + | 10 lines This patch fixes an issue with SRTP which makes + HOLD/UNHOLD impossible when too much time has passed between + sending audio. (closes issue #18206) Reported by: bernhardsi + Patches: res_srtp_unhold.patch uploaded by bernhards (license + 1138) Tested by: bernhards, notthematrix ........ + + * /, channels/chan_sip.c: Merged revisions 318917 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318917 | bbryant | 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) + | 11 lines This patch allows TCP peers into the ast_db where they + were previously restricted. (closes issue #18882) Reported by: + cmaj Patches: + patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt + uploaded by cmaj (license 830) Tested by: cmaj ........ + +2011-05-13 16:30 +0000 [r318869] Richard Mudgett + + * /, main/features.c: Merged revisions 318868 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) + | 19 lines CDR's are being written immediately on caller hangup. + CDR's are being written immediately on caller hangup. The + dialplan is not able to modify it in the h exten. The h exten in + the initial context is not run before closing CDR's when the + bridge is unlinked if a macro is active and does not have an h + exten. * Make ast_bridge_call() check for an h exten in the + current context and if a macro is active then the initial + context. The first h exten found is then run before closing the + CDR. (closes issue #18212) Reported by: leearcher Patches: + issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested + by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/ + ........ + +2011-05-13 08:33 +0000 [r318833] Damien Wedhorn + + * channels/chan_skinny.c: Move exten used for dialing from device + to subchannel. There were some issues where if a simple switch + was cancelled and a new switch started before the first had timed + out where the d->exten would be used for both subchannels. This + was bad leading to possible invalid extensions if some digits had + been entered in the abandoned simple switch and the second one + was completed before the first timed out, or the second would be + cancelled because d->exten would be set to nothing on the time + out of the first. + +2011-05-13 01:55 +0000 [r318785] Matthew Nicholson + + * /, channels/sip/reqresp_parser.c: Merged revisions 318720 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318720 | mnicholson | 2011-05-12 18:35:51 -0500 (Thu, 12 May + 2011) | 4 lines Handle ipv6 addresses in the sent-by Via: field. + This change fixes a regression in via header parsing and ipv6 + handling. (closes issue #18951) ........ + +2011-05-13 01:50 +0000 [r318784] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 318783 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) + | 14 lines PRI early media won't ring. And another way to pass + early media. Don't indicate that there is inband information + present, just assume that the B channel is connected. * Restore + clearing the dialing flag Rx squelch unconditionally when a + PROCEEDING message comes in. (closes issue #19268) Reported by: + tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett + (license 664) Tested by: tbsky ........ + +2011-05-12 22:56 +0000 [r318672] Alec L Davis + + * /, channels/chan_sip.c, apps/app_directed_pickup.c, + main/features.c, include/asterisk/features.h: Merged revisions + 318671 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May + 2011) | 30 lines Fix directed group pickup feature code *8 with + pickupsounds enabled Since 1.6.2, the new pickupsound and + pickupfailsound in features.conf cause many issues. 1). + chan_sip:handle_request_invite() shouldn't be playing out the + fail/success audio, as it has 'netlock' locked. 2). dialplan + applications for directed_pickups shouldn't beep. 3). feature + code for directed pickup should beep on success/failure if + configured. Created a sip_pickup() thread to handle the pickup + and playout the audio, spawned from handle_request_invite. Moved + app_directed:pickup_do() to features:ast_do_pickup(). Functions + below, all now use the new ast_do_pickup() app_directed_pickup.c: + pickup_by_channel() pickup_by_exten() pickup_by_mark() + pickup_by_part() features.c: ast_pickup_call() (closes issue + #18654) Reported by: Docent Patches: + ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license + 585) Tested by: lmadsen, francesco_r, amilcar, isis242, + alecdavis, irroot, rymkus, loloski, rmudgett Review: + https://reviewboard.asterisk.org/r/1185/ ........ + +2011-05-12 20:44 +0000 [r318600-318635] Damien Wedhorn + + * channels/chan_skinny.c: Consolidate setsubstate_* into + setsubstate and use a switch. Consolidate the functions and add + some debugging info. Allows to be able to set a substate without + explicitly knowing what the state is. + + * channels/chan_skinny.c: Add setsubstate_onhook. Add the + setsubstate_onhook to complete the initial substate handling + procedures. Added dumpsub(sub, forcehangup) which is the common + way of calling setsubstate_onhook. Dumpsub attempts to activate + another sub after setting the current one onhook. + +2011-05-11 18:52 +0000 [r318551-318552] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 318550 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318550 | twilson | 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) + | 2 lines Comment out the REF_DEBUG that slipped in during + debugging ........ + + * /, channels/chan_sip.c: Merged revisions 318549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r318549 | twilson | 2011-05-11 13:39:48 -0500 + (Wed, 11 May 2011) | 27 lines Merged revisions 318548 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) + | 19 lines Clean up several chan_sip reference leaks Several + situations in the code could lead to peers or sip_pvt references + being leaked. This would cause RTP ports to never be destroyed + (leading to exhaustion of all available RTP ports) and memory + leaks. The original patch for this issue from rgagnon was the + result of an obscene amount of testing and hard work, for which I + am very grateful. I did some cleanup and added a few additional + refcount fixes that I found. (closes issue #17255) Reported by: + kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff + uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, + wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ + Review: https://reviewboard.asterisk.org/r/1207/ Review: + https://reviewboard.asterisk.org/r/1210/ ........ + ................ + +2011-05-10 23:42 +0000 [r318500] Richard Mudgett + + * /, channels/sig_pri.c, channels/sig_ss7.c: Merged revisions + 318499 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) + | 15 lines Unable to pickup DAHDI/PRI call because call state is + reported as DIALING. The channel state is not updated to RINGING + when an ALERTING message is received. Regression caused when + sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added + missing channel state update to RINGING when the + AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes + issue #19257) Reported by: alecdavis Patches: + issue19257_v1.8_v2.patch uploaded by rmudgett (license 664) + Tested by: alecdavis, rmudgett ........ + +2011-05-10 15:16 +0000 [r318437] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 318436 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r318436 | russell | 2011-05-10 10:13:16 -0500 (Tue, 10 + May 2011) | 2 lines chan_iax2: change LOG_NOTICE to LOG_DEBUG in + iax2_read(). ........ + +2011-05-10 00:22 +0000 [r318400] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 318337 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r318337 | twilson | 2011-05-09 15:23:15 -0500 + (Mon, 09 May 2011) | 18 lines Merged revisions 318331 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) + | 12 lines Don't offer video to directmedia callee unless caller + offered it as well Make sure that when directmedia is enabled, + that video is not offered to the callee even if it supports it. + p->vrtp will not exist since the caller didn't offer video. + (closes issue #19195) Reported by: one47 Patches: + sip_cant_add_video_rtp uploaded by one47 (license 23) ........ + ................ + +2011-05-09 23:16 +0000 [r318283-318352] Richard Mudgett + + * /, res/Makefile, res/res_features.exports.in (removed): Merged + revisions 318351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) + | 6 lines Remove references to res_features and its export file. + The contents of res/res_features.c was moved to into + main/features.c awhile ago. There is no longer any need for the + res/Makefile to reference res_features or the res_features linker + exports file to exist. ........ + + * /, main/features.c: Merged revisions 318282 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) + | 24 lines Hangup extension executed twice. When a user hangs up + a call, in certain circumstances, the hangup extension can end up + being executed twice: 1) If a call is bridged and the 'h' + extension executes the Hangup application, then the 'h' extension + will be executed twice. 2) If a call is bridged within a macro + (Dial or Queue), it has its own 'h' extension, the main context + also has an 'h' extension, and the macro 'h' extension executes + the Hangup application, then both 'h' extensions will be + executed. * Revert originally commited fix for #16106 and just + set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in + ast_bridge_call(). The bridge code just executed an 'h' extension + so the main PBX loop does not need to execute one as well. (issue + #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd + ........ + +2011-05-09 17:13 +0000 [r318234] David Vossel + + * /, channels/chan_sip.c: Merged revisions 318233 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r318233 | dvossel | 2011-05-09 12:09:55 -0500 + (Mon, 09 May 2011) | 14 lines Merged revisions 318230 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) + | 7 lines Fixes cases where sip_set_rtp_peer can return too early + during media path reset. (closes issue #19225) Reported by: one47 + Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) + ........ ................ + +2011-05-09 17:00 +0000 [r318232] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 318231 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r318231 | rmudgett | 2011-05-09 11:57:18 -0500 + (Mon, 09 May 2011) | 41 lines Don't get early media for ISDN on + outgoing calls. It looks to be a long-standing misinterpretation + of the progress indicator ie values: 1 - Call is not end-to-end + ISDN; further call progress information may be available in-band. + 8 - In-band information or an appropriate pattern is now + available. Only value 8 is handled by chan_dahdi/sig_pri. The 1 + value is not handled as early media probably because the meaning + of the second half of it's description was overlooked. * Test to + see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or + PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path. + (closes issue #18868) Reported by: isrl Patches: + issue18868_19246_v1.8.patch uploaded by rmudgett (license 664) + Tested by: satish_lx .......... No inband progress on + PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider + sends an ALERTING with "Inband information or appropriate pattern + now available", but Asterisk only generates and passes the RING + to the SIP extension, not the inband message. Unfortunately, the + inband message is not a ringback tone but a prompt that says the + number is not in service. The SIP extension then hears two rings + and the call is hungup which confuses the caller. * Post an + AST_CONTROL_PROGRESS as well as opening the media path if inband + audio is indicated with an ALERTING message. (closes issue + #19246) Reported by: cristiandimache Patches: + issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested + by: cristiandimache ................ + +2011-05-09 14:41 +0000 [r318194] Leif Madsen + + * main/app.c: Increase prepend filename length. (closes issue + #19238) Reported by: byronclark Patches: + increase_prepend_filename_length.patch uploaded by byronclark + (license 1200) + +2011-05-09 14:37 +0000 [r318162-318193] Jonathan Rose + + * main/features.c: Minor change to 318141 to improve parsing + behavior. + + * /, configs/features.conf.sample: Merged revisions 318148 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | + 4 lines Documenting an observed behavior of features in + features.conf. Since parkinglots use an integer for the + parkinglot extensions, leading zeros specified in the + configuration file are ignored. ........ + +2011-05-09 14:11 +0000 [r318143] Matthew Nicholson + + * main/channel.c, /: Merged revisions 318142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318142 | mnicholson | 2011-05-09 09:09:38 -0500 (Mon, 09 May + 2011) | 9 lines Make indicate/control frames WRITE events on + framehooks. Also, if a framehook returns a non-control frame, + don't forward it to the channel. (closes issue #19251) Reported + by: irroot Patches: (modified) framehook_indicate.patch2 uploaded + by irroot (license 52) Tested by: irroot ........ + +2011-05-09 13:56 +0000 [r318141] Jonathan Rose + + * main/features.c, CHANGES: Allows ParkedCall application to + specify a parkinglot. When invoking the app parkedcall, the + argument can now include '@parkinglot' after the extension. + (closes issue #18777) Reported by: cartama Patches: 0018777.diff + uploaded by cartama (license 1157) Review: + https://reviewboard.asterisk.org/r/1209/ + +2011-05-09 07:40 +0000 [r318106] Damien Wedhorn + + * channels/chan_skinny.c: Add setsubstate_callwait. If a call is + made to a line that already has a call and the device is offhook + (ie activeish call), the call is set to CALLWAIT rather than + RINGIN. + +2011-05-07 23:36 +0000 [r318056-318058] Russell Bryant + + * res/res_config_curl.c, /: Merged revisions 318057 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r318057 | russell | 2011-05-07 18:35:37 -0500 (Sat, 07 + May 2011) | 8 lines res_config_curl: fix a crash with static + realtime. (closes issue #18413) Reported by: jmls Patches: + 20101202__issue18413.diff.txt uploaded by tilghman (license 14) + Tested by: jmls ........ + + * /, channels/chan_iax2.c: Merged revisions 318055 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r318055 | russell | 2011-05-07 18:24:18 -0500 (Sat, 07 + May 2011) | 7 lines chan_iax2: Don't overwrite port found with an + SRV lookup. (closes issue #17291) Reported by: jcovert Patches: + chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert + (license 551) ........ + +2011-05-06 23:07 +0000 [r317996-318019] Damien Wedhorn + + * channels/chan_skinny.c: Only allow voicemail if substate is + OFFHOOK or no channel active (UNSET). (closes issue #17901) + Reported by: salecha + + * channels/chan_skinny.c: Rename sub->parent to sub->line. Improve + readability of code, eg, (sub->parent == d->activeline) becomes + (sub->line == d->activeline). + + * channels/chan_skinny.c: Move the hookstate from line to device. + Long time coming, finally moving the hookstate from line to + device. This may fix some issues where a device has multiple + lines. Previously we had to run through all lines on a device to + see if it was actually onhook or not. + +2011-05-06 21:49 +0000 [r317968-317970] Russell Bryant + + * /, apps/app_meetme.c: Merged revisions 317969 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) + | 10 lines Use the right variable to print the time in a debug + message. The original patch also increased some buffer sizes, but + that was already done in this version. (closes issue #17034) + Reported by: sysreq Patches: asterisk-issue-17034.patch uploaded + by sysreq (license 1009) ........ + + * /, apps/app_meetme.c: Merged revisions 317967 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) + | 2 lines Fix some more "set but unused" compiler warnings. + ........ + +2011-05-06 21:10 +0000 [r317920] David Vossel + + * res/res_rtp_asterisk.c, /: Merged revisions 317918 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r317918 | dvossel | 2011-05-06 16:06:55 -0500 (Fri, 06 + May 2011) | 7 lines Fixes missing colon from To/From headers in + RTCP manager events. (closes issue #18221) Reported by: + clegall_proformatique Patches: 18221_1.patch uploaded by ebroad + (license 878) ........ + +2011-05-06 21:07 +0000 [r317843-317919] Russell Bryant + + * main/pbx.c, /: Merged revisions 317917 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317917 | russell | 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011) + | 7 lines Fix calculation of free RAM to make minmemfree option + work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff + uploaded by loic (license 1020) ........ + + * contrib/scripts/import-cdr-csv-mysql.pl (added): Add a cdr_csv to + MySQL import script to contrib/scripts. (closes issue #17036) + Reported by: precisenetworks Patches: import-cdr-csv-mysql.pl + uploaded by precisenetworks (license 1010) + + * apps/app_userevent.c, CHANGES: Add the Uniqueid header to + Userevent. (closes issue #16962) Reported by: jlpedrosa Patches: + patch.diff uploaded by jlpedrosa (license 1002) + + * /, channels/chan_sip.c: Merged revisions 317867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) + | 10 lines chan_sip: Destroy variables on a sip_pvt before + copying vars from the sip_peer. Don't duplicate variables on the + sip_pvt. Just reset the variable list each time. (closes issue + #19202) Reported by: wdoekes Patches: + issue19202_destroy_challenged_invite_chanvars.patch uploaded by + wdoekes (license 717) ........ + + * /, channels/chan_sip.c: Merged revisions 317865 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) + | 11 lines chan_sip: fix a deadlock in check_rtp_timeout. Don't + block doing silly deadlock avoidance. Just return and try again + later. The funciton gets called often enough that it's fine. + Also, this change was already made in trunk. (closes issue + #18791) Reported by: irroot Patches: chan_sip.rtptimeout.patch + uploaded by irroot (license 52) ........ + + * addons/app_mysql.c, /: Merged revisions 317837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) + | 11 lines Fix a crash in the MySQL() application. This code was + not handling channel datastores safely. The channel must be + locked. (closes issue #17964) Reported by: wuwu Patches: + issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license + 71) Tested by: wuwu ........ + +2011-05-06 19:23 +0000 [r317818-317833] Matthew Nicholson + + * CHANGES: Updated CHANGES to note the autoservice changes for + pbx_lua + + * configs/extensions.lua.sample: Updated the sample pbx_lua config + file to reflect autoservice changes. + +2011-05-06 19:15 +0000 [r317807] Russell Bryant + + * /, contrib/realtime/mysql/sipfriends.sql: Merged revisions 317805 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317805 | russell | 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011) + | 7 lines Add a new sipfriends.sql for MySQL that has more fields + in it. (closes issue #16399) Reported by: pabelanger Patches: + sipfriends.sql.v3 uploaded by pabelanger (license 224) ........ + +2011-05-06 19:14 +0000 [r317721-317806] Matthew Nicholson + + * pbx/pbx_lua.c, UPGRADE.txt: Default to starting an autoservice in + pbx_lua. The autoservice is automatically stopped when + applications are executed, so this shouldn't cause any problems. + + * pbx/pbx_lua.c, UPGRADE.txt: Make pbx_lua handle managing the + autoservice better. Make autoservice_start() and + autoservice_stop() return nothing. Also check if the autoservice + flag is set before starting or stopping the autoservice and stop + and start the autoservice when returning control to and getting + control from the pbx engine. + + * UPGRADE.txt: Added note about changes in pbx_lua's behavior when + applications do dialplan jumps + + * CHANGES: Use two spaces after periods for the recent pbx_lua + change descriptions + + * CHANGES: Updated CHANGES for hints support in pbx_lua + + * pbx/pbx_lua.c, CHANGES: Detect Goto in pbx_lua. This code will + actually detect any dialplan jump from any application that calls + ast_explicit_goto(). This change is only being done in trunk as + it may change the way some dialplans execute. + +2011-05-06 16:23 +0000 [r317671] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 317670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) + | 22 lines Fix SIP connected line updates. This patch fixes a + couple SIP connected line update problems: 1) The connected line + needs to be updated when the initial INVITE is sent if there is a + peer callerid configured. Previously, the connected line + information did not get reported until the call was connected so + SIP could not report connected line information in ringing or + progress messages. 2) The connected line should not be updated on + initial connect if there is no connected line information. + Previously, all it did was wipe out any default preset + CONNECTEDLINE information set by the dialplan with empty strings. + (closes issue #18367) Reported by: GeorgeKonopacki Patches: + issue18367_v1.8.patch uploaded by rmudgett (license 664) Tested + by: rmudgett Review: https://reviewboard.asterisk.org/r/1199/ + ........ + +2011-05-06 08:21 +0000 [r317596] Terry Wilson + + * /, apps/app_queue.c: Merged revisions 317584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r317584 | twilson | 2011-05-06 01:18:53 -0700 + (Fri, 06 May 2011) | 20 lines Merged revisions 317575 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r317575 | twilson | 2011-05-06 01:04:17 -0700 + (Fri, 06 May 2011) | 13 lines Merged revisions 317574 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) + | 6 lines Re-fix queue round-robin This part of the change for + r315596 was incorrect. No bridge occurs when doing a roundrobin + dial and no one answers, so this code shouldn't have been + removed. ........ ................ ................ + +2011-05-05 23:47 +0000 [r317426-317531] Russell Bryant + + * Makefile, /: Merged revisions 317530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317530 | russell | 2011-05-05 18:46:54 -0500 (Thu, 05 May 2011) + | 10 lines If the configure script runs, force a rebuild of + menuselect-tree. Some contents in the menuselect tree are + dependent on configure script parameters, namely + --enable-dev-mode. (closes issue #17219) Reported by: Nick_Lewis + Patches: issue_17219.rev1.txt uploaded by russell (license 2) + ........ + + * /, contrib/realtime/mysql/queue_log.sql, + contrib/realtime/mysql/sipfriends.sql: Merged revisions 317486 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317486 | russell | 2011-05-05 18:15:53 -0500 (Thu, 05 May 2011) + | 9 lines Fix some more realtime MySQL schema issues. (closes + issue #18537) Reported by: denzs Patches: sipfriends.sql.svndiff + uploaded by denzs (license 1182) queue_log.sql.svndiff uploaded + by denzs (license 1182) meetme.sql.svndiff uploaded by denzs + (license 1182) ........ + + * /, contrib/realtime/mysql/meetme.sql, + contrib/realtime/mysql/sipfriends.sql: Merged revisions 317484 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317484 | russell | 2011-05-05 18:12:35 -0500 (Thu, 05 May 2011) + | 8 lines Fix some errors in sample MySQL realtime schema files. + (closes issue #18915) Reported by: Dovid Patches: + sipfriends.patch uploaded by Dovid (license 652) meetme.patch + uploaded by Dovid (license 652) ........ + + * CHANGES, res/res_calendar.c: Add "calendar show types" CLI + command. (closes issue #18246) Reported by: junky Patches: + calendar_types.diff uploaded by junky (license 177) + + * cel/cel_pgsql.c, UPGRADE.txt, configs/cel_pgsql.conf.sample, + CHANGES: Add CEL extra field to cel_pgsql. (closes issue #18462) + Reported by: joscas Patches: bug_18462.diff uploaded by snuffy + (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by + joscas (license 1180) + + * /, cdr/cdr_syslog.c: Merged revisions 317480 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317480 | russell | 2011-05-05 18:00:55 -0500 (Thu, 05 May 2011) + | 8 lines Don't lose cdr_syslog config on a reload. (closes issue + #18679) Reported by: enegaard Patches: + issue18679_seanbright.patch uploaded by seanbright (license 71) + Tested by: enegaard ........ + + * channels/chan_unistim.c, channels/chan_usbradio.c, + channels/chan_dahdi.c, /, channels/chan_sip.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_alsa.c, channels/chan_console.c, + channels/chan_oss.c, channels/chan_mgcp.c, + channels/misdn_config.c: Merged revisions 317478 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 + May 2011) | 12 lines Fix some consistency issues with + jitterbuffer config. Store the defaults noted in the sample + config files in the jitterbuffer config data structure. This + makes the CLI commands that output these settings show the right + thing. Also only show the settings that are relevant in the + settings CLI commands, based on which jitterbuffer is selected + and whether it's enabled. (closes issue #19083) Reported by: + rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by + rgagnon (license 1202) ........ + + * /, pbx/pbx_lua.c: Merged revisions 317476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317476 | russell | 2011-05-05 17:47:57 -0500 (Thu, 05 May 2011) + | 8 lines Add a datastore fixup to fix a pbx_lua crash. (closes + issue #19055) Reported by: jamhed Patches: + lua_datastore_fixup1.diff uploaded by mnicholson (license 96) + Tested by: mnicholson, jamhed ........ + + * cel/cel_pgsql.c, channels/chan_jingle.c, + channels/sip/sdp_crypto.c, res/res_config_odbc.c, /, + channels/chan_sip.c, res/res_crypto.c, pbx/pbx_lua.c, + channels/iax2-provision.c, pbx/pbx_dundi.c, + channels/chan_console.c, cdr/cdr_radius.c, channels/chan_iax2.c, + res/res_jabber.c, res/res_config_sqlite.c: Merged revisions + 317474 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317474 | russell | 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) + | 2 lines Fix more "set but unused" warnings. ........ + + * /, main/dsp.c: Merged revisions 317429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317429 | russell | 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011) + | 5 lines Only display inband DTMF warning if inband DTMF + detection is enabled. (closes issue #18901) Reported by: irroot + ........ + + * /, apps/app_rpt.c: Merged revisions 317427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317427 | russell | 2011-05-05 16:58:45 -0500 (Thu, 05 May 2011) + | 7 lines Fix potential memory leak, and use of uninitialized + memory. (closes issue #16476) Reported by: junky Patches: + M16476.diff uploaded by junky (license 177) ........ + + * main/manager.c, /: Merged revisions 317425 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317425 | russell | 2011-05-05 16:53:13 -0500 (Thu, 05 May 2011) + | 7 lines Add missing ActioID handling to Events action. (closes + issue #18949) Reported by: edersohe Patches: 0018949.patch + uploaded by edersohe (license 1228) ........ + +2011-05-05 21:20 +0000 [r317395] Sean Bright + + * main/asterisk.c: Add some new editline bindings by default, and + allow for user specified configuration. I excluded the part of + this patch that used the HOME environment variable since the + built-in editline library goes to great lengths to disallow that. + Instead only settings the EDITRC environment variable will use a + user specified file. Also, the default environment variable use + to determine the edit more is AST_EDITMODE instead of AST_EDITOR + (although the latter is still supported). (closes issue #15929) + Reported by: kkm Patches: astcli-editrc-v2.diff uploaded by kkm + (license 888) 015929-astcli-editrc-trunk.240324.diff uploaded by + kkm (license 888) Tested by: seanbright + +2011-05-05 20:46 +0000 [r317382] Damien Wedhorn + + * channels/chan_skinny.c: Move hold stuff to the setsubstate + arrangement. skinny_hold moved to setsubstate_hold and + skinny_unhold integrated into setsubstate_connected. Removed + sub->onhold and replaced with SUBSTATE_HOLD. Also fixed inbound + call answering by queueing an AST_CONTROL_ANSWER on answering a + SUBSTATE_RINGIN sub (was a typo). + +2011-05-05 20:27 +0000 [r317377] Sean Bright + + * /, addons/res_config_mysql.c: Merged revisions 317370 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May + 2011) | 10 lines Don't duplicate our data on the stack and just + use the MYSQL_ROW directly. With large result sets we were + blowing out the stack. (closes issue #19090) Reported by: + mickecarlsson Patches: issue19090_trunk_svn.patch uploaded by + seanbright (license 71) Tested by: mickecarlsson ........ + +2011-05-05 19:56 +0000 [r317337] Russell Bryant + + * /, apps/app_queue.c: Merged revisions 317336 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317336 | russell | 2011-05-05 14:55:58 -0500 (Thu, 05 May 2011) + | 7 lines Increase buffer size to be PATH_MAX for a path. (closes + issue #19239) Reported by: byronclark Patches: + queue_announce_length.patch uploaded by byronclark (license 1200) + ........ + +2011-05-05 19:33 +0000 [r317334] Jonathan Rose + + * /, channels/chan_sip.c: Merged revisions 317283 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) | + 10 lines Resolves a deadlock that occurs during sip_new This is + based on an uncommitted patch by jpeeler for the issue. Instead + of relocking and then unlocking the channel though, we keep the + lock on the channel until we are finished doing what we need to + the channel. (closes issue #18441) Reported by: Alric ........ + +2011-05-05 18:46 +0000 [r317282] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 317281 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r317281 | russell | 2011-05-05 13:39:44 -0500 + (Thu, 05 May 2011) | 29 lines Merged revisions 317255 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r317255 | russell | 2011-05-05 13:29:53 -0500 + (Thu, 05 May 2011) | 22 lines Merged revisions 317211 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) + | 15 lines chan_sip: fix broken realtime peer count, fix memory + leak This patch addresses two bugs in chan_sip: 1) The count of + realtime peers and users was off. The increment checked the value + of the caching option, while the decrement did not. 2) Add a + missing regfree() for a regex. (closes issue #19108) Reported by: + vrban Patches: missing_regfree.patch uploaded by vrban (license + 756) sip_object_counter.patch uploaded by vrban (license 756) + ........ ................ ................ + +2011-05-05 18:09 +0000 [r317198] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 317196 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317196 | mnicholson | 2011-05-05 13:02:52 -0500 (Thu, 05 May + 2011) | 8 lines Set SO_KEEPALIVE on SIP TCP sockets so that they + eventually go away when a peer abruptly disappears. This mostly + occurs after a successful registration. (closes issue #17544) + Reported by: marcelloceschia Patches: (modified) tcptls.patch + uploaded by st (license 907) ........ + +2011-05-05 18:08 +0000 [r317197] David Vossel + + * bridges/bridge_softmix.c, funcs/func_jitterbuffer.c: Fixes + reliability issues with func_jitterbuffer's usage in the new + ConfBridge application. + +2011-05-05 15:06 +0000 [r317059-317105] Leif Madsen + + * /, contrib/scripts/safe_asterisk: Merged revisions 317104 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r317104 | lmadsen | 2011-05-05 11:04:24 -0400 + (Thu, 05 May 2011) | 15 lines Merged revisions 317102 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011) + | 8 lines Disable console colourization inside safe_asterisk + checks. (closes issue #19213) Reported by: lefoyer Patches: + issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by + wdoekes (license 717) Tested by: wdoekes, lefoyer ........ + ................ + + * Makefile, configs/cel.conf.sample, /: Merged revisions 317058 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317058 | lmadsen | 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011) + | 7 lines Remove unused directory and clear up some + documentation. (closes issue #19193) Reported by: bchia Patches: + cel-csv.diff uploaded by lathama (license 1028) Tested by: + lathama, Marquis42 ........ + +2011-05-05 09:03 +0000 [r316994-317026] Damien Wedhorn + + * channels/chan_skinny.c: Add setsubstate_congestion and + setsubstate_progress. Move handling of both state handling from + skinny_indicate to it's own sub. Also, modified behaviour to not + hangup the sub and let the dialplan have a chance in doing what + it wants for congestion. Added various states to substate2str and + added these states where applicable for other set_substate_ + procs. + + * channels/chan_skinny.c: Add setsubstate_busy. Move handling of + setting busy state from skinny_indicate to it's own sub. Also, + modified behaviour to not hangup the sub and let the dialplan + have a chance in doing what it wants (eg busy(10); hangup() in + the dialplan now gives a busy indication for 10 secs and then + hangs up. + +2011-05-05 07:09 +0000 [r316962] Stefan Schmidt + + * main/astobj2.c: Adding the Move to Front Hash functionality + Moving a found object to the front of its bucket to reduce the + necessary traversal steps to find an object. This change improves + the search time on large system with many data or in link lists. + (closes issue #19233) Reported by: schmidts Review: + https://reviewboard.asterisk.org/r/1201/ + +2011-05-05 02:34 +0000 [r316920] Sean Bright + + * main/manager.c, /, main/http.c, main/utils.c: Merged revisions + 316917-316919 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316917 | seanbright | 2011-05-04 22:23:28 -0400 (Wed, 04 May + 2011) | 5 lines Make sure that tcptls_session is properly + initialized. (issue #18598) Reported by: ksn ........ r316918 | + seanbright | 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) | 5 + lines Look at the correct buffer for our digest info instead of + an empty one. (issue #18598) Reported by: ksn ........ r316919 | + seanbright | 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) | 10 + lines Use the correct HTTP method when generating our digest, + otherwise we always fail. When calculating the 'A2' portion of + our digest for verification, we need the HTTP method that is + currently in use. Unfortunately our mapping function was + incorrect, resulting in invalid hashes being generated and, in + turn, failures in authentication. (closes issue #18598) Reported + by: ksn ........ + +2011-05-04 21:44 +0000 [r316885] Damien Wedhorn + + * channels/chan_skinny.c: Add setsubstate_ringout (equivalent to + AST_STATE ringing). Renamed previous setsubstate_ringout to + setsubstate_dialing for a state when attempting to dial a number, + substate ringout now for when core has indicated that the channel + is actually ringing on the other end. Also added substate2str for + debugging purposes. + +2011-05-04 18:57 +0000 [r316832] Richard Mudgett + + * /, apps/app_meetme.c: Merged revisions 316831 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) + | 9 lines Wait for leader with Music On Hold allows crosstalk + between participants. Parenthesis in the wrong position. + Regression from issue #14365 when expanding conference flags to + use 64 bits. (closes issue #18418) Reported by: MrHanMan Tested + by: rmudgett ........ + +2011-05-04 16:42 +0000 [r316798] David Vossel + + * channels/chan_sip.c, CHANGES: Reverts rev 316218 as it breaks + parsing the [general] section of sip.conf. The functionality this + patch attempts to achieve should already be possible using + [general](+) in the config file. issue #17957 + +2011-05-04 16:17 +0000 [r316664-316711] Sean Bright + + * /, apps/app_voicemail.c: Merged revisions 316709 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316709 | seanbright | 2011-05-04 12:15:32 -0400 + (Wed, 04 May 2011) | 22 lines Merged revisions 316708 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r316708 | seanbright | 2011-05-04 12:10:59 -0400 + (Wed, 04 May 2011) | 15 lines Merged revisions 316707 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May + 2011) | 8 lines If sox fails when processing a voicemail, don't + delete the original file. (closes issue #18111) Reported by: + sysreq Patches: issue18111_trunk.patch uploaded by seanbright + (license 71) Tested by: seanbright ........ ................ + ................ + + * main/manager.c, /: Merged revisions 316663 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316663 | seanbright | 2011-05-04 10:35:05 -0400 (Wed, 04 May + 2011) | 8 lines Only return a single error via AMI when + requesting a forbidden action. (closes issue #19216) Reported by: + oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright + (license 71) Tested by: seanbright ........ + +2011-05-04 14:26 +0000 [r316618-316657] David Vossel + + * /, apps/app_chanspy.c: Merged revisions 316650 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316650 | dvossel | 2011-05-04 09:25:03 -0500 + (Wed, 04 May 2011) | 15 lines Merged revisions 316644 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011) + | 9 lines Fixes one-way-audio when chanspy activated with the 'o' + option (closes issue #18382) Reported by: jkister Patches: + 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt + uploaded by malin (license ) Tested by: firstsip, Greenlightcrm, + malin, wdoekes, boroda, dvossel ........ ................ + + * /, channels/chan_sip.c: Merged revisions 316617 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316617 | dvossel | 2011-05-04 08:44:41 -0500 + (Wed, 04 May 2011) | 19 lines Merged revisions 316616 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) + | 12 lines Fixes session-timers=refuse not being enforced for + *caller* During handle_request_invite, the session timer mode was + retrieved from a cached variable. This patch forces a peer lookup + of the session timer mode in the case of an incoming invite. + (closes issue #18804) Reported by: wdoekes Patches: + issue18804_session_timer_refuse_caller.patch uploaded by wdoekes + (license 717) issue_18804_v2.diff uploaded by dvossel (license + 671) ........ ................ + +2011-05-04 08:25 +0000 [r316552-316584] Damien Wedhorn + + * channels/chan_skinny.c: Add setsubstate_ringin. Added + setsubstate_ringin. skinny_call now calls sss_ringin rather than + inline. Fixed previous issue so that setsubstate_connected now + use SUBSTATE_RINGIN to determine is an AST_CONTROL_ANSWER should + be queued. + + * channels/chan_skinny.c: Make skinny_answer use + setsubsate_connected. Cosolidated the code so that skinny_answer + now uses the setsubstate procedures rather than doing the + handling inline. + +2011-05-04 07:13 +0000 [r316520] Tzafrir Cohen + + * autoconf/ast_check_pwlib.m4, /, configure: Merged revisions + 316193 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316193 | tzafrir | 2011-05-03 13:57:16 +0300 (ג', 03 מאי 2011) | + 8 lines Re-fix bashism in ./configure: s/let/$(( ))/ A + forward-port in r278985 accidentally re-introduced issue 17485. + Fixing it. Thanks to Jilles Tjoelker for the good report. (closes + issue #17485) ........ + +2011-05-04 07:10 +0000 [r316519] Damien Wedhorn + + * channels/chan_skinny.c: Cleanup skinny callinfo. Cosolidated the + working out of the callinfo to be sent into transmit_callinfo. + Replaced ambiguous sub->outgoing with calldirection which can be + SKINNY_INCOMING or SKINNY_OUTGOING (same value as the skinny + protocol). + +2011-05-04 02:39 +0000 [r316477] Sean Bright + + * /, apps/app_meetme.c: Merged revisions 316476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316476 | seanbright | 2011-05-03 22:34:01 -0400 + (Tue, 03 May 2011) | 17 lines Merged revisions 316475 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May + 2011) | 10 lines Honor the C option to MeetMe when L is passed. + This fixes a case that r304773 and friends missed. (closes issue + #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff + uploaded by var (license 1227) Tested by: seanbright ........ + ................ + +2011-05-04 00:13 +0000 [r316428-316430] Tilghman Lesher + + * /, addons/cdr_mysql.c, addons/res_config_mysql.c: Merged + revisions 316429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316429 | tilghman | 2011-05-03 19:12:25 -0500 (Tue, 03 May 2011) + | 7 lines Escape column names in case they contain illegal + characters ('-') or reserved words. (closes issue #19063) + Reported by: festr Patches: patch uploaded by festr (license 443) + ........ + + * channels/chan_sip.c, CHANGES: If multiple [general] contexts + occur from sip.conf (usually due to external includes), merge + them. The original implementation of this did the merging of all + contexts with the same name in the realtime layer, but that + implementation severely breaks drivers which use the same context + name (e.g. iax.conf, type={peer,user}). Therefore, the + implementation needs to do the merging for particular entries + only, based upon what contexts would allow that in the channel + driver itself. This implementation is for chan_sip only, but + others could be added in the future. (closes issue #17957) + Reported by: marcelloceschia Patches: + chan-sip_parsing-general_branch162.patch uploaded by + marcelloceschia (license 1079) Tested by: tilghman + +2011-05-03 22:16 +0000 [r316337] Russell Bryant + + * /, channels/chan_skinny.c, pbx/pbx_dundi.c, channels/chan_mgcp.c: + Merged revisions 316336 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316336 | russell | 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011) + | 8 lines Use htons() instead of ntohs() in some places. (closes + issue #19200) Reported by: wdoekes Patches: + issue19200-trunk.patch uploaded by wdoekes (license 717) + issue19200-1.8.x.patch uploaded by wdoekes (license 717) ........ + +2011-05-03 22:07 +0000 [r316335] David Vossel + + * main/channel.c, /: Merged revisions 316334 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316334 | dvossel | 2011-05-03 17:05:59 -0500 (Tue, 03 May 2011) + | 8 lines Fixes framehook segfault on indicate (closes issue + #19215) Reported by: irroot Patches: framehook_indicate.patch + uploaded by irroot (license 52) ........ + +2011-05-03 21:48 +0000 [r316333] Russell Bryant + + * /, apps/app_minivm.c: Merged revisions 316331 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316331 | russell | 2011-05-03 16:41:11 -0500 (Tue, 03 May 2011) + | 2 lines Resolve another warning. ........ + +2011-05-03 21:45 +0000 [r316332] David Vossel + + * channels/chan_local.c, /: Merged revisions 316330 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316330 | dvossel | 2011-05-03 16:37:59 -0500 + (Tue, 03 May 2011) | 24 lines Merged revisions 316329 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r316329 | dvossel | 2011-05-03 16:29:55 -0500 + (Tue, 03 May 2011) | 17 lines Merged revisions 316328 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011) + | 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ + for tracking down the issue and submitting the patch. (closes + issue #19053) Reported by: oej Tested by: oej Review: + https://reviewboard.asterisk.org/r/1158/ ........ + ................ ................ + +2011-05-03 20:45 +0000 [r316293] Russell Bryant + + * channels/chan_unistim.c, main/udptl.c, main/fskmodem_float.c, + main/rtp_engine.c, /, res/res_musiconhold.c, apps/app_ices.c, + apps/app_followme.c, main/config.c, main/channel.c, main/cdr.c, + channels/chan_phone.c, funcs/func_enum.c, main/manager.c, + channels/chan_skinny.c, apps/app_minivm.c, main/features.c, + main/plc.c, res/res_agi.c, apps/app_amd.c, main/pbx.c, + res/res_fax.c, formats/format_wav.c, apps/app_festival.c, + channels/chan_agent.c, apps/app_originate.c, apps/app_queue.c, + codecs/lpc10/dyptrk.c, include/asterisk/linkedlists.h, + main/file.c, main/audiohook.c, pbx/pbx_config.c, main/asterisk.c, + main/dsp.c, res/res_calendar.c, apps/app_voicemail.c: Merged + revisions 316265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) + | 5 lines Fix a bunch of compiler warnings generated by gcc + 4.6.0. Most of these are -Wunused-but-set-variable, but there + were a few others mixed in here, as well. ........ + +2011-05-03 19:22 +0000 [r316240] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_pri.c: Merged revisions 316224 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316224 | rmudgett | 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011) + | 16 lines The dahdi_hangup() call does not clean up the channel + fully. After dahdi_hangup() has supposedly hungup an ISDN channel + there is still traffic on the S0-bus because the channel was not + cleaned up fully. Shuffled the hangup code to include some + missing cleanup. Also fixed some code formatting in the area. I + think the primary missing clean up code was the call to + tone_zone_play_tone() to turn off any active tones on the + channel. (closes issue #19188) Reported by: jg1234 Patches: + issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested + by: jg1234 ........ + +2011-05-03 19:00 +0000 [r316216-316218] David Vossel + + * /, channels/chan_sip.c: Merged revisions 316217 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316217 | dvossel | 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011) + | 9 lines Never put the Require: timer header in an Invite. This + has already been discussed and should have been resolved earlier. + View revsion 285565's log for more information about why it is + important to not put timer in the Require header. (closes issue + #18704) Reported by: mfrager ........ + + * /, res/res_odbc.c: Merged revisions 316215 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316215 | dvossel | 2011-05-03 13:49:48 -0500 (Tue, 03 May 2011) + | 9 lines Fixes a random crash (NULL reference) in res_odbc.c. + (closes issue #19180) Reported by: pruiz Patches: tmp.diff + uploaded by pruiz (license 1152) Tested by: pruiz, seanbright + ........ + +2011-05-03 18:23 +0000 [r316213] Sean Bright + + * main/manager.c, /: Merged revisions 316206 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316206 | seanbright | 2011-05-03 14:17:36 -0400 (Tue, 03 May + 2011) | 8 lines If we aren't interested in events, don't generate + the FullyBooted event on AMI login. (closes issue #19089) + Reported by: bklang Patches: issue19089-1.8-r316204.patch + uploaded by seanbright (license 71) Tested by: seanbright + ........ + +2011-05-02 19:15 +0000 [r316095] Tilghman Lesher + + * funcs/func_curl.c, /: Merged revisions 316094 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316094 | tilghman | 2011-05-02 14:09:55 -0500 + (Mon, 02 May 2011) | 15 lines Merged revisions 316093 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011) + | 8 lines More possible crashes based upon invalid inputs. + (closes issue #18161) Reported by: wdoekes Patches: + 20110301__issue18161.diff.txt uploaded by tilghman (license 14) + Tested by: wdoekes ........ ................ + +2011-05-02 15:58 +0000 [r316054] Paul Belanger + + * apps/app_meetme.c: Formatting change, remove red blobs + +2011-04-27 19:15 +0000 [r315895] Matthew Nicholson + + * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged + revisions 315894 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315894 | mnicholson | 2011-04-27 14:14:27 -0500 + (Wed, 27 Apr 2011) | 28 lines Merged revisions 315893 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315893 | mnicholson | 2011-04-27 14:03:05 -0500 + (Wed, 27 Apr 2011) | 21 lines Merged revisions 315891 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr + 2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2. + This change optimizes the free_via() function and removes some + redundant null checking. It also fixes compliance with RFC 3261 + section 18.2.2 by always using the port specified in the Via + header for routing responses (even when maddr is not set). Also + the htons() function is now used when setting the port. + Additional documentation comments have been added in various + places to make the logic in the code clearer. (closes issue + #18951) Reported by: jmls Patches: + issue18951_set_proper_port_from_via.patch uploaded by wdoekes + (license 717) (modified) ........ ................ + ................ + +2011-04-27 17:51 +0000 [r315855-315856] David Vossel + + * apps/app_confbridge.c: Makes the new ConfBridge join and leave + sounds be used by default rather than beep and beeperr. + + * main/channel.c: Clears exception flag during ast_read when + func_jitterbuffer is enabled + +2011-04-27 15:56 +0000 [r315811] Russell Bryant + + * /, main/asterisk.c: Merged revisions 315810 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r315810 | russell | 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011) + | 2 lines Set the copyright year to 2011 in the startup message. + ........ + +2011-04-27 12:37 +0000 [r315766] Leif Madsen + + * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 315765 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r315765 | lmadsen | 2011-04-27 07:36:17 -0500 (Wed, 27 Apr 2011) + | 4 lines Enable Russian core sound selection in menuselect. + (closes issue #18724) Reported by: pbxware ........ + +2011-04-26 23:10 +0000 [r315670-315675] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 315673 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315673 | twilson | 2011-04-26 15:56:19 -0700 + (Tue, 26 Apr 2011) | 25 lines Merged revisions 315672 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315672 | twilson | 2011-04-26 15:52:25 -0700 + (Tue, 26 Apr 2011) | 18 lines Merged revisions 315671 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) + | 11 lines Make sure unregistering a peer unlinks it from the + peer container Instead of mostly copying the code from + expire_register, just use the function that "does the right + thing". (closes issue #16033) Reported by: kkm Patches: + 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888) + Tested by: kkm, tilghman, twilson ........ ................ + ................ + + * channels/chan_sip.c: Make sure to create the caps structure for + autocreated peers Because crashing is bad. + + * apps/app_dial.c, main/features.c, apps/app_queue.c: Merged + revisions 315644 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315644 | twilson | 2011-04-26 14:39:01 -0700 + (Tue, 26 Apr 2011) | 32 lines Merged revisions 315643 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315643 | twilson | 2011-04-26 14:27:44 -0700 + (Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) + | 18 lines Allow transfer loops without allowing forwarding loops + We try to avoid the situation where two phones may be forwarded + to each other causing an infinite loop by storing each dialed + interface in a channel datastore and checking the list before + dialing out. This works, but currently breaks situations like A + calls B, A transfers B to C, B transfers C to A, and A transfers + C to B. Since human interaction is happening here and not an + automated forwarding loop, it should be allowed. This patch + removes the dialed_interfaces datastore when a call is bridged (a + suggestion from the brilliant mmichelson). If a call is being + bridged, it should be safe to assume that we aren't stuck in a + loop. Since we are now handling this is the bridge code, the + previous attempts at handling it in app_dial and app_queue are + removed. Review: https://reviewboard.asterisk.org/r/1195/ + ........ ................ ................ + +2011-04-26 22:18 +0000 [r315649] Richard Mudgett + + * main/pbx.c, /: Merged revisions 315645 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) + | 21 lines The 'e' special extension fails to trigger in at least + two cases. The 'e' extension is a fall back for the 'i', 't', or + 'T' extensions if any of them do not exist. Many of the places + the 'e' extension was supposed to be invoked fail because the + priority was set wrong. There were two places where the 'e' + extension was not even checked for fall back. * Made invoke the + 'e' extension similarly to the previous 'i', 't', or 'T' + extension check and added the 'e' extension as a fall back to the + two missing locations. * Prioritized and optimized some hangup + tests associated with the 'e' extension. (closes issue #19136) + Reported by: kshumard Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/1196/ ........ + +2011-04-26 19:38 +0000 [r315504] Tilghman Lesher + + * include/asterisk/select.h, /: Merged revisions 315503 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315503 | tilghman | 2011-04-26 14:32:50 -0500 + (Tue, 26 Apr 2011) | 28 lines Merged revisions 315502 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315502 | tilghman | 2011-04-26 14:22:52 -0500 + (Tue, 26 Apr 2011) | 21 lines Merged revisions 315501 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) + | 14 lines Fix the bounds-checking code. The code that set the + bit within the select bitfield was correct, but the + bounds-checking code was not. The change to that line uses the + new _bitsize macro for clarity. Also, FD_ZERO macro did not + zero-out anything but the first word of the bitfield, so this + could have caused problems with modules using that macro with the + expanded bitfield. (closes issue #18773) Reported by: jamicque + Patches: 20110423__issue18773.diff.txt uploaded by tilghman + (license 14) Tested by: chris-mac ........ ................ + ................ + +2011-04-26 18:02 +0000 [r315453] Richard Mudgett + + * apps/app_dial.c, /: Merged revisions 315452 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r315452 | rmudgett | 2011-04-26 13:00:34 -0500 (Tue, 26 Apr 2011) + | 1 line Add missing set of name valid flag when dialing. + ........ + +2011-04-26 17:41 +0000 [r315447] Russell Bryant + + * channels/chan_local.c, /: Merged revisions 315446 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r315446 | russell | 2011-04-26 12:40:23 -0500 (Tue, 26 + Apr 2011) | 14 lines chan_local: resolve a deadlock. This patch + resolves a fairly complex deadlock that can occur with the + combination of chan_local and a dialplan switch, such as dynamic + realtime extensions, which pulls autoservice into the picture + when doing a dialplan lookup. (closes issue #18818) Reported by: + nic Patches: issue18818.patch uploaded by jthurman (license 614) + 18818.v1.txt uploaded by russell (license 2) Tested by: nic, + jthurman, kterzi, steve-howes, sysreq, IshMalik ........ + +2011-04-26 02:21 +0000 [r315395] Paul Belanger + + * /, pbx/pbx_config.c: Merged revisions 315394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315394 | pabelanger | 2011-04-25 22:18:50 -0400 + (Mon, 25 Apr 2011) | 14 lines Merged revisions 315393 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr + 2011) | 7 lines Add back CLI command 'dialplan save' (closes + issue #19140) Reported by: lmadsen Patches: + __20110419_dialplan_save.patch.txt uploaded by lmadsen (license + 10) ........ ................ + +2011-04-25 21:55 +0000 [r315350] Richard Mudgett + + * /, channels/chan_mgcp.c: Merged revisions 315349 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r315349 | rmudgett | 2011-04-25 16:49:00 -0500 (Mon, 25 + Apr 2011) | 9 lines When using MGCP realtime gateway definitions, + random crashes occur. Fixed incorrect linked list node removal + for realtime gateways. (closes issue #18291) Reported by: + nahuelgreco Patches: dangling-pointers-when-pruning.patch + uploaded by nahuelgreco (license 162) ........ + +2011-04-25 19:40 +0000 [r315214-315260] Russell Bryant + + * /, formats/format_wav.c: Merged revisions 315259 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315259 | russell | 2011-04-25 14:37:32 -0500 + (Mon, 25 Apr 2011) | 24 lines Merged revisions 315258 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315258 | russell | 2011-04-25 14:31:44 -0500 + (Mon, 25 Apr 2011) | 17 lines Merged revisions 315257 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011) + | 10 lines Be more flexible with unknown chunks in wav files. + This patch makes format_wav ignore unknown chunks instead of + erroring out on them. (closes issue #18306) Reported by: jhirsch + Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch + (license 1156) ........ ................ ................ + + * /, channels/chan_sip.c: Merged revisions 315213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315213 | russell | 2011-04-25 14:04:28 -0500 + (Mon, 25 Apr 2011) | 14 lines Merged revisions 315212 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011) + | 7 lines Don't link non-cached realtime peers into the + peers_by_ip container. (closes issue #18924) Reported by: wdoekes + Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch + uploaded by wdoekes (license 717) ........ ................ + +2011-04-25 07:17 +0000 [r315054] Alec L Davis + + * channels/chan_local.c, /: Merged revisions 315053 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315053 | alecdavis | 2011-04-25 19:14:32 +1200 + (Mon, 25 Apr 2011) | 23 lines Merged revisions 315052 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315052 | alecdavis | 2011-04-25 19:11:12 +1200 + (Mon, 25 Apr 2011) | 16 lines Merged revisions 315051 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr + 2011) | 11 lines chan_local:check_bridge() misplaced misplaced + ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't + followed, brigde remains locked. (closes issue #19176) Reported + by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis + (license 585) ........ ................ ................ + +2011-04-22 23:01 +0000 [r315002] Alec L Davis + + * channels/chan_dahdi.c, /: Merged revisions 315001 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r315001 | alecdavis | 2011-04-23 10:59:18 +1200 (Sat, 23 + Apr 2011) | 12 lines chan_dahdi: Can't return to normal ring + after distinctive ring on FXS clear a previous distinctivering + pattern before each new call (closes issue #18985) Reported by: + bromont Patches: bug18985.diff.txt uploaded by alecdavis (license + 585) Tested by: alecdavis, bromont ........ + +2011-04-22 21:33 +0000 [r314960] Matthew Nicholson + + * /, channels/chan_agent.c: Merged revisions 314959 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314959 | mnicholson | 2011-04-22 16:20:08 -0500 + (Fri, 22 Apr 2011) | 24 lines Merged revisions 314958 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r314958 | mnicholson | 2011-04-22 15:49:45 -0500 + (Fri, 22 Apr 2011) | 17 lines Merged revisions 311203,314908 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar + 2011) | 4 lines Don't hold the pvt lock while streaming a file. + ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48 + -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and + the app threads from using the asterisk channel at the same time. + ABE-2756 ........ ................ ................ + +2011-04-22 14:49 +0000 [r314824] Tzafrir Cohen + + * channels/chan_unistim.c, /, res/res_fax_spandsp.c: Merged + revisions 314779 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314779 | tzafrir | 2011-04-22 16:59:43 +0300 (ו', 22 אפר 2011) | + 2 lines Fix a few typos (shown by Lintian) ........ + +2011-04-22 14:08 +0000 [r314781] Russell Bryant + + * /, res/res_agi.c: Merged revisions 314780 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314780 | russell | 2011-04-22 09:02:23 -0500 + (Fri, 22 Apr 2011) | 18 lines Merged revisions 314778 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011) + | 11 lines Initialize buffers in getvar and getvarfull. + Initialize the buffers used to hold the result from GET VARIABLE + or GET VARIABLE FULL. The bug report shows func_read returning + garbage in the result. It assumed that the buffer passed in was + initialized, like many other functions do. In the more common + code path (through the dialplan), it is initialized, so just + initialize it here too. (closes issue #19050) Reported by: johnz + ........ ................ + +2011-04-21 22:53 +0000 [r314733-314735] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Implement AMI action PRIShowSpans. PRIShowSpans works like the + AMI action DAHDIShowChannels but for PRI spans. It is similar to + the CLI command "pri show spans". (closes issue #15980) Reported + by: dwery + + * channels/sig_pri.c: Simplify sig_pri.c:build_status(). + + * channels/chan_dahdi.c, /: Merged revisions 314732 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r314732 | rmudgett | 2011-04-21 17:38:44 -0500 (Thu, 21 + Apr 2011) | 1 line Correct DAHDIShowChannels XML documentation. + ........ + +2011-04-21 18:32 +0000 [r314666] Matthew Nicholson + + * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c, + main/http.c, configs/sip.conf.sample, configs/skinny.conf.sample, + channels/sip/include/sip.h, configs/http.conf.sample: Merged + revisions 314628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314628 | mnicholson | 2011-04-21 13:24:05 -0500 + (Thu, 21 Apr 2011) | 27 lines Merged revisions 314620 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500 + (Thu, 21 Apr 2011) | 20 lines Merged revisions 314607 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr + 2011) | 14 lines Added limits to the number of unauthenticated + sessions TCP based protocols are allowed to have open + simultaneously. Also added timeouts for unauthenticated sessions + where it made sense to do so. Unrelated, the manager interface + now properly checks if the user has the "system" privilege before + executing shell commands via the Originate action. AST-2011-005 + AST-2011-006 (closes issue #18787) Reported by: kobaz (related to + issue #18996) Reported by: tzafrir ........ ................ + ................ + +2011-04-21 18:11 +0000 [r314598] David Vossel + + * configs/confbridge.conf.sample (added), apps/confbridge (added), + bridges/bridge_softmix.c, UPGRADE.txt, + include/asterisk/channel.h, res/res_musiconhold.c, CHANGES, + apps/confbridge/conf_config_parser.c (added), main/channel.c, + include/asterisk/bridging_technology.h, + bridges/bridge_builtin_features.c, + apps/confbridge/include/confbridge.h (added), apps/Makefile, + include/asterisk/bridging_features.h, + include/asterisk/bridging.h, include/asterisk/dsp.h, + apps/app_confbridge.c, apps/confbridge/include (added), + main/bridging.c, main/dsp.c: New HD ConfBridge conferencing + application. Includes a new highly optimized and customizable + ConfBridge application capable of mixing audio at sample rates + ranging from 8khz-192khz. Review: + https://reviewboard.asterisk.org/r/1147/ + +2011-04-21 00:29 +0000 [r314551] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 314550 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314550 | twilson | 2011-04-20 17:23:04 -0700 + (Wed, 20 Apr 2011) | 13 lines Merged revisions 314549 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) + | 6 lines Don't allocate more space than necessary for a sip_pkt + This extra allocation is a hold-over from when pkt->data was a + character array. Now that it is an allocated string, just + allocate enough for the sip_pkt. ........ ................ + +2011-04-20 20:52 +0000 [r314509] David Vossel + + * main/channel.c, main/abstract_jb.c, funcs/func_jitterbuffer.c + (added), include/asterisk/channel.h, CHANGES, + include/asterisk/abstract_jb.h: Introduction of the JITTERBUFFER + dialplan function. Review: + https://reviewboard.asterisk.org/r/1157/ + +2011-04-20 19:56 +0000 [r314471] Shaun Ruffell + + * codecs/codec_dahdi.c: codec_dahdi: DAHDI still advertises formats + using the old bitfields. Previously, the DAHDI format bit fields + matched up with the Asterisk bitfields. Since the Asterisk codec + bit fields were replaced in r306010, codec_dahdi needs to contain + the formats itself. In the future, the DAHDI formats should + either change to something other than bitfields, or the bitfields + need to move from include/dahdi/kernel.h to include/dahdi/user.h. + Signed-off-by: Shaun Ruffell + +2011-04-20 16:55 +0000 [r314418] Richard Mudgett + + * /, include/asterisk/frame.h: Merged revisions 314417 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r314417 | rmudgett | 2011-04-20 11:54:02 -0500 (Wed, 20 + Apr 2011) | 1 line AST_CONTROL_XXX comment changes. ........ + +2011-04-20 16:37 +0000 [r314415] David Vossel + + * codecs/codec_resample.c: Fixes error with frame datalen being + calculated from samples when this is not allwaya accurate. + +2011-04-20 05:28 +0000 [r314359] Terry Wilson + + * main/lock.c, /: Merged revisions 314358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314358 | twilson | 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011) + | 4 lines Initialize track pointer ast_reentrancy_init checks to + see if it is NULL before initializing with calloc ........ + +2011-04-19 15:42 +0000 [r314204-314252] Leif Madsen + + * main/tcptls.c, /: Merged revisions 314251 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314251 | lmadsen | 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011) + | 8 lines Use SSLv23_client_method instead of old SSLv2 only. + (closes issue #19095) (closes issue #19138) Reported by: tzafrir + Patches: no_ssl2.diff uploaded by tzafrir (license 46) Tested by: + russell, chazzam ........ + + * /, funcs/func_channel.c: Merged revisions 314206 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314206 | lmadsen | 2011-04-19 09:28:15 -0500 + (Tue, 19 Apr 2011) | 14 lines Merged revisions 314205 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011) + | 6 lines Remove duplicate documentation from func_channel.c + (closes issue #18970) Reported by: IgorG Patches: + func_channel.c.doc.diff uploaded by IgorG (license 20) ........ + ................ + + * apps/app_dial.c, /: Merged revisions 314203 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314203 | lmadsen | 2011-04-19 09:24:25 -0500 + (Tue, 19 Apr 2011) | 15 lines Merged revisions 314202 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011) + | 7 lines Update seconds to milliseconds in ast_verb output. + (closes issue #19084) Reported by: smurfix Patches: + app_dial.patch uploaded by smurfix (license 547) Tested by: + lmadsen, smurfix ........ ................ + +2011-04-19 08:22 +0000 [r314158] Olle Johansson + + * apps/app_meetme.c: Add explanation of strange flag setup in + app_meetme (stolen from Mark's message to asterisk-dev) + +2011-04-18 19:48 +0000 [r314079-314116] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c: Problems with ISDN MWI to phones. The + "controlling user number" is always the number of the voice mail + box which is identical with the subscriber number itself. This + number which is listed in the ISDN phone MWI menu cannot be + called back to contact the voice mail box. The controlling user + number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 + + * /, res/res_agi.c: Merged revisions 314069 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011) + | 22 lines The AsyncAGI command loop is lax in the value it + returns for the return status. * Return correct status: + SUCCESS/FAILED/HANGUP. Previously, abnormal exits from the + command loop such as hangup would return SUCCESS. * The "asyncagi + break" command now returns SUCCESS and is now the only way to + break the command loop with that status. Previously, it returned + FAILED. * The AMI event AsyncAGI End is no longer sent if the + AsyncAGI Start event is not sent. Previously, this happened + because of an error setting up the AGI pipes. * All executed AGI + commands now get an AsyncAGI Exec result event. Previously, if + the command returned failure (because of hangup), the command + loop just exited with FAILURE and did not send the AsyncAGI Exec + result event. * Makes sure that the channel frame queue is empty + on hangup. Review: https://reviewboard.asterisk.org/r/1183/ + ........ + + * apps/app_dial.c, /: Merged revisions 314068 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011) + | 7 lines Unclear code in app_dial.c. Make code formatting clear. + (closes issue #19134) Reported by: oej ........ + +2011-04-18 16:22 +0000 [r314018-314078] David Vossel + + * /, channels/chan_sip.c: Merged revisions 314067 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) + | 22 lines Remove the need for deadlock avoidance in chan_sip + do_monitor. Deadlock avoidance between the sip pvt and the + pvt->owner is very difficult. Now that channel's are ao2 objects, + this complication is no longer necessary. It turns out the pvt's + msg queue only exists because of deadlock avoidance (when + deadlock avoidance fails msgs were added to a queue to be + processed later), so this goes away as well. The technique used + in the new sip_lock_pvt_full() function should be used as a + template for replacing all locations where deadlock avoidance + occurs between a channel tech_pvt and the pvt's owner. My hope is + that this will begin a reversal of the invalid channel driver + locking architecture we have been using for so long. This patch + also resolves an issue where the pvt->owner gets unlocked during + processing the msg queue. (closes issue #18690) Reported by: + dvossel Review: https://reviewboard.asterisk.org/r/1182/ ........ + + * main/rtp_engine.c, /, channels/chan_sip.c, + include/asterisk/rtp_engine.h: Merged revisions 314017 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) + | 17 lines sip codec negotiation of dynamic rtp payloads error + fix This patch fixes how chan_sip handles dynamic rtp payload + types it does not understand. At the moment if a dynamic + payload's mime type does not match one we understand, the payload + does not get removed from our payload table. As a result of this, + the payload is set to whatever dynamic codec we use internally + for that payload number on outgoing INVITES. This is incorrect. + This patch fixes this by properly checking the rtpmap set + function's return code to make sure it was found. The function + can return both -1 and -2 depending on the source of the + mismatch. We were just checking -1 explicitly. Review: + https://reviewboard.asterisk.org/r/1169/ ........ + +2011-04-17 09:28 +0000 [r313980] Damien Wedhorn + + * channels/chan_skinny.c: Consolidate all new call calls to run + through new setsubstate_ringout. (closes issue #17907) Reported + by: wedhorn Patches: cleanup.stateringout.diff uploaded by + wedhorn (license 30) Tested by: salecha, wedhorn + +2011-04-17 01:28 +0000 [r313907-313944] Alexandr Anikin + + * addons/chan_ooh323.c: fix compile error from r313907 + + * addons/chan_ooh323.c: fix trivial error with set_max_datagram on + pvt->udptl + +2011-04-15 15:20 +0000 [r313867] Jonathan Rose + + * /, main/cli.c: Merged revisions 313860 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313860 | jrose | 2011-04-15 10:08:05 -0500 + (Fri, 15 Apr 2011) | 17 lines Merged revisions 313859 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) | + 10 lines Fix a Tab Completion bug that occurs due to multiple + matches on a substring. Makes word_match function in cli.c repeat + a search for a command string until a proper match is found or + the string is searched to the last point. (closes issue #17494) + Reported by: ffossard Review: + https://reviewboard.asterisk.org/r/1180/ ........ + ................ + +2011-04-14 21:53 +0000 [r313822] Terry Wilson + + * res/res_rtp_asterisk.c: Sets video mark bit on format field + correctly This fixes a regression in the media architecture + change where video frames did not have their video mark set + correctly. dvossel wrote this. twilson kindly committed this, + mmichelson found the bug. + +2011-04-14 21:02 +0000 [r313606-313781] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 313780 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 + Apr 2011) | 20 lines Leftover debug messages unconditionally sent + to the console. Executing Dial(DAHDI/1/18475551212,300,) with the + echotraining config option enabled outputs the following debug + messages unconditionally: Dialing T1847555121 on 1 Dialing www2w + on 1 * Made debug messages in my_dial_digits() normal debug + messages that do not get output unless enabled. * Reworded some + debug messages in my_dial_digits() to be clearer. * Replace + strncpy() with ast_copy_string() in my_dial_digits() which does + the same job better. (closes issue #18847) Reported by: + vmikhelson Tested by: rmudgett ........ + + * CREDITS, main/ccss.c, configs/ccss.conf.sample: Add Device State + Information CCSS for Generic Devices. Add Asterisk Device State + information and callbacks to the Call Completion Supplemental + Services for generic agents. There are currently not many devices + that have native support for CCSS. Even as the devices become + available there may be other reasons why one may choose to not + take advantage of the native abilities and stick with the generic + implementation. The generic implementation is quite capable and + could be greatly enhanced by adding device state capabilities. A + phone could then subscribe to the device state with a BLF key in + conjunction with Asterisk hints. The advantages of the device + state information would allow a single button to: request CCSS, + cancel a CCSS request, and display the current state of a CCSS + request. For example, you may have a single button that when not + lit, there is no active CCSS request. When you press that button, + the dialplan can query the DEVICE_STATE() associated with that + caller to determine whether they should be calling + CallCompletionRequest() or CallCompletionCancel(). If there is + currently a pending request, then the dialplan would cancel it. + This also has the advantage of showing the true state of a + request, which is an asynchronous call, even when + CallCompletionRequest() thinks it was successful. The actual + request could ultimately fail. Once lit, further feedback can be + provided to the caller about the current state of their request + since it will be updated by the CCSS State Machine as + appropriate. The DEVICE_STATE mapping is configurable since the + BLF being used on a given phone type may vary. The idea is to + allow some level of customization as to the phone's behavior. As + an example, you may want the BLF key to go solid once you have + requested a callback. You may then want the LED to blink + (typically ringing) when either the callback is in process, which + is a visual indication that the incoming call is the desired + callback. You may want it to blink when the callee is ready but + you are busy, giving you a visual indication that the target is + available as you may want to get off the line so that the + callback can be successful. Device state information is sent back + via the ast_devstate_prov_add() callback for any generic CCSS + device as it traverses through the state machine. You simply + provide a map between CC_STATE values and the corresponding + AST_DEVICE state values. You could then generate hints against + these states similar to what is possible today with Custom + Devstates or MeetMe states. For example, you may have an + extension 3000 that is currently associated with device SIP/3000. + You could then create a feature code for that extension that may + look something like: exten => *823000,hint,ccss:sip/3000 You + would then subscribe a BLF button to *823000 which would point to + the dialplan that handled CCSS requests/cancels using the + available DEVICE_STATE() information about ccss:sip/3000 to make + the decision about what to do. (closes issue #18788) Reported by: + p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p + lindheimer (license 558) Modified with final reviewboard + comments. Tested by: p_lindheimer, loloski Review: + https://reviewboard.asterisk.org/r/1105/ + + * /, res/res_agi.c: Merged revisions 313700 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313700 | rmudgett | 2011-04-13 17:52:47 -0500 (Wed, 13 Apr 2011) + | 5 lines Revert flushing stale AsyncAGI commands from -r313615. + It looks like it was intentional to leave any commands or + in-flight commands in the queue in case Async AGI is run again on + the call. ........ + + * /, res/res_agi.c: Merged revisions 313658 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313658 | rmudgett | 2011-04-13 12:47:43 -0500 (Wed, 13 Apr 2011) + | 2 lines Miscellaneous AGI diagnostic message cleanup and code + optimization. ........ + + * /, res/res_agi.c: Merged revisions 313615 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313615 | rmudgett | 2011-04-13 12:18:49 -0500 (Wed, 13 Apr 2011) + | 5 lines * Add missing channel lock to handle_cli_agi_add_cmd(). + * Flush any Async AGI commands left over from earlier Async AGI + control of the call. ........ + + * main/channel.c, /, res/res_agi.c: Merged revisions 313588 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 + (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 + (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) + | 41 lines Asterisk does not hangup a channel after endpoint + hangs up. If the call that the dialplan started an AGI script for + is hungup while the AGI script is in the middle of a command then + the AGI script is not notified of the hangup. There are many AGI + Exec commands that this can happen with. The reported + applications have been: Background, Wait, Read, and Dial. Also + the AGI Get Data command. * Don't wait on the Asterisk channel + after it has hung up. The channel is likely to never need + servicing again. * Restored the AGI script's ability to return + the AGI_RESULT_HANGUP value in run_agi(). It previously only + could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the + DeadAGI and AGI applications were merged. (closes issue #17954) + Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by + rmudgett (license 664) issue17954_v1.6.2.patch uploaded by + rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett + (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue + #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 + (closes issue #18935) Reported by: nvitaly Tested by: astmiv, + rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby + Tested by: rmudgett JIRA SWP-2727 Review: + https://reviewboard.asterisk.org/r/1165/ ........ + ................ ................ + +2011-04-13 15:49 +0000 [r313528] Leif Madsen + + * configs/iax.conf.sample, configs/users.conf.sample, + channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/chan_iax2.c, channels/sip/include/sip.h: Add + 'description' field for CLI and Manager output (closes issue + #19076) Reported by: lmadsen Patches: + __20110408-channel-description.txt uploaded by lmadsen (license + 10) Tested by: lmadsen Review: + https://reviewboard.asterisk.org/r/1163/ + +2011-04-13 15:23 +0000 [r313527] Richard Mudgett + + * /, apps/app_dumpchan.c: Merged revisions 313517 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011) + | 12 lines Bring the dumpchan application inline with "core show + channel". * Added fields that are in "core show channel" to + dumpchan output. * Fixed reuse of formatbuf before the previous + string stored there was used by snprintf. All output strings now + have their own buffer. * Adjusted the buffer sizes to not be so + abusive of the stack now that there are more buffers. Change + requested by oej. ........ + +2011-04-12 21:59 +0000 [r313482] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, + addons/ooh323c/src/ooLogChan.h, addons/chan_ooh323.c, + addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h, + addons/ooh323c/src/ooGkClient.h, addons/ooh323c/src/ooports.c, + addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooh323ep.h, + addons/ooh323c/src/ootypes.h, addons/ooh323c/src/ooLogChan.c, + addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooSocket.h, + addons/ooh323c/src/ooq931.c: IPv6 support for chan_ooh323 IPv6 + support for ooh323, bindaddr, peers and users ip can be IPv4 or + IPv6 addr correction for multi-homed mode (0.0.0.0 or :: + bindaddr) can work in dual 6/4 mode with :: bindaddr gatekeeper + mode isn't supported in v6 mode while (issue #18278) Reported by: + may213 Patches: ipv6-ooh323.patch uploaded by may213 (license + 454) Review: https://reviewboard.asterisk.org/r/1004/ + +2011-04-12 18:53 +0000 [r313437-313438] Jonathan Rose + + * /: blocking fix from 313436 that was already made in this commit + + * channels/chan_dahdi.c, /: Merged revisions 313435 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 also + went ahead and fixed the problem it introduces before committing. + ........ r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr + 2011) | 1 line fixing stupid mistake with putting code before + variable declaration ........ Merged revisions 313433 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | + 14 lines reload Chan_dahdi memory leak caused by variables + chan_dahdi reloading with variables set via setvar in + chan_dahdi.conf would stay in the dahdi_pvt structs for + individual channels (causing them to just continue adding the new + ones to the list) and also there was a memory leak causes by the + conf objects. This patch resolves both of these by using + ast_variables_destroy during the loading process. (closes issue + #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by + jrose (license 1225) Tested by: tilghman, jrose Review: + https://reviewboard.asterisk.org/r/1170/ ........ ........ + ........ + +2011-04-11 23:20 +0000 [r313367-313383] Richard Mudgett + + * apps/app_dial.c, /: Merged revisions 313368-313369 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 + Apr 2011) | 2 lines Backport a restructuring change from trunk to + make the next change stand out. ........ r313369 | rmudgett | + 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines Frames + from the inbound channel should go to all outbound channels in + app_dial.c. In app_dial.c:wait_for_answer() frames from the + inbound channel should be sent to all outbound channels instead + of only if there is just one outbound channel. Control frames + like AST_CONTROL_CONNECTED_LINE need to be passed to all of the + the outbound channels. This can happen if a blond transfer is + done by a remote switch on the inbound channel. JIRA AST-443 JIRA + SWP-2730 ........ + + * /, main/cli.c: Merged revisions 313366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313366 | rmudgett | 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) + | 2 lines Added "Connected Line ID" and "Connected Line ID Name" + to "core show channel" output. ........ + +2011-04-11 19:39 +0000 [r313280] Leif Madsen + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 313279 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313279 | lmadsen | 2011-04-11 14:36:40 -0500 + (Mon, 11 Apr 2011) | 21 lines Merged revisions 313278 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500 + (Mon, 11 Apr 2011) | 14 lines Merged revisions 313277 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011) + | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093) + Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by + tzafrir (license 46) ........ ................ ................ + +2011-04-11 15:47 +0000 [r313191] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 313190 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313190 | rmudgett | 2011-04-11 10:40:30 -0500 + (Mon, 11 Apr 2011) | 39 lines Merged revisions 313189 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r313189 | rmudgett | 2011-04-11 10:32:53 -0500 + (Mon, 11 Apr 2011) | 32 lines Merged revisions 313188 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) + | 25 lines Stuck channel using FEATD_MF if caller hangs up at the + right time. The cause was actually a caller hanging up just at + the end of the Feature Group D DTMF tones that setup the call. + The reason for this is a "guard timer" that's implemented using + ast_safe_sleep(100). If the caller happens to hang up AFTER the + final tone of the DTMF string but BEFORE the end of that + ast_safe_sleep(), then ast_safe_sleep() will return non-zero. + This causes the code to bounce to the end of ss_thread(), but it + does NOT tear down the call properly. This should be a rare + occurrence because the caller has to hang up at EXACTLY the right + time. Nonetheless, it was happening quite regularly on the + reporter's system. It's not easily reproducible, unless you + purposely increase the guard-time to 2000 or more. Once you do + that, you can reproduce it every time by watching the DTMF debug + and hanging up just as it ends. Simply add an ast_hangup() before + goto quit. (closes issue #15671) Reported by: jcromes Patches: + issue15671.patch uploaded by pabelanger (license 224) Tested by: + jcromes ........ ................ ................ + +2011-04-09 21:00 +0000 [r313143] Alexandr Anikin + + * addons/chan_ooh323.c, /: Merged revisions 313142 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r313142 | may | 2011-04-10 00:56:17 +0400 (Sun, 10 Apr + 2011) | 3 lines fix trivial bug in ooh323_indicate on + AST_CONTROL_SRC... check p->rtp is not null ........ + +2011-04-08 16:17 +0000 [r313100] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_ss7.h, channels/sig_pri.c, channels/sig_ss7.c: Add + private lock deadlock avoidance callback to PRI and SS7. Factor + out the equivalent function for analog. + +2011-04-07 13:42 +0000 [r313049] Jonathan Rose + + * /, main/features.c: Merged revisions 313048 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313048 | jrose | 2011-04-07 08:35:33 -0500 + (Thu, 07 Apr 2011) | 16 lines Merged revisions 313047 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) | + 9 lines Makes parking lots clear and rebuild properly when + features reload is invoked from CLI Before, default parkinglot in + context parkedcalls with ext 700 would always be present and when + reload was invoked, the previous parkinglots would not be + cleared. (closes issue #18801) Reported by: mickecarlsson Review: + https://reviewboard.asterisk.org/r/1161/ ........ + ................ + +2011-04-07 10:30 +0000 [r313003-313005] Alec L Davis + + * /, channels/sig_pri.c: Merged revisions 313001 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr + 2011) | 13 lines Fix ISDN calling subaddr User Specified Odd/Even + Flag Calculation of the Odd/Even flag was wrong. Implement + correct algo, and set odd/even=0 if data would be truncated. Only + allow automatic calculation of the O/E flag, don't let dialplan + influence. (closes issue #19062) Reported by: festr Patches: + bug19062.diff2.txt uploaded by alecdavis (license 585) Tested by: + festr, alecdavis, rmudgett ........ + + * apps/app_voicemail.c: app_voicemail: close_mailbox change + LOG_WARNING to LOG_NOTICE + +2011-04-05 18:47 +0000 [r312868-312950] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: + Merged revisions 312949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) + | 6 lines Crash if ISDN span layer 1 is down on initial load. + Regression from -r312575 B channel shifting during negotiation. * + Also combine updating the alarm flag with clearing the resetting + flag. ........ + + * /, channels/chan_sip.c: Merged revisions 312889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r312889 | rmudgett | 2011-04-05 11:19:35 -0500 (Tue, 05 Apr 2011) + | 5 lines Add 416 response to OPTIONS packet. RFC3261 Section + 11.2 says the response code to an OPTIONS packet needs to be the + same as if it were an INVITE. ........ + + * /, channels/chan_sip.c: Merged revisions 312866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r312866 | rmudgett | 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) + | 15 lines Responding to OPTIONS packet with 404 because Asterisk + not looking for "s" extension. The get_destination() function was + not using the "s" extension when the request URI did not specify + an extension. This is a regression caused when the URI parsing + code was extracted into parse_uri(). Made get_destination() + substitute the "s" extension when the parsed URI results in an + empty string. (closes issue #18348) Reported by: shmaize Patches: + issue18348_v1.8.patch uploaded by rmudgett (license 664) Tested + by: shmaize ........ + +2011-04-05 14:16 +0000 [r312767] Matthew Nicholson + + * main/manager.c, /, configs/manager.conf.sample: Merged revisions + 312766 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312766 | mnicholson | 2011-04-05 09:14:50 -0500 + (Tue, 05 Apr 2011) | 22 lines Merged revisions 312764 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312764 | mnicholson | 2011-04-05 09:13:07 -0500 + (Tue, 05 Apr 2011) | 15 lines Merged revisions 312761 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr + 2011) | 8 lines Limit the number of unauthenticated manager + sessions and also limit the time they have to authenticate. + AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested + by: mnicholson ........ ................ ................ + +2011-04-05 13:55 +0000 [r312756] Jonathan Rose + + * apps/app_meetme.c: Minor change to 'L' option for meetme to + include some verb statements for the option. + +2011-04-04 19:31 +0000 [r312716] Richard Mudgett + + * channels/sig_pri.c: Remove the channel parameter from + sig_pri_handle_subcmds(). It was only used in a debug message and + may not be correct anyway. + +2011-04-04 17:37 +0000 [r312678-312680] Jonathan Rose + + * pbx/pbx_config.c: In handle_cli_dialplan_add_extension, const + char pointer *into_context is used instead of a->argv[5] to + improve readability. + + * CHANGES, pbx/pbx_config.c: Makes 'dialplan add extension' create + the specified context if it does not already exist. If the user + invokes 'dialplan add extension' into a non-existing context, the + context will be created and a message informing the user of the + context being created will be issued in cli. (closes issue + #17431) Reported by: leearcher Patches: context_auto_create.diff + uploaded by kobaz (license 834) Tested by: leearcher, kobaz, + jrose + +2011-04-04 16:17 +0000 [r312579] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: + Merged revisions 312575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312575 | rmudgett | 2011-04-04 11:10:50 -0500 + (Mon, 04 Apr 2011) | 52 lines Merged revisions 312574 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500 + (Mon, 04 Apr 2011) | 45 lines Merged revisions 312573 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) + | 38 lines Issues with ISDN calls changing B channels during call + negotiations. The handling of the PROCEEDING message was not + using the correct call structure if the B channel was changed. + (The same for PROGRESS.) The call was also not hungup if the new + B channel is not provisioned or is busy. * Made all call + connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, + ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are + using the correct structure and B channel. If there is any + problem with the operations then the call is now hungup with an + appropriate cause code. * Made miscellaneous messages + (INFORMATION, FACILITY, NOTIFY) find the correct structure by + looking for the call and not using the channel ID. NOTIFY is an + exception with versions of libpri before v1.4.11 because a call + pointer is not available for Asterisk to use. * Made all hangup + messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct + structure by looking for the call and not using the channel ID. + (closes issue #18313) Reported by: destiny6628 Tested by: + rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: + destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue + #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The + issues fixed here are most likely causing this JIRA issue.) JIRA + DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) + ........ ................ ................ + +2011-04-01 23:17 +0000 [r312462-312510] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 312509 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01 + Apr 2011) | 22 lines When a call going out an NT-PTMP port gets + rejected, Asterisk crashes. If a call is sent to an ISDN phone + that rejects the call with RELEASE_COMPLETE(cause: call + reject(21), or busy(17)) Asterisk crashes. I could not get my + setup to crash. However, I could see the possibility from a race + condition between queuing an AST_CONTROL_BUSY to the core and + then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is + processed before the AST_CONTROL_HANGUP is queued, the + ast_channel could be destroyed out from under chan_misdn. Avoid + this particular crash scenario by not queueing the + AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes + issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch + uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy + JIRA SWP-2679 ........ + + * /, main/ccss.c: Merged revisions 312461 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) + | 25 lines CallCompletionRequest()/CallCompletionCancel() exit + non-zero if fail. The + CallCompletionRequest()/CallCompletionCancel() dialplan + applications exit nonzero on normal failure conditions. The + nonzero exit causes the dialplan to hangup immediately. The + dialplan author has no opportunity to report success/failure to + the user. * Made always return zero so the dialplan can continue. + * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and + CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. + Also documented the values set. * Reduced the warning about no + core instance in CallCompletionCancel() to a debug message. It is + a normal event and should not be output at the WARNING level. + (closes issue #18763) Reported by: p_lindheimer Patches: + ccss.patch uploaded by p lindheimer (license 558) Modified Tested + by: p_lindheimer, rmudgett JIRA SWP-3042 ........ + +2011-04-01 17:28 +0000 [r312384-312423] Jonathan Rose + + * channels/chan_dahdi.c: Fixing bad line break from 312384 + + * channels/chan_dahdi.c, include/asterisk/dsp.h, CHANGES, + main/dsp.c: New Feature for chan_dahdi. 4 length pattern + matching. In chan_dahdi.conf, the user can now use length 4 + patterns in addition to the usual length 2 patterns. The s ntax + remains the same and the method used to track the pattern history + will only change when using the length 4 patterns. (closes issue + SWP-3250) Code: jrose rmudgett + +2011-04-01 10:59 +0000 [r312289] Tilghman Lesher + + * include/asterisk/select.h, /, addons/cdr_mysql.c, + main/asterisk.c: Merged revisions 312286,312288 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312286 | tilghman | 2011-04-01 05:44:33 -0500 + (Fri, 01 Apr 2011) | 2 lines Reload must react correctly against + a possibly changed table, so dropping the conditional reload + flag. ................ r312288 | tilghman | 2011-04-01 05:58:45 + -0500 (Fri, 01 Apr 2011) | 21 lines Merged revisions 312287 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312287 | tilghman | 2011-04-01 05:51:24 -0500 + (Fri, 01 Apr 2011) | 14 lines Merged revisions 312285 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) + | 7 lines Found some leaking file descriptors while looking at + ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej + Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman + (license 14) ........ ................ ................ + +2011-04-01 09:08 +0000 [r312118-312212] Alec L Davis + + * /, apps/app_voicemail.c: Merged revisions 312211 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312211 | alecdavis | 2011-04-01 22:03:11 +1300 + (Fri, 01 Apr 2011) | 36 lines Merged revisions 312210 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300 + (Fri, 01 Apr 2011) | 29 lines Merged revisions 312174 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr + 2011) | 23 lines voicemail: get real last_message_index and + count_messages, ODBC resequence change last_message_index to read + the max msgnum stored in the database change count_messages to + actually count the number of messages. last_message_index change: + This fixed overwriting of the last message if msgnum=0 was + missing. Previously every incoming message would overwrite + msgnum=1. count_messages change: allows us to detect when + requencing is required in opneA_mailbox. resequence enabled for + ODBC storage: Assists with fixing up corrupt databases with gaps, + but only when a user actively opens there mailboxes. (closes + issue #18692,#18582,#19032) Reported by: elguero Patches: based + on odbc_resequence_mailbox2.1.diff uploaded by elguero (license + 37) Tested by: elguero, nivek, alecdavis Review: + https://reviewboard.asterisk.org/r/1153/ ........ + ................ ................ + + * /, apps/app_voicemail.c: Merged revisions 312117 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312117 | alecdavis | 2011-04-01 20:32:12 +1300 + (Fri, 01 Apr 2011) | 29 lines Merged revisions 312103 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300 + (Fri, 01 Apr 2011) | 22 lines Merged revisions 312070 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr + 2011) | 16 lines app_voicemail: close_mailbox needs to respect + additional messages while mailbox is open. close_mailbox leave + gaps in message sequence if messages are deleted and new messages + arrive during this time, this is because the shuffle down to slot + 0, only shuffles the number of pre-existing messages when mailbox + is opened, ignoring new arrivals. Fix: in close_mailbox + re-evaluate number of messages before the shuffle, this then + includes new arrivals. Happens on filebased or ODBC storage. + (issues #19032,#18582,#18692,#18998) Reported by: + alecdavis,tootai,afosorio Review: + https://reviewboard.asterisk.org/r/1153/ ........ + ................ ................ + +2011-03-31 20:12 +0000 [r311981-312023] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 312022 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 + Mar 2011) | 14 lines chan_misdn segfaults when DEBUG_THREADS is + enabled. The segfault happens because jb->mutexjb is + uninitialized from the ast_malloc(). The internals of + ast_mutex_init() were assuming a nonzero value meant mutex + tracking initialization had already happened. Recent changes to + mutex tracking code to reduce excessive memory consumption + exposed this uninitialized value. Converted misdn_jb_init() to + use ast_calloc() instead of ast_malloc(). Also eliminated + redundant zero initialization code in the routine. (closes issue + #18975) Reported by: irroot ........ + + * include/asterisk/channel.h: Fix function reference in comment. + +2011-03-31 06:44 +0000 [r311931] Tilghman Lesher + + * /, configs/cdr_mysql.conf.sample: Merged revisions 311930 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311930 | tilghman | 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011) + | 6 lines Incorrect default example; the field is actually + internally named "clid", not "callerid". (closes issue #19040) + Reported by: wcselby Tested by: tilghman ........ + +2011-03-30 01:57 +0000 [r311875] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 311874 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r311874 | rmudgett | 2011-03-29 20:56:05 -0500 (Tue, 29 + Mar 2011) | 1 line Update some setup_dahdi_int() comments. + ........ + +2011-03-29 08:33 +0000 [r311806] Tilghman Lesher + + * cel/cel_odbc.c, /: Merged revisions 311799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311799 | tilghman | 2011-03-29 02:08:39 -0500 (Tue, 29 Mar 2011) + | 7 lines Remove extraneous check from integer-type fields. + (closes issue #19027) Reported by: mlehner Review: + https://reviewboard.asterisk.org/r/1149/ ........ + +2011-03-28 22:00 +0000 [r311752] Russell Bryant + + * /, apps/app_voicemail.c: Merged revisions 311751 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r311751 | russell | 2011-03-28 17:00:01 -0500 (Mon, 28 + Mar 2011) | 2 lines Cross-reference VoiceMail() and + VoiceMailMain() in the xml docs. ........ + +2011-03-27 21:49 +0000 [r311688] Alexandr Anikin + + * addons/chan_ooh323.c, /: Merged revisions 311687 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r311687 | may | 2011-03-28 01:47:13 +0400 (Mon, 28 Mar + 2011) | 2 lines correct return values in ooh323_indicate for + AST_CONTROL_T38_PARAMETERS ........ + +2011-03-23 21:55 +0000 [r311613-311616] Brett Bryant + + * /, apps/app_meetme.c: Merged revisions 311615 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) + | 8 lines This patch fixes a bug with MeetMe behavior where the + 'P' option for always prompting for a pin is ignored for the + first caller. (closes issue #18070) Reported by: mav3rick Review: + https://reviewboard.asterisk.org/r/1132/ ........ + + * /, channels/sip/reqresp_parser.c: Merged revisions 311612 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) + | 9 lines Fix a possible crash in sip/reqresp_parser.c that is + caused by a possible null value. (closes issue #18821) Reported + by: cmaj Patches: + patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx + uploaded by cmaj (license 830) ........ + +2011-03-23 02:51 +0000 [r311559] Terry Wilson + + * /, channels/sip/reqresp_parser.c: Merged revisions 311558 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) + | 5 lines Don't use static declared buf in parse_name_andor_addr + This function isn't used anywhere yet, but we definitely don't + want to keep the same value for buf between calls to the + function. ........ + +2011-03-22 15:26 +0000 [r311498] David Vossel + + * /, apps/app_meetme.c: Merged revisions 311497 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r311497 | dvossel | 2011-03-22 10:25:24 -0500 + (Tue, 22 Mar 2011) | 9 lines Merged revisions 311496 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 + Mar 2011) | 2 lines Fixes memory leak in MeetMe AMI action + ........ ................ + +2011-03-18 19:05 +0000 [r311427] Jonathan Rose + + * CHANGES, apps/app_followme.c: Adds an option to FollowMe that + isn't useful for the bug it was made to solve. Still, due to the + nature of FollowMe, it makes sense to have this option since it + keeps apps bound to channels that would otherwise go away from + being lost. + +2011-03-18 16:27 +0000 [r311385] David Vossel + + * codecs/codec_resample.c: Remove libresample dependency from + codec_resample.c + +2011-03-18 16:24 +0000 [r311373] Jonathan Rose + + * /, channels/chan_sip.c, res/res_fax.c, res/res_jabber.c: Merged + revisions 311352 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | + 10 lines Changes some print statements/events to use a blank + string in place of NULL if the string in question is NULL. This + is supposed to improve Solaris compatibility since Solaris goes + berserk when trying to output NULL strings. (closes issue #18759) + Reported by: bklang Patches: null-strings.patch uploaded by + bklang (license 919) ........ + +2011-03-18 16:03 +0000 [r311343] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 311342 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311342 | mnicholson | 2011-03-18 11:02:50 -0500 (Fri, 18 Mar + 2011) | 2 lines Properly populate the LOCALSTATIONID channel + variable. ........ + +2011-03-18 03:00 +0000 [r311296-311298] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 311297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) + | 12 lines Race condition when ISDN CallRerouting/CallDeflection + invoked. The queued AST_CONTROL_BUSY could sometimes be processed + before the call_forward dial string is recognized. * Moved + setting the call_forwarding dial string after sending a response + to the initiator and just queue an empty frame to wake up the + media thread instead of an AST_CONTROL_BUSY. * Added check for + empty rerouting/deflection number and respond with an error. + ........ + + * apps/app_dial.c, /: Merged revisions 311295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r311295 | rmudgett | 2011-03-17 21:22:07 -0500 + (Thu, 17 Mar 2011) | 35 lines Merged revision 310986 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, + 16 Mar 2011) | 28 lines Dial() o option broke when connected line + feature added. The patch restores the o option behavior and adds + the ability to specify the CallerID. The Dial o and f options are + complementary to each other. The o option stores the CallerID on + the outgoing channel as the channel's CallerID. The f option + forces the CallerID sent by the outgoing channel. o(x) - The + argument 'x' is optional. If not present, then specify that the + CallerID that was present on the *calling* channel be stored as + the CallerID on the *called* channel. This was the behavior of + Asterisk 1.0 and earlier. If present, then specify the CallerID + stored on the *called* channel. Note that o(${CALLERID(all)}) is + similar to option o without parameters. f(x) - The argument 'x' + is optional and its presence changes the behavior of this option. + If not present, then force the outgoing CallerID on a + call-forward or deflection to the dialplan extension for this + Dial() using a dialplan 'hint'. For example, some PSTNs do not + allow CallerID to be set to anything other than the numbers + assigned to you. If present, then force the outgoing CallerID to + 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by + rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA + SWP-3096 .......... ................ + +2011-03-17 19:05 +0000 [r311198] Jonathan Rose + + * /, apps/app_chanspy.c: Merged revisions 311197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | + 11 lines This fixes a nasty chanspy bug which was causing a + channel leak every time a spied on channel made a call. In + addition to the above, it makes certain channel destruction + occurs so that applications don't get stuck waiting for datastore + destruction while monitored by chanspy. (closes issue #18742) + Reported by: jkister Tested by: jkister, jcovert, jrose Review: + http://reviewboard.digium.internal/r/106/ ........ + +2011-03-17 15:02 +0000 [r311142] Matthew Nicholson + + * main/manager.c, /: Merged revisions 311141 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r311141 | mnicholson | 2011-03-17 10:00:33 -0500 + (Thu, 17 Mar 2011) | 11 lines Merged revisions 311140 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar + 2011) | 4 lines Don't write items to the manager socket twice. + AST-2011-003 (closes issue 0018987) Reported by: ks-steven + ........ ................ + +2011-03-17 10:51 +0000 [r311051] Alec L Davis + + * /, configs/indications.conf.sample: Merged revisions 311050 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r311050 | alecdavis | 2011-03-17 23:49:41 +1300 + (Thu, 17 Mar 2011) | 24 lines Merged revisions 311049 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300 + (Thu, 17 Mar 2011) | 17 lines Merged revisions 311048 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar + 2011) | 12 lines Remove extra quote in indications.conf Picking + low hanging fruit. (closes issue #18971) Reported by: IgorG + Patches: based on indications.conf.sample.diff uploaded by IgorG + (license 20) Tested by: IgorG ........ ................ + ................ + +2011-03-16 19:51 +0000 [r310941-311001] Terry Wilson + + * main/tcptls.c, /: Merged revisions 310999 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310999 | twilson | 2011-03-16 14:47:59 -0500 + (Wed, 16 Mar 2011) | 18 lines Merged revisions 310998 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) + | 11 lines Fix crash on fdopen failure See security advisory + AST-2011-004 (closes issue #18845) Reported by: cmaj Patches: + patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt + uploaded by cmaj (license 830) + patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt + uploaded by cmaj (license 830) Tested by: cmaj, twilson ........ + ................ + + * main/manager.c, /: Merged revisions 310993 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310993 | twilson | 2011-03-16 14:26:57 -0500 + (Wed, 16 Mar 2011) | 11 lines Merged revisions 310992 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) + | 4 lines Don't keep trying to write to a closed connection See + security advisory AST-2011-003. ........ ................ + + * /, main/features.c: Merged revisions 310902 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310902 | twilson | 2011-03-16 12:19:57 -0500 + (Wed, 16 Mar 2011) | 43 lines Merged revisions 310889 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r310889 | twilson | 2011-03-16 12:03:27 -0500 + (Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) + | 29 lines Don't delay DTMF in core bridge while listening for + DTMF features This patch is mostly the work of Olle Johansson. I + did some cleanup and added the silence generating code if + transmit_silence is set. When a channel listens for DTMF in the + core bridge, the outbound DTMF is not sent until we have received + DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds + of DTMF to Asterisk, which sends no audio for those 4 seconds. + Some products see this delay and the time skew on RTP packets + that results and start ignoring the audio that is sent afterward. + With this change, the DTMF_BEGIN frame is inspected and checked. + If it matches a feature code, we wait for DTMF_END and activate + the feature as before. If transmit_silence=yes in asterisk.conf, + silence is sent if we paritally match a multi-digit feature. If + it doesn't match a feature, the frame is forwarded along with the + DTMF_END without delay. By doing it this way, DTMF is not + delayed. (closes issue #15642) Reported by: jasonshugart Patches: + issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license + 396) Tested by: globalnetinc, jde (closes issue #16625) Reported + by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ + Review: https://reviewboard.asterisk.org/r/1125/ ........ + ................ ................ + +2011-03-15 01:49 +0000 [r310835] Tilghman Lesher + + * addons/chan_ooh323.c, /: Merged revisions 310834 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r310834 | tilghman | 2011-03-14 20:48:25 -0500 (Mon, 14 + Mar 2011) | 2 lines Fix branch compile. ........ + +2011-03-15 01:36 +0000 [r310833] Alec L Davis + + * /, main/utils.c: Merged revisions 310781 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310781 | alecdavis | 2011-03-15 14:00:55 +1300 (Tue, 15 Mar + 2011) | 10 lines core show locks: display ThreadID in hexadecimal + Allow easier cross referencing of thread ID's with GDB backtraces + (closes issue #18968) Reported by: alecdavis Patches: + bug18968.diff.txt uploaded by alecdavis (license 585) ........ + +2011-03-14 21:51 +0000 [r310735] Alexandr Anikin + + * addons/chan_ooh323.c, addons/ooh323c/src/ooCapability.c, /, + addons/ooh323c/src/ooCalls.h: Merged revisions 310734 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 (closes + issue #18693) ........ r310734 | may | 2011-03-15 00:45:53 +0300 + (Tue, 15 Mar 2011) | 12 lines Introduce t.38 parameters control + functionality not full but enough for Send/RcvFax support + Introduce t.38 controls between asterisk core and channel/proto + layers. Not all parameters are transferred from proto layers but + *Fax apps tested and work ok. (issue #18693) Reported by: + benngard2 Patches: issue-18693.patch uploaded by may213 (license + 454) ........ + +2011-03-14 16:55 +0000 [r310637] Richard Mudgett + + * /, main/callerid.c: Merged revisions 310636 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310636 | rmudgett | 2011-03-14 11:50:59 -0500 + (Mon, 14 Mar 2011) | 39 lines Merged revisions 310635 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500 + (Mon, 14 Mar 2011) | 32 lines Merged revisions 310633 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) + | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and + TDM410 The last character in the caller id message is getting a + framing error. The checksum is the last character in the message. + A framing error in the checksum could be because: 1) The sender + did not send a full stop bit. 2) The sender cut off the FSK + carrier too soon. 3) The sender opted to send zero of the + specified zero to 10 trailing mark bits and round-off errors in + the code resulted in the code not being where it thought it was + in the demodulated bit stream. Bit 8 of 'b' is set when parity + error. Bit 9 of 'b' is set when framing error. Made ignore the + framing and parity error bits if the errored character is the + checksum. We can tolerate a framing/parity error there. The + checksum character validates the message. (closes issue #18474) + Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek + (license 636) (with modifications) Tested by: nivek ........ + ................ ................ + +2011-03-14 15:40 +0000 [r310547-310588] Jonathan Rose + + * /, funcs/func_volume.c: Merged revisions 310587 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310587 | jrose | 2011-03-14 10:27:57 -0500 + (Mon, 14 Mar 2011) | 15 lines Merged revisions 310585 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | + 8 lines Adds 'p' as an option to func_volume. When it is on, the + old behavior with DTMF controlling volume adjustment will be + enforced. When it is off, DTMF will not be processed by the + function. Programmed by Jonathan Rose Reviewed by David Vossel, + Leif Madsen, and Russell Bryant + http://reviewboard.digium.internal/r/93/ ........ + ................ + + * main/audiohook.c: Fixes null reference bug introduced by audio + hook changes that affects various OS distributions. Thanks David. + +2011-03-12 20:42 +0000 [r310416-310500] Tilghman Lesher + + * /, pbx/pbx_ael.c: Merged revisions 310462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310462 | tilghman | 2011-03-12 14:27:54 -0600 + (Sat, 12 Mar 2011) | 45 lines Merged revisions 310448 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r310448 | tilghman | 2011-03-12 14:24:54 -0600 + (Sat, 12 Mar 2011) | 38 lines Recorded merge of revisions 310435 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011) + | 31 lines Add AELSub, which provides a stable entry point into + AEL subroutines. This commit needs some explanation, given that + we're adding a new application into an existing release branch. + This is generally a violation of our release policy, except in + very limited circumstances, and I believe this is one of those + circumstances. The problem that this solves is one of the sanity + of using multiple dialplan languages to define a dialplan. In the + case of the reporter, he or she is using AEL is define + subroutines, while using Realtime extensions to invoke those + subroutines. While you can do this, it's based upon the reality + of AEL using actual dialplan extensions; however, there is no + guarantee that the details of _how_ AEL is compiled into + extensions will remain stable. In fact, at the time of this + commit, it has already changed twice, once in a fundamental way. + Now normally, a new application would only be added to trunk. + However, this application is explicitly to create a stable + user-level API between versions, and adding it to trunk only will + not solve the user's problem of switching between 1.6.2 and 1.8, + nor will it help anybody switching from 1.8 to 1.10. Therefore, + it needs to go into existing release branches. For the sake of + consistency, and also because one of the changes was between 1.4 + and 1.6.x, I am also electing to commit this to 1.4. (closes + issue #18910) Reported by: alexandrekeller Patches: + 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman + (license 14) 20110304__issue18919__1.4.diff.txt uploaded by + tilghman (license 14) Tested by: alexandrekeller ........ + ................ ................ + + * /, funcs/func_odbc.c: Merged revisions 310415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310415 | tilghman | 2011-03-12 14:05:46 -0600 + (Sat, 12 Mar 2011) | 14 lines Merged revisions 310414 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011) + | 7 lines Transactional handles should be used for the insertbuf, + if available. Also, fix a possible resource leak. (closes issue + #18943) Reported by: irroot ........ ................ + +2011-03-11 18:54 +0000 [r310373] Jonathan Rose + + * include/asterisk/audiohook.h, main/audiohook.c, CHANGES, + apps/app_mixmonitor.c: Mix Monitor: Now with r and t options. + +2011-03-11 15:09 +0000 [r310332] Kevin P. Fleming + + * Makefile, configure, codecs/gsm/Makefile, configure.ac, + makeopts.in, codecs/lpc10/Makefile: Use "-march=native" when + possible. Recent versions of GCC have a tuning option value of + 'native', which causes the compiler to optimize the build for the + CPU the compile is performed on. Since most people are building + Asterisk on the machine they plan to run it on, the configure + script and build system will now use this value unless a + different value is specified by the user in CFLAGS when the + configure script is executed. In addition, this value will be + used for building the GSM and LPC10 codecs as well, in preference + to the logic that has been in their Makefiles forever to optimize + for certain types of CPUs. + +2011-03-11 06:56 +0000 [r310288] Alec L Davis + + * main/rtp_engine.c, /: Merged revisions 310287 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar + 2011) | 17 lines remote_bridge_loop: prevent segfault when after + transfer of IAX2 of DAHDI call If the channel condition is one of + the following after breaking out of the loop, don't try to + update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx + 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes + issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt + uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81 + Review: https://reviewboard.asterisk.org/r/1128/ ........ + +2011-03-10 16:09 +0000 [r310241] Terry Wilson + + * main/manager.c, /, res/res_phoneprov.c: Merged revisions 310240 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) + | 13 lines Add \r\n to remaining http headers passed to + ast_http_send r309204 changed the behavior of ast_http_send. It + now requires headers to be passed with trailing \r\n. This change + updates the remaining instances in the code that did not pass the + \r\n. (closes issue #18186) Reported by: nivaldomjunior Patches: + res_phoneprov.c.diff uploaded by lathama (license 1028) + manager.diff.txt uploaded by twilson (license 396) Tested by: + lathama ........ + +2011-03-10 15:28 +0000 [r310238] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 310231 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar + 2011) | 9 lines Be more tolerant of what URI we accept for call + completion PUBLISH requests. (closes issue #18946) Reported by: + GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson + (license 60) Tested by: GeorgeKonopacki ........ + +2011-03-10 05:54 +0000 [r310143] Tilghman Lesher + + * res/res_config_odbc.c, /, funcs/func_odbc.c, + apps/app_voicemail.c: Merged revisions 310142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310142 | tilghman | 2011-03-09 23:53:29 -0600 + (Wed, 09 Mar 2011) | 19 lines Merged revisions 310141 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600 + (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) + | 5 lines Initialize column size to 0 to deal with a potential + UnixODBC bug on 64-bit systems. (closes issue #18295) Reported + by: pruiz ........ ................ ................ + +2011-03-08 20:34 +0000 [r310089] Jonathan Rose + + * /, channels/sip/dialplan_functions.c: Merged revisions 310088 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) | + 9 lines Returns with an error notice if CHANNEL function of SIP + channel is read without arguments. (Closes issue #18653) Reported + by: wuwu Patches: diff.patch uploaded by jrose (license 1225) + Tested by: jrose ........ + +2011-03-08 18:19 +0000 [r310045] Terry Wilson + + * /, res/res_calendar.c: Merged revisions 310039 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310039 | twilson | 2011-03-08 10:10:50 -0800 (Tue, 08 Mar 2011) + | 11 lines Spelling fix in "calendar show calendar" + s/Cartegories/Catagories/ (closes issue #18931) Reported by: + pdugas Patches: res_calendar.c.patch uploaded by pdugas (license + 1222) Review: [full review board URL with trailing slash] + ........ + +2011-03-08 16:46 +0000 [r309996] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 309994 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) + | 1 line Make pri parameter description consistent. ........ + +2011-03-07 22:16 +0000 [r309859] Jonathan Rose + + * /, apps/app_mixmonitor.c: Merged revisions 309858 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309858 | jrose | 2011-03-07 16:07:25 -0600 + (Mon, 07 Mar 2011) | 22 lines Merged revisions 309857 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r309857 | jrose | 2011-03-07 16:04:44 -0600 + (Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | + 8 lines Bug fix for MixMonitor involving filenames with '.' not + in the extension Closes issue #18391) Reported by: pabelanger + Patches: bugfix.patch uploaded by jrose (license 1225) Tested by: + jrose ........ ................ ................ + +2011-03-07 01:01 +0000 [r309809] Tilghman Lesher + + * channels/chan_dahdi.c, /, configure, + include/asterisk/autoconfig.h.in, main/ast_expr2f.c, + configure.ac, main/ast_expr2.fl: Merged revisions 309808 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309808 | tilghman | 2011-03-06 18:54:42 -0600 + (Sun, 06 Mar 2011) | 14 lines Merged revisions 309251 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) + | 7 lines Revert previous 2 commits, and instead conditionally + redefine the same macro used in flex 2.5.35 that clashed with our + workaround. Not surprisingly, the workaround was exactly the same + code as was provided by the Flex maintainers, albeit in two + different places, in different macros. This should fix the + FreeBSD builds, which have an older version of Flex. ........ + ................ + +2011-03-07 00:14 +0000 [r309766] Mark Michelson + + * /, configs/sip.conf.sample: Merged revisions 309765 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, + 06 Mar 2011) | 3 lines Indicate that Asterisk uses the Allow + header to determine if MESSAGE requests should be sent. ........ + +2011-03-05 17:53 +0000 [r309721] Moises Silva + + * channels/chan_dahdi.c, /: Merged revisions 309720 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar + 2011) | 6 lines Fix caller id passed to openr2_chan_make_call + (closes issue #18894) Reported by: malufrj Tested by: moy + ........ + +2011-03-05 10:30 +0000 [r309679] Tilghman Lesher + + * /, main/asterisk.c: Merged revisions 309678 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309678 | tilghman | 2011-03-05 04:29:30 -0600 + (Sat, 05 Mar 2011) | 14 lines Merged revisions 309677 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) + | 7 lines Missed part of the conversion when we started passing + ppid to astcanary. (closes issue #18850) Reported by: viraptor + Patches: canary_ppid.patch uploaded by viraptor (license 543) + ........ ................ + +2011-03-04 23:22 +0000 [r309640] Terry Wilson + + * configs/calendar.conf.sample, include/asterisk/calendar.h, + CHANGES, res/res_calendar.c: Add setvar option to calendaring + Adding the setvar option with variable substitution on the value + allows things like setting the outbound caller id name to the + summary of a calendar event, etc. Values could be chained + together as they are appended in order to do some scripting if + necessary. Review: https://reviewboard.asterisk.org/r/1134/ + +2011-03-04 19:38 +0000 [r309493-309587] Matthew Nicholson + + * /, pbx/pbx_lua.c: Merged revisions 309585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309585 | mnicholson | 2011-03-04 13:38:25 -0600 + (Fri, 04 Mar 2011) | 9 lines Merged revisions 309584 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, + 04 Mar 2011) | 2 lines Restore mysterious lua_pushvalue() call + removed in r309494. The mystery has been solved. ........ + ................ + + * /, pbx/pbx_lua.c: Merged revisions 309542 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309542 | mnicholson | 2011-03-04 13:00:33 -0600 + (Fri, 04 Mar 2011) | 11 lines Merged revisions 309541 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar + 2011) | 4 lines Check for errors from fseek() when loading config + file, properly abort on errors from fread(), and supply a + traceback for errors generated when loading the config file. + Also, prepend a newline to traceback output so that the main + error message is on it's own line. ........ ................ + + * /, pbx/pbx_lua.c: Merged revisions 309495 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309495 | mnicholson | 2011-03-04 12:10:23 -0600 + (Fri, 04 Mar 2011) | 9 lines Merged revisions 309494 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, + 04 Mar 2011) | 2 lines remove mysterious lua_pushvalue() that is + never used ........ ................ + + * pbx/pbx_lua.c, configs/extensions.lua.sample: Add support for + defining hints from pbx_lua (closes issue #16024) Reported by: + mnicholson + +2011-03-04 17:40 +0000 [r309491] Russell Bryant + + * channels/chan_nbs.c: Fix a buglet that prevented chan_nbs from + loading (and subsequently stopped Asterisk). In passing, convert + the return codes to be the proper AST_MODULE_LOAD_* constants. + +2011-03-04 16:00 +0000 [r309449] Matthew Nicholson + + * /, pbx/pbx_lua.c: Merged revisions 309448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r309448 | mnicholson | 2011-03-04 09:59:25 -0600 (Fri, 04 Mar + 2011) | 8 lines Export global symbols from pbx_lua to allow + modules to be loaded. Fixes a regression introduced in r278132. + (closes issue #18671) Reported by: Igels Patches: + pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96) + Tested by: Igels ........ + +2011-03-04 15:28 +0000 [r309446] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, + funcs/func_channel.c, channels/sig_pri.c, UPGRADE-1.8.txt: Merged + revisions 309445 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) + | 46 lines Get real channel of a DAHDI call. Starting with + Asterisk v1.8, the DAHDI channel name format was changed for ISDN + calls to: DAHDI/i/[:]- + There were several reasons that the channel name had to change. + 1) Call completion requires a device state for ISDN phones. The + generic device state uses the channel name. 2) Calls do not + necessarily have B channels. Calls placed on hold by an ISDN + phone do not have B channels. 3) The B channel a call initially + requests may not be the B channel the call ultimately uses. + Changes to the internal implementation of the Asterisk master + channel list caused deadlock problems for chan_dahdi if it needed + to change the channel name. Chan_dahdi no longer changes the + channel name. 4) DTMF attended transfers now work with ISDN + phones because the channel name is "dialable" like the chan_sip + channel names. For various reasons, some people need to know + which B channel a DAHDI call is using. * Added + CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and + CHANNEL(dahdi_type) so the dialplan can determine the B channel + currently in use by the channel. Use CHANNEL(no_media_path) to + determine if the channel even has a B channel. * Added AMI event + DAHDIChannel to associate a DAHDI channel with an Asterisk + channel so AMI applications can passively determine the B channel + currently in use. Calls with "no-media" as the DAHDIChannel do + not have an associated B channel. No-media calls are either on + hold or call-waiting. (closes issue #17683) Reported by: mrwho + Tested by: rmudgett (closes issue #18603) Reported by: arjankroon + Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett + (license 664) Tested by: stever28, rmudgett ........ + +2011-03-04 01:52 +0000 [r309404] David Ruggles + + * /, apps/app_externalivr.c: Merged revisions 309403 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309403 | diruggles | 2011-03-03 20:50:44 -0500 + (Thu, 03 Mar 2011) | 23 lines Merged revisions 309356 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500 + (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar + 2011) | 9 lines fix small memory leak fix small memory leak + caused by a string allocation that wasn't freed (closes issue + #18907) Reported by: andy11 Patches: + asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 + (license 1224) ........ ................ ................ + +2011-03-02 21:08 +0000 [r309209-309300] Jason Parker + + * main/channel.c: Add HangupRequest manager event, to specify + when/where a channel gets hung up. (closes issue #18226) Reported + by: clegall_proformatique Patches: + asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall + proformatique (license 1139) + + * /, channels/chan_sip.c: Merged revisions 309256 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309256 | qwell | 2011-03-02 13:54:20 -0600 + (Wed, 02 Mar 2011) | 15 lines Merged revisions 309255 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | + 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in + users.conf for SIP. Since it's a duplicate, nothing is going to + be done, so delme doesn't need to be set at all. Strangely, when + this was added, this was being set to 1 in 1.6, and 0 in trunk. + (issue AST-439) ........ ................ + + * /, main/http.c: Merged revisions 309204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | + 7 lines Fix consistency of CRLFs on HTTP headers that get sent + out. (closes issue #18186) Reported by: nivaldomjunior Patches: + 18186-httpheadernewline.diff uploaded by qwell (license 4) + ........ + +2011-03-01 21:57 +0000 [r309127-309171] Richard Mudgett + + * /, funcs/func_channel.c: Merged revisions 309170 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 + Mar 2011) | 7 lines Document CHANNEL(keypad_digits) and + CHANNEL(no_media_path). * Added XML documentation for + CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML + documentation for CHANNEL(reversecharge). ........ + + * channels/sig_analog.c, /: Merged revisions 309126 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 + Mar 2011) | 16 lines Chan_dahdi does not retain CID when + detecting DTMF CID without polarity reversal. Looks like an + unintended change when sig_analog.c was extracted from + chan_dahdi.c. Removed useless conditional around needed code and + fixed resulting compiler warning. (closes issue #18667) Reported + by: enegaard Patches: issue18667.patch uploaded by enegaard + (license 1197) Tested by: enegaard JIRA SWP-2965 ........ + +2011-03-01 16:22 +0000 [r309090] David Vossel + + * /, channels/chan_sip.c: Merged revisions 309084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309084 | dvossel | 2011-03-01 10:09:11 -0600 + (Tue, 01 Mar 2011) | 15 lines Merged revisions 309083 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) + | 9 lines Fixes thread blocking issue in the sip TCP/TLS + implementation. (closes issue #18497) Reported by: vois Patches: + issues_18497.diff uploaded by dvossel (license 671) Tested by: + vois, rossbeer, kowalma, Freddi_Fonet ........ ................ + +2011-02-28 11:16 +0000 [r308992-309036] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, + main/ast_expr2f.c, configure.ac, main/ast_expr2.fl: Merged + revisions 309035 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309035 | tilghman | 2011-02-28 05:10:28 -0600 + (Mon, 28 Feb 2011) | 15 lines Merged revisions 309033-309034 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) + | 4 lines A later version of flex already includes the fwrite + workaround code, which if used twice causes a compilation error. + Detect whether Flex will compile without the workaround; if so, + suppress our workaround code. ........ r309034 | tilghman | + 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify + meaning, removing double negative (stupid!) ........ + ................ + + * /, funcs/func_odbc.c: Merged revisions 308991 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308991 | tilghman | 2011-02-28 03:33:22 -0600 + (Mon, 28 Feb 2011) | 14 lines Merged revisions 308990 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) + | 7 lines Statements updating zero rows may return SQL_NO_DATA. + This is fine; it's handled. (closes issue #18815) Reported by: + irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot + (license 52) ........ ................ + +2011-02-25 18:58 +0000 [r308946] Alec L Davis + + * /, channels/chan_sip.c: Merged revisions 308945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb + 2011) | 21 lines Fix Deadlock with attended transfer of SIP call + Call path sip_set_rtp_peer (locks chan then pvt) + transmit_reinvite_with_sdp try_suggested_sip_codec + pbx_builtin_getvar_helper (locks p->owner) But by the time + p->owner lock was attempted, seems as though chan and p->owner + were different. So in sip_set_rtp_peer, lock pvt first then lock + p->owner using deadlocking methods. (closes issue #18837) + Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded + by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81, + cmaj Review: [https://reviewboard.asterisk.org/r/1126/] ........ + +2011-02-24 21:43 +0000 [r308904] Richard Mudgett + + * main/channel.c, /: Merged revisions 308903 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) + | 9 lines Invalid read in ast_channel_set_caller_event(). + Valgrind reported that ast_channel_set_caller_event() was reading + data from a freed buffer when using the pre_set structure. + Rearange things to pre-calculate the name and number pointer + before updating the caller party structure to see if the name or + number was changed. ........ + +2011-02-24 17:59 +0000 [r308816] Terry Wilson + + * main/manager.c, /: Merged revisions 308815 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308815 | twilson | 2011-02-24 11:57:18 -0600 + (Thu, 24 Feb 2011) | 26 lines Merged revisions 308814 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r308814 | twilson | 2011-02-24 11:54:49 -0600 + (Thu, 24 Feb 2011) | 19 lines Merged revisions 308813 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) + | 12 lines Don't broadcast FullyBooted to every AMI connection + The FullyBooted event should not be sent to every AMI connection + every time someone connects via AMI. It should only be sent to + the user who just connected. (closes issue #18168) Reported by: + FeyFre Patches: bug0018168.patch uploaded by FeyFre (license + 1142) Tested by: FeyFre, twilson ........ ................ + ................ + +2011-02-24 15:10 +0000 [r308724] Matthew Nicholson + + * main/udptl.c, /: Merged revisions 308723 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308723 | mnicholson | 2011-02-24 09:06:14 -0600 + (Thu, 24 Feb 2011) | 16 lines Merged revisions 308722 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600 + (Thu, 24 Feb 2011) | 9 lines Merged revisions 308721 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, + 24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........ + ................ ................ + +2011-02-24 03:49 +0000 [r308680] Terry Wilson + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 308679 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308679 | twilson | 2011-02-23 21:41:34 -0600 + (Wed, 23 Feb 2011) | 15 lines Merged revisions 308678 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) + | 8 lines Use remotesecret to authenticate with a remote party + The remotesecret option was only being used for outbound + registration and not for placing calls. This patch uses + remotesecret on outbound calls if it is set, otherwise secret is + still used. Review: https://reviewboard.asterisk.org/r/1107/ + ........ ................ + +2011-02-23 23:55 +0000 [r308623-308624] Richard Mudgett + + * main/translate.c: Fix compiler warning. + + * /, channels/sig_pri.c: Merged revisions 308622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) + | 9 lines sig_pri_new_ast_channel() should return NULL when + new_ast_channel() fails. (closes issue #18874) Reported by: cmaj + Patches: + patch-sig_pri-crash-possible-null-channel-pointer.diff.txt + uploaded by cmaj (license 830) JIRA SWP-3172 ........ + +2011-02-22 23:04 +0000 [r308582] David Vossel + + * main/format.c, funcs/func_speex.c, main/frame.c, + main/rtp_engine.c, include/asterisk/silk.h (added), + codecs/speex/fixed_generic.h (added), bridges/bridge_softmix.c, + channels/chan_gtalk.c, bridges/bridge_multiplexed.c, + channels/chan_iax2.c, main/format_pref.c, codecs/speex/resample.c + (added), main/channel.c, funcs/func_pitchshift.c, + include/asterisk/audiohook.h, channels/chan_skinny.c, + main/format_cap.c, funcs/func_volume.c, codecs/speex (added), + codecs/codec_resample.c, include/asterisk/format.h, + codecs/speex/arch.h (added), include/asterisk/frame.h, + include/asterisk/rtp_engine.h, codecs/speex/stack_alloc.h + (added), main/bridging.c, apps/app_jack.c, + configs/codecs.conf.sample, res/res_rtp_asterisk.c, + formats/format_attr_silk.c (added), channels/chan_sip.c, + main/translate.c, main/slinfactory.c, codecs/codec_speex.c, + include/asterisk/_private.h, CHANGES, + codecs/speex/speex_resampler.h (added), res/res_mutestream.c, + include/asterisk/format_cap.h, codecs/Makefile, + channels/chan_jingle.c, main/data.c, channels/iax2.h, + main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c, + main/asterisk.c, include/asterisk/slinfactory.h, + include/asterisk/translate.h, codecs/speex/resample_sse.h + (added), include/asterisk/time.h: Media Project Phase2: SILK + 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio + ConfBridge, and other stuff -Functional changes 1. Dynamic global + format list build by codecs defined in codecs.conf 2. SILK 8khz, + 12khz, 16khz, and 24khz with custom attributes defined in + codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. + SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, + 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using + codec_resample.c 6. Various changes to RTP code required to + properly handle the dynamic format list and formats with + attributes. 7. ConfBridge now dynamically jumps to the best + possible sample rate. This allows for conferences to take + advantage of HD audio (Which sounds awesome) 8. Audiohooks are no + longer limited to 8khz audio, and most effects have been updated + to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. + 9. codec_resample now uses its own code rather than depending on + libresample. -Organizational changes Global format list is moved + from frame.c to format.c Various format specific functions moved + from frame.c to format.c Review: + https://reviewboard.asterisk.org/r/1104/ + +2011-02-22 15:33 +0000 [r308527] Andrew Latham + + * main/http.c: Use ast_debug for console logging Guessed the log + levels based on info that level 3 is the soft roof. Can we create + a page / document to define the levels? + +2011-02-21 15:04 +0000 [r308417] Matthew Nicholson + + * main/udptl.c, /: Merged revisions 308416 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308416 | mnicholson | 2011-02-21 09:02:20 -0600 + (Mon, 21 Feb 2011) | 19 lines Merged revisions 308414 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600 + (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb + 2011) | 5 lines Properly check the bounds of arrays when decoding + UDPTL packets. Also, remove broken support for receiving UDPTL + packets larger than 16k. That shouldn't ever happen anyway. + AST-2011-002 FAX-281 ........ ................ ................ + +2011-02-21 14:14 +0000 [r308372] Andrew Latham + + * main/http.c: Add HTTP URI Debug logging and update notice enable + reporting of the request URI / URL in debugging change funny + debug note to a serious note. + +2011-02-21 13:58 +0000 [r308371] Tzafrir Cohen + + * main/pbx.c: fix a memory leak in device state The callback + handle_statechange (pbx.c) fails to release its data pointer, + leaking memory in the process. Reported by: tzafrir Patches: + 18735_pbx_free_callback.diff uploaded by tzafrir (license 46) + Review: https://reviewboard.asterisk.org/r/1110/ + +2011-02-19 14:07 +0000 [r308331] Andrew Latham + + * main/http.c: Add CSS MIME Type Modern browsers are checking for + the MIME Type of pages and in some cases will not load a file if + the type is wrong. + +2011-02-19 11:03 +0000 [r308289] Tilghman Lesher + + * utils, /: Merged revisions 308288 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308288 | tilghman | 2011-02-19 05:02:49 -0600 (Sat, 19 Feb 2011) + | 2 lines A few more (copies of) files to ignore in this + directory. ........ + +2011-02-18 00:11 +0000 [r308243] Alexandr Anikin + + * addons/chan_ooh323.c, /, addons/ooh323cDriver.c, + addons/ooh323cDriver.h: Merged revisions 308242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 + lines added g729onlyA option for announce only AnnexA g.729 codec + in h.323 capabilities. Option can be global or per user/peer. + ........ + +2011-02-17 20:21 +0000 [r308205] Richard Mudgett + + * channels/chan_dahdi.c: Add more verbage to CLI command 'pri show + channels' usage. + +2011-02-16 22:02 +0000 [r308157] Paul Belanger + + * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308150 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb + 2011) | 2 lines Fix FreeBSD builds. ........ + +2011-02-16 08:06 +0000 [r308099] Alexandr Anikin + + * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308098 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 + lines ifdef __linux__ keepalive variables also ........ + +2011-02-15 23:34 +0000 [r308013] Jason Parker + + * /, apps/app_queue.c: Merged revisions 308010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308010 | qwell | 2011-02-15 17:34:03 -0600 + (Tue, 15 Feb 2011) | 24 lines Merged revisions 308007 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r308007 | qwell | 2011-02-15 17:33:24 -0600 + (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | + 10 lines Fix regression that changed behavior of queues when + ringing a queue member. This reverts r298596, which was to fix a + highly bizarre and contrived issue with a queue member that + called into his own queue being transferred back into his own + queue. I couldn't reproduce that issue in any way. I think one of + the other recent transfer fixes actually fixed this. (closes + issue #18747) Reported by: vrban ........ ................ + ................ + +2011-02-15 23:07 +0000 [r307969] Alexandr Anikin + + * addons/ooh323c/src/ooSocket.c: include tcp keepalive socket calls + only on linux, freebsd and others don't have these options on + sockets. + +2011-02-15 21:42 +0000 [r307963-307964] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Add CLI "pri show channels" command. List the current mapping of + DAHDI B channels to Asterisk channel names and which calls are on + hold or call-waiting. Calls on hold or call-waiting are not + associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 + + * apps/app_dial.c, /: Merged revisions 307962 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) + | 1 line Don't crash when forcing caller id. ........ + +2011-02-15 18:09 +0000 [r307927] David Vossel + + * channels/chan_phone.c: Fixes compile error in chan_phone for big + endian + +2011-02-15 16:18 +0000 [r307883] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, + channels/chan_sip.c, main/ccss.c, channels/sig_pri.c, + include/asterisk/ccss.h: Merged revisions 307879 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 + Feb 2011) | 37 lines No response sent for SIP CC + subscribe/resubscribe request. Asterisk does not send a response + if we try to subscribe for call completion after we have received + a 180 Ringing. You can only subscribe for call completion when + the call has been cleared. When we receive the 180 Ringing, for + this call, its call-completion state is 'CC_AVAILABLE'. If we + then send a subscribe message to Asterisk, it trys to change the + call-completion state to 'CC_CALLER_REQUESTED'. Because this is + an invalid state change, it just ignores the message. The only + state Asterisk will accept our subscribe message is in the + 'CC_CALLER_OFFERED' state. Asterisk will go into the + 'CC_CALLER_OFFERED' when the SIP client clears the call by + sending a CANCEL. Asterisk should always send a response. Even if + its a negative one. The fix is to allow for the CCSS core to + notify a CC agent that a failure has occurred when CC is + requested. The "ack" callback is replaced with a "respond" + callback. The "respond" callback has a parameter indicating + either a successful response or a specific type of failure that + may need to be communicated to the requester. (closes issue + #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, + rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: + GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ + +2011-02-15 07:03 +0000 [r307751-307838] Tilghman Lesher + + * /, funcs/func_odbc.c: Merged revisions 307837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r307837 | tilghman | 2011-02-15 01:02:45 -0600 + (Tue, 15 Feb 2011) | 15 lines Merged revisions 307836 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) + | 8 lines Need to retrieve the rows affected before using the + associated variable. (closes issue #18795) Reported by: irroot + Patches: 20110211__issue18795.diff.txt uploaded by tilghman + (license 14) Tested by: tilghman ........ ................ + + * /, res/res_odbc.c: Merged revisions 307793 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r307793 | tilghman | 2011-02-14 14:16:55 -0600 + (Mon, 14 Feb 2011) | 15 lines Merged revisions 307792 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011) + | 8 lines Increment usage count at first reference, to avoid a + race condition with many threads creating connections all at + once. (issue #18156) Reported by: asgaroth Patches: + 20110214__issue18156.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ ................ + + * addons/chan_ooh323.c, addons/ooh323c/src/ooCmdChannel.c: Making + trunk compile again. + + * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 307750 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) + | 23 lines Calling a gosub routine defined in AEL from Dial/Queue + ceased to work. A bug in AEL did not distinguish between the "s" + extension generated by AEL and an "s" extension that was required + to exist by the chan_dahdi (or another channel) that was not + supplied with a starting extension. Therefore, AEL made incorrect + assumptions about what commands were permissable in the context. + This was fixed by making AEL generate a different extension name. + However, Dial and Queue make additional assumptions about the + name of the default gosub extension. Therefore, they needed to be + brought into line with a "macro" rendered by AEL (as a gosub), + without breaking traditional dialplans written without the aid of + AEL. Related to (issue #18480) Reported by: nivek (closes issue + #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt + uploaded by tilghman (license 14) + 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) + Tested by: kkm ........ + +2011-02-13 10:50 +0000 [r307677-307713] Alexandr Anikin + + * addons/ooh323c/src/ooLogChan.c, + addons/ooh323c/src/ooCmdChannel.c: lc not found - it's warning, + not error, change malloc to ast_calloc again + + * addons/chan_ooh323.c, addons/ooh323cDriver.c: change malloc to + ast_calloc calls to prevent crash of asterisk + +2011-02-10 22:43 +0000 [r307537] Jason Parker + + * contrib/init.d/rc.debian.asterisk, /, main/asterisk.c: Merged + revisions 307536 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r307536 | qwell | 2011-02-10 16:39:30 -0600 + (Thu, 10 Feb 2011) | 22 lines Merged revisions 307535 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r307535 | qwell | 2011-02-10 16:35:49 -0600 + (Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | + 8 lines Remove color when executing commands via a remote + console. Essentially this makes '-x' imply '-n' on rasterisk. + This was done in a different and incomplete way previously, which + I'm reverting here. (issue #18776) Reported by: alecdavis + ........ ................ ................ + +2011-02-10 17:45 +0000 [r307468] Mark Michelson + + * /, configs/ccss.conf.sample: Merged revisions 307467 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu, + 10 Feb 2011) | 5 lines Fix a gaffe in the CCSS sample + configuration. Discovered by Philippe Lindheimer and pointed out + on #asterisk-dev ........ + +2011-02-10 17:12 +0000 [r307433] David Vossel + + * channels/chan_sip.c, main/format_cap.c, + include/asterisk/format_cap.h: Fixes bug in chan_sip where + nativeformats are not set correctly. The nativeformats field was + being overwritten when it should have been appended too. This + caused some format capabilities to be lost briefly and some log + warnings to be output. + +2011-02-10 13:29 +0000 [r307396] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, + addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h, + addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c: + Corrections for properly work with H.323v2 (older) endpoints and + other small fixes. Interpret remote side H.225 version. + Corrections for H.323v2 endpoints: don't start TCS and MSD before + connect, don't start TCS and MSD by accepting H.245 connection, + start TCS and MSD by StartH245 facility message. Other fixes: fix + non zeroended remoteDisplayName issue, small fixes in call + clearing by closing H.245 connection, tcp keepalive introduced on + TCP connections (now is hardcoded, will be configurable in the + future), don't force H.245tunneling if FastStart is active, don't + send Alerting singal more than once per call. (closes issue + #18542) Reported by: vmikhelson Patches: issue18542-final-3.patch + uploaded by may213 (license 454) Tested by: vmikhelson + +2011-02-09 22:48 +0000 [r307359] Jeff Peeler + + * apps/app_meetme.c, CHANGES: Add new manager action + MeetmeListRooms. From the submitter: I've added a new manager + action to list only the active conferences on an Asterisk system. + It shows the same data displayed when you run a 'meetme list' on + the Asterisk CLI. (closes issue #17905) Reported by: rcasas + Patches: app_meetme.c.patch uploaded by rcasas (license 641) + Review: https://reviewboard.asterisk.org/r/874/ + +2011-02-09 21:46 +0000 [r307315] Andrew Latham + + * contrib/init.d/rc.debian.asterisk: Disable color during running + test (closes issue #18776) Reported by: alecdavis Patches: + ast_deb_init.diff uploaded by lathama (license 1028) Tested by: + andrel, lathama + +2011-02-09 21:08 +0000 [r307229-307274] Jeff Peeler + + * /, main/astobj2.c: Merged revisions 307273 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307273 | jpeeler | 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011) + | 8 lines Add missing debug info for ao2_link for use with + REF_DEBUG in ao2 callback. (closes issue #18758) Reported by: + rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by + rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by + rgagnon (license 1202) ........ + + * main/features.c, CHANGES: Allow parkedmusicclass to be settable + for non-default parking lots. (closes issue #17946) Reported by: + bluecrow76 Patches: + asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff + + * /, main/features.c: Merged revisions 307228 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r307228 | jpeeler | 2011-02-09 13:52:51 -0600 + (Wed, 09 Feb 2011) | 17 lines Merged revisions 307227 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) + | 11 lines Make sure to set parking dial context for non-default + parking lots. Since parking_con_dial isn't settable, set all + parking lots to "park-dial". (closes issue #17946) Reported by: + bluecrow76 Patches: + asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by + bluecrow76 (license 270) modified by me ........ ................ + +2011-02-09 19:17 +0000 [r307192] Tzafrir Cohen + + * main/loader.c: clarify warning when no loadable module support + Clarify warning message when LOADABLE_MODULES is disabled but we + still try to load a module. + +2011-02-09 05:53 +0000 [r307143] Tilghman Lesher + + * main/lock.c, /: Merged revisions 307142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307142 | tilghman | 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011) + | 3 lines Initialize tracking variable in structure properly. + Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by + me.) ........ + +2011-02-08 21:24 +0000 [r307097] Jason Parker + + * /, main/logger.c: Merged revisions 307092 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | + 9 lines Fix issue with verbose messages not showing on remote + console. This code was reworked recently, and since the + logchannel list hadn't been created yet at this point, and it was + a verbose message, it was being dropped on the floor. Now it'll + continue on to where it should be handled. (closes issue #18580) + Reported by: pabelanger ........ + +2011-02-08 21:18 +0000 [r307071] Mark Michelson + + * /, main/ccss.c: Merged revisions 307065 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb + 2011) | 6 lines Add a couple of useful channel variables for the + CC recall macro. CC_EXTEN and CC_CONTEXT will allow you to + determine the channel and context that will be called when the + recall occurs. ........ + +2011-02-08 20:42 +0000 [r307061] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 306979 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306979 | twilson | 2011-02-08 12:18:08 -0800 + (Tue, 08 Feb 2011) | 16 lines Merged revisions 306973 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306973 | twilson | 2011-02-08 12:14:09 -0800 + (Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 + Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with + pedantic=yes ........ ................ ................ + +2011-02-08 20:31 +0000 [r307041] Andrew Latham + + * /, doc/asterisk.8, configs/asterisk.conf.sample, + configs/voicemail.conf.sample, doc/asterisk.sgml: Documentation + Updates Note default polling setting in voicemail.conf Add + missing config to asterisk.conf Update manpage (issue #16505) + Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff + uploaded by tzafrir (license 46) Tested by: lathama, tzafrir + +2011-02-08 19:42 +0000 [r306867-306968] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 306967 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306967 | jpeeler | 2011-02-08 13:41:42 -0600 + (Tue, 08 Feb 2011) | 16 lines Merged revisions 306966 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600 + (Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 + Feb 2011) | 1 line fix this line again ........ ................ + ................ + + * /, apps/app_voicemail.c: Merged revisions 306962 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306962 | jpeeler | 2011-02-08 13:25:38 -0600 + (Tue, 08 Feb 2011) | 22 lines Merged revisions 306961 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600 + (Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) + | 9 lines Backup file storing message duration is not used with + IMAP_STORAGE, remove code. The message duration is stored in the + body of the email when using IMAP_STORAGE, so nothing needs to + happen with the backup file. (closes issue #18718) Reported by: + kerframil ........ ................ ................ + + * /, apps/app_voicemail.c: Merged revisions 306866 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306866 | jpeeler | 2011-02-08 10:21:45 -0600 + (Tue, 08 Feb 2011) | 16 lines Merged revisions 306865 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600 + (Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 + Feb 2011) | 1 line make this safer and fully correct, pointed out + by Steve Davis ........ ................ ................ + +2011-02-08 02:05 +0000 [r306827] Andrew Latham + + * doc/asterisk.sgml: Documentation Updates. Start updates to the + man pages. (issue #16505) Reported by: tzafrir Tested by: lathama + +2011-02-08 00:43 +0000 [r306755-306793] Richard Mudgett + + * configs/chan_dahdi.conf.sample: Define the MCID acronym in + chan_dahdi.conf.sample. + + * channels/sig_pri.h: Use correct conditional for MCID send. + + * channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, funcs/func_frame_trace.c, + main/features.c, CHANGES, channels/sig_pri.c, + include/asterisk/frame.h: Pass a MCID request to the bridged + channel. Pass a MCID request to the bridged channel so the + bridged channel can send it to the network. The ability to send + the MCID request on an ISDN span is enabled with the new + chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 + +2011-02-07 22:46 +0000 [r306670-306675] Terry Wilson + + * /, main/features.c: Merged revisions 306674 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306674 | twilson | 2011-02-07 14:43:22 -0800 + (Mon, 07 Feb 2011) | 24 lines Merged revisions 306673 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306673 | twilson | 2011-02-07 14:40:20 -0800 + (Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) + | 10 lines Don't try to pickup a call in the middle of a + masquerade If A calls B which doesn't answer and C & D both try + to do a call pickup, it is possible for ast_pickup_call to answer + the call, then fail to masquerade one of the calls because the + other one is already in the process of masquerading. This patch + checks to see if the channel is in the process of masquerading + before call before selecting it for a pickup. Review: + https://reviewboard.asterisk.org/r/1094/ ........ + ................ ................ + + * /, channels/chan_sip.c: Merged revisions 306619 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306619 | twilson | 2011-02-07 14:15:27 -0800 + (Mon, 07 Feb 2011) | 24 lines Merged revisions 306618 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306618 | twilson | 2011-02-07 13:59:54 -0800 + (Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) + | 10 lines Don't allow a REFER w/replaces to replace its own + dialog Asterisk currently accepts a REFER with a Refer-To with an + embedded Replaces header that matches the dialog of the REFER. + This would be a situation like A calls B, A calls C, A transfers + B to A, which is just silly. This patch makes the transfer fail + instead of making Asterisk freak out and forget to hang other + channels up. Review: https://reviewboard.asterisk.org/r/1093/ + ........ ................ ................ + +2011-02-07 17:55 +0000 [r306576] Mark Michelson + + * /, main/ccss.c: Merged revisions 306575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb + 2011) | 9 lines Rearrange a bit of code in the generic CC recall + operation. By waiting to call the callback macro after the + CC_INTERFACES, extension, priority, and context have been set, + this information can be accessed more easily within the callback + macro. Reported by Philippe Lindheimer. ........ + +2011-02-07 16:33 +0000 [r306541] David Vossel + + * channels/chan_sip.c: Fixes use of ast_format_cap_append where + ast_format_cap_copy is necessary. + +2011-02-05 22:16 +0000 [r306499] Alexandr Anikin + + * addons/chan_ooh323.c: fix trivial issue after dvossel patch, + initial zero fill user and peer structure before cap structure + allocated. + +2011-02-05 02:55 +0000 [r306464] Richard Mudgett + + * channels/chan_dahdi.c: Ignore voice frames in chan_dahdi native + bridging. Hardware is handling them. + +2011-02-04 22:37 +0000 [r306432] Jeff Peeler + + * main/manager.c: Send manager event for blackfilter only if it + DOES NOT match. The logic got reversed, oops. Works properly now + when multiple blackfilters are present. (closes issue #18283) + Reported by: telecos82 Patches: ast_managereventfilter.patch + uploaded by telecos82 (license 687) + +2011-02-04 20:30 +0000 [r306396] Richard Mudgett + + * apps/app_dial.c, channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac, CHANGES, + channels/sig_pri.c: Add ISDN display ie text handling options to + chan_dahdi.conf. The display ie handling can be controlled + independently in the send and receive directions with the + following options: * Block display text data. * Use display text + in SETUP/CONNECT messages for name. * Use display text for COLP + name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary + display text during a call. Sent in INFORMATION messages. + Received from any message that the display text was not used as a + name. If the display options are not set then the options default + to legacy behavior. The arbitrary display text is exchanged + between bridged channels using the AST_FRAME_TEXT frame type. To + send display text from the dialplan use the SendText() + application when the arbitrary display text option is enabled. + JIRA SWP-2688 JIRA ABE-2693 + +2011-02-04 19:24 +0000 [r306359] Jason Parker + + * /, apps/app_queue.c: Merged revisions 306356 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306356 | qwell | 2011-02-04 13:24:29 -0600 + (Fri, 04 Feb 2011) | 16 lines Merged revisions 306346 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | + 9 lines Don't fallthrough to 'unknown' in the 'ringing' case. + This could cause improper exits from the queue. (closes issue + #18499) Reported by: zaltar Patches: app_queue.patch uploaded by + zaltar (license 1148) ........ ................ + +2011-02-04 19:09 +0000 [r306325-306326] Richard Mudgett + + * tests/test_format_api.c: Fix compiler warning. + + * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 306324 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) + | 9 lines Don't send redirecting updates to the caller if the + dialplan forked the call. Each fork in the dial could be + redirected and confuse the caller. For ISDN the DivLeg1 and + DivLeg3 messages would get confused because ISDN redirects calls + in sequence not in parallel. * Also fixed a formatting + inconsistency in app_dial.c and make a warning message more + useful about what frame type could not be written. ........ + +2011-02-04 18:16 +0000 [r306258-306292] Paul Belanger + + * utils/extconf.c: Revert changes to extconf.c It seems extconf.c + already defines some local ast_debug() functions. Theses should + be removed and replaced with logger.h. A patch will be added to + reviewboard shortly. + + * cel/cel_radius.c, addons/chan_ooh323.c, apps/app_meetme.c, + main/say.c, channels/chan_gtalk.c, main/taskprocessor.c, + res/res_http_post.c, res/res_musiconhold.c, channels/chan_iax2.c, + res/res_jabber.c, pbx/pbx_loopback.c, main/channel.c, + channels/chan_dahdi.c, pbx/pbx_spool.c, main/manager.c, + res/res_smdi.c, channels/chan_skinny.c, main/features.c, + res/res_agi.c, main/http.c, main/logger.c, res/ais/evt.c, + main/app.c, res/res_config_ldap.c, apps/app_rpt.c, + res/res_rtp_asterisk.c, main/pbx.c, channels/chan_sip.c, + apps/app_fax.c, include/asterisk/channel.h, channels/sig_pri.c, + channels/chan_misdn.c, include/asterisk/sched.h, utils/extconf.c, + codecs/codec_ilbc.c, main/audiohook.c, res/res_odbc.c, + main/xmldoc.c, apps/app_voicemail.c: Replace ast_log(LOG_DEBUG, + ...) with ast_debug() (closes issue #18556) Reported by: kkm + Review: https://reviewboard.asterisk.org/r/1071/ + +2011-02-04 16:42 +0000 [r306257] David Vossel + + * codecs/codec_ilbc.c, codecs/ex_ilbc.h: Fix compile error in codec + ilbc translator. + +2011-02-03 23:50 +0000 [r306216] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 306215 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) + | 20 lines Fix SIP deadlock involving state changes. Once again a + call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper) + has caused locking problems. Both of these functions lock the + channel when the channel argument is passed in! In this case, the + suspected problem (the backtrace makes it impossible to tell) was + the private being locked in sip_set_rtp_peer and then: + transmit_reinvite_with_sdp try_suggested_sip_codec + pbx_builtin_getvar_helper (Traced to verify that the fix was only + required in 1.8 and later.) (closes issue #18491) Reported by: + cmaj Patches: chan_sip_fix_deadlocks_bug_18491.txt uploaded by + cmaj (license 830) Tested by: cmaj ........ + +2011-02-03 21:13 +0000 [r306128] Terry Wilson + + * channels/chan_local.c, /: Merged revisions 306127 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306127 | twilson | 2011-02-03 13:03:26 -0800 + (Thu, 03 Feb 2011) | 23 lines Merged revisions 306126 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306126 | twilson | 2011-02-03 12:56:00 -0800 + (Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) + | 9 lines Set hangup cause in local_hangup When a call involves a + local channel (like SIP -> Local -> SIP), the hangup cause was + not being set. This resulted in SIP channels sometimes getting a + 503 error instead of a 486 when the far side sent a busy. In + Asterisk 1.8+ this also can cause issues with CCSS that involve a + local channel. This patch sets the hangupcause for one side of + the local channel to the other in local_hangup for outbound + calls. ........ ................ ................ + +2011-02-03 20:51 +0000 [r306125] Jeff Peeler + + * /, main/features.c: Merged revisions 306124 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306124 | jpeeler | 2011-02-03 14:50:48 -0600 + (Thu, 03 Feb 2011) | 17 lines Merged revisions 306123 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) + | 10 lines Set exception on channel in parking thread when + POLLPRI event detected. This is done just to make the code be + equivalent to the old select code. As noted in 303106 the same + issue was already fixed in this branch, but the exception was not + set on the channel in the case of POLLPRI. The reason that this + did not cause a problem here is because in 122923 the check in + __ast_read to check the exception flag was removed. (related to + #18637) ........ ................ + +2011-02-03 18:37 +0000 [r306086] Jason Parker + + * main/frame.c: Modify alignment of 'core show codecs', since the + ID is no longer a huge int. + +2011-02-03 18:12 +0000 [r306010-306053] David Vossel + + * main/frame.c: Fixes output of "core show codecs" to display image + types correctly. + + * apps/app_dahdibarge.c, channels/chan_local.c, main/frame.c, + apps/app_record.c, apps/app_alarmreceiver.c, + bridges/bridge_softmix.c, formats/format_sln16.c, + apps/app_ices.c, bridges/bridge_multiplexed.c, + channels/chan_iax2.c, main/astobj2.c, res/res_rtp_multicast.c, + channels/chan_dahdi.c, include/asterisk/bridging_technology.h, + funcs/func_pitchshift.c, pbx/pbx_spool.c, + include/asterisk/audiohook.h, channels/chan_skinny.c, + channels/sip/include/globals.h, apps/app_dumpchan.c, + formats/format_pcm.c, formats/format_h263.c, main/bridging.c, + codecs/ex_ulaw.h, channels/sip/include/sip.h, main/pbx.c, + codecs/codec_g722.c, formats/format_wav.c, codecs/codec_g726.c, + formats/format_ogg_vorbis.c, bridges/bridge_simple.c, + include/asterisk/channel.h, apps/app_talkdetect.c, + channels/iax2-parser.c, include/asterisk/format_cap.h (added), + apps/app_speech_utils.c, channels/iax2-parser.h, main/data.c, + funcs/func_channel.c, main/audiohook.c, codecs/codec_dahdi.c, + include/asterisk/frame_defs.h, formats/format_g726.c, + apps/app_mixmonitor.c, main/asterisk.c, res/res_calendar.c, + apps/app_voicemail.c, channels/chan_vpb.cc, addons/format_mp3.c, + formats/format_sln.c, apps/app_dictate.c, codecs/ex_g722.h, + codecs/codec_gsm.c, codecs/ex_g726.h, channels/chan_gtalk.c, + include/asterisk/abstract_jb.h, main/channel.c, apps/app_mp3.c, + codecs/codec_resample.c, formats/format_h264.c, + formats/format_siren14.c, apps/app_rpt.c, channels/chan_mgcp.c, + codecs/codec_lpc10.c, channels/chan_sip.c, codecs/ex_lpc10.h, + include/asterisk/format_pref.h (added), codecs/codec_alaw.c, + res/res_adsi.c, tests/test_format_api.c (added), + apps/app_originate.c, channels/chan_jingle.c, + formats/format_vox.c, main/abstract_jb.c, + include/asterisk/bridging.h, main/callerid.c, main/file.c, + apps/app_sms.c, formats/format_g723.c, main/dsp.c, main/format.c + (added), main/udptl.c, main/rtp_engine.c, addons/chan_ooh323.c, + codecs/codec_adpcm.c, apps/app_test.c, addons/chan_ooh323.h, + include/asterisk/speech.h, codecs/ex_adpcm.h, codecs/ex_alaw.h, + formats/format_wav_gsm.c, include/asterisk/data.h, + codecs/ex_gsm.h, main/indications.c, main/format_pref.c (added), + main/cli.c, main/features.c, include/asterisk/mod_format.h, + apps/app_amd.c, addons/ooh323cDriver.c, channels/chan_alsa.c, + formats/format_jpeg.c, addons/ooh323cDriver.h, + formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c, + formats/format_g719.c, channels/h323/ast_h323.cxx, + channels/chan_bridge.c, apps/app_echo.c, apps/app_fax.c, + codecs/codec_speex.c, include/asterisk/slin.h, + channels/chan_agent.c, channels/iax2-provision.c, + codecs/ex_speex.h, channels/chan_misdn.c, + include/asterisk/image.h, channels/iax2.h, codecs/codec_ilbc.c, + apps/app_chanspy.c, res/res_fax_spandsp.c, + include/asterisk/slinfactory.h, include/asterisk/translate.h, + channels/chan_unistim.c, channels/chan_multicast_rtp.c, + main/ccss.c, apps/app_meetme.c, res/res_musiconhold.c, + apps/app_followme.c, formats/format_siren7.c, + formats/format_ilbc.c, include/asterisk/file.h, + include/asterisk/callerid.h, channels/chan_phone.c, main/dial.c, + main/manager.c, main/format_cap.c (added), + funcs/func_frame_trace.c, res/res_agi.c, main/app.c, + apps/app_confbridge.c, include/asterisk/format.h (added), + main/image.c, include/asterisk/rtp_engine.h, + include/asterisk/frame.h, addons/chan_mobile.c, + apps/app_parkandannounce.c, apps/app_jack.c, + res/res_clioriginate.c, res/res_rtp_asterisk.c, + apps/app_nbscat.c, codecs/codec_a_mu.c, res/res_fax.c, + apps/app_festival.c, apps/app_waitforsilence.c, + include/asterisk/astobj2.h, main/slinfactory.c, main/translate.c, + channels/chan_console.c, channels/h323/chan_h323.h, + channels/chan_oss.c, channels/chan_usbradio.c, + channels/chan_h323.c, codecs/codec_ulaw.c, + include/asterisk/pbx.h, channels/chan_nbs.c, + formats/format_g729.c: Asterisk media architecture conversion - + no more format bitfields This patch is the foundation of an + entire new way of looking at media in Asterisk. The code present + in this patch is everything required to complete phase1 of my + Media Architecture proposal. For more information about this + project visit the link below. + https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal + The primary function of this patch is to convert all the usages + of format bitfields in Asterisk to use the new format and + format_cap APIs. Functionally no change in behavior should be + present in this patch. Thanks to twilson and russell for all the + time they spent reviewing these changes. Review: + https://reviewboard.asterisk.org/r/1083/ + +2011-02-03 16:13 +0000 [r305988] Andrew Latham + + * phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample: + res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support + (issue #18713) Reported by: lathama Patches: snom_dir.diff + uploaded by lathama (license 1028) Tested by: lathama + +2011-02-03 00:29 +0000 [r305939] Richard Mudgett + + * main/channel.c, main/manager.c, /, channels/chan_sip.c, + apps/app_sendtext.c: Merged revisions 305923 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600 + (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 + (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) + | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null + terminator in the buffer length. When the frame is queued it is + copied. If the null terminator is not part of the frame buffer + length, the receiver could see garbage appended onto it. * Add + channel lock protection with ast_sendtext(). * Fixed AMI SendText + action ast_sendtext() return value check. ........ + ................ ................ + +2011-02-02 20:06 +0000 [r305845] Tilghman Lesher + + * /, funcs/func_env.c: Merged revisions 305844 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r305844 | tilghman | 2011-02-02 14:05:43 -0600 (Wed, 02 Feb 2011) + | 5 lines Eliminate a file descriptor leak when using the FILE() + dialplan function. (closes issue #18731) Reported by: marioabajo + ........ + +2011-02-02 19:30 +0000 [r305759-305843] Andrew Latham + + * configs/iax.conf.sample, funcs/func_enum.c, + configs/dundi.conf.sample, funcs/func_callcompletion.c, /, + configs/mgcp.conf.sample, configs/iaxprov.conf.sample, + configs/unistim.conf.sample, apps/app_externalivr.c, + configs/sip.conf.sample, configs/skinny.conf.sample, + configs/h323.conf.sample, configs/sla.conf.sample, + apps/app_voicemail.c: Replacing doc/* and asterisk.pdf with wiki + links Adding links to http(s)://wiki.asterisk.org + + * configs/chan_dahdi.conf.sample, /, configs/extconfig.conf.sample, + configs/res_snmp.conf.sample, main/ast_expr2f.c, + res/res_timing_dahdi.c, configs/ccss.conf.sample, + configs/sip.conf.sample, configs/skinny.conf.sample, + main/config.c, configs/h323.conf.sample, configs/sla.conf.sample, + main/ast_expr2.fl, res/res_srtp.c: Replacing doc/* with wiki + links Adding links to http(s)://wiki.asterisk.org + + * /, channels/chan_sip.c: Replace link to old doc with new wiki + page. Link to + https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions + +2011-02-01 22:48 +0000 [r305693] Jason Parker + + * /, channels/chan_iax2.c: Merged revisions 305692 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r305692 | qwell | 2011-02-01 16:48:16 -0600 (Tue, 01 Feb + 2011) | 7 lines Reverse sense of an error test when reading from + astdb. (closes issue #18545) Reported by: jcovert Patches: + chan_iax2.c.patch uploaded by jcovert (license 551) ........ + +2011-02-01 21:16 +0000 [r305650] Andrew Latham + + * configs/sip.conf.sample: SIP Configuration Documentation sip show + settings reports qualifyfreq in milliseconds. sip.conf configures + qualifyfreg in seconds. + +2011-02-01 19:27 +0000 [r305604] Brett Bryant + + * cel/cel_pgsql.c, /: Merged revisions 305603 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r305603 | bbryant | 2011-02-01 14:23:20 -0500 (Tue, 01 Feb 2011) + | 4 lines Add a possible solution to a customer problem with + reloading cel_pgsql.so quickly. ........ + +2011-02-01 18:03 +0000 [r305561] Andrew Latham + + * /: doc/tex dir removed, but corresponding entries still exists + Update README, CHANGES, and Makefile. Direct users to + http://wiki.asterisk.org for documentation or to the AST.txt and + AST.pdf included in the tarball. (closes issue #18443) Reported + by: bas Patches: changes.diff uploaded by lathama (license 1028) + readme.diff uploaded by lathama (license 1028) Tested by: lathama + bas + +2011-02-01 17:05 +0000 [r305474] Jason Parker + + * /, res/res_musiconhold.c: Merged revisions 305473 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305473 | qwell | 2011-02-01 11:04:23 -0600 + (Tue, 01 Feb 2011) | 23 lines Merged revisions 305472 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305472 | qwell | 2011-02-01 11:02:09 -0600 + (Tue, 01 Feb 2011) | 16 lines Merged revisions 305471 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) | + 9 lines Close file descriptor for timing source when a MOH class + gets destroyed. (closes issue #18457) Reported by: mcallist + Patches: 18457-closetimer.diff uploaded by qwell (license 4) + 18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested + by: qwell, loloski ........ ................ ................ + +2011-02-01 16:05 +0000 [r305433] Brett Bryant + + * apps/app_confbridge.c: Add's two features to confbridge: + confbridge kick, and confbridge list. (closes issue #14389) + (closes issue #18007) Reported by: jcollie Patches: + 0001-Fix-up-bridging-module-so-that-menuselect-works.patch + uploaded by jcollie (license 412) + 0002-Add-confbridge-list-and-confbridge-kick-CLI-comm.patch + uploaded by jcollie (license 412) Tested by: file Review: + https://reviewboard.asterisk.org/r/1084/ + +2011-02-01 00:07 +0000 [r305344] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 305343 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305343 | rmudgett | 2011-01-31 18:01:09 -0600 + (Mon, 31 Jan 2011) | 21 lines Merged revisions 305342 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600 + (Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) + | 7 lines Obtain the pri lock for PRI queue counters. Need to + obtain the pri lock when calling pri_dump_info_str() to avoid a + reentrancy problem when calculating the Q.921 Q count statistic. + JIRA AST-484 ........ ................ ................ + +2011-01-31 23:08 +0000 [r305132-305255] Jason Parker + + * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305254 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305254 | qwell | 2011-01-31 17:07:00 -0600 + (Mon, 31 Jan 2011) | 24 lines Merged revisions 305253 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305253 | qwell | 2011-01-31 16:59:34 -0600 + (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | + 10 lines Prevent a crash when dialing a technology with no + destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers + already had code to prevent this. The attempt that app_dial was + making to prevent it was not correct, so I fixed that. (closes + issue #18371) Reported by: gbour Patches: 18371.patch uploaded by + gbour (license 1162) ........ ................ ................ + + * main/tcptls.c, /, configs/sip.conf.sample: Merged revisions + 305247 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | + 7 lines Add alternative name for config option. The SIP sample + configuration had "tlscadir" as the option name, but chan_sip + used the more correct "tlscapath". Now both are accepted. + Discovered (sort of) by a user on IRC in #asterisk ........ + + * /, res/res_musiconhold.c: Merged revisions 305198 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r305198 | qwell | 2011-01-31 15:30:44 -0600 (Mon, 31 Jan + 2011) | 2 lines Fix compile error. pseudofd no longer exists. + ........ + + * /, res/res_musiconhold.c: Merged revisions 305131 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305131 | qwell | 2011-01-31 15:00:25 -0600 + (Mon, 31 Jan 2011) | 16 lines Merged revisions 305130 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305130 | qwell | 2011-01-31 14:59:37 -0600 + (Mon, 31 Jan 2011) | 9 lines Merged revisions 305129 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan + 2011) | 2 lines Set file descriptors to -1 on creation, so that + we don't see weirdness later. ........ ................ + ................ + +2011-01-31 13:57 +0000 [r305084] Andrew Latham + + * main/http.c: Asterisk HTTP response Content-type Address content + type for BSD and other platforms (closes issue #18456) Reported + by: alexo Patches: asterisk18_http.patch uploaded by alexo + (license 1175) Tested by: alexo + +2011-01-31 07:52 +0000 [r304951-305041] Tilghman Lesher + + * /, include/asterisk/lock.h: Merged revisions 305040 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r305040 | tilghman | 2011-01-31 01:51:40 -0600 (Mon, 31 + Jan 2011) | 2 lines Use the non-specific API aliases, to avoid a + problem with building the utils directory. ........ + + * /, apps/app_voicemail.c: Merged revisions 304985 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304985 | tilghman | 2011-01-31 01:27:13 -0600 + (Mon, 31 Jan 2011) | 16 lines Merged revisions 304978 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600 + (Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31 + Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined. + ........ ................ ................ + + * main/lock.c, /, main/heap.c, main/utils.c, + include/asterisk/lock.h, .cleancount: Merged revisions 304950 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r304950 | tilghman | 2011-01-31 00:41:36 -0600 (Mon, 31 Jan 2011) + | 18 lines Change mutex tracking so that it only consumes memory + in the core mutex object when it's actually being used. This + reduces the overall size of a mutex which was 3016 bytes before + this back down to 216 bytes (this is on 64-bit Linux with a + glibc-implemented mutex). The exactness of the numbers here may + vary slightly based upon how mutexes are implemented on a + platform, but the long and short of it is that prior to this + commit, chan_iax2 held down 98MB of memory on a 64-bit system for + nothing more than a table of 32767 locks. After this commit, the + same table occupies a mere 7MB of memory. (closes issue #18194) + Reported by: job Patches: 20110124__issue18194.diff.txt uploaded + by tilghman (license 14) Tested by: tilghman Review: + https://reviewboard.asterisk.org/r/1066 ........ + +2011-01-30 00:22 +0000 [r304913] Andrew Latham + + * funcs/func_callcompletion.c, /, apps/app_externalivr.c, + apps/app_queue.c, apps/app_voicemail.c, funcs/func_realtime.c, + res/res_calendar.c: Add Function and Application Relationships to + documentation Add and extend the see-also sections to the + documentation for applications and functions in an effort to + expand the online documentation of the wiki. Also check for and + update any links to moved documentation in the doc folder. + +2011-01-29 23:10 +0000 [r304639-304867] Sean Bright + + * /, res/res_config_ldap.c: Merged revisions 304866 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304866 | seanbright | 2011-01-29 18:07:18 -0500 + (Sat, 29 Jan 2011) | 14 lines Merged revisions 304865 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304865 | seanbright | 2011-01-29 18:05:25 -0500 (Sat, 29 Jan + 2011) | 7 lines Plug some memory leaks in the LDAP realtime + driver. (closes issue #18435) Reported by: zaltar Patches: + res_config_ldap.patch uploaded by zaltar (license 1148) ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 304777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304777 | seanbright | 2011-01-29 13:09:37 -0500 + (Sat, 29 Jan 2011) | 22 lines Merged revisions 304776 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan + 2011) | 15 lines If we fail to allocate our announcement objects, + make sure we don't leak objects. The majority of this patch was + committed already in r304726 and r304729. (issue #18225) Reported + by: kenji (issue #18444) Reported by: junky (closes issue #18343) + Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz + (license 834) ........ ................ + + * /, apps/app_meetme.c: Merged revisions 304774 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304774 | seanbright | 2011-01-29 12:54:43 -0500 + (Sat, 29 Jan 2011) | 16 lines Merged revisions 304773 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan + 2011) | 9 lines When we pass the S() or L() options to MeetMe, + make sure that we honor C as well. Without this patch, if the + user was kicked from the conference via the S() or L() mechanism, + we would just hang up on them even if we also passed C (continue + in dialplan when kicked). With this patch we honor the C flag in + those cases. (closes issue #17317) Reported by: var ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 304730 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304730 | seanbright | 2011-01-29 12:15:27 -0500 + (Sat, 29 Jan 2011) | 22 lines Merged revisions 304729 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan + 2011) | 15 lines Make sure that we unref the correct object when + ejecting the most recent caller. Currently, when we kick the last + user to enter, we decrement our own reference count which results + in a crash when we kick another user or when we exit the + conference ourselves. This will fix #18225 in 1.8 and trunk, but + that particular bug does not exist in 1.6.2. (closes issue + #18225) Reported by: kenji Patches: issue18225.patch uploaded by + seanbright (license 71) Tested by: seanbright ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 304727 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304727 | seanbright | 2011-01-29 11:28:27 -0500 + (Sat, 29 Jan 2011) | 16 lines Merged revisions 304726 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan + 2011) | 9 lines Fix user reference leak in MeetMe. We were + unlinking the user from the conferences user container, but not + decrementing the reference count of the user as well, resulting + in a leak. (closes issue #18444) Reported by: junky Tested by: + seanbright ........ ................ + + * /, apps/app_meetme.c: Merged revisions 304683 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304683 | seanbright | 2011-01-28 17:54:23 -0500 + (Fri, 28 Jan 2011) | 16 lines Merged revisions 304659,304682 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan + 2011) | 5 lines Don't leak references if we can't create a pseudo + channel for mixing in MeetMe. If there was a problem allocating a + pseudo channel when building our meetme, we weren't destroying + our user container or destroying the mutexes that we created. + ........ r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri, + 28 Jan 2011) | 2 lines Revert part of the previous commit that + snuck in. ........ ................ + + * /, main/acl.c: Merged revisions 304638 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r304638 | seanbright | 2011-01-28 15:19:08 -0500 (Fri, 28 Jan + 2011) | 11 lines Restore some conditionals that we lost in + r277814. There are some cases where ast_append_ha() is called + with a NULL instead of a valid int pointer. So if we get a NULL, + don't try to dereference it. (closes issue #18162) Reported by: + imcdona Patches: issue0018162.patch uploaded by pabelanger + (license 224) Tested by: enegaard ........ + +2011-01-27 20:09 +0000 [r304600] Brett Bryant + + * res/res_config_pgsql.c: Patch that fixes the "realtime show pgsql + cache" command crash when giving a table name, because of the use + of an uninitialized variable. Fixes an error introduced in + r300882. (closes issue #18605) Reported by: romain_proformatique + Patches: res_config_pgsql_fix.patch uploaded by romain + proformatique (license 975) Tested by: romain_proformatique + +2011-01-27 20:07 +0000 [r304599] Kevin P. Fleming + + * res/res_fax.c: Fix bug with 'F' option for ReceiveFAX and + SendFAX. Skipping the call to set_t38_fax_caps() caused the FAX + session details to not be marked as supporting audio FAX + either... the function's name is a bit misleading. This patch + restores the single bit of non-T.38 behavior from that function + when audio mode is forced. + +2011-01-27 19:12 +0000 [r304555] Richard Mudgett + + * /, main/ccss.c: Merged revisions 304554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r304554 | rmudgett | 2011-01-27 13:08:14 -0600 (Thu, 27 Jan 2011) + | 4 lines Warning message if CALLCOMPLETION(cc_callback_macro or + cc_agent_dialstring) are empty. Test if the value pointer is not + NULL instead of not ast_strlen_zero(). ........ + +2011-01-27 17:03 +0000 [r304463-304467] Jason Parker + + * /, configure, configure.ac: Merged revisions 304466 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304466 | qwell | 2011-01-27 11:03:01 -0600 + (Thu, 27 Jan 2011) | 23 lines Merged revisions 304465 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304465 | qwell | 2011-01-27 11:01:24 -0600 + (Thu, 27 Jan 2011) | 16 lines Merged revisions 304464 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) | + 9 lines Fix default prefix=/usr regression on non-Linux systems. + This partially reverts a change made in branches/1.4/ r267759, + which will cause issue #17013 to be reopened. This issue was + pointed out by a user on #asterisk, who helpfully discovered that + paths were being set incorrectly. To truly understand what was + wrong, one should run: svn diff --force -c + configure ........ ................ ................ + + * /, configure: Merged revisions 304462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304462 | qwell | 2011-01-27 10:48:44 -0600 + (Thu, 27 Jan 2011) | 16 lines Merged revisions 304461 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304461 | qwell | 2011-01-27 10:48:00 -0600 + (Thu, 27 Jan 2011) | 9 lines Merged revisions 304460 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan + 2011) | 1 line Rerun bootstrap.sh with no changes, so that it is + more obvious what my next commit changes. ........ + ................ ................ + +2011-01-27 15:57 +0000 [r304422] Kevin P. Fleming + + * res/res_fax.c: Rename the SendFAX/ReceiveFAX 'force audio' + option. The recently added option to disable usage of T.38 for a + single session should have been named 'F' for 'force audio', + since that is really what the user is asking to happen (and it's + a positive option instead of a negative option that way). + +2011-01-27 00:06 +0000 [r304385] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, channels/sig_pri.c: Merged from + revision 304341 + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, + 26 Jan 2011) | 7 lines Add connected line chan_dahdi.conf + pricpndialplan option. * Added from_channel value to + prilocaldialplan option. JIRA ABE-2731 JIRA SWP-2842 .......... + +2011-01-26 23:41 +0000 [r304384] Jeff Peeler + + * apps/app_followme.c: Add option to followme to delay answer until + ready to bridge call. Followme answers an incoming call if it + hasn't already been answered and starts MOH. Some poorly designed + autodialers see the answer and start playing their message to the + hold music. The 'N' option has been added to indicate ringing and + not answer until the call is accepted. (closes issue #18479) + Reported by: ianc Patches: trunk_followme.diff uploaded by ianc + (license 998) + +2011-01-26 22:39 +0000 [r304342] Kevin P. Fleming + + * res/res_fax.c: Add ability to disable T.38 usage for specific + SendFAX/ReceiveFAX sessions. Sometimes during troubleshooting it + can be useful to disable T.38 usage in order to narrow down a + problem. This patch adds an 'n' option to SendFAX and ReceiveFAX + so that can be done without having to disable T.38 usage entirely + for the peer that Asterisk is communicating with. (inspired by + trying to assist Bryant Zimmerman on asterisk-users) + +2011-01-26 22:27 +0000 [r304340] Jeff Peeler + + * /, main/features.c: Merged revisions 304339 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304339 | jpeeler | 2011-01-26 16:27:30 -0600 + (Wed, 26 Jan 2011) | 9 lines Merged revisions 304338 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 + Jan 2011) | 2 lines Change delimiter used internally for + GOTO_ON_BLINDXFR to commas to match 76703. ........ + ................ + +2011-01-26 21:03 +0000 [r304252] Mark Michelson + + * main/udptl.c, /: Merged revisions 304250 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600 + (Wed, 26 Jan 2011) | 9 lines Merged revisions 304242 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed, + 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in + ast_udptl ........ ................ + +2011-01-26 20:44 +0000 [r304246] Matthew Nicholson + + * main/netsock2.c, /, channels/chan_sip.c, + channels/sip/reqresp_parser.c, include/asterisk/netsock2.h, + channels/sip/include/sip.h, + channels/sip/include/reqresp_parser.h: Merged revisions 304245 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304245 | mnicholson | 2011-01-26 14:43:27 -0600 + (Wed, 26 Jan 2011) | 20 lines Merged revisions 304244 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600 + (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan + 2011) | 6 lines This patch modifies chan_sip to route responses + to the address the request came from. It also modifies chan_sip + to respect the maddr parameter in the Via header. ABE-2664 + Review: https://reviewboard.asterisk.org/r/1059/ ........ + ................ ................ + +2011-01-26 20:25 +0000 [r304195] Sean Bright + + * /, configs/queues.conf.sample: Merged revisions 304186 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304186 | seanbright | 2011-01-26 15:23:48 -0500 + (Wed, 26 Jan 2011) | 16 lines Merged revisions 304181 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304181 | seanbright | 2011-01-26 15:22:47 -0500 + (Wed, 26 Jan 2011) | 9 lines Merged revisions 304159 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed, + 26 Jan 2011) | 1 line Make sure the sample queues.conf is + properly commented. ........ ................ ................ + +2011-01-26 19:58 +0000 [r304152] Matthew Nicholson + + * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions + 303907 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r303907 | mnicholson | 2011-01-25 14:56:12 -0600 (Tue, 25 Jan + 2011) | 2 lines Reimplemented fax session reservation to reverse + the ABI breakage introduced in r297486. ........ + +2011-01-26 19:40 +0000 [r304151] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 304150 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304150 | rmudgett | 2011-01-26 13:39:35 -0600 + (Wed, 26 Jan 2011) | 16 lines Merged revisions 304149 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600 + (Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, + 26 Jan 2011) | 2 lines Update documentation for + DAHDISendCallreroutingFacility() application. .......... + ................ ................ + +2011-01-26 01:27 +0000 [r304098] Sean Bright + + * /, main/file.c: Merged revisions 304097 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304097 | seanbright | 2011-01-25 20:26:26 -0500 + (Tue, 25 Jan 2011) | 19 lines Merged revisions 304096 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan + 2011) | 12 lines Per the man page, setvbuf() must be called + before any other operation on an open file. We use setvbuf() to + associate a buffer with a stream, but we have already written to + the open file. This works (by chance) on Linux, but fails on + other platforms, such as OpenSolaris. (closes issue #16610) + Reported by: bklang Patches: setvbuf.patch uploaded by crjw + (license 963) Tested by: bklang, asgaroth, efutch ........ + ................ + +2011-01-25 23:31 +0000 [r304008] Richard Mudgett + + * /, main/features.c: Merged revisions 304007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304007 | rmudgett | 2011-01-25 17:28:25 -0600 + (Tue, 25 Jan 2011) | 22 lines Merged revisions 304006 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304006 | rmudgett | 2011-01-25 17:25:32 -0600 + (Tue, 25 Jan 2011) | 15 lines Merged revisions 304005 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) + | 8 lines DTMF attended transfers sometimes fail for no apparent + reason. The loop in feature_request_and_dial() can exit when + Party C has answered without processing an AST_CONTROL_ANSWER. + Also sometimes an AST_CONTROL_ANSWER never happens even though + Party C has answered. Don't hangup Party C if he is up or we + receive an AST_CONTROL_ANSWER. ........ ................ + ................ + +2011-01-25 22:15 +0000 [r303963] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 303962 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303962 | twilson | 2011-01-25 16:09:01 -0600 + (Tue, 25 Jan 2011) | 30 lines Merged revisions 303960 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303960 | twilson | 2011-01-25 16:02:42 -0600 + (Tue, 25 Jan 2011) | 23 lines Merged revisions 303906 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) + | 16 lines Guard against retransmitting BYEs indefinitely In the + case of an attended transfer (A calls B, A atxfers to C) where A + becomes unreachable before replying to Asterisk's BYE, Asterisk + can sometimes retransmit the BYE indefinitely. This is because + __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], + SIP_ALREADYGONE and will then transmit a BYE. When this BYE times + out, it will not ever be marked as ALREADYGONE, so when + __sip_autodestruct is called again, we end up starting the cycle + over. This patch adds a call to sip_alreadygone(pkt->owner) in + retrans_pkt in the case of a BYE that has timed out. This should + prevent Asterisk from trying to transmit new BYE messages in the + future. Review: https://reviewboard.asterisk.org/r/1077/ ........ + ................ ................ + +2011-01-25 18:56 +0000 [r303861] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 303860 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303860 | tilghman | 2011-01-25 12:55:27 -0600 + (Tue, 25 Jan 2011) | 12 lines Merged revisions 303858 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011) + | 5 lines Fix "sip show user ", so that it actually shows + results, instead of just completing the last entry. (closes issue + #16675) Reported by: pj ........ ................ + +2011-01-25 17:58 +0000 [r303772] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h, /, + channels/sig_pri.c, channels/sig_ss7.c: Merged revisions 303771 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 + (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 + (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) + | 40 lines Sending out unnecessary PROCEEDING messages breaks + overlap dialing. Issue #16789 was a good idea. Unfortunately, it + breaks overlap dialing through Asterisk. There is not enough + information available at this point to know if dialing is + complete. The ast_exists_extension(), ast_matchmore_extension(), + and ast_canmatch_extension() calls are not adequate to detect a + dial through extension pattern of "_9!". Workaround is to use the + dialplan Proceeding() application early in non-dial through + extensions. * Effectively revert issue #16789. * Allow outgoing + overlap dialing to hear dialtone and other early media. A + PROGRESS "inband-information is now available" message is now + sent after the SETUP_ACKNOWLEDGE message for non-digital calls. + An AST_CONTROL_PROGRESS is now generated for incoming + SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of + the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent + with the cause codes. * Added better protection from sending out + of sequence messages by combining several flags into a single + enum value representing call progress level. * Added diagnostic + messages for deferred overlap digits handling corner cases. + (closes issue #17085) Reported by: shawkris (closes issue #18509) + Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch + uploaded by rmudgett (license 664) Expanded upon + issue18509_early_media_v1.8_v3.patch to include analog and SS7 + because of backporting requirements. Tested by: wimpy, rmudgett + ........ ................ ................ + +2011-01-25 17:05 +0000 [r303679] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 303678 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303678 | jpeeler | 2011-01-25 11:02:38 -0600 + (Tue, 25 Jan 2011) | 33 lines Merged revisions 303677 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303677 | jpeeler | 2011-01-25 10:59:28 -0600 + (Tue, 25 Jan 2011) | 26 lines Merged revisions 303676 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) + | 20 lines Fix voicemail sequencing for file based storage. A + previous change was made to account for when the number of + voicemail messages exceeds the max limit to be handled properly, + but it caused gaps in the messages to not be properly handled. + This has now been resolved. In later non 1.4 branches, it appears + that resequencing wasn't even occurring due from what appears and + accidental code removal. (closes issue #18498) Reported by: + JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license + 325) (closes issue #18486) Reported by: bluefox Patches: + bug18486.patch uploaded by jpeeler (license 325) ........ + ................ ................ + +2011-01-25 15:52 +0000 [r303638] Matthew Nicholson + + * main/utils.c: Use unsigned char in comparison for UTF8 check to + quiet a compiler warning. + +2011-01-24 20:57 +0000 [r303547-303551] Russell Bryant + + * main/channel.c, main/pbx.c, /, apps/app_meetme.c, + main/features.c, include/asterisk/channel.h: Merged revisions + 303549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303549 | russell | 2011-01-24 14:51:37 -0600 + (Mon, 24 Jan 2011) | 45 lines Merged revisions 303548 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303548 | russell | 2011-01-24 14:49:53 -0600 + (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) + | 31 lines Fix channel redirect out of MeetMe() and other issues + with channel softhangup. Mantis issue #18585 reports that a + channel redirect out of MeetMe() stopped working properly. This + issue includes a patch that resolves the issue by removing a call + to ast_check_hangup() from app_meetme.c. I left that in my patch, + as it doesn't need to be there. However, the rest of the patch + fixes this problem with or without the change to app_meetme. The + key difference between what happens before and after this patch + is the effect of the END_OF_Q control frame. After END_OF_Q is + hit in ast_read(), ast_read() will return NULL. With the + ast_check_hangup() removed, app_meetme sees this which causes it + to exit as intended. Checking ast_check_hangup() caused + app_meetme to exit earlier in the process, and the target of the + redirect saw the condition where ast_read() returned NULL. + Removing ast_check_hangup() works around the issue in app_meetme, + but doesn't solve the issue if another application did the same + thing. There are also other edge cases where if an application + finishes at the same time that a redirect happens, the target of + the redirect will think that the channel hung up. So, I made some + changes in pbx.c to resolve it at a deeper level. There are + already places that unset the SOFTHANGUP_ASYNCGOTO flag in an + attempt to abort the hangup process. My patch extends this to + remove the END_OF_Q frame from the channel's read queue, making + the "abort hangup" more complete. This same technique was used in + every place where a softhangup flag was cleared. (closes issue + #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: + https://reviewboard.asterisk.org/r/1082/ ........ + ................ ................ + + * contrib/scripts/install_prereq: Add gsm-devel as a package to + install on redhat based systems. + +2011-01-24 18:59 +0000 [r303509] Matthew Nicholson + + * res/res_config_curl.c, include/asterisk/utils.h, + funcs/func_curl.c, channels/chan_sip.c, tests/test_utils.c, + res/res_agi.c, channels/sip/reqresp_parser.c, main/http.c, + main/utils.c, funcs/func_uri.c: According to section 19.1.2 of + RFC 3261: For each component, the set of valid BNF expansions + defines exactly which characters may appear unescaped. All other + characters MUST be escaped. This patch modifies ast_uri_encode() + to encode strings in line with this recommendation. This patch + also adds an ast_escape_quoted() function which escapes '"' and + '\' characters in quoted strings in accordance with section 25.1 + of RFC 3261. The ast_uri_encode() function has also been modified + to take an ast_flags struct describing the set of rules it should + use when escaping characters to allow for it to escape SIP URIs + in addition to HTTP URIs and other types of URIs or variations of + those two URI types in the future. The ast_uri_decode() function + has also been modified to accept an ast_flags struct describing + the set of rules to use when decoding to enable decoding '+' as ' + ' in legacy http URLs. The unit tests for these functions have + also been updated. ABE-2705 Review: + https://reviewboard.asterisk.org/r/1081/ + +2011-01-24 17:21 +0000 [r303468] Jason Parker + + * channels/chan_dahdi.c, /: Merged revisions 303467 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303467 | qwell | 2011-01-24 11:20:03 -0600 + (Mon, 24 Jan 2011) | 22 lines Merged revisions 303285 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 + (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | + 8 lines Reset configuration before parsing users.conf. Some + values configured in chan_dahdi.conf were able to leak in to + users.conf configuration. This was surprising users, and + potentially setting non-sane "defaults". ASTNOW-125 ........ + ................ ................ + +2011-01-22 04:13 +0000 [r303418] Russell Bryant + + * configure, configure.ac: Revert default compiler change. If + someone wishes to do so, it is trivial to set your own default + when running the configure script. + +2011-01-21 23:11 +0000 [r303288-303376] Jason Parker + + * channels/chan_dahdi.c, /: Temporarily revert r303288 + + * channels/chan_dahdi.c, /: Merged revisions 303286 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303286 | qwell | 2011-01-21 15:50:11 -0600 + (Fri, 21 Jan 2011) | 22 lines Merged revisions 303285 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 + (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | + 8 lines Reset configuration before parsing users.conf. Some + values configured in chan_dahdi.conf were able to leak in to + users.conf configuration. This was surprising users, and + potentially setting non-sane "defaults". ASTNOW-125 ........ + ................ ................ + +2011-01-21 09:09 +0000 [r303198-303235] Tilghman Lesher + + * configure, configure.ac: Really use llvm-gcc, when available. + + * funcs/func_db.c, CHANGES: Add DB_KEYS. Discussion on #asterisk on + 2011-01-19: (02:07:03 PM) boch: i wonder how to cycle all entries + in a tree (02:07:11 PM) leifmadsen: use While() (02:07:17 PM) + leifmadsen: you need to know the tree structure already though + (02:07:36 PM) boch: what you mean? (02:09:02 PM) leifmadsen: you + need to know the structure prior to looping, because you can't + just return the structure from the dialplan (02:09:43 PM) + leifmadsen: the only way I can think of doing that is via + something like writing the output of: asterisk -rx "database + show" to a file, then looping through that to know the structure + of the database and check everything (02:09:59 PM) leifmadsen: + but at that point you're better off just using either a + relational database or an external script (02:10:13 PM) boch: for + example i need to know all entries in the tree (02:10:15 PM) + boch: got it (02:10:20 PM) leifmadsen: exactly (02:10:22 PM) + leifmadsen: that's the problem (02:10:22 PM) boch: thank you + (02:13:09 PM) mateu: yeah, i'm surprised there isn't something + from the dialplan like 'database show family' so one can get all + keys in a family to loop over. (02:15:35 PM) leifmadsen: database + shows everything (02:16:22 PM) mateu: i mean something from the + dial plan that mimics 'database show ' (02:16:41 PM) + leifmadsen: guess no one has found that important enough to + program :) (02:16:52 PM) leifmadsen: at that point you should + probably just use a relational database... (02:17:10 PM) mateu: i + dunno (02:17:16 PM) mateu: seems pretty basic to me. (02:17:16 + PM) leifmadsen: me either (02:17:19 PM) leifmadsen: sure does + (02:17:24 PM) leifmadsen: no one has programmed it though + (02:17:28 PM) ***leifmadsen shrugs (02:17:43 PM) mateu: ok, well + at least we know how it currently stands. thanks leifmadsen + (02:28:52 PM) Corydon76-home: leifmadsen: something like + HASHKEYS() ? (02:30:11 PM) leifmadsen: Corydon76-home: ummm, I + was thinking more like DUNDI_QUERY() and DUNDI_RESULT() (02:30:31 + PM) leifmadsen: although HASHKEYS() might work (02:30:58 PM) + leifmadsen: actually ya, looking at it, similar to HASHKEYS() + (02:31:01 PM) leifmadsen: DBKEYS() I guess? (02:31:45 PM) + Corydon76-home: So with no argument, retrieves families, with an + argument, retrieves keys of that family? (02:34:02 PM) + leifmadsen: ya (02:34:16 PM) leifmadsen: how would you iterate + through layers of them? (02:34:30 PM) leifmadsen: i.e. + family/key/key/key ? (02:34:43 PM) Corydon76-home: Essentially, + yes + +2011-01-20 20:35 +0000 [r303154] Richard Mudgett + + * /, main/ccss.c: Merged revisions 303153 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303153 | rmudgett | 2011-01-20 14:31:20 -0600 + (Thu, 20 Jan 2011) | 22 lines Merged revision 303098 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, + 20 Jan 2011) | 15 lines CC_INTERFACES does not get built + correctly with local channels. If local channels are used with + CCSS, CC_INTERFACES gets garbage prepended to it so the CC recall + fails. Also CC_INTERFACES gets "&(null)" appended to it. * + Initialize the buffer to eliminate the prepended garbage. * + Filter out the empty interface strings to eliminate the latter. * + Added a diagnostic message if the CC_INTERFACES is ever empty. + JIRA ABE-2740 JIRA SWP-2848 .......... ................ + +2011-01-20 19:58 +0000 [r303108] Shaun Ruffell + + * /, main/features.c: Merged revisions 303107 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303107 | sruffell | 2011-01-20 13:57:31 -0600 + (Thu, 20 Jan 2011) | 23 lines Merged revisions 303106 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011) + | 15 lines main/features: Use POLLPRI when waiting for events on + parked channels. This change resolves a regression in the 1.6.2 + when converting from select to poll. The DAHDI timers use POLLPRI + to indicate that the timer fired, but features was not waiting + for that flag. The result was no audio for MOH when a call was + parked and res_timing_dahdi was in use. This patch is slightly + modified from the one on the mantis issue. It does not set an + exception on the channel if the POLLPRI flag is set. (closes + issue #18262) Reported by: francesco_r Patches: + patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029) + Tested by: francesco_r, rfrantik, one47 ........ ................ + +2011-01-20 17:14 +0000 [r303011] Jeff Peeler + + * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions + 303009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303009 | jpeeler | 2011-01-20 11:10:32 -0600 + (Thu, 20 Jan 2011) | 21 lines Merged revisions 303008 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600 + (Thu, 20 Jan 2011) | 14 lines Merged revisions 303007 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) + | 8 lines Add new queue strategy to preserve behavior for when + queue members moved to ao2. Add queue strategy called "rrordered" + to mimic old behavior from when queue members were stored in a + linked list. ABE-2707 ........ ................ ................ + +2011-01-20 16:12 +0000 [r302922] Russell Bryant + + * /, apps/app_privacy.c: Merged revisions 302921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302921 | russell | 2011-01-20 10:12:15 -0600 + (Thu, 20 Jan 2011) | 9 lines Merged revisions 302920 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 + Jan 2011) | 2 lines Resolve a compiler warning. ........ + ................ + +2011-01-20 15:46 +0000 [r302919] Leif Madsen + + * apps/app_dial.c, /: Merged revisions 302918 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302918 | lmadsen | 2011-01-20 09:45:39 -0600 + (Thu, 20 Jan 2011) | 16 lines Merged revisions 302917 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) + | 8 lines Option L() is milliseconds, not seconds. > Change the + verbose output of option L() to say milliseconds and not seconds + > as the value is in milliseconds. > > (closes issue #18264) > + Reported by: jacco > Patches: > app_dial_patch.txt uploaded by + lmadsen (license 10) ........ ................ + +2011-01-20 09:07 +0000 [r302879] Tilghman Lesher + + * configure, configure.ac: On systems which have LLVM, use that + compiler. Should result in a massive speed increase. + +2011-01-19 23:57 +0000 [r302838] Russell Bryant + + * main/manager.c, /: Merged revisions 302837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) + | 2 lines Only check container count if it exists. ........ + +2011-01-19 23:53 +0000 [r302835-302836] Sean Bright + + * main/config.c: Clarify a source comment about configuration + template categories. (closes issue #18578) Reported by: astmiv + Patches: asterisk.main.config.2.patch uploaded by astmiv (license + 1189) + + * /, apps/app_voicemail.c: Merged revisions 302834 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302834 | seanbright | 2011-01-19 18:49:00 -0500 + (Wed, 19 Jan 2011) | 14 lines Merged revisions 302833 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed, 19 Jan + 2011) | 7 lines Support greetingsfolder as documented in + voicemail.conf.sample. (closes issue #17870) Reported by: + edhorton Patches: + __20100816-app_voicemail-greetingsfolder-support.txt uploaded by + lmadsen (license 10) ........ ................ + +2011-01-19 23:33 +0000 [r302832] Paul Belanger + + * /, contrib/scripts/install_prereq: Merged revisions 302831 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302831 | pabelanger | 2011-01-19 18:29:45 -0500 (Wed, 19 Jan + 2011) | 2 lines Add binutils-dev for BETTER_BACKTRACES ........ + +2011-01-19 23:07 +0000 [r302786-302790] Russell Bryant + + * main/manager.c, /: Merged revisions 302789 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302789 | russell | 2011-01-19 17:06:46 -0600 + (Wed, 19 Jan 2011) | 11 lines Merged revisions 302788 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011) + | 4 lines Turn a noisy verbose message into a debug message. This + can drown your console if you're using the AMI over HTTP. + ........ ................ + + * main/manager.c, /: Merged revisions 302785 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) + | 15 lines Resolve a memory leak with the manager interface is + disabled. The intent of this check as it stands in previous + versions of Asterisk was to check if there are any active + sessions. If there were no sessions, then the function would + return immediately and not bother with queueing up the manager + event to be processed. Since the conversion of storing sessions + in an astobj2 container, this check will always pass. I changed + it to go back to checking what was intended. The side effect of + this was that if the AMI is disabled, the manager event queue is + populated anyway, but the code that runs to clear out the queue + never runs. A producer with no consumer is a bad thing. Reported + internally by kmorgan. ........ + +2011-01-19 21:35 +0000 [r302732] Richard Mudgett + + * /, main/features.c: Merged revisions 302713 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302713 | rmudgett | 2011-01-19 15:29:22 -0600 + (Wed, 19 Jan 2011) | 29 lines Merged revisions 302693 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600 + (Wed, 19 Jan 2011) | 22 lines Merged revisions 302671 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) + | 15 lines DTMF transfer plays the wrong sounds for wrong number + or other call failure. * Set the default for features.conf.sample + xferfailsound option to "beeperr" as documented instead of + "pbx-invalid" and corrected the use of it in DTMF blind transfer + (#1). * Improved DTMF blind transfer handling of wrong numbers. + Most of the concerns in this issue were taken care of by the + patch for issue 17999: Issues with DTMF triggered attended + transfers. (closes issue #18379) Reported by: gincantalupo Tested + by: rmudgett ........ ................ ................ + +2011-01-19 21:24 +0000 [r302644-302686] Tilghman Lesher + + * /, include/asterisk/astdb.h: Merged revisions 302680 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302680 | tilghman | 2011-01-19 15:23:31 -0600 + (Wed, 19 Jan 2011) | 16 lines Merged revisions 302675 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r302675 | tilghman | 2011-01-19 15:22:45 -0600 + (Wed, 19 Jan 2011) | 9 lines Merged revisions 302663 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 + Jan 2011) | 2 lines Add some API documentation ........ + ................ ................ + + * /, main/app.c: Merged revisions 302634 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302634 | tilghman | 2011-01-19 14:24:57 -0600 + (Wed, 19 Jan 2011) | 22 lines Merged revisions 302599 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) + | 15 lines Kill zombies. When we ast_safe_fork() with a non-zero + argument, we're expected to reap our own zombies. On a zero + argument, however, the zombies are only reaped when there aren't + any non-zero forked children alive. At other times, we accumulate + zombies. This code is forward ported from res_agi in 1.4, so that + forked children are always reaped, thus preventing an + accumulation of zombie processes. (closes issue #18515) Reported + by: ernied Patches: 20101221__issue18515.diff.txt uploaded by + tilghman (license 14) Tested by: ernied ........ ................ + +2011-01-19 20:15 +0000 [r302601] Jason Parker + + * /, res/res_fax.c: Merged revisions 302600 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302600 | qwell | 2011-01-19 14:14:40 -0600 (Wed, 19 Jan 2011) | + 1 line Fix typo pointed out on asterisk-users list. ........ + +2011-01-19 19:04 +0000 [r302507-302556] Sean Bright + + * /, main/utils.c: Merged revisions 302555 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302555 | seanbright | 2011-01-19 14:03:32 -0500 + (Wed, 19 Jan 2011) | 14 lines Merged revisions 302554 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan + 2011) | 7 lines Don't call strlen() when we only need to look at + the next character or two. (closes issue #18042) Reported by: + wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded + by wdoekes (license 717) ........ ................ + + * /, main/features.c: Merged revisions 302552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302552 | seanbright | 2011-01-19 13:54:47 -0500 + (Wed, 19 Jan 2011) | 14 lines Merged revisions 302551 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan + 2011) | 7 lines Remove an extraneous \r\n at the end of a parking + manager events. (closes issue #18363) Reported by: + clegall_proformatique Patches: + asterisk_1.8_295998_parking_manager_events_format.patch uploaded + by clegall proformatique (license 1139) ........ ................ + + * /, res/res_agi.c: Merged revisions 302549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302549 | seanbright | 2011-01-19 13:43:11 -0500 + (Wed, 19 Jan 2011) | 17 lines Merged revisions 302548 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan + 2011) | 10 lines Properly handle partial reads from fgets() when + handling AGIs. When fgets() failed with EAGAIN, we were + continually decrementing the available space left in our buffer, + resulting in botched command handling. (closes issue #16032) + Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by + fnordian (license 110) ........ ................ + + * /, main/utils.c: Merged revisions 302505 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302505 | seanbright | 2011-01-19 12:58:11 -0500 + (Wed, 19 Jan 2011) | 14 lines Merged revisions 302504 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan + 2011) | 7 lines Make sure that h_length is set when we + short-circuit out of ast_gethostbyname. (closes issue #16135) + Reported by: thedavidfactor Patches: utils.patch uploaded by + thedavidfactor (license 903) ........ ................ + +2011-01-19 17:15 +0000 [r302463] Paul Belanger + + * /, res/res_timing_timerfd.c: Merged revisions 302462 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302462 | pabelanger | 2011-01-19 12:09:35 -0500 + (Wed, 19 Jan 2011) | 9 lines Merged revisions 302461 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed, + 19 Jan 2011) | 2 lines Handle 'Resource temporarily unavailable' + error more gracefully. ........ ................ + +2011-01-19 15:54 +0000 [r302413-302418] Sean Bright + + * /, configs/extensions.conf.sample: Merged revisions 302417 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302417 | seanbright | 2011-01-19 10:53:20 -0500 + (Wed, 19 Jan 2011) | 16 lines Merged revisions 302416 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan + 2011) | 9 lines Remove references to priorityjumping from the + sample extensions.conf. Priority jumping was removed from + pbx_config in r68970. (closes issue #18622) Reported by: kshumard + Patches: extensions.conf.sample.patch uploaded by kshumard + (license 92) ........ ................ + + * /, channels/chan_sip.c: Merged revisions 302414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302414 | seanbright | 2011-01-19 10:45:17 -0500 (Wed, 19 Jan + 2011) | 7 lines Initialize an uninitialized variable. (closes + issue #18640) Reported by: jcovert Patches: chan_sip.c.patch + uploaded by jcovert (license 551) ........ + + * channels/chan_local.c, /: Merged revisions 302412 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, + 19 Jan 2011) | 10 lines Use appropriate type for requested format + in chan_local. We were passing and storing the requested format + as an int instead of format_t resulting in truncation. (closes + issue #18238) Reported by: whizemen Patches: + 0018238_speex16.patch uploaded by whizemen (license 1143) + ........ + +2011-01-18 22:06 +0000 [r302319] Richard Mudgett + + * /, main/features.c: Merged revisions 302318 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302318 | rmudgett | 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011) + | 1 line Use the expanded format type instead of plain int. + ........ + +2011-01-18 21:44 +0000 [r302315] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 302314 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302314 | mnicholson | 2011-01-18 15:43:21 -0600 + (Tue, 18 Jan 2011) | 18 lines Merged revisions 302313 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600 + (Tue, 18 Jan 2011) | 11 lines Merged revisions 302311 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan + 2011) | 4 lines URI encode the user part of the contact header. + ABE-2705 ........ ................ ................ + +2011-01-18 20:40 +0000 [r302270] Jeff Peeler + + * main/pbx.c, /: Merged revisions 302266 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302266 | jpeeler | 2011-01-18 14:19:57 -0600 + (Tue, 18 Jan 2011) | 34 lines Merged revisions 302265 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) + | 27 lines Convert device state callbacks to ao2 objects to fix a + deadlock in chan_sip. Lock scenario presented here: Thread 1 + holds ast_rdlock_contexts &conlock holds handle_statechange hints + holds handle_statechange hint waiting for cb_extensionstate + Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds + handle_request_do &netlock holds find_call sip_pvt_ptr waiting + for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911 + (ast_rdlock_contexts) Chan_sip has an established locking order + of locking the sip_pvt and then getting the context lock. So the + as stated by the summary, the operations in thread 2 have been + modified to no longer require the context lock. (closes issue + #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch + uploaded by one47 (license 23), modified by me Review: + https://reviewboard.asterisk.org/r/1072/ ........ + ................ + +2011-01-18 20:21 +0000 [r302268] Russell Bryant + + * /, main/astobj2.c: Merged revisions 302267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302267 | russell | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) + | 5 lines Don't enable AO2_DEBUG by default if AST_DEVMODE is on. + AO2_DEBUG is not important and is causing a false compiler + warning to be generated on my Ubuntu Natty dev box. ........ + +2011-01-18 18:17 +0000 [r302178] Richard Mudgett + + * /, main/features.c: Merged revisions 302174 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302174 | rmudgett | 2011-01-18 12:11:43 -0600 + (Tue, 18 Jan 2011) | 102 lines Merged revisions 302173 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600 + (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) + | 88 lines Issues with DTMF triggered attended transfers. Issue + #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in + features.conf for attended transfer). 3) A hears MOH. B dial + number C 4) C ringing. A hears MOH. 5) B hangup. A still hears + MOH. C ringing. 6) A hangup. C still ringing until + "atxfernoanswertimeout" expires. For v1.4 C will ring forever + until C answers the dead line. (Issue #17096) Problem: When A and + B hangup, C is still ringing. Issue #18395 SIP call limit of B is + 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C + ringing 4. Timeout waiting for C to answer 5. Recall to B fails + because B has reached its call limit. Because B reached its call + limit, it cannot do anything until the transfer it started + completes. Issue #17273 Same scenario as issue 18395 but party B + is an FXS port. Party B cannot do anything until the transfer it + started completes. If B goes back off hook before C answers, B + hears ringback instead of the expected dialtone. ********** Note + for the issue #17273 and #18395 fix: DTMF attended transfer works + within the channel bridge. Unfortunately, when either party A or + B in the channel bridge hangs up, that channel is not completely + hung up until the transfer completes. This is a real problem + depending upon the channel technology involved. For chan_dahdi, + the channel is crippled until the hangup is complete. Either the + channel is not useable (analog) or the protocol disconnect + messages are held up (PRI/BRI/SS7) and the media is not released. + For chan_sip, a call limit of one is going to block that endpoint + from any further calls until the hangup is complete. For party A + this is a minor problem. The party A channel will only be in this + condition while party B is dialing and when party B and C are + conferring. The conversation between party B and C is expected to + be a short one. Party B is either asking a question of party C or + announcing party A. Also party A does not have much incentive to + hangup at this point. For party B this can be a major problem + during a blonde transfer. (A blonde transfer is our term for an + attended transfer that is converted into a blind transfer. :)) + Party B could be the operator. When party B hangs up, he assumes + that he is out of the original call entirely. The party B channel + will be in this condition while party C is ringing, while + attempting to recall party B, and while waiting between call + attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to + fix the problem. It will replace the party B channel technology + with a NULL channel driver to complete hanging up the party B + channel technology. The consequences of this code is that the 'h' + extension will not be able to access any channel technology + specific information like SIP statistics for the call. + ATXFER_NULL_TECH is not defined by default. ********** (closes + issue #17999) Reported by: iskatel Tested by: rmudgett JIRA + SWP-2246 (closes issue #17096) Reported by: gelo Tested by: + rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: + shihchuan Tested by: rmudgett (closes issue #17273) Reported by: + grecco Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/1047/ ........ + ................ ................ + +2011-01-17 16:38 +0000 [r302006-302048] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 293493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) + | 14 lines Only offer codecs both sides support for directmedia + When using directmedia, Asterisk needs to limit the codecs + offered to just the ones that both sides recognize, otherwise + they may end up sending audio that the other side doesn't + understand. (closes issue #17403) Reported by: one47 Patches: + sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: + one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ + ........ + + * /, configs/sip.conf.sample: Merged revisions 302005 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17 + Jan 2011) | 2 lines Document "encryption" option in + sip.conf.sample ........ + +2011-01-14 21:13 +0000 [r301947] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 301946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) + | 13 lines Deadlock between dahdi_request() and pri_dchannel() + processing an incomming call. The sig_pri_new_ast_channel() is + called with the channel private lock held when pri_dchannel() + calls it and no channel private lock held when dahdi_request() + calls it. The use of pri_grab() in sig_pri_new_ast_channel() + could leave the channel private lock held when it returns if the + lock was not held before calling it. Make + sig_pri_new_ast_channel() just lock the PRI span lock instead of + using pri_grab(). It is safe to do this because dahdi_request() + does not have the channel private lock and the deadlock potential + with the PRI span lock is only between pri_dchannel() and other + threads. ........ + +2011-01-14 20:18 +0000 [r301858] Brett Bryant + + * channels/chan_multicast_rtp.c, /: Merged revisions 301851 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301851 | bbryant | 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) + | 6 lines Changing previous revisions 301845/301847 to use + ast_sockaddr_setnull() instead of setting the field manually to + avoid uninitialized data. Review: + https://reviewboard.asterisk.org/r/1076/ ........ + +2011-01-14 20:07 +0000 [r301850] Andrew Latham + + * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to + function documentation. Fix amatuer type mistake + +2011-01-14 19:44 +0000 [r301847] Brett Bryant + + * channels/chan_multicast_rtp.c, /: Merged revisions 301845 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301845 | bbryant | 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) + | 9 lines Fix for a consistent MulticastRTP channel driver crash + due to use of unitilized data. (closes issue #18290) (closes + issue #18602) Reported by: voipgate, wybecom Review: + https://reviewboard.asterisk.org/r/1076/ ........ + +2011-01-14 19:39 +0000 [r301846] Andrew Latham + + * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to + function documentation. + +2011-01-14 17:34 +0000 [r301791] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 301790 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301790 | jpeeler | 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) + | 42 lines Resolve deadlock involving REFER. Two fixes: 1) One + must always have the private unlocked before calling + pbx_builtin_setvar_helper to not invalidate locking order since + it locks the channel. 2) Unlock the channel before calling + pbx_find_extension, which starts and stops autoservice during the + lookup. The problem scenario as illustrated by the reporter: + Thread: do_monitor ----------------------- handle_request_do + handle_incoming handle_request_refer ast_parking_ext_valid + pbx_find_extension ast_autoservice_stop while (chan_list_state == + as_chan_list_state) { usleep(1000); } Thread: autoservice_run + ----------------------- autoservice_run chan = ast_waitfor_n + ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple / + complex (depending on your system) ast_channel_lock(c[x]); + handle_request_do and schedule_process_request_queue locks the + owner if it exists. The autoservice thread is waiting for the + channel lock, which wasn't ever released since the do_monitor + thread was waiting for autoservice operations to complete. Solved + by unlocking the channel but keeping a reference to guarantee + safety. (closes issue #18403) Reported by: jthurman Patches: + 20110103-blind_deadlock.diff uploaded by jthurman (license 614) + issue18403.patch uploaded by jpeeler (license 325) Tested by: + jthurman ........ + +2011-01-13 17:02 +0000 [r301732] Leif Madsen + + * /, configs/phoneprov.conf.sample: Merged revisions 301731 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301731 | lmadsen | 2011-01-13 11:01:43 -0600 + (Thu, 13 Jan 2011) | 15 lines Merged revisions 301730 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011) + | 7 lines Add static entry for split Polycom 332 firmware. + (closes issue #18607) Reported by: cjacobsen Patches: + polycom_331.diff uploaded by cjacobsen (license 1029) Tested by: + lathama ........ ................ + +2011-01-13 16:27 +0000 [r301729] Paul Belanger + + * main/pbx.c, CHANGES: Add dialplan variables for asterisk.conf + directories Review: https://reviewboard.asterisk.org/r/1075/ + +2011-01-12 21:24 +0000 [r301684] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 301683 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301683 | twilson | 2011-01-12 15:19:48 -0600 + (Wed, 12 Jan 2011) | 15 lines Merged revisions 301682 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) + | 9 lines Don't reject all SUBSCRIBE auth requests When merging + another SUBSCRIBE fix from 1.4, some braces were put in the wrong + place. This patch fixes that. (closes issue #18597) Reported by: + thsgmbh ........ ................ + +2011-01-12 18:52 +0000 [r301596] Matthew Nicholson + + * main/manager.c, /: Merged revisions 301595 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301595 | mnicholson | 2011-01-12 12:51:37 -0600 + (Wed, 12 Jan 2011) | 22 lines Merged revisions 301594 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600 + (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't + be necessary in session_do, and removed the ms_t member from the + mansession_session structure. Merged revisions 301591 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan + 2011) | 5 lines Don't store the thread id for the manager session + in the structure we pass to the thread for the manager session. + ABE-2543 ........ ................ ................ + +2011-01-12 18:12 +0000 [r301505] Jeff Peeler + + * main/channel.c, /: Merged revisions 301504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301504 | jpeeler | 2011-01-12 12:12:08 -0600 + (Wed, 12 Jan 2011) | 26 lines Merged revisions 301503 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600 + (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) + | 12 lines Fix CPU spike when pressing DTMF after agent login. + The problem here is that DTMF was being continuously deferred and + requeued since ast_safe_sleep is called in a loop. There are + serveral other places in the code that sleeps and then loops in a + similar fashion. Because of this fact I opted to not defer DTMF + any more, which will not affect the original fix: + https://reviewboard.asterisk.org/r/674 (closes issue #18130) + Reported by: rgj ........ ................ ................ + +2011-01-12 16:05 +0000 [r301447] David Vossel + + * /, main/file.c: Merged revisions 301446 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301446 | dvossel | 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011) + | 2 lines Removal of unused variables so Asterisk will compile. + ........ + +2011-01-12 15:59 +0000 [r301445] Stefan Schmidt + + * Makefile: fix wrong text of rerun menuselect after user interface + warning the warning, if no user interface for menuselect warning + was found is not right. you have to rerun configure before make + menuselect after installing a proper user interface. (closes + issue 0018594) Reported by: Dovid + +2011-01-12 00:27 +0000 [r301403] Tilghman Lesher + + * /, main/file.c: Merged revisions 301402 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301402 | tilghman | 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011) + | 7 lines Call execl() directly for a better solution for paths + with spaces. (closes issue #18600) Reported by: ebroad Patches: + 20110111__issue18600__2.diff.txt uploaded by tilghman (license + 14) ........ + +2011-01-11 19:19 +0000 [r301319] Paul Belanger + + * /, configs/extensions.conf.sample: Merged revisions 301311 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301311 | pabelanger | 2011-01-11 14:16:06 -0500 + (Tue, 11 Jan 2011) | 9 lines Merged revisions 301310 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, + 11 Jan 2011) | 2 lines Fix a logic issue when passing context ARG + ........ ................ + +2011-01-11 18:55 +0000 [r301309] Matthew Nicholson + + * /, main/utils.c: Merged revisions 301308 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301308 | mnicholson | 2011-01-11 12:51:40 -0600 + (Tue, 11 Jan 2011) | 18 lines Merged revisions 301307 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600 + (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan + 2011) | 4 lines Prevent buffer overflows in ast_uri_encode() + ABE-2705 ........ ................ ................ + +2011-01-10 22:40 +0000 [r301264] Tilghman Lesher + + * /, main/strcompat.c: Merged revisions 301263 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301263 | tilghman | 2011-01-10 16:39:31 -0600 (Mon, 10 Jan 2011) + | 8 lines Little endian machines were not converted properly. + (closes issue #18583) Reported by: jcovert Patches: + 20110110__issue18583.diff.txt uploaded by tilghman (license 14) + Tested by: jcovert ........ + +2011-01-09 21:42 +0000 [r301178-301222] Paul Belanger + + * /, configure, configure.ac, autoconf/ast_ext_lib.m4: Merged + revisions 301221 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301221 | pabelanger | 2011-01-09 16:40:34 -0500 + (Sun, 09 Jan 2011) | 21 lines Merged revisions 301220 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan + 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds + files included in the Asterisk tarball were being ignored and + re-downloaded. Users wanting to cache the files can still + override the setting using the --with-sounds-cache option. + (closes issue #18589) Reported by: pabelanger Patches: + issue18589.patch uploaded by pabelanger (license 224) Tested by: + pabelanger Review: https://reviewboard.asterisk.org/r/1074/ + ........ ................ + + * /, apps/app_verbose.c: Merged revisions 301177 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301177 | pabelanger | 2011-01-08 17:00:12 -0500 + (Sat, 08 Jan 2011) | 14 lines Merged revisions 301176 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan + 2011) | 7 lines Indicate log level argument for Log() is not + optional (closes issue #18586) Reported by: kshumard Patches: + app_verbose.c.patch uploaded by kshumard (license 92) ........ + ................ + +2011-01-08 01:13 +0000 [r301135] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 301134 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r301134 | rmudgett | 2011-01-07 19:11:31 -0600 (Fri, 07 + Jan 2011) | 7 lines The DTMF attended transfer feature cannot + callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is + not dialable. Make a channel name like DAHDI/i3/400-12 dialable + when the sequence number is stripped off of the name. ........ + +2011-01-07 20:53 +0000 [r301091] Jason Parker + + * /, apps/app_meetme.c: Merged revisions 301090 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301090 | qwell | 2011-01-07 14:53:02 -0600 + (Fri, 07 Jan 2011) | 15 lines Merged revisions 301089 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | + 8 lines Initialize useropts/adminopts in case there is no column + in the realtime DB. (closes issue #18182) Reported by: dimas + Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: + dimas ........ ................ + +2011-01-07 19:58 +0000 [r301048] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 301047 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301047 | jpeeler | 2011-01-07 13:58:30 -0600 + (Fri, 07 Jan 2011) | 15 lines Merged revisions 301046 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) + | 8 lines Fix regression causing forwarding voicemails to not + work with file storage. I had actually already fixed this in + 295200 in 1.4 and thought it wasn't missing in the other branches + for some reason. (closes issue #18358) Reported by: cabal95 + ........ ................ + +2011-01-07 18:23 +0000 [r301008] Tilghman Lesher + + * funcs/func_curl.c: Oops, missed the actual decoding part. (closes + issue #18046) Reported by: wdoekes + +2011-01-07 17:24 +0000 [r300959] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 300955 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300955 | jpeeler | 2011-01-07 11:24:14 -0600 + (Fri, 07 Jan 2011) | 21 lines Merged revisions 300951 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r300951 | jpeeler | 2011-01-07 11:23:37 -0600 + (Fri, 07 Jan 2011) | 14 lines Merged revisions 300918 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) + | 7 lines Ensure good bye prompt in voicemail is played at the + correct time. Specifically in the case of timing out but not + leaving voicemail nothing should be heard. And when leaving + voicemail it should be heard. ABE-2647 ........ ................ + ................ + +2011-01-07 07:47 +0000 [r300882] Mark Murawki + + * res/res_config_pgsql.c: Added support for postgres database retry + query on disconnection to res_config_pgsql If your postgres + connection died suddenly in between res_config_pgsql queries, the + next query will fail because the query is executed on a + disconnected/disconnecting handle. The query is abandoned and is + returned from in error. Now we will reconnect and try again if a + query was run on a disconnected connection. (closes issue #18071) + +2011-01-06 17:50 +0000 [r300799-300841] Tilghman Lesher + + * Makefile, funcs/func_curl.c: XML validation + + * funcs/func_curl.c: Add a hashcompat mode called "legacy", which + translates a literal plus sign to a space. (closes issue #18046) + Reported by: wdoekes Patches: 20100930__issue18046.diff.txt + uploaded by tilghman (license 14) + + * /, addons/res_config_mysql.c: Merged revisions 300798 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r300798 | tilghman | 2011-01-06 00:28:18 -0600 (Thu, 06 Jan 2011) + | 8 lines Don't destroy handle not created by use (because the + caller will). (closes issue #18526) Reported by: makoto Patches: + res-config-mysql-include.patch uploaded by makoto (license 38) + Tested by: makoto ........ + +2011-01-06 01:41 +0000 [r300761] David Ruggles + + * Makefile, contrib/scripts/safe_asterisk: update safe_asterisk + script change defaults to make a little more sense. Default log + location is now asterisk log location and default email + notification has been changed to root on the local machine + Review: https://reviewboard.asterisk.org/r/1067/ + +2011-01-05 21:07 +0000 [r300716] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 300714 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300714 | rmudgett | 2011-01-05 14:54:21 -0600 + (Wed, 05 Jan 2011) | 21 lines Merged revision 300711 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed, + 05 Jan 2011) | 14 lines A call retrieved from hold may wind up + with no audio. If the retrieved call is natively bridged then the + call may not have any audio path. The following warning message + is given: "Failed to add to conference /: + Invalid argument". * Open the media on a B channel when + pri_fixup_principle() moves the call from a no_b_channel channel + to a real channel. * Added lock protection while + pri_fixup_principle() moves a call from one private structure to + another. * Made some pri_fixup_principle() messages more + meaningful. .......... ................ + +2011-01-05 18:57 +0000 [r300624] Tilghman Lesher + + * /, res/res_odbc.c: Merged revisions 300623 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300623 | tilghman | 2011-01-05 12:56:12 -0600 + (Wed, 05 Jan 2011) | 24 lines Merged revisions 300622 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r300622 | tilghman | 2011-01-05 12:54:58 -0600 + (Wed, 05 Jan 2011) | 17 lines Merged revisions 300621 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011) + | 10 lines Use the sanity check in place of the + disconnect/connect cycle. The disconnect/connect cycle has the + potential to cause random crashes. (closes issue #18243) Reported + by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147) + Tested by: ks3 ........ ................ ................ + +2011-01-05 16:30 +0000 [r300576] Paul Belanger + + * /, cdr/cdr_sqlite.c: Merged revisions 300575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300575 | pabelanger | 2011-01-05 11:29:19 -0500 + (Wed, 05 Jan 2011) | 13 lines Merged revisions 300574 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan + 2011) | 6 lines Change deprecated message to LOG_WARNING Also + removed latter part of message Discussed on #asterisk-dev + ........ ................ + +2011-01-04 21:54 +0000 [r300434-300522] Leif Madsen + + * /, channels/chan_sip.c, channels/chan_agent.c, + channels/chan_iax2.c, main/xmldoc.c: Merged revisions 300521 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300521 | lmadsen | 2011-01-04 15:53:27 -0600 + (Tue, 04 Jan 2011) | 17 lines Merged revisions 300520 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) + | 9 lines Fix backwards and broken XML documentation. (closes + issue #18547) Reported by: jcovert Patches: xmldoc.c.patch + uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded + by jcovert (license 551) chan_sip.c.patch uploaded by jcovert + (license 551) chan_agent.c.patch uploaded by jcovert (license + 551) ........ ................ + + * configs/users.conf.sample, /: Merged revisions 300433 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300433 | lmadsen | 2011-01-04 15:00:55 -0600 + (Tue, 04 Jan 2011) | 15 lines Merged revisions 300431 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011) + | 7 lines Add some documentation to users.conf.sample. (closes + issue #18531) Reported by: lathama Patches: + users.conf.sample2.diff uploaded by lathama (license 1028) Tested + by: lathama ........ ................ + +2011-01-04 21:00 +0000 [r300432] Russell Bryant + + * /, contrib/scripts/autosupport.8, contrib/scripts/autosupport: + Merged revisions 300430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300430 | russell | 2011-01-04 15:00:16 -0600 + (Tue, 04 Jan 2011) | 18 lines Merged revisions 300429 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r300429 | russell | 2011-01-04 14:59:56 -0600 + (Tue, 04 Jan 2011) | 11 lines Merged revisions 300428 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011) + | 4 lines Update the autosupport script from Digium support. + (closes AST-395) ........ ................ ................ + +2011-01-04 19:45 +0000 [r300385] Leif Madsen + + * phoneprov/000000000000.cfg, /: Merged revisions 300384 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r300384 | lmadsen | 2011-01-04 13:45:22 -0600 (Tue, 04 Jan 2011) + | 7 lines Update STAT() to use the comma instead of the pipe. + (closes issue #18503) Reported by: cjacobsen Patches: + old_separator.diff uploaded by cjacobsen (license 1029) Tested + by: lathama ........ + +2011-01-04 18:51 +0000 [r300345] Moises Silva + + * channels/chan_dahdi.c: Update MFC-R2 code to use new DTMF-R2 + functionality in OpenR2 (closes issue #18576) + +2011-01-04 18:06 +0000 [r300302] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 300301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300301 | twilson | 2011-01-04 11:54:41 -0600 + (Tue, 04 Jan 2011) | 29 lines Merged revisions 300298 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r300298 | twilson | 2011-01-04 11:37:26 -0600 + (Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) + | 15 lines Don't authenticate SUBSCRIBE re-transmissions This + only skips authentication on retransmissions that are already + authenticated. A similar method is already used for INVITES. This + is the kind of thing we end up having to do when we don't have a + transaction layer... (closes issue #18075) Reported by: mdu113 + Patches: diff.txt uploaded by twilson (license 396) Tested by: + twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ + ........ ................ ................ + +2011-01-04 17:04 +0000 [r300215] Jan Kalab + + * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, /: + Merged revisions 300214 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r300214 | pitel | 2011-01-04 18:01:52 +0100 (Út, 04 led 2011) | 7 + lines Memory leaking in calendars ne_request_destroy() was + missing in icalendar and exchange calendar modules, causing + memory leak. (closes issue #18521) Review: + https://reviewboard.asterisk.org/r/1068/ ........ + +2011-01-04 16:38 +0000 [r300168-300212] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, UPGRADE.txt, CHANGES, + channels/sig_pri.c: Optional HOLD/RETRIEVE signaling for PTMP TE + when the bridge goes on and off hold. Added the moh_signaling + option to specify what to do when the channel's bridged peer puts + the ISDN channel on and off of hold. Implemented as a FSM to + control libpri ISDN signaling when the bridged peer places the + channel on and off of hold with the AST_CONTROL_HOLD and + AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 + Review: https://reviewboard.asterisk.org/r/1063/ + + * /, main/features.c: Merged revisions 300166 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300166 | rmudgett | 2011-01-03 17:14:55 -0600 + (Mon, 03 Jan 2011) | 11 lines Merged revisions 300165 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011) + | 4 lines Use correct variable for atxfercallbackretries config + option. * Misc formatting changes. ........ ................ + +2011-01-03 14:09 +0000 [r300121] David Ruggles + + * apps/app_externalivr.c: initialize playing_silence in struct + initialization playing_silence was not initialized with the + struct was initialized, it was being set after the fact which + caused problems if something that relied on playing_silence being + set was called too quickly (closes issue #18430) Reported by: + stevebrandli Patches: externalivr.patch uploaded by + thedavidfactor (license 903) Tested by: thedavidfactor, + stevebrandli + +2011-01-03 13:15 +0000 [r300083] Leif Madsen + + * /, pbx/pbx_dundi.c: Merged revisions 300082 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r300082 | lmadsen | 2011-01-03 07:14:25 -0600 (Mon, 03 Jan 2011) + | 11 lines Increase side of mapping response field. I've + increased the size of the response field in a DUNDi mapping + because of some documentation I'm writing. Previously it was set + to AST_MAX_EXTENSION which is only 80 characters, which is far + too small when you're using some dialplan functions to craft a + response. The example I'm using is: extensions => + RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial + ........ + +2010-12-31 09:29 +0000 [r300044-300045] Tilghman Lesher + + * cdr/cdr_adaptive_odbc.c, CHANGES, + configs/cdr_adaptive_odbc.conf.sample: Support negative filters. + (closes issue #17979) Reported by: tilghman Patches: + 20100911__for_blitzrage.diff.txt uploaded by tilghman (license + 14) Tested by: lmadsen + + * main/logger.c, CHANGES: Support an alternate configuration file + for the 'logger reload' command. (closes issue #17668) Reported + by: tilghman Patches: 20100718__logger_reload_altconf__2.diff.txt + uploaded by tilghman (license 14) Review: (by lmadsen, russell + within comments on issue tracker) + +2010-12-29 22:19 +0000 [r300000] Sean Bright + + * main/asterisk.c: Remove some trailing whitespace and steal + revision 300000. + +2010-12-29 22:03 +0000 [r299990] Tilghman Lesher + + * /, main/file.c, apps/app_voicemail.c: Merged revisions 299989 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) + | 4 lines Quote arguments, just in case there's a space in a + pathname. (Diagnosed by pabelanger on #asterisk-dev, fixed by + me.) ........ + +2010-12-29 19:29 +0000 [r299866-299949] Paul Belanger + + * /, sounds/Makefile: Merged revisions 299948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299948 | pabelanger | 2010-12-29 14:28:36 -0500 (Wed, 29 Dec + 2010) | 2 lines Only remove /tmp/astdatadir, not + /var/lib/asterisk ........ + + * Makefile, /, build_tools/make_sample_voicemail, sounds/Makefile: + Merged revisions 299907 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299907 | pabelanger | 2010-12-29 13:22:23 -0500 (Wed, 29 Dec + 2010) | 2 lines Properly quote varibles for MAC OS X ........ + + * /, apps/app_chanspy.c: Merged revisions 299865 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299865 | pabelanger | 2010-12-28 13:53:37 -0500 + (Tue, 28 Dec 2010) | 9 lines Merged revisions 299864 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, + 28 Dec 2010) | 2 lines Documentation typo ........ + ................ + +2010-12-27 21:23 +0000 [r299754-299824] Tilghman Lesher + + * /, sounds/Makefile: Merged revisions 299820 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299820 | tilghman | 2010-12-27 15:23:10 -0600 (Mon, 27 Dec 2010) + | 2 lines More space-in-pathname issues. ........ + + * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged + revisions 299794 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299794 | tilghman | 2010-12-27 14:41:04 -0600 (Mon, 27 Dec 2010) + | 2 lines Mac OS X spaces-in-pathnames fix. ........ + + * /, configure, configure.ac: Merged revisions 299752 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r299752 | tilghman | 2010-12-26 15:15:58 -0600 (Sun, 26 + Dec 2010) | 2 lines Properly quote path on Darwin. ........ + +2010-12-25 16:35 +0000 [r299715] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c, + addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c: + Change order of sending TCS and MSD packets Change order of + sending Terminal Capability Set and MasterSlave Determination + packets, MSD send when TCS exchange procedure is done (we send + tcs ack to remote and we have remote tcs ack already or we + receive tcs ack from remote and we have send our tcs ack to + remote already). Some endpoints can work in this sequence only, i + suggest they can't work with both (tcs and msd) exchange + procedures simultaneously. Also changed StartH245 facility + message sending. It send on incoming calls only due to some + endpoints can't proccess properly this facility messages on their + incoming calls. (closes issue #18433) Reported by: MrHanMan + Patches: tcs-msd-h245-3.patch uploaded by may213 (license 454) + Tested by: MrHanMan, may213 + +2010-12-25 10:08 +0000 [r299584-299627] Tilghman Lesher + + * channels/chan_local.c, /: Merged revisions 299626 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299626 | tilghman | 2010-12-25 04:07:15 -0600 + (Sat, 25 Dec 2010) | 19 lines Merged revisions 299625 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r299625 | tilghman | 2010-12-25 04:05:00 -0600 + (Sat, 25 Dec 2010) | 12 lines Merged revisions 299624 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) + | 5 lines Move check for extension existence below variable + inheritance, due to the possible use of an eswitch. (closes issue + #16228) Reported by: jlaguilar ........ ................ + ................ + + * /, addons/res_config_mysql.c: Merged revisions 299583 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299583 | tilghman | 2010-12-24 11:58:30 -0600 (Fri, 24 Dec 2010) + | 7 lines Reset 'first' variable after usage. (closes issue + #18525) Reported by: makoto Patches: + res-config-mysql-update2.patch uploaded by makoto (license 38) + ........ + +2010-12-23 01:46 +0000 [r299493] Moises Silva + + * channels/chan_dahdi.c: Enqueue AST_CONTROL_PROGRESS after + AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue + #18438) Reported by: mariner7 Tested by: moy + +2010-12-22 20:10 +0000 [r299450] Tilghman Lesher + + * res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest25, + pbx/ael/ael-test/ref.ael-vtest17, /, + pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 299449 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299449 | tilghman | 2010-12-22 14:05:02 -0600 + (Wed, 22 Dec 2010) | 15 lines Merged revisions 299448 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010) + | 8 lines Resolve warnings by disambiguating the "s" extension as + used by chan_dahdi from the "s" extension as used by the AEL + macros. (closes issue #18480) Reported by: nivek Patches: + 20101215__issue18480__2.diff.txt uploaded by tilghman (license + 14) Tested by: nivek ........ ................ + +2010-12-22 02:12 +0000 [r299406] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 299405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299405 | rmudgett | 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010) + | 17 lines Chan_dahdi sends an empty COLP on the bridged channel. + Chan_dahdi always inserts a connected party IE when you call from + one dahdi channel to another dahdi channel, even if no such + information was received on the 2nd channel. This clears the + display of many phones. * Removed leftover artifact from before + the valid flag was added. * Updated all of the channel's caller + id information with the new connected line information instead of + just the string parts. (closes issue #18508) Reported by: wimpy + Patches: issue18508_trunk.patch uploaded by rmudgett (license + 664) Tested by: wimpy, rmudgett ........ + +2010-12-21 16:02 +0000 [r299355] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 299353 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299353 | mnicholson | 2010-12-21 09:25:03 -0600 + (Tue, 21 Dec 2010) | 30 lines Merged revisions 299242 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r299242 | mnicholson | 2010-12-20 15:25:35 -0600 + (Mon, 20 Dec 2010) | 23 lines Merged revisions + 299194,299198,299220 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec + 2010) | 6 lines Respond as soon as possible with a 202 Accepted + to refer requests. This change also plugs a few memory leaks that + can occur when parking sip calls. ABE-2656 ........ r299198 | + mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 + lines Remove changes to via processing that were not supposed to + go into the last commit. ........ r299220 | mnicholson | + 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use + ast_free() instead of free() ABE-2656 ........ ................ + ................ + +2010-12-21 00:45 +0000 [r299313] Paul Belanger + + * configs/cel.conf.sample, /: Merged revisions 299312 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r299312 | pabelanger | 2010-12-20 19:44:08 -0500 (Mon, + 20 Dec 2010) | 8 lines Correct typo with USER_DEFINED event. + (closes issue #18461) Reported by: joscas Patches: + cel.conf.sample.diff uploaded by lathama (license 1028) Tested + by: lathama, joscas ........ + +2010-12-20 21:40 +0000 [r299249] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 299248 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299248 | mmichelson | 2010-12-20 15:38:30 -0600 (Mon, 20 Dec + 2010) | 20 lines Fix a couple of CCSS issues. * Make sure to + allocate a cc_params structure when creating autopeers. * Use + sip_uri_cmp when retrieving SIP CC agents and monitors in case + parameters appear in the URI. (closes issue #18504) Reported by: + kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches: + 18338.diff uploaded by mmichelson (license 60) Tested by: + GeorgeKonopacki ........ + +2010-12-20 18:18 +0000 [r299142] Tilghman Lesher + + * /, sample.call: Merged revisions 299138 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299138 | tilghman | 2010-12-20 12:17:28 -0600 + (Mon, 20 Dec 2010) | 9 lines Merged revisions 299136 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r299136 | tilghman | 2010-12-20 12:16:37 -0600 (Mon, 20 + Dec 2010) | 2 lines Documentation fix ........ ................ + +2010-12-20 18:03 +0000 [r299135] David Vossel + + * include/asterisk/astobj2.h, main/astobj2.c: New astobj2 flag for + issuing a callback without locking the container. + +2010-12-20 17:59 +0000 [r299133-299134] Russell Bryant + + * channels/chan_misdn.c: Fix chan_misdn build after sched API + changes. + + * addons/chan_ooh323.c, addons/chan_mobile.c: Fix some build errors + in addons due to sched API changes. + +2010-12-20 17:48 +0000 [r299132] Tilghman Lesher + + * /, cdr/cdr_pgsql.c: Merged revisions 299131 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299131 | tilghman | 2010-12-20 11:47:10 -0600 + (Mon, 20 Dec 2010) | 18 lines Merged revisions 299130 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010) + | 11 lines If a call was not answered, then the billsec was + calculated unusually large. Also, due to a copy and paste error, + a request for the answer field would have given the start value, + instead. (closes issue #18460) Reported by: joscas Patches: + 20101215__issue18460.diff.txt uploaded by tilghman (license 14) + Tested by: joscas ........ ................ + +2010-12-20 17:15 +0000 [r299091] Russell Bryant + + * channels/chan_unistim.c, main/udptl.c, res/res_rtp_asterisk.c, + include/asterisk.h, main/rtp_engine.c, main/dnsmgr.c, + channels/chan_sip.c, main/ccss.c, include/asterisk/channel.h, + channels/chan_gtalk.c, tests/test_sched.c, channels/chan_iax2.c, + res/res_rtp_multicast.c, main/channel.c, main/cdr.c, + channels/chan_jingle.c, channels/chan_skinny.c, + channels/sip/include/globals.h, res/res_stun_monitor.c, + channels/sip/dialplan_functions.c, channels/chan_h323.c, + include/asterisk/sched.h, pbx/pbx_dundi.c, + include/asterisk/udptl.h, include/asterisk/rtp_engine.h, + main/sched.c, channels/chan_mgcp.c, res/res_calendar.c: Some + scheduler API cleanup and improvements. Previously, I had added + the ast_sched_thread stuff that was a generic scheduler thread + implementation. However, if you used it, it required using + different functions for modifying scheduler contents. This patch + reworks how this is done and just allows you to optionally start + a thread on the original scheduler context structure that has + always been there. This makes it trivial to switch to the generic + scheduler thread implementation without having to touch any of + the other code that adds or removes scheduler entries. In + passing, I made some naming tweaks to add ast_ prefixes where + they were not there before. Review: + https://reviewboard.asterisk.org/r/1007/ + +2010-12-20 16:19 +0000 [r299089] Leif Madsen + + * /, main/features.c: Merged revisions 299088 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299088 | lmadsen | 2010-12-20 10:18:26 -0600 + (Mon, 20 Dec 2010) | 13 lines Merged revisions 299087 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010) + | 5 lines Note that Park() timeout is milliseconds. (closes issue + #15758) Reported by: mmurdock Tested by: mmurdock, seanbright + ........ ................ + +2010-12-20 09:14 +0000 [r299005] Tzafrir Cohen + + * channels/sig_pri.h, channels/chan_sip.c, main/aoc.c: Typos: + recieved => received + +2010-12-18 00:08 +0000 [r298819-298961] Tilghman Lesher + + * utils/refcounter.c, include/asterisk/utils.h, + build_tools/cflags-devmode.xml, /, main/Makefile, + include/asterisk/autoconfig.h.in, configure.ac, utils/hashtest.c, + main/utils.c, main/astobj2.c, utils/conf2ael.c, + include/asterisk/logger.h, configure, + build_tools/menuselect-deps.in, main/logger.c, utils/hashtest2.c, + utils/ael_main.c, makeopts.in, utils/check_expr.c: Merged + revisions 298960 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298960 | tilghman | 2010-12-17 17:52:04 -0600 + (Fri, 17 Dec 2010) | 20 lines Merged revisions 298957 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600 + (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) + | 6 lines Let Asterisk find better backtrace information with + libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will + use libbfd to search for better symbol information within both + the Asterisk binary, as well as loaded modules, to assist when + using inline backtraces to track down problems. Review: + https://reviewboard.asterisk.org/r/1055/ ........ + ................ ................ + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 298827 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r298827 | tilghman | 2010-12-17 15:18:18 -0600 (Fri, 17 Dec 2010) + | 8 lines -v implies -f, so override with -F. (closes issue + #18446) Reported by: lathama Patches: rc.debian.asterisk.diff + uploaded by lathama (license 1028) Tested by: lathama ........ + + * /, configure, configure.ac: Merged revisions 298818 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298818 | tilghman | 2010-12-17 15:04:21 -0600 + (Fri, 17 Dec 2010) | 15 lines Merged revisions 298817 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r298817 | tilghman | 2010-12-17 15:03:06 -0600 (Fri, 17 Dec 2010) + | 8 lines Also include PTHREAD_LIBS and PTHREAD_CFLAGS for SQLite + 3, as it's needed on some platforms. (closes issue #18493) + Reported by: pprindeville Patches: asterisk-1.8-sqlite3.patch + uploaded by pprindeville (license 347) Tested by: pprindeville + ........ ................ + +2010-12-17 17:29 +0000 [r298774] Brad Watkins + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 298773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r298773 | marquis | 2010-12-17 12:26:31 -0500 (Fri, 17 Dec 2010) + | 10 lines Fix parsing of mwi => lines in sip.conf Reworking + parsing of mwi => lines to resolve a segfault. Also add a set of + unit tests for the function that does the parsing. (closes issue + #18350) Reported by: gbour Tested by: Marquis, gbour Review: + https://reviewboard.asterisk.org/r/1053/ ........ + +2010-12-16 23:33 +0000 [r298599-298686] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 298685 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298685 | jpeeler | 2010-12-16 17:31:50 -0600 + (Thu, 16 Dec 2010) | 16 lines Merged revisions 298684 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298684 | jpeeler | 2010-12-16 17:30:59 -0600 + (Thu, 16 Dec 2010) | 9 lines Merged revisions 298683 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 + Dec 2010) | 2 lines After recording only silence for a voicemail + prepending, restore backup files. ........ ................ + ................ + + * /, apps/app_queue.c: Merged revisions 298598 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298598 | jpeeler | 2010-12-16 14:51:44 -0600 + (Thu, 16 Dec 2010) | 21 lines Merged revisions 298597 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298597 | jpeeler | 2010-12-16 14:49:33 -0600 + (Thu, 16 Dec 2010) | 14 lines Merged revisions 298596 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) + | 7 lines Fix improper hangup when doing an attended transfer to + queue. Had to indicate ringing in wait_for_answer so the attended + transfer code would not try and hang up the local channel it + created, which would kill the call. ABE-2624 ........ + ................ ................ + +2010-12-16 09:29 +0000 [r298441-298545] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 298539 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r298539 | tilghman | 2010-12-16 03:28:17 -0600 (Thu, 16 Dec 2010) + | 8 lines Ensure the ipaddr field in realtime is large enough to + handle IPv6 addresses. (closes issue #18464) Reported by: IgorG + Patches: realtime_ipv6store.diff uploaded by IgorG (license 20) + (plus a few additional lines by tilghman) ........ + + * res/res_config_odbc.c, /: Merged revisions 298482 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298482 | tilghman | 2010-12-16 03:05:28 -0600 + (Thu, 16 Dec 2010) | 28 lines Merged revisions 298481 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298481 | tilghman | 2010-12-16 03:04:38 -0600 + (Thu, 16 Dec 2010) | 21 lines Merged revisions 298480 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010) + | 14 lines Only increment the pointer once per loop, otherwise we + corrupt the value. (closes issue #18251) Reported by: bcnit + Patches: 20101110__issue18251.diff.txt uploaded by tilghman + (license 14) Tested by: trev, jthurman, elguero (closes issue + #18279) Reported by: zerohalo Patches: + 20101109__issue18279.diff.txt uploaded by tilghman (license 14) + Tested by: zerohalo ........ ................ ................ + + * /, funcs/func_dialgroup.c: Merged revisions 298478 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298478 | tilghman | 2010-12-16 02:56:13 -0600 + (Thu, 16 Dec 2010) | 15 lines Merged revisions 298477 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16 Dec 2010) + | 8 lines Eliminate duplicates from container. (closes issue + #18091) Reported by: bunny Patches: 20101006__issue18091.diff.txt + uploaded by tilghman (license 14) Tested by: bunny ........ + ................ + + * /, cdr/cdr_sqlite.c: Merged revisions 298394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298394 | tilghman | 2010-12-15 18:30:04 -0600 + (Wed, 15 Dec 2010) | 22 lines Merged revisions 298393 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298393 | tilghman | 2010-12-15 18:29:10 -0600 + (Wed, 15 Dec 2010) | 15 lines Merged revisions 298392 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010) + | 8 lines Unregister before shutting down the connection, to + avoid a race. (closes issue #18481) Reported by: pabelanger + Patches: 20101215__issue18481.diff.txt uploaded by tilghman + (license 14) Tested by: pabelanger ........ ................ + ................ + +2010-12-13 22:10 +0000 [r298201-298288] Richard Mudgett + + * channels/sig_pri.c: Post AMI hold events on PRI spans when the + remote party HOLD/RETRIEVEs the call. Part of JIRA + SWP-2687/ABE-2691. + + * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions + 298195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298195 | rmudgett | 2010-12-13 11:11:43 -0600 + (Mon, 13 Dec 2010) | 33 lines Merged revisions 298194 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298194 | rmudgett | 2010-12-13 11:04:41 -0600 + (Mon, 13 Dec 2010) | 26 lines Merged revisions 298193 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) + | 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered + transfers. Outgoing PRI/BRI calls cannot do DTMF triggered + transfers if a PROCEEDING message is not received. The debug + output shows that the DTMF begin event is seen, but the DTMF end + event is missing. When the DTMF begin happens, the call is muted + so we now have one way audio (until a DTMF end event is somehow + seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER + event is received. * Made absorb the DTMF begin and DTMF end + events if we are overlap dialing and have not seen a PROCEEDING + message. * Added a debug message when absorbing a DTMF event. + JIRA SWP-2690 JIRA ABE-2697 ........ ................ + ................ + +2010-12-12 03:58 +0000 [r298137] Jeff Peeler + + * include/asterisk/utils.h, configure, + include/asterisk/autoconfig.h.in, configure.ac, main/logger.c, + main/utils.c, main/asterisk.c: Add support for several platforms + to obtain the real thread ID. Already had the pthread ID which is + not the same. The most obvious enhancement is in the "core show + threads" output. As stated in the utils header, if the platform + isn't supported -1 is reported (instead of the process ID + previously). + +2010-12-11 21:47 +0000 [r298100] Alexandr Anikin + + * addons/ooh323c/src/ooGkClient.c: Correction to work with + gatekeeper which don't send GK ID Don't use GK ID if it's not + presented in GK replies Extract GK ID not only in GK confirm but + in GK register confirm also (closes issue #18401) Reported by: + MrHanMan Patches: no-gkid-2.patch uploaded by may213 (license + 454) Tested by: may213, MrHanMan + +2010-12-10 16:53 +0000 [r298055] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 298054 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r298054 | mnicholson | 2010-12-10 10:52:11 -0600 (Fri, 10 Dec + 2010) | 2 lines Prevent a memcpy overlap in + GENERIC_FAX_EXEC_SET_VARS ........ + +2010-12-10 16:28 +0000 [r298052] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + main/netsock.c: Merged revisions 298051 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298051 | tilghman | 2010-12-10 10:26:46 -0600 + (Fri, 10 Dec 2010) | 18 lines Merged revisions 298050 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010) + | 11 lines Portability issue on OpenSolaris. Also detect the + required structure element, because OpenSolaris defines + SIOCGIFHWADDR, but without support for IP sockets. (closes issue + #18442) Reported by: ranjtech Patches: + 20101209__issue18442.diff.txt uploaded by tilghman (license 14) + Tested by: ranjtech ........ ................ + +2010-12-09 22:19 +0000 [r297972] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 297965 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297965 | twilson | 2010-12-09 16:18:19 -0600 + (Thu, 09 Dec 2010) | 28 lines Merged revisions 297960 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297960 | twilson | 2010-12-09 16:10:31 -0600 + (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) + | 14 lines Ignore spurious REGISTER requests If a REGISTER + request with a Call-ID matching an existing transaction is + received it was possible that the REGISTER request would + overwrite the initreq of the private structure. This info is used + to generate messages for other responses in the transaction. This + patch ignores REGISTER requests that match non-REGISTER + transactions. (closes issue #18051) Reported by: eeman Tested by: + twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ + ................ ................ + +2010-12-09 21:33 +0000 [r297958] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 297957 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r297957 | dvossel | 2010-12-09 15:32:20 -0600 (Thu, 09 + Dec 2010) | 11 lines Fixes issue with outbound google voice calls + not working. Thanks to az1234 and nevermind_quack for their input + in helping debug the issue. (closes issue #18412) Reported by: + nevermind_quack Patches: fix uploaded by dvossel (license 671) + ........ + +2010-12-09 21:26 +0000 [r297956] Terry Wilson + + * /, main/features.c: Merged revisions 297952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r297952 | twilson | 2010-12-09 14:48:44 -0600 (Thu, 09 Dec 2010) + | 10 lines Don't crash after Set(CDR(userfield)=...) in + ast_bridge_call Instead of setting peer->cdr = NULL, set it to + not post. (closes issue #18415) Reported by: macbrody Patches: + patch-18415 uploaded by jsolares (license 1167) Tested by: + jsolares, twilson ........ + +2010-12-08 18:08 +0000 [r297910] Tilghman Lesher + + * /, configs/extensions.conf.sample: Merged revisions 297909 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297909 | tilghman | 2010-12-08 12:06:04 -0600 + (Wed, 08 Dec 2010) | 11 lines Merged revisions 297908 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r297908 | tilghman | 2010-12-08 12:04:38 -0600 (Wed, 08 Dec 2010) + | 4 lines Use inheritance to get correct results for + SIPFROMDOMAIN. (from an internal Digium discussion) ........ + ................ + +2010-12-07 23:00 +0000 [r297826] Jeff Peeler + + * main/channel.c, /: Merged revisions 297825 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297825 | jpeeler | 2010-12-07 16:59:30 -0600 + (Tue, 07 Dec 2010) | 26 lines Merged revisions 297824 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600 + (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) + | 12 lines Revert code that changed SSRC for DTMF. Some previous + behavior was attempted to be restored, but mistakingly I did not + realize that the previous behavior was incorrect. This fixes DTMF + not being detected since DTMF shouldn't cause the SSRC to change. + (related to issue #17404) (closes issue #18189) (closes issue + #18352) Reported by: marcbou Tested by: cmbaker82 ........ + ................ ................ + +2010-12-07 22:54 +0000 [r297734-297822] Tilghman Lesher + + * Makefile, contrib/init.d/org.asterisk.asterisk.plist, + utils/muted.c, /, contrib/init.d/org.asterisk.muted.plist + (added): Merged revisions 297821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297821 | tilghman | 2010-12-07 16:51:05 -0600 + (Tue, 07 Dec 2010) | 18 lines Merged revisions 297819 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297819 | tilghman | 2010-12-07 16:40:45 -0600 + (Tue, 07 Dec 2010) | 11 lines Merged revisions 297818 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010) + | 4 lines Use non-deprecated APIs for CoreAudio Review: + https://reviewboard.asterisk.org/r/1040/ ........ + ................ ................ + + * /, apps/app_followme.c: Merged revisions 297733 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297733 | tilghman | 2010-12-06 18:29:26 -0600 + (Mon, 06 Dec 2010) | 22 lines Merged revisions 297713 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600 + (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) + | 8 lines Don't create a Local channel if the target extension + does not exist. (closes issue #18126) Reported by: junky Patches: + followme.diff uploaded by junky (license 177) (partially + restructured by me to avoid a possible memory leak) ........ + ................ ................ + +2010-12-06 22:10 +0000 [r297608] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 297607 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297607 | jpeeler | 2010-12-06 16:06:37 -0600 + (Mon, 06 Dec 2010) | 25 lines Merged revisions 297605 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600 + (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) + | 12 lines Improve handling of REGISTER requests with multiple + contact headers. The changes here attempt to more strictly follow + RFC 3261 section 10.3. Basically the following will now cause a + 400 Bad Response to be returned, if: - multiple Contact headers + are present with one set to expire all bindings ("*") - wildcard + parameter is specified for Contact without Expires header or + Expires header is not set to zero. ABE-2442 ABE-2443 ........ + ................ ................ + +2010-12-03 17:42 +0000 [r297536] Sean Bright + + * /, channels/chan_console.c: Merged revisions 297535 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297535 | seanbright | 2010-12-03 12:41:30 -0500 + (Fri, 03 Dec 2010) | 9 lines Merged revisions 297534 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri, + 03 Dec 2010) | 3 lines The CLI command should not contain + s, these are for descriptions. ........ + ................ + +2010-12-03 15:32 +0000 [r297496] Matthew Nicholson + + * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions + 297157,297486,297495 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r297157 | mnicholson | 2010-12-01 13:47:33 -0600 (Wed, 01 Dec + 2010) | 2 lines Changed some NOTICE and WARNING messages to DEBUG + messages. ........ r297486 | mnicholson | 2010-12-02 15:30:47 + -0600 (Thu, 02 Dec 2010) | 6 lines Add support for reserving a + fax session before answering the channel. Note: this change + breaks ABI compatibility. FAX-217 ........ r297495 | mnicholson | + 2010-12-03 09:21:52 -0600 (Fri, 03 Dec 2010) | 4 lines Print a + DEBUG message instead of a WARNING message when the selected fax + tech does not support reserving sessions. Answer the channel + before quering it for t.38 support. This is necessary for the + query to work properly over local channels. ........ + +2010-12-02 20:11 +0000 [r297407] Paul Belanger + + * Makefile, /: Merged revisions 297406 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297406 | pabelanger | 2010-12-02 15:09:29 -0500 + (Thu, 02 Dec 2010) | 21 lines Merged revisions 297405 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297405 | pabelanger | 2010-12-02 15:06:43 -0500 + (Thu, 02 Dec 2010) | 14 lines Merged revisions 297404 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec + 2010) | 7 lines Resolve compile error under FreeBSD We now set + _ASTCFLAGS+=-march=i686 for i386 processors, still allowing + ASTCFLAGS to override the setting. Review: + https://reviewboard.asterisk.org/r/1043/ ........ + ................ ................ + +2010-12-02 18:28 +0000 [r297356] Terry Wilson + + * /, main/abstract_jb.c: Merged revisions 297312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297312 | twilson | 2010-12-02 12:13:49 -0600 + (Thu, 02 Dec 2010) | 28 lines Merged revisions 297311 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297311 | twilson | 2010-12-02 12:07:39 -0600 + (Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010) + | 12 lines Initialize offset for adaptive jitter buffer When the + adaptive jitter buffer is enabled in sip.conf, the first frame + placed in the jitter buffer fails with something like: + jb_warning_output: Resyncing the jb. last_delay 0, this delay + -215886466, threshold 1000, new offset 215886466 This happens + because the offset is not initialized before calling jb_put(). + This patch modifies jb_put_first_adaptive() to set the offset to + the frame's timestamp. Review: + https://reviewboard.asterisk.org/r/1041/ ........ + ................ ................ + +2010-12-02 13:20 +0000 [r297248] Russell Bryant + + * /, apps/app_meetme.c: Merged revisions 297245 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297245 | russell | 2010-12-02 07:20:19 -0600 + (Thu, 02 Dec 2010) | 20 lines Merged revisions 297229 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297229 | russell | 2010-12-02 07:16:47 -0600 + (Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) + | 6 lines Add "DAHDI" to a couple of app_meetme error messages. + This is in response to some questions on IRC. To the user, there + was nothing that made it obvious that this error had anything to + do with DAHDI not being loaded. ........ ................ + ................ + +2010-12-01 17:53 +0000 [r297076] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 297075 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297075 | jpeeler | 2010-12-01 11:53:13 -0600 + (Wed, 01 Dec 2010) | 37 lines Merged revisions 297073 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600 + (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) + | 23 lines Fix not stopping MOH when transfered local channel + queue member is answered. The problem here is only present when + local channels are used with the MOH passthru option as well as + no optimization (/nm). I will describe the slightly bizarre + scenario that was used to test, where phones B and C are queue + members: Phone A dials into a queue with two members using local + channels and the above options. Phone B answers. Phone A blind + transfers phone B into the same queue. Phone A hangs up. Phone C + answers, but phone B didn't stop playing MOH. In this scenario, + the unhold frame that should have gotten to phone B never arrived + due to the masquerade from the blind transfer. This is usually + fine since app_queue manages the starting and stopping of MOH. + However, with the passthrough option enabled when app_queue + attempts to stop MOH it tries to do so on the local channel + rather than the real channel. The easiest solution was to just + make sure to send an unhold frame during the transfer since it + wouldn't make sense to have MOH playing after a transfer anyway. + This only modifies SIP transfers, but the other transfers did not + seem to be a problem. If DTMF based transfers were a problem it + might be okay to add ast_moh_stop to finishup, but I didn't want + to have to add that unless required. ABE-2624 ........ + ................ ................ + +2010-12-01 17:03 +0000 [r296952-296993] Tilghman Lesher + + * /, include/asterisk/frame.h: Merged revisions 296992 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296992 | tilghman | 2010-12-01 11:01:56 -0600 + (Wed, 01 Dec 2010) | 19 lines Merged revisions 296991 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296991 | tilghman | 2010-12-01 11:01:00 -0600 + (Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010) + | 5 lines Clarify documentation on how we store codec preference + lists. (closes issue #18397) Reported by: birgita ........ + ................ ................ + + * /, channels/chan_iax2.c: Merged revisions 296951 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296951 | tilghman | 2010-11-30 19:46:32 -0600 + (Tue, 30 Nov 2010) | 9 lines Merged revisions 296950 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30 + Nov 2010) | 2 lines Missed initializations caused startup errors + on Mac OS X (and possibly others, too). ........ ................ + +2010-12-01 00:28 +0000 [r296871] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 296870 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296870 | jpeeler | 2010-11-30 18:28:16 -0600 + (Tue, 30 Nov 2010) | 18 lines Merged revisions 296869 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600 + (Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) + | 4 lines Properly restore backup information file when hanging + up during message prepending. ABE-2654 ........ ................ + ................ + +2010-11-30 22:32 +0000 [r296788-296826] Tilghman Lesher + + * include/asterisk/frame.h: Add a comment on why the reserved bit + is reserved. Came up when reviewing discussion on the CODEC PREFS + IE in IAX2. + + * /, apps/app_meetme.c: Merged revisions 296787 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296787 | tilghman | 2010-11-30 13:12:48 -0600 (Tue, 30 Nov 2010) + | 2 lines DOC: Conference number can be omitted; if omitted, all + users in a meetme are listed. ........ + +2010-11-30 09:49 +0000 [r296752] Stefan Schmidt + + * include/asterisk.h, main/pbx.c, main/asterisk.c: move devices + from hints into an ao2_container by splitting up devices from + hints into an own ao2_container the callback to get these devices + for statechange handling is faster. with this changes the length + of a device used in a hint isnt longer restricted to 80 + characters. Tests showed that calling handle_statechange is 40 + times faster if no hints are used and 25 times faster if there + are any hints. (closes issue #17928) Reported by: mdu113 Tested + by: schmidts Review: https://reviewboard.asterisk.org/r/1003/ + +2010-11-29 23:07 +0000 [r296674] Paul Belanger + + * /, channels/chan_iax2.c: Merged revisions 296673 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296673 | pabelanger | 2010-11-29 18:05:45 -0500 + (Mon, 29 Nov 2010) | 19 lines Merged revisions 296671 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500 + (Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov + 2010) | 5 lines Make sure nothing else is needed before + destroying the scheduler. (closes issue #18398) Reported by: + pabelanger ........ ................ ................ + +2010-11-29 21:31 +0000 [r296630] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 296628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296628 | russell | 2010-11-29 15:26:44 -0600 (Mon, 29 Nov 2010) + | 6 lines Complete some error handling in transmit_publish() in + chan_sip.c. This error handling block caught my eye. It was + missing a couple of things, but it should be safe now. Thanks to + mmichelson for the quick peer review on IRC. ........ + +2010-11-29 20:54 +0000 [r296585] Richard Mudgett + + * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 296582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296582 | rmudgett | 2010-11-29 14:46:03 -0600 + (Mon, 29 Nov 2010) | 24 lines Merged revision 296575 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon, + 29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling + as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY + redirecting number and notification code, SETUP redirecting + number) is also sent in PTMP/TE mode. It should only apply in + PTMP/NT mode. The call setup proceeds but the network (Deutsche + Telekom) reacts with ugly ISDN STATUS messages. Also don't send + the redirecting number ie when PTP is also sending the + DivertingLegInformation2 facility. The redirecting number ie is + redundant and the network (Deutsche Telekom) complains about it. + Patches: abe_2651_v4.patch uploaded by rmudgett (license 664) + JIRA ABE-2651 JIRA SWP-2537 .......... ................ + +2010-11-29 07:30 +0000 [r296535] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + main/asterisk.c: Merged revisions 296534 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296534 | tilghman | 2010-11-29 01:28:44 -0600 + (Mon, 29 Nov 2010) | 20 lines Merged revisions 296533 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010) + | 13 lines I love standards. There are so many to choose from. + Except when there isn't one. Linux and *BSD disagree on the + elements within the ucred structure. Detect which one is in use + on the system. (closes issue #18384) Reported by: bjm Patches: + cred-diffs uploaded by bjm (license 473) + 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman + (license 14) 20101127__issue18384__1.8.diff.txt uploaded by + tilghman (license 14) Tested by: tilghman, bjm ........ + ................ + +2010-11-27 10:41 +0000 [r296430-296468] Tilghman Lesher + + * /, apps/app_meetme.c: Merged revisions 296467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296467 | tilghman | 2010-11-27 04:40:22 -0600 + (Sat, 27 Nov 2010) | 12 lines Merged revisions 296466 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) + | 5 lines 18 characters is too short for most date/times (20 is + the usual, but we add more in case of greater precision). (closes + issue #18369) Reported by: tnakonz ........ ................ + + * include/asterisk.h, /: Merged revisions 296429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296429 | tilghman | 2010-11-27 03:58:57 -0600 (Sat, 27 Nov 2010) + | 5 lines Also don't build DEBUG_FD_LEAKS when STANDALONE2 is + defined. (closes issue #18385) Reported by: cmaj ........ + +2010-11-26 22:02 +0000 [r296393] Olle Johansson + + * /, main/say.c: Merged revisions 296391 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296391 | oej | 2010-11-26 22:37:21 +0100 (Fre, + 26 Nov 2010) | 24 lines Merged revisions 296351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre, + 26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11 + lines Fix bugs in saying numbers using the Swedish language + syntax (closes issue #18355) Reported by: oej Patch by: oej Much + help from Peter Lindahl. Testing by the ClearIT team during a + coffee break. Review: https://reviewboard.asterisk.org/r/1033/ + ........ ................ ................ + +2010-11-26 18:31 +0000 [r296353-296355] Brad Watkins + + * /, res/res_jabber.c: Merged revisions 296354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296354 | marquis | 2010-11-26 13:31:17 -0500 (Fri, 26 Nov 2010) + | 12 lines Fix XMPP PubSub-based distributed device state. + Initialize pubsubflags to 0 so res_jabber doesn't think there is + already an XMPP connection sending device state. Also clean up + CLI commands a bit. (closes issue #18272) Reported by: klaus3000 + Patches: res_jabber_fix_pubsubflags_and_CLI-patch.txt uploaded by + klaus3000 (license 65) Tested by: klaus3000, Marquis Review: + https://reviewboard.asterisk.org/r/1030/ ........ + + * /, channels/chan_sip.c: Merged revisions 296352 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296352 | marquis | 2010-11-26 13:19:02 -0500 (Fri, 26 Nov 2010) + | 12 lines Fix reloading of peer when a user is requested. + Prevent peer reloading from causing multiple MWI subscriptions to + be created when using realtime. This had the effect of sending + one NOTIFY for every time a sip peer made a call, in one case + eventually overwhelming the phone and causing it to reboot. + (closes issue #18342) Reported by: nivek Patches: + issue0018342p1.patch uploaded by nivek (license 636) Tested by: + nivek Review: https://reviewboard.asterisk.org/r/1029/ ........ + +2010-11-24 23:46 +0000 [r296249] Andrew Parisio + + * apps/app_meetme.c, CHANGES: Meetme use voicemail greet for + join/leave announce Added option v(mailbox@[context]) which tells + MeetMe where to look for a users greet file. If one does not + exist it clears the v option and defers to the functionality of + i/I as/if set by the MeetMe() command. Review: + https://reviewboard.asterisk.org/r/1009/ (closes issue #18297) + Reported by: parisioa Patches: meetme_final_patch_v.diff uploaded + by parisioa (license 1153) + +2010-11-24 23:30 +0000 [r296235] Russell Bryant + + * main/channel.c, /: Merged revisions 296230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296230 | russell | 2010-11-24 17:29:44 -0600 + (Wed, 24 Nov 2010) | 20 lines Merged revisions 296221 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296221 | russell | 2010-11-24 17:28:19 -0600 + (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) + | 6 lines Make Asterisk less crashy. Since we might not put a new + translation path on the channel, go ahead and set it to NULL + right after destroying the old one to ensure we don't try to free + an invalid translation path later on. ........ ................ + ................ + +2010-11-24 22:52 +0000 [r296168] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + /, channels/sig_analog.h: Merged revisions 296167 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296167 | rmudgett | 2010-11-24 16:49:48 -0600 + (Wed, 24 Nov 2010) | 57 lines Merged revisions 296166 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600 + (Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) + | 43 lines Oneway audio to SIP phone from FXS port after FXS port + gets a CallWaiting pip. The FXS connected phone has to have + CW/CID support to fail, as it will send back a DTMF 'A' or 'D' + when it's ready to receive CallerID. A normal phone with no CID + never fails. Also the SIP phone does not hear MOH when the CW + call is answered. The DTMF end frame is suppressed when the phone + acknowledges the CW signal for CID. The problem is the DTMF begin + frame needs to be suppressed as well. The DTMF begin frame is + causing SIP to start sending the DTMF RTP frames. Since the DTMF + end frame is suppressed, SIP will not stop sending those DTMF RTP + packets. * Suppress the DTMF begin and end frames when the + channel driver is looking for DTMF digits. * Fixed a couple + issues caused by not cleaning up the CID spill if you answer the + CW call while it is sending the CID spill. * Fixed not sending + CW/CID spill to the phone when the call is natively bridged. + (Fixed by not using native bridge if CW/CID is possible.) * + Suppress received audio when sending CW/CID spills. The other + parties involved do not need to hear the CW/CID spills and may be + confused if the CW call is for them. (closes issue #18129) + Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch + uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett + NOTE: * v1.4 does not have the main problem fixed by suppressing + the DTMF start frames. The other three items fixed are relevant. + * If you really must restore native bridging between analog + ports, you need to disable CW/CID either by configuring + chan_dahdi.conf callwaitingcallerid=no or dialing *70 before + dialing the number to temporarily disable CW. ........ + ................ ................ + +2010-11-24 20:24 +0000 [r296034-296085] Russell Bryant + + * main/channel.c, /: Merged revisions 296084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296084 | russell | 2010-11-24 14:23:46 -0600 + (Wed, 24 Nov 2010) | 26 lines Merged revisions 296083 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296083 | russell | 2010-11-24 14:23:11 -0600 + (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) + | 12 lines Fix false reporting of an error by set_format(). In + the case that the native format was able to be changed to match + the new requested format, the code proceeded to attempt to build + a translation path, anyway. The result would be NULL, since no + translation path is necessary and resulted in this function + thinking an error has occurred. This case is now specifically + caught and no attempt to build a translation path is attempted. + Thanks to our automated tests and bamboo.asterisk.org for + catching this problem and making a whole lot of noise when things + started failing. :-) ........ ................ ................ + + * apps/app_dial.c, main/channel.c, /: Merged revisions 296002 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296002 | russell | 2010-11-24 11:13:08 -0600 + (Wed, 24 Nov 2010) | 52 lines Merged revisions 296001 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296001 | russell | 2010-11-24 11:03:16 -0600 + (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) + | 38 lines Handle failures building translation paths more + effectively. The problem scenario occurred on a heavily loaded + system that was using the codec_dahdi module and exceeded the + hardware transcoding capacity. The failure mode at that point was + not good. The report came in to us as an Asterisk lock-up. The + "core show locks" shows a ton of threads locked up (but no + obvious deadlock). Upon deeper investigation, when the system is + in this state, the CPU was maxed out. The CPU was being consumed + by the Asterisk logger spewing messages on every audio frame for + calls set up after transcoder capacity was reached. The purpose + of this patch is to make Asterisk handle failures to create a + translation path in a more graceful manner. If we can't + translate, then the call just needs to be dropped, as it's not + going to work. These are the changes: 1) In set_format() of + channel.c (which is called by set_read_format() and + set_write_format()), it was ignoring if + ast_translator_build_path() failed and returned NULL. It now pays + attention to that case and returns a result reflecting failure. + With this change in place, the bridging code will immediately + detect a failure and end the bridge instead of proceeding to try + to bridge frames that can't be translated and making channel + drivers freak out by sending them frames in a format they weren't + expecting. 2) In ast_indicate_data() of channel.c, failure of + ast_playtones_start() was ignored. It is now reflected in the + return value of the function. This didn't turn out to have any + affect on the bug, but seemed like a good change to leave in. 3) + In app_dial(), when only sending a call to a single endpoint, it + will attempt to do some bridging of its own of early audio. It + uses make_compatible() when it's going to do this. However, it + ignored failure from make compatible. So, even with the fix from + #1, if there was early audio going through app_dial, there would + still be a period of invalid frames passing through. After + detecting failure here, Dial() exits. ABE-2658 ........ + ................ ................ + +2010-11-23 10:34 +0000 [r295950] Olle Johansson + + * /, main/say.c: Merged revisions 295949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295949 | oej | 2010-11-23 11:30:05 +0100 (Tis, + 23 Nov 2010) | 21 lines Merged revisions 295907 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis, + 23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8 + lines Fix support of saynumber(1,n) in the Swedish language + (closes issue #18353) Reported by: oej Review: + https://reviewboard.asterisk.org/r/1031/ ........ + ................ ................ + +2010-11-22 20:05 +0000 [r295870] Sean Bright + + * configs/chan_dahdi.conf.sample, /: Merged revisions 295869 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295869 | seanbright | 2010-11-22 15:03:49 -0500 + (Mon, 22 Nov 2010) | 9 lines Merged revisions 295868 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, + 22 Nov 2010) | 2 lines Change some documentation to suggest + dahdi_monitor instead of ztmonitor. ........ ................ + +2010-11-22 19:42 +0000 [r295867] Richard Mudgett + + * main/channel.c, main/pbx.c, /, apps/app_macro.c, + include/asterisk/channel.h, include/asterisk/frame.h: Merged + revisions 295866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295866 | rmudgett | 2010-11-22 13:36:10 -0600 + (Mon, 22 Nov 2010) | 60 lines Merged revisions 295843 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600 + (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) + | 46 lines The channel redirect function (CLI or AMI) hangs up + the call instead of redirecting the call. To recreate the + problem: 1) Party A calls Party B 2) Invoke CLI "channel + redirect" command to redirect channel call leg associated with A. + 3) All associated channels are hung up. Note that if the CLI + command were done on the channel call leg associated with B it + works. This regression was a result of the fix for issue #16946 + (https://reviewboard.asterisk.org/r/740/). The regression affects + all features that use an async goto to execute the dialplan + because of an external event: Channel redirect, AMI redirect, SIP + REFER, and FAX detection. The struct ast_channel._softhangup code + is a mess. The variable is used for several purposes that do not + necessarily result in the call being hung up. I have added + doxygen comments to describe how the various _softhangup bits are + used. I have corrected all the places where the variable was + tested in a non-bit oriented manner. The primary fix is the new + AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so + the soft hangup requests that do not normally result in a hangup + do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) + Reported by: SantaFox (closes issue #18185) Reported by: + kwemheuer (closes issue #18211) Reported by: zahir_koradia + (closes issue #18230) Reported by: vmarrone (closes issue #18299) + Reported by: mbrevda (closes issue #18322) Reported by: nerbos + Review: https://reviewboard.asterisk.org/r/1013/ ........ + ................ ................ + +2010-11-22 18:43 +0000 [r295789] Erin Spiceland + + * res/res_agi.c: Revert to the previous behavior of AGI command + WAIT FOR DIGIT, since the behavior of the command with this patch + is almost exactly like that of GET DATA. + +2010-11-20 03:13 +0000 [r295748] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 295747 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295747 | rmudgett | 2010-11-19 21:11:15 -0600 (Fri, 19 Nov 2010) + | 13 lines One way audio before answering call waiting call on + analog port. * Analog call waiting Caller ID spills could get + stuck resulting in one way audio until the waiting call is + answered. This only happens on the second (and later) call + waiting call if the active call is not the first call. * The + CLI/AMI "dahdi show channel" command could report the wrong + channel information. Must keep the struct analog_pvt.owner and + struct dahdi_pvt.owner pointer in sync. ........ + +2010-11-20 00:52 +0000 [r295712] Russell Bryant + + * include/asterisk/event.h, /, main/event.c: Merged revisions + 295711 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295711 | russell | 2010-11-19 18:50:00 -0600 + (Fri, 19 Nov 2010) | 36 lines Merged revisions 295710 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) + | 29 lines Fix cache of device state changes for multiple + servers. This patch addresses a regression where device states + across multiple servers were not being processing completely + correctly. The code works to determine the overall state by + looking at the last known state of a device on each server. + However, there was a regression due to some invasive rewrites of + how the cache works that led to the cache only storing the last + device state change for a device, regardless of which server it + was on. The code is set up to cache device state change events by + ensuring that each event in the cache has a unique device name + + entity ID (server ID). The code that was responsible for + comparing raw information elements (which EID is) always returned + a match due to a memcmp() with a length of 0. There isn't much + code to fix the actual bug. This patch also introduces a new CLI + command that was very useful for debugging this problem. The + command allows you to dump the contents of the event cache. + (closes issue #18284) Reported by: klaus3000 Patches: + issue18284.rev1.txt uploaded by russell (license 2) Tested by: + russell, klaus3000 (closes issue #18280) Reported by: klaus3000 + Review: https://reviewboard.asterisk.org/r/1012/ ........ + ................ + +2010-11-19 22:15 +0000 [r295674] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 295673 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295673 | twilson | 2010-11-19 14:06:10 -0800 + (Fri, 19 Nov 2010) | 22 lines Merged revisions 295672 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295672 | twilson | 2010-11-19 13:55:48 -0800 + (Fri, 19 Nov 2010) | 15 lines Merged revisions 295628 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) + | 8 lines Discard responses with more than one Via This is not a + perfect solution as headers that are joined via commas are not + detected. This is a parsing issue that to fix "correctly" would + necessitate a new SIP parser. Review: + https://reviewboard.asterisk.org/r/1019/ ........ + ................ ................ + +2010-11-19 21:42 +0000 [r295671] Brett Bryant + + * /, apps/app_queue.c: Merged revisions 295670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295670 | bbryant | 2010-11-19 16:40:21 -0500 (Fri, 19 Nov 2010) + | 8 lines Patch for deadlock from ordering issue between + channel/queue locks in app_queue (set_queue_variables). (closes + issue #18031) Reported by: rain Review: + https://reviewboard.asterisk.org/r/1018/ ........ + +2010-11-19 19:32 +0000 [r295554] Erin Spiceland + + * res/res_agi.c: Add extra functionality to AGI command WAIT FOR + DIGIT. Add the ability to play a sound file, listen for more than + just one digit, specify escape characters. Backwards compatible + (to work with only timeout specified). (closes issue #15531) + Reported by: diLLec Patches: + asterisk-res_agi-203638-patched.patch uploaded by diLLec (license + 839) Tested by: diLLec, espiceland + +2010-11-19 16:49 +0000 [r295517] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 295516 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295516 | rmudgett | 2010-11-19 10:47:11 -0600 (Fri, 19 Nov 2010) + | 13 lines Bring sig_analog extraction more into alignment with + orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI support. * Fixed + initial value of struct analog_pvt.use_callerid. It may get + forced on depending upon other config options. * Call + analog_dnd() instead of manual inlined code. * Removed unused + struct analog_pvt.usedistinctiveringdetection. * Removed the + struct analog_pvt.unknown_alarm flag. It was really the struct + analog_pvt.inalarm flag. * Use ast_debug() instead of + ast_log(LOG_DEBUG). * Rename several function's index variable to + idx. * Some formatting tweaks. ........ + +2010-11-18 20:31 +0000 [r295478] Leif Madsen + + * configs/sip_notify.conf.sample, /: Merged revisions 295477 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295477 | lmadsen | 2010-11-18 14:30:35 -0600 (Thu, 18 Nov 2010) + | 6 lines 'sip notify clear-mwi' needs terminating CRLF. (closes + issue #18275) Reported by: klaus3000 Patches: + fix_body_CRLF_patch.txt uploaded by klaus3000 (license 65) + ........ + +2010-11-18 18:08 +0000 [r295364-295442] Paul Belanger + + * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions + 295441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295441 | pabelanger | 2010-11-18 13:02:12 -0500 + (Thu, 18 Nov 2010) | 11 lines Merged revisions 295440 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r295440 | pabelanger | 2010-11-18 12:51:34 -0500 (Thu, 18 Nov + 2010) | 4 lines Fix compiler warnings when using openssl-dev + 1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........ + ................ + + * /, contrib/scripts/install_prereq: Merged revisions 295404 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295404 | pabelanger | 2010-11-18 00:12:05 -0500 (Thu, 18 Nov + 2010) | 2 lines Add RedHat specific dependencies ........ + + * /, configs/res_curl.conf.sample: Merged revisions 295361 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295361 | pabelanger | 2010-11-17 09:09:38 -0500 (Wed, 17 Nov + 2010) | 2 lines Uncomment settings under [global], to surpress + warning when loading Asterisk. ........ + +2010-11-16 23:04 +0000 [r295283] Richard Mudgett + + * main/channel.c, /: Merged revisions 295282 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295282 | rmudgett | 2010-11-16 17:02:36 -0600 + (Tue, 16 Nov 2010) | 16 lines Merged revisions 295281 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600 + (Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 + Nov 2010) | 1 line Dead code elimination in + channel.c:ast_channel_bridge() variable who. ........ + ................ ................ + +2010-11-16 22:41 +0000 [r295125-295279] Russell Bryant + + * /, build_tools/prep_tarball: Merged revisions 295278 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r295278 | russell | 2010-11-16 16:41:11 -0600 (Tue, 16 + Nov 2010) | 2 lines Check for pdftotext and give a useful error + if not found. ........ + + * /, build_tools/prep_tarball: Merged revisions 295201 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r295201 | russell | 2010-11-16 15:46:18 -0600 (Tue, 16 + Nov 2010) | 2 lines Remove intentional typo I had added when + testing the check. oops. ........ + + * /, build_tools/prep_tarball: Merged revisions 295164 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r295164 | russell | 2010-11-16 14:50:03 -0600 (Tue, 16 + Nov 2010) | 2 lines Check for wikiexport.py in PATH and give a + useful error message if not found. ........ + + * main/app.c: Remove a trailing space. (testing something with + bamboo ...) + +2010-11-15 19:11 +0000 [r294990-295079] Tilghman Lesher + + * tests/test_expr.c (added), /: Merged revisions 295078 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295078 | tilghman | 2010-11-15 12:30:13 -0600 + (Mon, 15 Nov 2010) | 16 lines Merged revisions 295062 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295062 | tilghman | 2010-11-15 12:24:02 -0600 + (Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 + Nov 2010) | 2 lines Create test verifying results of expression + parser ........ ................ ................ + + * funcs/func_curl.c, /: Merged revisions 294989 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294989 | tilghman | 2010-11-15 01:44:38 -0600 + (Mon, 15 Nov 2010) | 15 lines Merged revisions 294988 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) + | 8 lines It is possible to crash Asterisk by feeding the curl + engine invalid data. (closes issue #18161) Reported by: wdoekes + Patches: 20101029__issue18161.diff.txt uploaded by tilghman + (license 14) Tested by: tilghman ........ ................ + +2010-11-12 21:15 +0000 [r294907-294912] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 294911 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294911 | jpeeler | 2010-11-12 15:14:43 -0600 + (Fri, 12 Nov 2010) | 11 lines Merged revisions 294910 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) + | 4 lines Return correct error code if lock path fails. The + recent changes to open_mailbox actually caused it to be fixed, + but let's be consistent. Reported by alecdavis in asterisk-dev. + ........ ................ + + * /, apps/app_voicemail.c: Merged revisions 294905 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294905 | jpeeler | 2010-11-12 14:52:06 -0600 + (Fri, 12 Nov 2010) | 30 lines Merged revisions 294904 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600 + (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) + | 16 lines Fix regression causing abort in voicemail after + opening a mailbox with no mesgs. In order to be more safe, some + error handling code was changed to respect more error conditions + including the potential memory allocation failure for deleted and + heard message tracking introduced in 293004. However, + last_message_index returns -1 for zero messages (perhaps as + expected) and was triggering the stricter error checking. Because + last_message_index is only called directly in one place, just + return 0 from open_mailbox (for file based storage) when no + messages are detected unless a real error has occurred. (closes + issue #18240) Reported by: leobrown Patches: + bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) + Tested by: pabelanger ........ ................ ................ + +2010-11-12 02:46 +0000 [r294824] Richard Mudgett + + * channels/sig_pri.h, /, channels/sig_pri.c: Merged revisions + 294823 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294823 | rmudgett | 2010-11-11 20:45:22 -0600 + (Thu, 11 Nov 2010) | 25 lines Merged revisions 294822 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600 + (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) + | 11 lines Asterisk is getting a "No D-channels available!" + warning message every 4 seconds. Asterisk is just whining too + much with this message: "No D-channels available! Using Primary + channel XXX as D-channel anyway!". Filtered the message so it + only comes out once if there is no D channel available without an + intervening D channel available period. (closes issue #17270) + Reported by: jmls ........ ................ ................ + +2010-11-11 22:18 +0000 [r294741-294749] Russell Bryant + + * /, doc/CCSS_architecture.pdf (removed): Merged revisions 294745 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294745 | russell | 2010-11-11 16:17:57 -0600 (Thu, 11 Nov 2010) + | 6 lines Remove CCSS architecture PDF. It has been moved to: + https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture + ........ + + * doc/CODING-GUIDELINES (removed), doc/ss7.txt (removed), /, + doc/backtrace.txt (removed), doc/India-CID.txt (removed), + doc/digium-mib.txt (removed), doc/followme.txt (removed), + doc/building_queues.txt (removed), doc/timing.txt (removed), + doc/advice_of_charge.txt (removed), doc/unistim.txt (removed), + doc/video_console.txt (removed), doc/macroexclusive.txt + (removed), doc/google-soc2009-ideas.txt (removed), doc/README.txt + (added), doc/callfiles.txt (removed), build_tools/prep_tarball, + doc/codec-64bit.txt (removed), doc/externalivr.txt (removed), + doc/video.txt (removed), doc/jingle.txt (removed), + doc/modules.txt (removed), doc/manager_1_1.txt (removed), + doc/PEERING (removed), doc/snmp.txt (removed), doc/siptls.txt + (removed), doc/HOWTO_collect_debug_information.txt (removed), + doc/ldap.txt (removed), doc/sip-retransmit.txt (removed), + doc/distributed_devstate.txt (removed), + doc/voicemail_odbc_postgresql.txt (removed), doc/tex (removed), + doc/queue.txt (removed), doc/jabber.txt (removed), + doc/chan_sip-perf-testing.txt (removed), doc/asterisk-mib.txt + (removed), Makefile, doc/database_transactions.txt (removed), + doc/smdi.txt (removed), doc/janitor-projects.txt (removed), + doc/hoard.txt (removed), doc/res_config_sqlite.txt (removed), + doc/osp.txt (removed), doc/speechrec.txt (removed), doc/sms.txt + (removed), doc/distributed_devstate-XMPP.txt (removed), + doc/valgrind.txt (removed), doc/realtimetext.txt (removed), + doc/cli.txt (removed), doc/rtp-packetization.txt (removed), + doc/datastores.txt (removed): Merged revisions 294740 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010) + | 11 lines Remove most of the contents of the doc dir in favor of + the wiki content. This merge does the following things: * Removes + most of the contents from the doc/ directory in favor of the wiki + - http://wiki.asterisk.org/ * Updates the + build_tools/prep_tarball script to know how to export the + contents of the wiki in both PDF and plain text formats so that + the documentation is still included in Asterisk release tarballs. + ........ + +2010-11-11 22:01 +0000 [r294735] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 294734 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294734 | jpeeler | 2010-11-11 15:58:25 -0600 + (Thu, 11 Nov 2010) | 32 lines Merged revisions 294733 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600 + (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) + | 18 lines Fix problem with qualify option packets for realtime + peers never stopping. The option packets not only never stopped, + but if a realtime peer was not in the peer list multiple options + dialogs could accumulate over time. This scenario has the + potential to progress to the point of saturating a link just from + options packets. The fix was to ensure that the poke scheduler + checks to see if a peer is in the peer list before continuing to + poke. The reason a peer must be in the peer list to be able to + properly manage an options dialog is because otherwise the call + pointer is lost when the peer is regenerated from the database, + which is how existing qualify dialogs are detected. (closes issue + #16382) (closes issue #17779) Reported by: lftsy Patches: + bug16382-3.patch uploaded by jpeeler (license 325) Tested by: + zerohalo ........ ................ ................ + +2010-11-10 23:27 +0000 [r294570-294606] Tilghman Lesher + + * pbx/pbx_spool.c, /: Merged revisions 294605 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294605 | tilghman | 2010-11-10 17:26:39 -0600 (Wed, 10 Nov 2010) + | 2 lines Fixing the Mac OS X build (bamboo warning) ........ + + * pbx/pbx_spool.c, /: Merged revisions 294569 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294569 | tilghman | 2010-11-10 17:13:37 -0600 (Wed, 10 Nov 2010) + | 8 lines Properly queue files with inotify(7). (closes issue + #18089) Reported by: abelbeck Patches: + 20101021__issue18089.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ + +2010-11-10 14:15 +0000 [r294502-294536] Russell Bryant + + * /, res/ais/clm.c, res/ais/evt.c, UPGRADE-1.8.txt: Merged + revisions 294535 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294535 | russell | 2010-11-10 08:14:51 -0600 (Wed, 10 Nov 2010) + | 5 lines Tweak a couple of CLI commands back to their original + form. The "module" in this case is two parts, so there are two + words before the verb of the CLI command. ........ + + * /, main/devicestate.c: Merged revisions 294501 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294501 | russell | 2010-11-10 06:46:27 -0600 + (Wed, 10 Nov 2010) | 14 lines Merged revisions 294500 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010) + | 7 lines Improve a debug message to be more readable and + consistent. (closes issue #18282) Reported by: klaus3000 Patches: + ast_devstate2str-patch.txt uploaded by klaus3000 (license 65) + ........ ................ + +2010-11-09 22:52 +0000 [r294467] Richard Mudgett + + * main/channel.c, /: Merged revisions 294466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294466 | rmudgett | 2010-11-09 16:46:45 -0600 (Tue, 09 Nov 2010) + | 1 line Allow ast_do_masquerade() failure to be reported again. + ........ + +2010-11-09 20:35 +0000 [r294431] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 294430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294430 | tilghman | 2010-11-09 14:33:05 -0600 + (Tue, 09 Nov 2010) | 15 lines Merged revisions 294429 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) + | 8 lines Detect GMime properly on systems where gmime flags and + libs are configured with pkg-config. (closes issue #16155) + Reported by: jcollie Patches: 20100917__issue16155.diff.txt + uploaded by tilghman (license 14) Tested by: tilghman ........ + ................ + +2010-11-09 17:00 +0000 [r294351] Richard Mudgett + + * main/channel.c, channels/chan_misdn.c, channels/sig_analog.c, /, + include/asterisk/channel.h, channels/sig_pri.c: Merged revisions + 294349 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) + | 17 lines Analog lines do not transfer CONNECTED LINE or execute + the interception macros. Add connected line update for sig_analog + transfers and simplify the corresponding sig_pri and chan_misdn + transfer code. Note that if you create a three-way call in + sig_analog before transferring the call, the distinction of the + caller/callee interception macros make little sense. The + interception macro writer needs to be prepared for either + caller/callee macro to be executed. The current implementation + swaps which caller/callee interception macro is executed after a + three-way call is created. Review: + https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA + SWP-2372 ........ + +2010-11-08 22:33 +0000 [r294279-294314] Jeff Peeler + + * /, res/res_timing_timerfd.c: Merged revisions 294313 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294313 | jpeeler | 2010-11-08 16:32:13 -0600 + (Mon, 08 Nov 2010) | 9 lines Merged revisions 294312 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08 + Nov 2010) | 1 line add missing unlock not present in 294277 + ........ ................ + + * main/channel.c, /, res/res_timing_timerfd.c, + include/asterisk/timing.h, main/timing.c: Merged revisions 294278 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294278 | jpeeler | 2010-11-08 15:59:45 -0600 + (Mon, 08 Nov 2010) | 23 lines Merged revisions 294277 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) + | 16 lines Fix playback failure when using IAX with the timerfd + module. To fix this issue the alert pipe will now be used when + the timerfd module is in use. There appeared to be a race that + was not solved by adding locking in the timerfd module, but + needed to be there anyway. The race was between the timer being + put in non-continuous mode in ast_read on the channel thread and + the IAX frame scheduler queuing a frame which would enable + continuous mode before the non-continuous mode event was read. + This race for now is simply avoided. (closes issue #18110) + Reported by: tpanton Tested by: tpanton I put tested by tpanton + because it was tested on his hardware. Thanks for the remote + access to debug this issue! ........ ................ + +2010-11-08 21:04 +0000 [r294244] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 294243 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294243 | mnicholson | 2010-11-08 14:56:30 -0600 + (Mon, 08 Nov 2010) | 15 lines Merged revisions 294242 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov + 2010) | 8 lines Go off hold when we get an empty reinvite telling + us to. (closes issue 0014448) Reported by: frawd (closes issue + #17878) Reported by: frawd ........ ................ + +2010-11-08 19:59 +0000 [r294208] Terry Wilson + + * /, configs/calendar.conf.sample, res/res_calendar.c: Merged + revisions 294207 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294207 | twilson | 2010-11-08 13:56:10 -0600 (Mon, 08 Nov 2010) + | 2 lines Set a default waittime, and make sure to convert it to + milliseconds ........ + +2010-11-08 17:19 +0000 [r294127] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 294125 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 + Nov 2010) | 33 lines valgrind reported references to freed memory + during a mISDN hangup collision. Bad things have been happening + in chan_misdn because the chan_misdn channel private struct + chan_list is not protected from reentrancy. Hangup collisions + have be causing read and write accesses to freed memory. + Converted chan_misdn struct chan_list to an ao2 object for its + reference counting feature. ********** Removed an impediment to + converting chan_list to an ao2 object. The use of the other_ch + member in chan_list is shaky at best. It is set if the incoming + and outgoing call legs are mISDN. The use of the other_ch member + goes against the Asterisk architecture and can even cause + problems. 1) It is used to disable echo cancellation. This could + be bad if the call is forked and the winning call leg is not + mISDN or the winning call leg is not the last mISDN channel + called by the fork. The other_ch would become a dangling pointer. + 2) It is used when the far end is alerting to hear the far end's + inband audio instead of Asterisk's generated ringback tone. This + is bad if the call is forked. You would only hear the last forked + mISDN channel and it may not be ringing yet. The other_ch would + become a dangling pointer if the call is later transferred. + ********** JIRA SWP-2423 JIRA ABE-2614 ........ + +2010-11-05 22:17 +0000 [r294086] Brett Bryant + + * /, channels/chan_sip.c: Merged revisions 294084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294084 | bbryant | 2010-11-05 18:03:11 -0400 (Fri, 05 Nov 2010) + | 9 lines Fixed deadlock avoidance issues while locking channel + when adding the Max-Forwards header to a request. (closes issue + #17949) (closes issue #18200) Reported by: bwg Review: + https://reviewboard.asterisk.org/r/997/ ........ + +2010-11-05 21:56 +0000 [r294083] David Vossel + + * channels/chan_sip.c: Perform proper handling of forked outbound + INVITE requests. RFC3261 section 12 about dialog creation says an + INVITE transaction results in an established dialog once it + receives the 200 OK response. It is possible to receive multiple + differing 200 OK responses for a single outbound INVITE Request, + and this should result in establishing multiple dialogs. This + patch allows for all differing 200 OK responses to an INVITE + request to establish a separate dialog, but only the first dialog + is kept. All other resulting dialogs from the initial request are + immediately ACKed and then immediately terminated with a BYE + request. Review: https://reviewboard.asterisk.org/r/946/ + +2010-11-05 16:07 +0000 [r294048-294050] Terry Wilson + + * contrib/scripts/ast_tls_cert, /: Merged revisions 294049 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294049 | twilson | 2010-11-05 09:05:50 -0700 (Fri, 05 Nov 2010) + | 2 lines Corret spelling and example ........ + + * contrib/scripts/ast_tls_cert, /: Merged revisions 294047 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294047 | twilson | 2010-11-05 08:36:20 -0700 (Fri, 05 Nov 2010) + | 2 lines Tell people to use the correct common name for clients + as well ........ + +2010-11-05 15:26 +0000 [r294046] David Vossel + + * /, channels/chan_sip.c: Merged revisions 293924 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293924 | dvossel | 2010-11-04 16:39:51 -0500 (Thu, 04 Nov 2010) + | 4 lines Fixes ringback tone on sip semi-attended transfer. + ABE-2168 ........ + +2010-11-05 00:08 +0000 [r293971] Shaun Ruffell + + * /, codecs/codec_dahdi.c: Merged revisions 293970 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293970 | sruffell | 2010-11-04 19:07:11 -0500 + (Thu, 04 Nov 2010) | 32 lines Merged revisions 293969 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500 + (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) + | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when + receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically + commit 9034) added the capability for the wctc4xxp to return more + than a single packet of data in response to a read. However, when + decoding packets, codec_dahdi was still assuming that the default + number of samples was in each read. In other words, each packet + your provider sent you, regardless of size, would result in 20 ms + of decoded data (30 ms if decoding G723). If your provider was + sending 60 ms packets then codec_dahdi would end up stripping 40 + ms of data from each transcoded frame resulting in "choppy" + audio. This would only affect systems where G729 packets are + arriving in sizes greater than 20ms or G723 packets arriving in + sizes greater than 30ms. DAHDI-744. ........ ................ + ................ + +2010-11-04 13:29 +0000 [r293888] Paul Belanger + + * /, channels/chan_sip.c: Merged revisions 293887 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293887 | pabelanger | 2010-11-04 09:27:54 -0400 (Thu, 04 Nov + 2010) | 8 lines Do not output port in IPaddress for AMI sippeers. + (closes issue #18248) Reported by: orn Patches: + ami_sippeers.patch uploaded by pabelanger (license 224) Tested + by: orn ........ + +2010-11-03 18:43 +0000 [r293809] Terry Wilson + + * main/rtp_engine.c, /, channels/chan_sip.c, + include/asterisk/rtp_engine.h: Merged revisions 293803 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) + | 25 lines Avoid valgrind warnings for + ast_rtp_instance_get_xxx_address The documentation for + ast_rtp_instance_get_(local/remote)_address stated that they + returned 0 for success and -1 on failure. Instead, they returned + 0 if the address structure passed in was already equivalent to + the address instance local/remote address or 1 otherwise. 90% of + the calls to these functions completely ignored the return + address and passed in an uninitialized struct, which would make + valgrind complain even though the operation was technically safe. + This patch fixes the documentation and converts the + get_xxx_address functions to void since all they really do is + copy the address and cannot fail. Additionally two new functions + (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created + for the 3 times where the return value was actually checked. The + get_and_cmp_local_address function is currently unused, but + exists for the sake of symmetry. The only functional change as a + result of this change is that we will not do an + ast_sockaddr_cmp() on (mostly uninitialized) addresses before + doing the ast_sockaddr_copy() in the get_*_address functions. So, + even though it is an API change, it shouldn't have a noticeable + change in behavior. Review: + https://reviewboard.asterisk.org/r/995/ ........ + +2010-11-03 18:38 +0000 [r293808] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 293807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293807 | rmudgett | 2010-11-03 13:35:19 -0500 + (Wed, 03 Nov 2010) | 34 lines Merged revisions 293806 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500 + (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) + | 20 lines Party A in an analog 3-way call would continue to hear + ringback after party C answers. All parties are analog FXS ports. + 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to + bring C into 3-way call before C answers. (A and B hear ringback) + 4) C answers 5) A continues to hear ringback during the 3-way + call. (All parties can hear each other.) * Fixed use of wrong + variable in dahdi_bridge() that stopped ringback on the wrong + subchannel. * Made several debug messages have more information. + A similar issue happens if B and C are SIP channels. B continues + to hear ringback. For some reason this only affects v1.8 and + trunk. * Don't start ringback on the real and 3-way subchannels + when creating the 3-way conference. Removing this code is benign + on v1.6.2 and earlier. ........ ................ ................ + +2010-11-02 23:10 +0000 [r293725] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 293724 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293724 | jpeeler | 2010-11-02 18:09:06 -0500 + (Tue, 02 Nov 2010) | 22 lines Merged revisions 293723 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500 + (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) + | 8 lines Add enabled/disabled information for rtautoclear sip + show settings output. When setting to zero/"no", the numeric + default was shown making it not obvious the disabled setting was + respected. (closes issue #18123) Reported by: zerohalo ........ + ................ ................ + +2010-11-02 21:31 +0000 [r293649] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 293648 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293648 | rmudgett | 2010-11-02 16:29:25 -0500 + (Tue, 02 Nov 2010) | 20 lines Merged revisions 293647 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500 + (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) + | 6 lines Make warning message have more useful information in + it. Change "Unable to get index, and nullok is not asserted" to + "Unable to get index for '' on channel + ((), line )". ........ ................ + ................ + +2010-11-02 20:47 +0000 [r293578-293612] Paul Belanger + + * main/manager.c, /: Merged revisions 293611 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293611 | pabelanger | 2010-11-02 16:45:09 -0400 (Tue, 02 Nov + 2010) | 2 lines If manager and tls are disabled, do not display + TCP/TLS Bindaddress. ........ + + * configs/gtalk.conf.sample, UPGRADE.txt, channels/chan_gtalk.c, + CHANGES: New CLI command 'gtalk show settings'. Review: + https://reviewboard.asterisk.org/r/984/ + +2010-11-02 14:43 +0000 [r293577] Mark Michelson + + * CHANGES: Add to the CHANGES file that the HTTP server supports + IPv6 addressing. + +2010-11-01 17:32 +0000 [r293531] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 293530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293530 | rmudgett | 2010-11-01 12:29:30 -0500 (Mon, 01 Nov 2010) + | 10 lines Analog 3-way call would not connect all parties if one + was using sig_pri. Also the "dahdi show channel" would not show + the correct 3-way call status. * Synchronized the inthreeway flag + between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() + sign error and made take an analog sub channel enum. ........ + +2010-11-01 16:11 +0000 [r293497] Paul Belanger + + * /, channels/chan_iax2.c: Merged revisions 293496 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r293496 | pabelanger | 2010-11-01 12:09:05 -0400 (Mon, + 01 Nov 2010) | 13 lines Use ast_sockaddr_from_sin function not + memcpy This resolves some IAX2 registration issue report on the + asterisk-users mailing list. (closes issue #18202) Reported by: + pabelanger Patches: update_registry.patch.v2 uploaded by + pabelanger (license 224) Tested by: pabelanger, Nic Colledge + (mailing list) Review: https://reviewboard.asterisk.org/r/993 + ........ + +2010-10-30 01:55 +0000 [r293342-293419] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 293418 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293418 | rmudgett | 2010-10-29 20:53:29 -0500 + (Fri, 29 Oct 2010) | 16 lines Merged revisions 293417 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500 + (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 + Oct 2010) | 1 line Remove some more code that serves no purpose. + ........ ................ ................ + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 293341 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293341 | rmudgett | 2010-10-29 19:46:41 -0500 + (Fri, 29 Oct 2010) | 16 lines Merged revisions 293340 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500 + (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 + Oct 2010) | 1 line Remove some code that serves no purpose. + ........ ................ ................ + +2010-10-29 21:50 +0000 [r293306] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 293305 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293305 | jpeeler | 2010-10-29 16:48:38 -0500 (Fri, 29 Oct 2010) + | 9 lines Modify sip_setoption to not complain about unknown + options. This now behaves just like the other setoption + callbacks. For the curious the offending option for the reporter + was AST_OPTION_CHANNEL_WRITE which was getting passed due to a + fix for chan_local in 286189. (closes issue #17985) Reported by: + globalnetinc ........ + +2010-10-29 20:46 +0000 [r293273] Mark Michelson + + * main/http.c, UPGRADE.txt, configs/http.conf.sample: Enable IPv6 + for the built-in HTTP server. Review: + https://reviewboard.asterisk.org/r/986 + +2010-10-28 20:01 +0000 [r293198] Tilghman Lesher + + * /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y, + res/ael/ael_lex.c, res/ael/ael.tab.h, main/ast_expr2.c: Merged + revisions 293197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293197 | tilghman | 2010-10-28 15:00:06 -0500 + (Thu, 28 Oct 2010) | 33 lines Merged revisions 293195-293196 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500 + (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) + | 5 lines "!00" evaluated as false, which is incorrect. Fixing. + Reported (though the reporter did not understand he was reporting + a bug) on the asterisk-users list: + http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html + ........ ................ r293196 | tilghman | 2010-10-28 + 14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions + 293194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) + | 5 lines "!00" evaluated as false, which is incorrect. Fixing. + Reported (though the reporter did not understand he was reporting + a bug) on the asterisk-users list: + http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html + ........ ................ ................ + +2010-10-28 16:11 +0000 [r293160] Jeff Peeler + + * /, funcs/func_strings.c: Merged revisions 293159 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293159 | jpeeler | 2010-10-28 11:11:08 -0500 + (Thu, 28 Oct 2010) | 18 lines Merged revisions 293158 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) + | 11 lines Fix infinite loop in FILTER(). Specifically when + you're using characters above \x7f or invalid character escapes + (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: + issue18060_func_strings_filter_infinite_loop.patch uploaded by + wdoekes (license 717) Tested by: wdoekes ........ + ................ + +2010-10-26 18:54 +0000 [r293120] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 293119 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293119 | jpeeler | 2010-10-26 13:49:08 -0500 + (Tue, 26 Oct 2010) | 43 lines Merged revisions 293118 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500 + (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) + | 29 lines Fix inprocess_container in voicemail to correctly + restrict max messages. The comparison function logic was off, so + the number of sessions for a given mailbox were not being + incremented properly. This problem caused the maximum number of + messages per folder to not be respected when simultaneously + leaving multiple voicemails just below the threshold. These + problems should be fixed by the above, but just in case: Fixed + resequence_mailbox to rely on the actual number of detected + number of files in a directory rather than just assuming only 10 + messages more than the maximum had been left. Also if more + messages than the maximum are deleted they are actually removed + now. The second purpose of this commit should have been separated + out probably, but is related to the above. Again, if the number + of messages in a given voicemail folder exceeds the maximum set + limit make sure to allocate enough space for the deleted and + heard index tracking array. A few random fixes: There was a + forgotten decrement of the inprocess count in imap_store_file. + When using IMAP storage, do not look in the directory where file + based storage messages may still reside and influence the message + count. Ensure to use only the first format in sendmail. ABE-2516 + ........ ................ ................ + +2010-10-26 16:33 +0000 [r293047-293082] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 293081 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293081 | rmudgett | 2010-10-26 11:32:59 -0500 (Tue, 26 Oct 2010) + | 1 line No need to define the struct if there are no users. + ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c: Merged revisions 293046 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293046 | rmudgett | 2010-10-26 10:53:58 -0500 (Tue, 26 Oct 2010) + | 4 lines Allow the DAHDI driver to compile, even with a + sufficiently older version of libpri. Fixes our Bamboo builds. + ........ + +2010-10-25 21:16 +0000 [r292915-292970] Tilghman Lesher + + * /, channels/sig_pri.c: Merged revisions 292969 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292969 | tilghman | 2010-10-25 16:15:19 -0500 (Mon, 25 Oct 2010) + | 2 lines Several more defines that need to be altered for + compiling against an older version of libpri ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c: Merged revisions 292906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292906 | tilghman | 2010-10-25 14:28:35 -0500 (Mon, 25 Oct 2010) + | 4 lines Allow the DAHDI driver to compile, even with a + sufficiently older version of libpri. Fixes our Bamboo builds. + ........ + +2010-10-25 19:11 +0000 [r292869] David Vossel + + * channels/chan_local.c, /: Merged revisions 292868 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292868 | dvossel | 2010-10-25 14:07:50 -0500 + (Mon, 25 Oct 2010) | 39 lines Merged revisions 292867 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500 + (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) + | 27 lines This patch turns chan_local pvts into astobj2 objects. + chan_local does some dangerous things involving deadlock + avoidance. tech_pvt functions like hangup and queue_frame are + provided with a locked channel upon entry. Those functions are + completely safe as long as you don't attempt to give up that + channel lock, but that is impossible to guarantee due to the + required deadlock avoidance necessary to lock both the tech_pvt + and both channels involved. In the past, we have tried to account + for this by doing things like setting a "glare" flag that + indicates what function should destroy the pvt. This was used in + local_hangup and local_queue_frame to decided who should destroy + the pvt if they collided in separate threads. I have removed the + need to do this by converting all chan_local tech_pvts to + astobj2. This means we can ref a pvt before deadlock avoidance + and not have to worry about that pvt possibly getting destroyed + under us. It also cleans up where we destroy the tech_pvt. The + only unlink from the tech_pvt container occurs in local_hangup + now, which is where it should occur. Since there still may be + thread collisions on some functions like local_hangup after + deadlock avoidance, I have added some checks to detect those + collisions and exit appropriately. I think this patch is going to + solve quite a bit of weirdness we have had with local channels in + the past. ........ ................ ................ + +2010-10-22 22:40 +0000 [r292808-292826] Terry Wilson + + * contrib/scripts/ast_tls_cert, /: Merged revisions 292825 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292825 | twilson | 2010-10-22 15:35:29 -0700 (Fri, 22 Oct 2010) + | 4 lines Don't create directories without at least o+x Also, + making files that you are going to modify read-only is dumb. + ........ + + * contrib/scripts/ast_tls_cert, /: Merged revisions 292794 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292794 | twilson | 2010-10-22 15:18:36 -0700 (Fri, 22 Oct 2010) + | 2 lines Make files readable only by the owner ........ + +2010-10-22 21:29 +0000 [r292788] Leif Madsen + + * /, channels/chan_sip.c, configs/res_ldap.conf.sample, + contrib/scripts/asterisk.ldif: Merged revisions 292787 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292787 | lmadsen | 2010-10-22 16:28:43 -0500 + (Fri, 22 Oct 2010) | 21 lines Merged revisions 292786 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) + | 13 lines Update the LDIF file for LDAP. The LDIF file + asterisk.ldif was quite a bit out of date from the + asterisk.ldap-schema file, so I've now updated that to be in + sync. The asterisk.ldif file being out of sync was a problem on + my systems where I was doing an ldapadd to import the schema into + the LDAP database, and the existing file would cause problems and + ERROR messages when registering. Additional documention has been + added based on feedback in the issue I'm closing. (closes issue + #13861) Reported by: scramatte Patches: ldap-update.txt uploaded + by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, + rgenthner ........ ................ + +2010-10-22 17:16 +0000 [r292743] Terry Wilson + + * contrib/scripts/ast_tls_cert (added), /: Merged revisions 292740 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292740 | twilson | 2010-10-22 09:49:34 -0700 (Fri, 22 Oct 2010) + | 45 lines Add TLS cert helper script This script is useful for + quickly generating self-signed CA, server, and client + certificates for use with Asterisk. It is still recommended to + obtain certificates from a recognized Certificate Authority and + to develop an understanding how SSL certificates work. Real + security is hard work. OPTIONS: -h Show this message -m Type of + cert "client" or "server". Defaults to server. -f Config filename + (openssl config file format) -c CA cert filename (creates new CA + cert/key as ca.crt/ca.key if not passed) -k CA key filename -C + Common name (cert field) For a server cert, this should be the + same address that clients attempt to connect to. Usually this + will be the Fully Qualified Domain Name, but might be the IP of + the server. For a CA or client cert, it is merely informational. + Make sure your certs have unique common names. -O Org name (cert + field) An informational string (company name) -o Output filename + base (defaults to asterisk) -d Output directory (defaults to the + current directory) Example: To create a CA and a server + (pbx.mycompany.com) cert with output in /tmp: ast_tls_cert -C + pbx.mycompany.com -O "My Company" -d /tmp This will create a CA + cert and key as well as asterisk.pem and the the two files that + it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem + and ca.crt somewhere (like /etc/asterisk) and set + tlscertfile=/etc/asterisk.pem and tlscafile=/etc/ca.crt. Since + this is a self-signed key, many devices will require you to + import the ca.crt file as a trusted cert. To create a client cert + using the CA cert created by the example above: ast_tls_cert -m + client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \ "My + Company" -d /tmp -o joe_user This will create client.crt/key/pem + in /tmp. Use this if your device supports a client certificate. + Make sure that you have the ca.crt file set up as a tlscafile in + the necessary Asterisk configs. Make backups of all .key files in + case you need them later. ........ + +2010-10-22 17:10 +0000 [r292742] Mark Michelson + + * /, tests/test_event.c: Merged revisions 292741 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292741 | mmichelson | 2010-10-22 12:09:52 -0500 (Fri, 22 Oct + 2010) | 12 lines Prevent multiple runs of event_sub_test from + producing false failure results. The array of test subscriptions + was declared "static," meaning that the data.count field would + retain its value between runs of the test. After the first test + run, this would result in false reports of test failures. I chose + to just remove the "static" keyword from the structure since it's + not a huge deal to construct this structure during each run of + the test. Another alternative would have been to zero out the + data.count fields of each test subscription instead. ........ + +2010-10-22 15:47 +0000 [r292705] Richard Mudgett + + * main/channel.c, channels/chan_misdn.c, /, channels/sig_pri.c: + Merged revisions 292704 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) + | 19 lines Connected line is not updated when chan_dahdi/sig_pri + or chan_misdn transfers a call. When a call is transfered by ECT + or implicitly by disconnect in sig_pri or implicitly by + disconnect in chan_misdn, the connected line information is not + exchanged. The connected line interception macros also need to be + executed if defined. The CALLER interception macro is executed + for the held call. The CALLEE interception macro is executed for + the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: + abe_2589_c3bier.patch uploaded by rmudgett (license 664) + abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: + https://reviewboard.asterisk.org/r/958/ ........ + +2010-10-21 22:11 +0000 [r292668] Tilghman Lesher + + * /, channels/misdn/ie.c: Merged revisions 292667 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292667 | tilghman | 2010-10-21 17:09:25 -0500 (Thu, 21 Oct 2010) + | 2 lines Compile correctly on Linux (asterisk/localtime.h + depends upon asterisk/autoconfig.h loading first). ........ + +2010-10-21 18:23 +0000 [r292630] Paul Belanger + + * /, contrib/init.d/rc.suse.asterisk: Merged revisions 292628 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292628 | pabelanger | 2010-10-21 14:13:18 -0400 (Thu, 21 Oct + 2010) | 5 lines Fix typo in SUSE init script. Reported by: Dave + Cotton on asterisk-users list. ........ + +2010-10-21 16:46 +0000 [r292597] David Vossel + + * main/manager.c, /: Merged revisions 292595 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292595 | dvossel | 2010-10-21 11:14:33 -0500 (Thu, 21 Oct 2010) + | 14 lines Fixes recursive lock problem in manager.c It is + possible for a AMI session to freeze because of invalid use of + recursive locks during the EVENT processing. This patch removes + the unnecessary locks. (closes issue #18167) Reported by: sustav + Patches: manager_locking_v1.diff uploaded by dvossel (license + 671) Tested by: sustav ........ + +2010-10-21 13:17 +0000 [r292559] Leif Madsen + + * /, configs/res_ldap.conf.sample: Merged revisions 292557 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292557 | lmadsen | 2010-10-21 08:12:19 -0500 + (Thu, 21 Oct 2010) | 14 lines Merged revisions 292556 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r292556 | lmadsen | 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010) + | 6 lines Change res_ldap.sample.conf to match the schema. + (closes issue #17376) Reported by: jcovert Patches: + res_ldap.conf.sample.patch uploaded by jcovert (license 551) + ........ ................ + +2010-10-21 11:38 +0000 [r292524] Russell Bryant + + * /, res/res_config_ldap.c: Merged revisions 292523 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r292523 | russell | 2010-10-21 06:36:47 -0500 (Thu, 21 + Oct 2010) | 4 lines Add var=value to log message on update + failure, and add newline. ... just for you, Leif. ........ + +2010-10-21 01:03 +0000 [r292490] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 292489 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292489 | rmudgett | 2010-10-20 20:02:50 -0500 (Wed, 20 Oct 2010) + | 7 lines Send CONNECT_ACKNOWLEDGE for CIS calls too. The + originator of the Q.SIG call completion signaling link was not + changed to the active state when the CONNECT message came in. The + T309 processing would immediately kill the signaling link because + it was not in the active state. ........ + +2010-10-21 00:23 +0000 [r292414-292443] Paul Belanger + + * /, apps/app_voicemail.c: Merged revisions 292436 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r292436 | pabelanger | 2010-10-20 20:21:59 -0400 (Wed, + 20 Oct 2010) | 8 lines Application not properly unregister in + voicemail (closes issue #18128) Reported by: junky Patches: + vm_unregister.diff uploaded by junky (license 177) Tested by: + pabelanger, lmadsen ........ + + * apps/app_dial.c, /: Merged revisions 292413 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292413 | pabelanger | 2010-10-20 20:07:17 -0400 + (Wed, 20 Oct 2010) | 24 lines Merged revisions 292412 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r292412 | pabelanger | 2010-10-20 20:05:45 -0400 + (Wed, 20 Oct 2010) | 17 lines Merged revisions 292411 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct + 2010) | 10 lines Record priv-recordintro as sln, not gsm This + removes the gsm->sln step when transcoding priv-recordintro. + (closes issue #18176) Reported by: pabelanger Patches: + chan_sip.diff uploaded by pabelanger (license 224) ........ + ................ ................ + +2010-10-20 00:41 +0000 [r292377] Tilghman Lesher + + * /, res/res_musiconhold.c: Merged revisions 292376 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r292376 | tilghman | 2010-10-19 19:40:29 -0500 (Tue, 19 + Oct 2010) | 5 lines Oops. This module uses the generic timer and + no longer uses DAHDI. This causes a problem with the Solaris and + other system builds that have gcc 4.1 (where optional_api is + non-optional). ........ + +2010-10-19 22:19 +0000 [r292345] Paul Belanger + + * /, contrib/scripts/install_prereq: Merged revisions 292343 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292343 | pabelanger | 2010-10-19 18:14:23 -0400 (Tue, 19 Oct + 2010) | 2 lines Add resample and imap_tk dependencies. ........ + +2010-10-19 19:35 +0000 [r292310] Terry Wilson + + * /, channels/chan_sip.c, res/res_srtp.c: Merged revisions 292309 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) + | 10 lines Add sip show peer info about crypto and remove dated + comment This patch adds information about the encryption setting + to 'sip show peers' and removes an out-of-date comment from + res_srtp.c and instead directs users to the proper documentation. + (closes issue #18140) Reported by: chodorenko ........ + +2010-10-18 22:14 +0000 [r292231] Leif Madsen + + * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 292225 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292225 | lmadsen | 2010-10-18 16:51:23 -0500 + (Mon, 18 Oct 2010) | 24 lines Merged revisions 292224 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500 + (Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010) + | 9 lines Add support for the new English (Australian Accent) + sound files. (closes issue #17426) Reported by: camsown Patches: + core-sounds-en_AU.txt uploaded by camsown (license 1050) + add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested + by: camsown, lmadsen, jtodd, qwell ........ ................ + ................ + +2010-10-18 21:56 +0000 [r292228] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 292227 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292227 | jpeeler | 2010-10-18 16:55:46 -0500 + (Mon, 18 Oct 2010) | 25 lines Merged revisions 292226 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500 + (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) + | 11 lines Fix improper operator key acceptance and clean up temp + recording files. This is a fix for when pressing the operator key + after recording an unavailable, busy, name, or temporary message + in mailbox options. The operator key should not be accepted here, + but should be allowed during the message recording. If the + operator key is pressed during ensure the file is saved or + deleted as apporopriate. Also, ensure removal of temporary + recorded files after an early hang up or when message acceptance + confirmation times out. ABE-2518 ........ ................ + ................ + +2010-10-18 19:52 +0000 [r292189] Russell Bryant + + * main/netsock2.c, /: Merged revisions 292188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292188 | russell | 2010-10-18 14:50:04 -0500 (Mon, 18 Oct 2010) + | 9 lines Resolve some compiler errors in ast_sockaddr_is_any(). + These errors came up once this function was used from within + netsock2.c. The errors were like the following: netsock2.c:393: + error: dereferencing pointer ‘({anonymous})’ does break + strict-aliasing rules The usage of a union here avoids this + problem. ........ + +2010-10-18 19:16 +0000 [r292156] David Vossel + + * main/netsock2.c, /: Merged revisions 292155 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292155 | dvossel | 2010-10-18 14:16:00 -0500 (Mon, 18 Oct 2010) + | 2 lines Fixes build error for systems not supporting + IPV6_TCLASS. ........ + +2010-10-18 17:18 +0000 [r292124] Matthew Nicholson + + * /, addons/chan_mobile.c: Merged revisions 292122 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r292122 | mnicholson | 2010-10-18 12:15:24 -0500 (Mon, + 18 Oct 2010) | 5 lines Fix the cmgr parser. (closes issue + 0018152) Reported by: menschentier ........ + +2010-10-18 16:03 +0000 [r292086] David Vossel + + * main/netsock2.c, /: Merged revisions 292085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292085 | dvossel | 2010-10-18 11:02:17 -0500 (Mon, 18 Oct 2010) + | 7 lines Fixes qos settings for sockets bound to any IPv6 or + IPv4 address. (closes issue #18099) Reported by: jamesnet + Patches: issues_18099_v3.diff uploaded by dvossel (license 671 + ........ + +2010-10-18 15:33 +0000 [r292084] Jeff Peeler + + * pbx/pbx_spool.c, /: Merged revisions 292083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292083 | jpeeler | 2010-10-18 10:32:40 -0500 (Mon, 18 Oct 2010) + | 4 lines Disable use of inotify for call file handling as it is + not working properly. (related to #18089) ........ + +2010-10-16 11:51 +0000 [r292052] Tzafrir Cohen + + * /, configs/musiconhold.conf.sample, res/res_musiconhold.c: Merged + revisions 292050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292050 | tzafrir | 2010-10-16 12:47:00 +0200 + (ש', 16 אוק 2010) | 22 lines Merged revisions 292049 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 + אוק 2010) | 15 lines Base directory for MOH should be ASTDATADIR + If the directive 'directory' is relative, make it relative to the + datadir, rather than to the varlibdir. In the sample + configuration it is relative ('moh'). This has no effect unless + you have actively set the datadir explicitly (at build time or at + run time). (closes issue #16906) Patches: moh_datadir uploaded by + tzafrir (license 46) Review: + https://reviewboard.asterisk.org/r/974/ ........ ................ + +2010-10-15 21:49 +0000 [r292017] Terry Wilson + + * /, res/res_srtp.c: Merged revisions 292016 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292016 | twilson | 2010-10-15 16:40:56 -0500 (Fri, 15 Oct 2010) + | 5 lines Ref/unref res_srtp when we create/destroy a session + This avoids unhappy crashing when we try to 'core stop + gracefully' and res_srtp tries to unload before chan_sip does. + Thanks, Russell! ........ + +2010-10-15 20:12 +0000 [r291943] David Vossel + + * /, channels/chan_sip.c: Merged revisions 291942 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291942 | dvossel | 2010-10-15 15:12:04 -0500 (Fri, 15 Oct 2010) + | 8 lines Fixes peer's host port information being lost on sip + reload. (closes issue #18135) Reported by: lmadsen Patches: + crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: + lmadsen ........ + +2010-10-15 19:53 +0000 [r291941] Paul Belanger + + * configs/gtalk.conf.sample, /: Merged revisions 291940 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291940 | pabelanger | 2010-10-15 15:50:22 -0400 + (Fri, 15 Oct 2010) | 16 lines Merged revisions 291939 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400 + (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, + 15 Oct 2010) | 2 lines Clean up formatting. ........ + ................ ................ + +2010-10-15 16:54 +0000 [r291906] Terry Wilson + + * /, res/res_jabber.c: Merged revisions 291905 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291905 | twilson | 2010-10-15 09:39:58 -0700 + (Fri, 15 Oct 2010) | 14 lines Merged revisions 291904 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010) + | 7 lines Don't crash or deadlock on module unload We can't hold + the lock while pthread_join is called since aji_log_hook will + attempt to lock from the other therad. We reorder the + pthread_join and ast_aji_disconnect so that we don't do an + SSL_read() while SSL_shutdown is running, causing a crash. + ........ ................ + +2010-10-14 22:10 +0000 [r291828-291830] David Vossel + + * main/netsock2.c, /: Merged revisions 291829 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291829 | dvossel | 2010-10-14 17:09:32 -0500 (Thu, 14 Oct 2010) + | 8 lines Set TCLASS field of IPv6 header when sip qos options + are set. (closes issue #18099) Reported by: jamesnet Patches: + issues_18099_v2.diff uploaded by dvossel (license 671) Tested by: + dvossel, jamesnet ........ + + * /, channels/chan_gtalk.c: Merged revisions 291827 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291827 | dvossel | 2010-10-14 16:27:42 -0500 (Thu, 14 + Oct 2010) | 18 lines Safer xml parsing, treat all clients the + same, and better local candidate selection. The gtalk channel + driver was doing several unsafe operations in regards to how it + parsed incoming XML messages. I have cleaned that code up so it + should be much safer now. We now treat all clients types the + same. We have no reason to distinguish between GMAIL and GOOGLE + VOICE clients anymore because they all work the same way. I also + modified how the local ip is found. If no bindaddress is provided + in the config file, we attempt to determine the local ip we would + use to connect to google.com. If that fails, then we fall back to + the ast_find_ourip() function as a last resort. Using the new + method makes it much less likely that we would ever advertise a + local RTP candidate as a loopback address. ........ + +2010-10-14 18:46 +0000 [r291792] Jeff Peeler + + * /, main/stdtime/localtime.c: Merged revisions 291791 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291791 | jpeeler | 2010-10-14 13:45:02 -0500 (Thu, 14 + Oct 2010) | 10 lines Add missing ifdefs for test framework and + new locale code. (closes issue #18137) Reported by: ovi Patches: + 18137_test_framework_ifdef.patch uploaded by wdoekes (license + 717) 18137_localelist_warning.patch uploaded by wdoekes (license + 717) Tested by: ovi ........ + +2010-10-14 15:21 +0000 [r291760] Paul Belanger + + * channels/chan_jingle.c, include/asterisk/acl.h, /, + channels/chan_sip.c, channels/chan_h323.c, main/acl.c, + channels/chan_gtalk.c: Merged revisions 291758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct + 2010) | 11 lines Add the ability for ast_find_ourip to return + IPv4, IPv6 or both. While testing chan_gtalk I noticed jabber was + using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 + it is possible for ast_find_ourip() to return both IPv6 and IPv4 + results. Adding a family parameter gives you the ablility to + choose. Since jabber/gtalk/h323 do not support IPv6, we should + only return IPv4 results. Review: + https://reviewboard.asterisk.org/r/973/ ........ + +2010-10-14 12:10 +0000 [r291726] Russell Bryant + + * /, doc/tex/secure-calls.tex: Merged revisions 291725 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291725 | russell | 2010-10-14 07:08:43 -0500 (Thu, 14 + Oct 2010) | 2 lines Fix a typo - s/seucre/secure/ ........ + +2010-10-13 23:52 +0000 [r291658] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 291656 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291656 | rmudgett | 2010-10-13 18:45:11 -0500 + (Wed, 13 Oct 2010) | 34 lines Merged revisions 291655 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500 + (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) + | 20 lines Deadlock between dahdi_exception() and + dahdi_indicate(). There is a deadlock between dahdi_exception() + and dahdi_indicate() for analog ports. The call-waiting and + three-way-calling feature can experience deadlock if these + features are trying to do something and an event from the bridged + channel happens at the same time. Deadlock avoidance code added + to obtain necessary channel locks before attemting an operation + with call-waiting and three-way-calling. (closes issue #16847) + Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch + uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch + uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch + uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett + Review: https://reviewboard.asterisk.org/r/971/ ........ + ................ ................ + +2010-10-13 23:47 +0000 [r291657] Terry Wilson + + * main/channel.c, /: Merged revisions 291581 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291581 | twilson | 2010-10-13 16:01:56 -0700 + (Wed, 13 Oct 2010) | 35 lines Merged revisions 291580 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291580 | twilson | 2010-10-13 15:58:43 -0700 + (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) + | 21 lines Don't ignore frames that have been queued when + softhangup'd When an outgoing call is answered and hung up by the + far end *very* quickly, we may not read any frames and therefor + end up with a call that displays the wrong + disposition/DIALSTATUS. The reason is because ast_queue_hangup() + immediately sets the _softhangup flag on the channel and then + queues the HANGUP control frame, but __ast_read refuses to read + any frames if ast_check_hangup() indicates that a hangup request + has been made (which it will if _softhangup is set). So, we end + up losing control frames. This change makes __ast_read continue + to read frames even if a soft hangup has been requested. It + queues a hangup frame to make sure that __ast_read() will still + eventually return NULL. Much thanks to David Vossel for all of + the reviews, discussion, and help! (closes issue #16946) Reported + by: davidw Review: https://reviewboard.asterisk.org/r/740/ + ........ ................ ................ + +2010-10-13 22:47 +0000 [r291579] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 291578 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291578 | dvossel | 2010-10-13 17:46:34 -0500 (Wed, 13 + Oct 2010) | 4 lines More fixup for chan_gtalk. This patch makes + the xml parsing safer. ........ + +2010-10-13 22:34 +0000 [r291576] Terry Wilson + + * Makefile, /, static-http/mantest.html (added): Merged revisions + 291575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291575 | twilson | 2010-10-13 15:24:44 -0700 (Wed, 13 Oct 2010) + | 8 lines Add a simple AMI client web page This patch uses the + XML docs to parse all of the available AMI commands and allows + you to enter the command name and be presented with a form with + the available fields. You can then rapidly tab through the fields + and submit the command and view the response. It is much + faster/easier than having to use telnet for testing purposes. + ........ + +2010-10-13 20:24 +0000 [r291470-291542] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 291541 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291541 | rmudgett | 2010-10-13 15:21:02 -0500 (Wed, 13 + Oct 2010) | 26 lines The chan_dahdi faxdetect option only works + for the first FAX call. The chan_dahdi faxdetect option only + works for the first call. After that the option no longer works. + The struct dahdi_pvt.callprogress member is the encoded user + config setting for the callprogress and faxdetect config options. + Changing this value alters the configuration for all following + calls until the chan_dahdi.conf file is reloaded. * Fixed the + chan_dahdi ast_channel_setoption callback to not change the users + faxdetect config setting except for the current call. * Fixed the + chan_dahdi ast_channel_queryoption callback to read the active + DSP setting of the faxdetect option. * Made actually disable the + active faxdetect DSP setting for the current call on the analog + port. my_handle_dtmfup() is used for normal analog ports. + dahdi_handle_dtmfup() is the legacy code and is no longer used + unless in a radio mode. (closes issue #18116) Reported by: + seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett + (license 664) Review: https://reviewboard.asterisk.org/r/972/ + ........ + + * channels/chan_misdn.c, /: Merged revisions 291507 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291507 | rmudgett | 2010-10-13 14:01:48 -0500 + (Wed, 13 Oct 2010) | 18 lines Merged revision 291504 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed, + 13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the + ast_channel. Must get the ast_channel lock before proceeding with + release_chan() and release_chan_early() to hold off ast_hangup() + from destroying the ast_channel. Missed this change for -r291468. + JIRA ABE-2598 JIRA SWP-2317 .......... ................ + + * channels/chan_misdn.c, /: Merged revisions 291469 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291469 | rmudgett | 2010-10-13 13:10:21 -0500 + (Wed, 13 Oct 2010) | 23 lines Merge revision 291468 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed, + 13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN + call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE + --> RELEASE_COMPLETE * Add lock protection around channel list + for find/add/delete operations. * Protect misdn_hangup() from + release_chan() and vise versa using the release_lock. JIRA + ABE-2598 JIRA SWP-2317 .......... ................ + +2010-10-13 15:51 +0000 [r291395] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 291394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291394 | russell | 2010-10-13 10:46:39 -0500 + (Wed, 13 Oct 2010) | 20 lines Merged revisions 291393 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291393 | russell | 2010-10-13 10:29:21 -0500 + (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) + | 6 lines Lock pvt so pvt->owner can't disappear when queueing up + a frame. This fixes a crash due to a hangup race condition. + ABE-2601 ........ ................ ................ + +2010-10-13 08:58 +0000 [r291361] Stefan Schmidt + + * apps/app_macro.c: Report what extension called a failed macro Add + the extension and context of the calling channel to the log + output if a macro could not be found. (closes issue #18112) + Reported by: prado Patches: app_macro-info.diff uploaded by prado + (license 510) Tested by: schmidts + +2010-10-12 17:21 +0000 [r291287] Leif Madsen + + * /, configs/phoneprov.conf.sample: Merged revisions 291284 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291284 | lmadsen | 2010-10-12 12:20:43 -0500 + (Tue, 12 Oct 2010) | 15 lines Merged revisions 291280 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010) + | 7 lines Add undocumented variables to phoneprov.conf.sample + (closes issue #18107) Reported by: lathama Patches: + phoneprov.conf.sample.diff uploaded by lathama (license 1028) + ........ ................ + +2010-10-12 17:07 +0000 [r291266] Tilghman Lesher + + * /, main/acl.c: Merged revisions 291265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291265 | tilghman | 2010-10-12 12:06:23 -0500 + (Tue, 12 Oct 2010) | 16 lines Merged revisions 291264 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500 + (Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 + Oct 2010) | 2 lines Oops, incorrect range (although unallocated + at ARIN) ........ ................ ................ + +2010-10-12 16:08 +0000 [r291231] Leif Madsen + + * /, configs/manager.conf.sample: Merged revisions 291230 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291230 | lmadsen | 2010-10-12 11:08:04 -0500 + (Tue, 12 Oct 2010) | 10 lines Merged revisions 291229 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010) + | 2 lines Add documention that mentions options are defined but + not used. (Issue #18101) ........ ................ + +2010-10-12 16:00 +0000 [r291193-291228] David Vossel + + * main/manager.c, /: Merged revisions 291227 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291227 | dvossel | 2010-10-12 10:58:56 -0500 (Tue, 12 Oct 2010) + | 16 lines Fixes manager.c crash. This issue was caused by + improper use of the mansession lock and manession_session lock. + These two structures are confusing to begin with so I'm not + surprised this occurred. I fixed this by consistently making sure + we use each of these locks only to protect the data in the + corresponding structure. We had mismatched usage of these locks + which resulted in no mutual exclusivity occurring at all. (closes + issue #17994) Reported by: vrban Patches: + mansession_locking_fix.diff uploaded by dvossel (license 671) + Tested by: vrban ........ + + * /, CHANGES: Merged revisions 291194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291194 | dvossel | 2010-10-11 16:44:04 -0500 (Mon, 11 Oct 2010) + | 2 lines Update CHANGES to reflect new gtalk.conf options. + ........ + + * configs/gtalk.conf.sample, /, res/res_stun_monitor.c, + channels/chan_gtalk.c, include/asterisk/stun.h: Merged revisions + 291192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) + | 19 lines Gtalk enhancements and general code cleanup. This + patch includes several chan_gtalk enhancements. Two new + gtalk.conf options have been added, externip and stunadd. Setting + externip allows us to manually specify what the external IP + address is outside of a NAT environment. Setting the stunaddr + option to a valid stun server allows for that external ip to be + retrieved via a STUN server automatically. This external IP is + then advertised during call setup as a possible candidate. I have + also attempted to clean up chan_gtalk's code so it meets our + coding guidelines. During this cleanup I noticed several things + that need to be done in the code and made a TODO section at the + top of the file. ........ + +2010-10-11 19:07 +0000 [r291076-291115] Richard Mudgett + + * channels/chan_sip.c: Add todo comment about handle_incoming() + calling assumption. + + * /, channels/chan_sip.c: Merged revisions 291112-291113 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291112 | rmudgett | 2010-10-11 13:48:15 -0500 + (Mon, 11 Oct 2010) | 20 lines Merged revisions 291110-291111 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500 + (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 + Oct 2010) | 1 line Add missing unlock to an exception condition + in reload_config(). ........ ................ r291111 | rmudgett + | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit + from handle_request_do() consistent. ................ + ................ r291113 | rmudgett | 2010-10-11 13:51:13 -0500 + (Mon, 11 Oct 2010) | 1 line Move declaration closer to where now + used. ................ + + * /, main/cli.c: Merged revisions 291075 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291075 | rmudgett | 2010-10-11 11:42:54 -0500 + (Mon, 11 Oct 2010) | 22 lines Merged revisions 291073 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010) + | 15 lines Fixed infinite loop in verbose/debug message output. + Setting the module/filename specific message level and then + changing it resulted in the linked list being looped on itself. + Traversing this linked list is an infinite loop if what you are + looking for is not in the list. Also plugged some CLI parsing + holes in the associated CLI command: * Removing a nonexistent + module from the list actually added it with a level of zero. * + Setting the non-module specific level to zero is now equivalent + to setting it to "off" as documented. ........ ................ + +2010-10-11 03:20 +0000 [r291039] Tilghman Lesher + + * /, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Merged + revisions 291038 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291038 | tilghman | 2010-10-09 18:25:37 -0500 (Sat, 09 Oct 2010) + | 2 lines Add missing option to set calls to be logged in + GMT/UTC. ........ + +2010-10-09 14:04 +0000 [r291006] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, + addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling + options per user and peer. Added options for faststart/h.245 + tunneling per user/peer, properly handle these and global + options, correction of handling fs/tunneling fields in signalling + responses (closes issue #17972) Reported by: salecha Patches: + fs-tunnel-per-point-3.patch uploaded by may213 (license 454) + Tested by: may213, salecha + +2010-10-08 20:45 +0000 [r290974] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 290973 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290973 | dvossel | 2010-10-08 15:44:59 -0500 (Fri, 08 + Oct 2010) | 12 lines Make outbound Google Voice calls. This patch + allows for outbound Google Voice calls to be dialed from Asterisk + using chan_gtalk. Below is an example dialstring. exten -> + blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In + this example, 'asterisk' is the jabber.conf profile configured to + connect to your gmail account. In order to receive Google Voice + calls make sure to enable 'allowguest=yes' in gtalk.conf. + ........ + +2010-10-08 16:27 +0000 [r290939] Erin Spiceland + + * addons/app_mysql.c, configs/res_config_mysql.conf.sample, /, + addons/res_config_mysql.c: Add option to res_config_mysql and + app_mysql to specify a character set that MySQL should use. + (closes issue 17948) Reported by qmax. + +2010-10-08 03:00 +0000 [r290865] Jeff Peeler + + * /, main/asterisk.c: Merged revisions 290864 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290864 | jpeeler | 2010-10-07 21:56:24 -0500 + (Thu, 07 Oct 2010) | 23 lines Merged revisions 290863 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500 + (Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) + | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed + at control console. A recent change was made to avoid a race + condition on shutdown which only called the end functions from + the console thread. However, when pressing Ctrl-C the quit + handler is called from the signal handler thread. (closes issue + #17698) Reported by: jmls ........ ................ + ................ + +2010-10-07 22:39 +0000 [r290830-290831] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 290829 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290829 | dvossel | 2010-10-07 17:38:05 -0500 (Thu, 07 + Oct 2010) | 6 lines Add Philippe Sultan to chan_gtalk author + list. Philippe has made some notable contributions to the gtalk + channel driver. His name deserves to be listed amoung the authors + of that file. Thanks Philippe! ........ + + * /, channels/chan_gtalk.c: Merged revisions 290828 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290828 | dvossel | 2010-10-07 16:44:58 -0500 (Thu, 07 + Oct 2010) | 5 lines Outbound gtalk calls now work correctly. + There was a problem with how the candidates were being built on + an outbound call. This patch fixes that. ........ + +2010-10-07 20:59 +0000 [r290753] Jason Parker + + * /, configure, include/asterisk/autoconfig.h.in, + autoconf/ast_ext_lib.m4: Merged revisions 290752 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290752 | qwell | 2010-10-07 15:58:47 -0500 + (Thu, 07 Oct 2010) | 23 lines Merged revisions 290751 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290751 | qwell | 2010-10-07 15:57:14 -0500 + (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | + 9 lines Allow PRI to build properly when using --with-pri. Use + the directories found for the parent when using lib dependencies. + (closes issue #17314) Reported by: tzafrir Patches: + 17314-withdeps.diff uploaded by qwell (license 4) ........ + ................ ................ + +2010-10-07 11:12 +0000 [r290714] Russell Bryant + + * main/pbx.c, /: Merged revisions 290713 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290713 | russell | 2010-10-07 13:00:52 +0200 + (Thu, 07 Oct 2010) | 11 lines Merged revisions 290712 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010) + | 4 lines Don't crash when Set() is called without a value. + Review: https://reviewboard.asterisk.org/r/949/ ........ + ................ + +2010-10-06 21:23 +0000 [r290649-290677] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 290674 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290674 | dvossel | 2010-10-06 16:22:51 -0500 (Wed, 06 + Oct 2010) | 4 lines Fixes commented out code to use #if 0 + instead. Thanks to rmudgett for catching this! ........ + + * /, channels/chan_gtalk.c: Merged revisions 290648 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290648 | dvossel | 2010-10-06 16:08:19 -0500 (Wed, 06 + Oct 2010) | 12 lines Fixes gtalk outbound DTMF to work properly. + Outbound DTMF with gtalk needs to be done within the RTP stream. + I discovered this after investigating a packet capture from the + gmail client. Instead of performing jingle signaling DTMF, the + gtalk servers expect all DTMF to arrive on the RTP stream using + RFC2833 way of doing things. Chan_gtalk also had an issue with + negotiating RTP payload type 106 for the telephony-event and then + sending DTMF as payload 101. This has been resolved by always + negotiating 101 as the payload type like we do everywhere else. + With this patch, incoming google voice calls forwarded to + Asterisk via gtalk work. ........ + +2010-10-06 18:56 +0000 [r290615] Richard Mudgett + + * apps/app_dial.c, /: Merged revisions 290614 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290614 | rmudgett | 2010-10-06 13:50:37 -0500 + (Wed, 06 Oct 2010) | 12 lines Merged revision 290613 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed, + 06 Oct 2010) | 5 lines Eliminate a redundant test for + AST_CONTROL_REDIRECTING. Eliminate redundant test for + AST_CONTROL_REDIRECTING that prevents running the redirecting + interception macro if it is defined. .......... ................ + +2010-10-06 13:50 +0000 [r290577] Tilghman Lesher + + * /, main/file.c: Merged revisions 290576 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290576 | tilghman | 2010-10-06 08:49:19 -0500 + (Wed, 06 Oct 2010) | 15 lines Merged revisions 290575 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010) + | 8 lines Allow streaming audio from a pipe. (closes issue + #18001) Reported by: jamicque Patches: + 20100926__issue18001.diff.txt uploaded by tilghman (license 14) + Tested by: jamicque ........ ................ + +2010-10-06 04:47 +0000 [r290543] Terry Wilson + + * res/res_rtp_asterisk.c, /: Merged revisions 290542 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290542 | twilson | 2010-10-05 21:35:51 -0700 (Tue, 05 + Oct 2010) | 6 lines Don't try to send RTP when remote_address is + null It is possible for ast_rtp_stop() to be called which will + clear the remote address and cause the sendto to fail and spam + warnings. Don't send in this case. ........ + +2010-10-05 22:23 +0000 [r290480-290509] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 290506 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290506 | dvossel | 2010-10-05 17:23:00 -0500 (Tue, 05 + Oct 2010) | 2 lines Fixes uninitialized memory problem in 'iax2 + set debug peer' option. ........ + + * /, include/asterisk/jabber.h, include/asterisk/jingle.h, + channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 290479 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r290479 | dvossel | 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010) + | 6 lines Fixes chan_gtalk to work with gmail client This patch + was written by Philippe Sultan (phsultan). Thanks for keeping + this up to date! ........ + +2010-10-05 20:24 +0000 [r290414] Tilghman Lesher + + * /, res/res_jabber.c: Merged revisions 290408 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290408 | tilghman | 2010-10-05 15:23:33 -0500 + (Tue, 05 Oct 2010) | 22 lines Merged revisions 290396 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500 + (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010) + | 8 lines Fix a crash by ensuring that we don't alter memory + after it's freed. (closes issue #17387) Reported by: jmls + Patches: 20100726__issue17387.diff.txt uploaded by tilghman + (license 14) Tested by: jmls ........ ................ + ................ + +2010-10-05 20:10 +0000 [r290377-290379] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 290378 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290378 | dvossel | 2010-10-05 15:09:06 -0500 (Tue, 05 + Oct 2010) | 11 lines Resolves dnsmgr memory corruption in + chan_iax2. (closes issue #17902) Reported by: afried Patches: + issue_17902.rev1.txt uploaded by russell (license 2) Tested by: + afried, russell, dvossel Review: + https://reviewboard.asterisk.org/r/965/ ........ + + * /, apps/app_directed_pickup.c: Merged revisions 290376 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290376 | dvossel | 2010-10-05 14:56:29 -0500 + (Tue, 05 Oct 2010) | 16 lines Merged revisions 290375 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010) + | 10 lines Fixes PickupChan() not working with full channel name. + (closes issue #18011) Reported by: schern Patches: + app_directed_pickup.c.2.patch uploaded by schern (license 995) + app_directed_pickup.c.trunk.patch uploaded by schern (license + 995) Tested by: schern, dvossel ........ ................ + +2010-10-05 14:17 +0000 [r290067-290291] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 290289 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290289 | tilghman | 2010-10-05 09:15:46 -0500 (Tue, 05 + Oct 2010) | 2 lines Restore run directory for OS X, as well as + standardizing some other paths to Mac OS X. ........ + + * res/ael/pval.c, main/pbx.c, pbx/ael/ael-test/ref.ael-vtest17, /, + pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, + pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, + pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, + pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 290255 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290255 | tilghman | 2010-10-04 18:23:11 -0500 + (Mon, 04 Oct 2010) | 18 lines Merged revisions 290254 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010) + | 11 lines Change new pattern matcher to regard dashes the same + as the old pattern matcher -- as visual candy to be ignored. Also + change the AEL parser to not generate dashes within extensions, + as those dashes would be ignored. Update the AEL tests to match + this behavior. (closes issue #17366) Reported by: murf Patches: + 20100727__issue17366.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ ................ + + * /, configure, configure.ac: Merged revisions 290209 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290209 | tilghman | 2010-10-04 15:23:13 -0500 + (Mon, 04 Oct 2010) | 16 lines Merged revisions 290201 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500 + (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04 + Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........ + ................ ................ + + * /, configure, configure.ac: Merged revisions 290102 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290102 | tilghman | 2010-10-03 16:08:45 -0500 + (Sun, 03 Oct 2010) | 16 lines Merged revisions 290101 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500 + (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03 + Oct 2010) | 2 lines Automatically re-run configure test for + menuselect, when the relevant makeopts settings change. ........ + ................ ................ + + * pbx/pbx_spool.c, /: Merged revisions 290066 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r290066 | tilghman | 2010-10-03 15:02:29 -0500 (Sun, 03 Oct 2010) + | 8 lines Get notification only when file is closed, not when + created. (closes issue #17924) Reported by: mkeuter Patches: + asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946) + Tested by: abelbeck ........ + +2010-10-02 18:06 +0000 [r290027] Kevin P. Fleming + + * contrib/scripts/get_mp3_source.sh, /: Merged revisions 290026 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r290026 | kpfleming | 2010-10-02 12:57:13 -0500 (Sat, 02 Oct + 2010) | 6 lines Allow users to pass additional arguments to the + Subversion command that obtains the MP-3 source code. (reported + on IRC by jmls) ........ + +2010-10-02 08:58 +0000 [r289952] Olle Johansson + + * main/manager.c, /: Merged revisions 289951 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289951 | oej | 2010-10-02 10:56:08 +0200 (Lör, + 02 Okt 2010) | 16 lines Merged revisions 289950 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör, + 02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 + lines Add documentation for undocumented option to AMI action + originate ........ ................ ................ + +2010-10-02 04:54 +0000 [r289876] Tilghman Lesher + + * /, apps/app_voicemail.c: Merged revisions 289875 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289875 | tilghman | 2010-10-01 23:46:43 -0500 + (Fri, 01 Oct 2010) | 22 lines Merged revisions 289874 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500 + (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) + | 8 lines When forwarding a message, a prepend means that the + filesystem will always have a better copy. (closes issue #17803) + Reported by: dpetersen Patches: 20100923__issue17803.diff.txt + uploaded by tilghman (license 14) Tested by: dpetersen ........ + ................ ................ + +2010-10-02 02:46 +0000 [r289841] Jeff Peeler + + * res/res_rtp_asterisk.c, main/rtp_engine.c, /, + channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged + revisions 289840 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289840 | jpeeler | 2010-10-01 21:43:45 -0500 + (Fri, 01 Oct 2010) | 29 lines Merged revisions 289798 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 + (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) + | 15 lines Change RFC2833 DTMF event duration on end to report + actual elapsed time. The scenario here is with a non P2P early + media session. The reported time length of DTMF presses are + coming up short when sending to the remote side. Currently the + event duration is a running total that is incremented when + sending continuation packets. These continuation packets are only + triggered upon incoming media from the remote side, which means + that the running total probably is not going to end up matching + the actual length of time Asterisk received DTMF. This patch + changes the end event duration to be lengthened if it is detected + that the end event is going to come up short. Review: + https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ + ................ ................ + +2010-10-01 17:22 +0000 [r289732] Paul Belanger + + * /, configs/jabber.conf.sample, res/res_jabber.c: Merged revisions + 289718 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289718 | pabelanger | 2010-10-01 13:19:49 -0400 + (Fri, 01 Oct 2010) | 20 lines Merged revisions 289704 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400 + (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct + 2010) | 6 lines Disable debugging by default and reformat .config + file. Review: https://reviewboard.asterisk.org/r/929/ ........ + ................ ................ + +2010-10-01 16:23 +0000 [r289702] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 289701 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289701 | jpeeler | 2010-10-01 11:22:19 -0500 + (Fri, 01 Oct 2010) | 28 lines Merged revisions 289700 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500 + (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) + | 14 lines Ensure user portion of SIP URI matches dialplan when + using encoded characters. This commit takes a simliar approach to + 288112 and checks the dialplan to determine the proper action for + an incoming contact header as to whether or not it should be + decoded or not. sip_new was blindly always decoding the + extension, which also caused the outgoing contact header to be + incorrect as well as failing to match the encoded extension in + the dialplan. (closes issue #17892) Reported by: wdoekes Patches: + bug17892-1.patch uploaded by jpeeler (license 325) Tested by: + wdoekes ........ ................ ................ + +2010-10-01 10:04 +0000 [r289623] Stefan Schmidt + + * channels/chan_sip.c: don't iterate through all dialogs to find + and delete old subscribes On every incoming subscribe there is a + iteration through all dialogs to find old subscribes and delete + them. This is slow and not RFC conform. This was only needed in + 1.2 cause a subscribe was not deleted when a dialog was + destroyed, after 1.4 a subscribe get removed when its dialog is + destroyed. Review: https://reviewboard.asterisk.org/r/901/ + +2010-09-30 20:40 +0000 [r289588] Tilghman Lesher + + * /, tests/test_time.c, funcs/func_env.c, tests/test_utils.c, + res/res_agi.c, include/asterisk/localtime.h, + main/stdtime/localtime.c: Merged revisions 289543,289581 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289543 | tilghman | 2010-09-30 12:50:52 -0500 (Thu, 30 Sep 2010) + | 2 lines More Solaris compatibility fixes ........ r289581 | + tilghman | 2010-09-30 15:23:10 -0500 (Thu, 30 Sep 2010) | 2 lines + Solaris fixes. ........ + +2010-09-30 19:54 +0000 [r289555] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 289554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289554 | mnicholson | 2010-09-30 14:53:10 -0500 + (Thu, 30 Sep 2010) | 11 lines Merged revisions 289553 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep + 2010) | 4 lines Properly handle channel allocation failures duing + invites with replaces. ABE-2588 ........ ................ + +2010-09-30 19:35 +0000 [r289552] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 289549 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289549 | rmudgett | 2010-09-30 14:28:36 -0500 + (Thu, 30 Sep 2010) | 17 lines Merged revision 289547 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu, + 30 Sep 2010) | 10 lines In chan_misdn, the + DivertingLegInformation2 DivertingNr is garbage when the number + is restricted. The same thing happens with + DivertingLegInformation1 DivertedTo number. The + misdn_PresentedNumberUnscreened_extract() extracted the + Unscreened PartyNumber field unconditionally. It now checks the + presented number unscreened type to see if the PartyNumber was + even present. JIRA ABE-2595 .......... ................ + +2010-09-30 15:40 +0000 [r289427] Russell Bryant + + * /, apps/app_sms.c: Merged revisions 289426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289426 | russell | 2010-09-30 10:39:45 -0500 + (Thu, 30 Sep 2010) | 22 lines Merged revisions 289425 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289425 | russell | 2010-09-30 10:37:29 -0500 + (Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) + | 8 lines Fix a crash in app_sms. Since the data being passed to + the generator callback is on the stack of the SMS() application, + we must ensure that the generator is stopped before the + application exits. ABE-2587 ........ ................ + ................ + +2010-09-29 21:19 +0000 [r289354] Jason Parker + + * main/channel.c, /, main/features.c: Merged revisions 289340 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289340 | qwell | 2010-09-29 16:12:43 -0500 + (Wed, 29 Sep 2010) | 22 lines Merged revisions 289339 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289339 | qwell | 2010-09-29 16:03:47 -0500 + (Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | + 8 lines Allow a manager originate to succeed on forwarded + devices. The timeout to wait for an answer was being set to 0 + when a device forwarded to another extension. We don't always + need the timeout set like this, so make it an optional parameter, + and don't use it in this case. ABE-2544 ........ ................ + ................ + +2010-09-29 20:29 +0000 [r289337] Leif Madsen + + * /, configs/res_ldap.conf.sample: Merged revisions 289336 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289336 | lmadsen | 2010-09-29 15:27:25 -0500 + (Wed, 29 Sep 2010) | 9 lines Merged revisions 289334 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 + Sep 2010) | 1 line Update sample documentation to note md5secret + requirements. ........ ................ + +2010-09-29 20:24 +0000 [r289335] Russell Bryant + + * /, res/res_config_ldap.c: Merged revisions 289333 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289333 | russell | 2010-09-29 15:20:23 -0500 + (Wed, 29 Sep 2010) | 11 lines Merged revisions 289332 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29 Sep 2010) + | 4 lines Don't completely ignore md5secret from LDAP if the + value does not begin with {md5}. This fixes a problem that + lmadsen ran in to where md5secret was not working for him. + ........ ................ + +2010-09-29 17:54 +0000 [r289269-289301] Matthew Nicholson + + * /, configs/res_fax.conf.sample: Merged revisions 289300 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289300 | mnicholson | 2010-09-29 12:53:54 -0500 (Wed, 29 Sep + 2010) | 2 lines Add 'ecm' to the sample fax config file ........ + + * main/channel.c, /: Merged revisions 289268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289268 | mnicholson | 2010-09-29 12:08:20 -0500 (Wed, 29 Sep + 2010) | 5 lines Update the CDR record when + ast_channel_set_caller_event() is called (related to issue + #17569) Reported by: tbelder ........ + +2010-09-29 16:17 +0000 [r289254] Richard Mudgett + + * main/channel.c, /: Merged revisions 289253 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289253 | rmudgett | 2010-09-29 11:16:47 -0500 (Wed, 29 Sep 2010) + | 1 line Make development error message indicate which channel. + ........ + +2010-09-29 15:07 +0000 [r289180] Matthew Nicholson + + * main/channel.c, /: Merged revisions 289179 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289179 | mnicholson | 2010-09-29 10:04:56 -0500 + (Wed, 29 Sep 2010) | 22 lines Merged revisions 289178 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500 + (Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep + 2010) | 8 lines Set the caller id on CDRs when it is set on the + parent channel. (closes issue #17569) Reported by: tbelder + Patches: 17569.diff uploaded by tbelder (license 618) Tested by: + tbelder ........ ................ ................ + +2010-09-28 18:24 +0000 [r289131] Brett Bryant + + * main/channel.c, /: Merged revisions 289099 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289099 | bbryant | 2010-09-28 14:18:02 -0400 + (Tue, 28 Sep 2010) | 28 lines Merged revisions 289095 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289095 | bbryant | 2010-09-28 14:14:19 -0400 + (Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) + | 14 lines Fixes an issue with the Newchannel AMI event during + the Masquerading process. Fixes an issue with the Newchannel AMI + event during the Masquerading process, where no Newchannel AMI + event was generated for the psuedo channel used during the + masquerading process. (closes issue #17987) Reported by: + RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish + (license 1122) Tested by: RadicAlish Review: + https://reviewboard.asterisk.org/r/937/ ........ ................ + ................ + +2010-09-28 18:20 +0000 [r289112] Tilghman Lesher + + * Makefile, /, tests/test_time.c, configure, + include/asterisk/autoconfig.h.in, include/asterisk/compat.h, + main/strcompat.c, tests/test_utils.c, configure.ac, makeopts.in, + apps/app_voicemail.c: Merged revisions 289104 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289104 | tilghman | 2010-09-28 13:18:43 -0500 (Tue, 28 Sep 2010) + | 4 lines Solaris compatibility fixes Review: + https://reviewboard.asterisk.org/r/942/ ........ + +2010-09-28 01:10 +0000 [r289056-289058] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 289057 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289057 | rmudgett | 2010-09-27 20:04:37 -0500 (Mon, 27 Sep 2010) + | 5 lines Avoid deadlock processing incoming AOC-E messages. + Deadlock avoidance for the owner channel was not done when + processing incoming AOC-E messages. ........ + + * /, channels/chan_sip.c: Merged revisions 289054-289055 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289054 | rmudgett | 2010-09-27 19:32:18 -0500 (Mon, 27 Sep 2010) + | 1 line Break up long ast_manager_event_multichan() event lines. + ........ r289055 | rmudgett | 2010-09-27 19:35:25 -0500 (Mon, 27 + Sep 2010) | 1 line Revert stuff not ready for commit in -r289054. + ........ + +2010-09-27 22:03 +0000 [r289023] David Vossel + + * channels/chan_sip.c: For an INVITE transaction, treat all 2XX + responses the same as a 200. ABE-2305 + +2010-09-27 19:45 +0000 [r288992-288993] Olle Johansson + + * channels/chan_sip.c: Formatting fixes + + * cdr/cdr_pgsql.c: Formating changes + +2010-09-27 18:39 +0000 [r288962] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 288961 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288961 | tilghman | 2010-09-27 13:37:41 -0500 (Mon, 27 Sep 2010) + | 5 lines Still build SIP, even if res_crypto cannot be built + (use, not depend). (closes issue #18062) Reported by: a user on + the mailing list ........ + +2010-09-27 13:04 +0000 [r288926-288928] Russell Bryant + + * /, res/res_agi.c: Merged revisions 288927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288927 | russell | 2010-09-27 08:03:43 -0500 (Mon, 27 Sep 2010) + | 2 lines Fix some documentation typos and spelling errors. + ........ + + * /, res/res_agi.c: Merged revisions 288925 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288925 | russell | 2010-09-27 07:42:10 -0500 (Mon, 27 Sep 2010) + | 2 lines Fix a documentation spelling error. ........ + +2010-09-25 07:58 +0000 [r288893] Alexandr Anikin + + * addons/ooh323c/src/oochannels.c: small correction for verbose + print h.323 packets + +2010-09-24 17:59 +0000 [r288822-288853] David Vossel + + * /, channels/chan_sip.c: Merged revisions 288852 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288852 | dvossel | 2010-09-24 12:58:57 -0500 (Fri, 24 Sep 2010) + | 5 lines Append Retry-After header on 500 error response to + Re-INVITE according to RFC3261 section 14.2. ABE-2301 ........ + + * /, channels/chan_sip.c: Merged revisions 288821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288821 | dvossel | 2010-09-24 12:05:12 -0500 (Fri, 24 Sep 2010) + | 4 lines Inspect Require header on BYE transaction according to + RFC3261 section 8.2.2.3. ABE-2293 ........ + +2010-09-24 16:11 +0000 [r288749] Terry Wilson + + * channels/chan_local.c, /: Merged revisions 288748 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288748 | twilson | 2010-09-24 09:02:27 -0700 + (Fri, 24 Sep 2010) | 19 lines Merged revisions 288747 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288747 | twilson | 2010-09-24 08:37:39 -0700 + (Fri, 24 Sep 2010) | 12 lines Merged revisions 288746 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) + | 5 lines Don't fail a masquerade if it is already being hung up + This avoids noise on some Local channel situations where we don't + use /n. Thanks to Alec Davis for the suggestion. ........ + ................ ................ + +2010-09-24 13:55 +0000 [r288607-288714] Tilghman Lesher + + * /, funcs/func_strings.c: Merged revisions 288713 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288713 | tilghman | 2010-09-24 08:54:17 -0500 + (Fri, 24 Sep 2010) | 12 lines Merged revisions 288712 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24 Sep 2010) + | 5 lines Solaris won't printf a NULL. (closes issue #18041) + Reported by: asgaroth ........ ................ + + * /, main/asterisk.exports.in: Merged revisions 288640 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r288640 | tilghman | 2010-09-23 22:42:37 -0500 (Thu, 23 + Sep 2010) | 2 lines Export timersub for platforms which do not + have it ........ + + * /, configure, include/asterisk/autoconfig.h.in, + include/asterisk/compat.h, main/strcompat.c, configure.ac, + include/asterisk/channel.h, cdr/cdr_pgsql.c: Merged revisions + 288638 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288638 | tilghman | 2010-09-23 22:39:29 -0500 + (Thu, 23 Sep 2010) | 16 lines Merged revisions 288637 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500 + (Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 + Sep 2010) | 2 lines Solaris compatibility fixes ........ + ................ ................ + + * /, CHANGES: Merged revisions 288606 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288606 | tilghman | 2010-09-23 13:44:44 -0500 (Thu, 23 Sep 2010) + | 2 lines Add note about the checkhangup option of ${CHANNEL()} + ........ + +2010-09-23 18:08 +0000 [r288519-288573] Terry Wilson + + * main/manager.c, /: Merged revisions 288572 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288572 | twilson | 2010-09-23 13:05:16 -0500 (Thu, 23 Sep 2010) + | 2 lines Make AMI honor enabled=no ........ + + * channels/chan_local.c, /: Merged revisions 288507 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288507 | twilson | 2010-09-22 16:18:27 -0700 + (Wed, 22 Sep 2010) | 22 lines Merged revisions 288500 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288500 | twilson | 2010-09-22 16:10:09 -0700 + (Wed, 22 Sep 2010) | 15 lines Merged revisions 288499 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) + | 8 lines Don't let a Local channel get bridged to itself If a + local channel gets bridged to itself, it becomes orphaned with no + devices left to actually tell it to hang up. This patch modifies + local_fixup() to detect this case and deny it. Review: + https://reviewboard.asterisk.org/r/934 ........ ................ + ................ + +2010-09-22 17:50 +0000 [r288346-288419] David Vossel + + * /, channels/chan_sip.c: Merged revisions 288418 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288418 | dvossel | 2010-09-22 12:49:56 -0500 + (Wed, 22 Sep 2010) | 18 lines Merged revisions 288417 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500 + (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) + | 5 lines RFC3261 section 12.2 explicitly says out of order + requests are responded with a 500 Server Internal Error response. + ABE-2458 ........ ................ ................ + + * /, channels/chan_sip.c: Merged revisions 288345 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288345 | dvossel | 2010-09-22 11:59:14 -0500 + (Wed, 22 Sep 2010) | 16 lines Merged revisions 288344 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500 + (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 + Sep 2010) | 2 lines During check_pendings, if the dialog is + terminated with a CANCEL, change the invitestate to INV_CANCEL + like in sip_hangup. ........ ................ ................ + +2010-09-22 16:46 +0000 [r288342] Russell Bryant + + * /, main/asterisk.c: Merged revisions 288341 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288341 | russell | 2010-09-22 11:45:18 -0500 + (Wed, 22 Sep 2010) | 25 lines Merged revisions 288340 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288340 | russell | 2010-09-22 11:44:13 -0500 + (Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) + | 11 lines Fix a 100% CPU consumption problem when setting + console=yes in asterisk.conf. The handling of -c and console=yes + should be the same, but they were not. When you specify -c, it + sets both a flag for console module and for asterisk not to + fork() off into the background. The handling of console=yes only + set console mode, so you would end up with a background process() + trying to run the Asterisk console and freaking out since it + didn't have anything to read input from. Thanks to beagles for + reporting and helping debug the problem! ........ + ................ ................ + +2010-09-22 15:18 +0000 [r288278] Tilghman Lesher + + * /, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Merged + revisions 288268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288268 | tilghman | 2010-09-22 10:14:02 -0500 + (Wed, 22 Sep 2010) | 30 lines Merged revisions 288267 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500 + (Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010) + | 9 lines Allow the encoding to be set, in case local charset + does not agree with database. (closes issue #16940) Reported by: + jamicque Patches: 20100827__issue16940.diff.txt uploaded by + tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt + uploaded by tilghman (license 14) Tested by: jamicque ........ + r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010) + | 5 lines Document addition of encoding parameter. (issue #16940) + Reported by: jamicque ........ ................ ................ + +2010-09-22 00:08 +0000 [r288195] Richard Mudgett + + * /, channels/chan_iax2.c: Merged revisions 288194 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288194 | rmudgett | 2010-09-21 19:06:21 -0500 + (Tue, 21 Sep 2010) | 40 lines Merged revisions 288193 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500 + (Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) + | 26 lines In chan_iax2.c:schedule_delivery() calls + ast_bridged_channel() on an unlocked channel. Near the beginning + of schedule_delivery(), ast_bridged_channel() is called on + iaxs[fr->callno]->owner. However, the channel is not locked, + which can result in ast_bridged_channel() crashing should + owner->tech change to a technology that doesn't implement + bridged_channel. I also fixed the other calls to + ast_bridged_channel() in chan_iax2.c since the owner lock was not + held there either. Converted the existing channel deadlock + avoidance to use iax2_lock_owner(). Using the new function + simplified some awkward code. In the process of fixing the + locking on ast_bridged_channel(), I also found a memory leak in + socket_process() for v1.6.2 and v1.8. The local struct variable + ies.vars is not freed on early/abnormal function exits. (closes + issue #17919) Reported by: rain Patches: issue17919_v1.4.patch + uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch + uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch + uploaded by rmudgett (license 664) Review: + https://reviewboard.asterisk.org/r/926/ ........ ................ + ................ + +2010-09-21 22:58 +0000 [r288160] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 288159 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288159 | tilghman | 2010-09-21 17:57:22 -0500 + (Tue, 21 Sep 2010) | 29 lines Merged revisions 288113 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500 + (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) + | 15 lines Try both the encoded and unencoded subscription URI + for a match in hints. When a phone sends an encoded URI for a + subscription, the URI is not matched with the actual hint that is + in decoded format. For example, if we have an extension with a + hint that is named: "#5601" or "*5601", the subscription will + work fine if the phone subscribes with an already decoded URI, + but when it's decoded like "%255601" or "%2A5601", Asterisk is + unable to match it with the correct hint. (closes issue #17785) + Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt + uploaded by tilghman (license 14) Tested by: ramonpeek ........ + ................ ................ + +2010-09-21 22:28 +0000 [r288158] Paul Belanger + + * /, channels/chan_iax2.c: Merged revisions 288157 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288157 | pabelanger | 2010-09-21 18:26:15 -0400 + (Tue, 21 Sep 2010) | 15 lines Merged revisions 288147 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue, 21 Sep + 2010) | 9 lines Setup timer before set_config(). (closes issue + #18019) Reported by: Netview Patches: issue_0018019.patch + uploaded by pabelanger (license 224) Tested by: Netview ........ + ................ + +2010-09-21 21:04 +0000 [r288081-288083] Richard Mudgett + + * /, doc/tex/partymanip.tex: Merged revisions 288082 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r288082 | rmudgett | 2010-09-21 16:03:28 -0500 (Tue, 21 + Sep 2010) | 1 line Add note in party manipulation chapter on + interception macros. ........ + + * apps/app_dial.c, main/channel.c, /, apps/app_queue.c: Merged + revisions 288079-288080 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) + | 2 lines Protect channel access in CONNECTED_LINE and + REDIRECTING interception macro launch code. ........ r288080 | + rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines + Simplify locking code for REDIRECTING interception macro when + forwarding a call. Simplified the locking code by using a local + copy of the redirecting party information in + app_dial.c:do_forward() and app_queue.c:wait_for_answer() for + launching the REDIRECTING interception macro when a call is + forwarded. Reduced the lock time of the 'o->chan' and 'in' + channels. ........ + +2010-09-21 20:27 +0000 [r288063] Stefan Schmidt + + * channels/chan_sip.c: Instead of iterate through all dialogs, add + two separte container for needdestroy and rtptimeout adding two + dialog container, one for dialogs which need destroy, another for + rtptimeout checks. both container will be checked on every loop + of do_monitor instead of iterate through all dialogs. (closes + issue #17912) Reported by: schmidts Tested by: schmidts Review: + https://reviewboard.asterisk.org/r/917/ + +2010-09-21 19:50 +0000 [r288008] Brett Bryant + + * main/channel.c, /: Merged revisions 288007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288007 | bbryant | 2010-09-21 15:48:53 -0400 + (Tue, 21 Sep 2010) | 21 lines Merged revisions 288006 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400 + (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) + | 8 lines Add a check to fix a rare segmentation fault you'd get + if ast_frdup couldn't allocate memory on the first frame being + queued in ast_queue_frame. (closes issue #17882) Reported by: + seanbright Tested by: seanbright ........ ................ + ................ + +2010-09-21 19:09 +0000 [r287936] Tilghman Lesher + + * /, main/asterisk.c: Merged revisions 287935 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287935 | tilghman | 2010-09-21 14:08:36 -0500 + (Tue, 21 Sep 2010) | 16 lines Merged revisions 287934 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500 + (Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 + Sep 2010) | 2 lines Less than zero is an error, not any non-zero + value. ........ ................ ................ + +2010-09-21 19:04 +0000 [r287932] Terry Wilson + + * main/channel.c, /: Merged revisions 287931 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287931 | twilson | 2010-09-21 14:02:40 -0500 (Tue, 21 Sep 2010) + | 2 lines Revert change in favor of a more targeted fix ........ + +2010-09-21 18:33 +0000 [r287930] David Vossel + + * /, channels/chan_sip.c: Merged revisions 287929 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287929 | dvossel | 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010) + | 4 lines Send a "415 Unsupported Media Type" after failure to + process sdp due to unknown Content-Encoding header. ABE-2258 + ........ + +2010-09-21 15:54 +0000 [r287898] Richard Mudgett + + * /, main/features.c: Merged revisions 287897 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287897 | rmudgett | 2010-09-21 10:53:19 -0500 (Tue, 21 Sep 2010) + | 1 line Cut-n-paste error in builtin_blindtransfer(). ........ + +2010-09-21 15:45 +0000 [r287896] Russell Bryant + + * res/res_rtp_asterisk.c, main/dnsmgr.c, /, channels/chan_sip.c, + main/acl.c: Merged revisions 287895 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287895 | russell | 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010) + | 10 lines Don't use ast_strdupa() from within the arguments to a + function. (closes issue #17902) Reported by: afried Patches: + issue_17902.rev1.txt uploaded by russell (license 2) Tested by: + russell Review: https://reviewboard.asterisk.org/r/927/ ........ + +2010-09-21 15:27 +0000 [r287894] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 287893 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287893 | tilghman | 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010) + | 9 lines Anonymous callerid needs a "sip:" uri prefix. (closes + issue #17981) Reported by: avalentin Patches: + sip-anonymous-aastra.patch uploaded by avalentin (license 1107) + (plus an additional fix by me) Tested by: avalentin ........ + +2010-09-21 13:45 +0000 [r287864] Russell Bryant + + * /, main/logger.c: Merged revisions 287863 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287863 | russell | 2010-09-21 08:41:41 -0500 (Tue, 21 Sep 2010) + | 2 lines Fix a regression in verbose logger processing. ........ + +2010-09-21 04:39 +0000 [r287764-287834] Terry Wilson + + * main/channel.c, /: Merged revisions 287833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287833 | twilson | 2010-09-20 23:37:44 -0500 (Mon, 20 Sep 2010) + | 3 lines Don't generate connected line buffer twice for + comparison ........ + + * main/channel.c, /: Merged revisions 287757 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287757 | twilson | 2010-09-20 18:51:38 -0500 (Mon, 20 Sep 2010) + | 7 lines Avoid infinite loop with certain local channel + connected line updates Compare connected line data before sending + a connected line indication to avoid possible loops. Review: + https://reviewboard.asterisk.org/r/932/ ........ + +2010-09-21 00:04 +0000 [r287763] Brett Bryant + + * /, apps/app_meetme.c: Merged revisions 287760 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287760 | bbryant | 2010-09-20 20:00:23 -0400 + (Mon, 20 Sep 2010) | 30 lines Merged revisions 287759 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400 + (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) + | 16 lines Fix misvalidation of meetme pins in conjunction with + the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a + user and admin pin setup for your conference, using the user pin + would gain you admin priviledges. Also, when no user pin was set, + an admin pin was, the 'a' MeetMe flag wasn't used, and the user + tried to enter a conference then they were still prompted for a + pin and forced to hit #. (closes issue #17908) Reported by: kuj + Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: + kuj Review: [full review board URL with trailing slash] ........ + ................ ................ + +2010-09-21 00:01 +0000 [r287761-287762] Terry Wilson + + * /: Add alecdavis' commit to merged props + + * /: Add merge properties back. + +2010-09-20 23:42 +0000 [r287756] Alec L Davis + + * main/channel.c, /: Merged revisions 287685 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep + 2010) | 18 lines ast_channel_masquerade: Avoid recursive + masquerades. Check all 4 combinations of (original/clonechan) * + (masq/masqr). Initially original->masq and clonechan->masqr were + only checked. It's possible with multiple masq's planned - and + not yet executed, that the 'original' chan could already have + another masq'd into it - thus original->masqr would be set, that + masqr would lost. Likewise for the clonechan->masq. (closes issue + #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches: + based on bug16057.diff4.txt uploaded by alecdavis (license 585) + Tested by: ramonpeek, davidw, alecdavis ........ + +2010-09-20 23:18 +0000 [r287693] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 287683 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r287683 | rmudgett | 2010-09-20 18:14:42 -0500 (Mon, 20 + Sep 2010) | 9 lines The inalarm flag was not set in sig_analog + struct if the port is initially in alarm. Fixed initial inalarm + value for sig_analog ports. Along with -r261007, this gets the + inalarm flag in sync with chan_dahdi for sig_analog ports. + (closes issue #16983) ........ + +2010-09-20 22:24 +0000 [r287671] Alec L Davis + + * main/channel.c, /: Merged revisions 287661 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287661 | alecdavis | 2010-09-21 10:21:50 +1200 (Tue, 21 Sep + 2010) | 14 lines ast_do_masquerade. Keep channels ao2_container + locked while unlink and linking channels. Previously, Masquerade + would unlock 'original' and 'clonechan' and allow another masq + thread to run. End result would be corrupted memory, and the + frequent report 'Bad Magic Number'. (closes issue #17801,#17710) + Reported by: notthematrix Patches: Based on bug17801.diff1.txt + uploaded by alecdavis (license 585) Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/928 ........ + +2010-09-20 22:16 +0000 [r287646-287648] David Vossel + + * main/channel.c, main/framehook.c (added), /, + funcs/func_frame_trace.c (added), include/asterisk/channel.h, + CHANGES, include/asterisk/framehook.h (added): Merged revisions + 287647 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) + | 21 lines Addition of the FrameHook API (AKA AwesomeHooks) So + far all our tools for viewing and manipulating media streams + within Asterisk have been entirely focused on audio. That made + sense then, but is not scalable now. The FrameHook API lets us + tap into and manipulate _ANY_ type of media or signaling passed + on a channel present today or in the future. This tool is a step + in the direction of expanding Asterisk's boundaries and will help + generate some rather interesting applications in the future. In + addition to the FrameHook API, a simple dialplan function + exercising the api has been included as well. This function is + called FRAME_TRACE(). FRAME_TRACE() allows for the internal + ast_frames read and written to a channel to be output. Filters + can be placed on this function to debug only certain types of + frames. This function could be thought of as an internal way of + doing ast_frame packet captures. Review: + https://reviewboard.asterisk.org/r/925/ ........ + + * /, channels/chan_sip.c: Merged revisions 287645 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287645 | dvossel | 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010) + | 9 lines Fixes issue with registrations not working properly + with pedantic=yes. (closes issue #18017) Reported by: schmidts + Patches: issues_18017_v1.diff uploaded by dvossel (license 671) + Tested by: schmidts ........ + +2010-09-20 21:30 +0000 [r287644] Jason Parker + + * /, channels/chan_skinny.c: Merged revisions 287643 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287643 | qwell | 2010-09-20 16:29:46 -0500 + (Mon, 20 Sep 2010) | 15 lines Merged revisions 287642 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) | + 8 lines Don't crash when parking a non-bridged call. (closes + issue #17680) Reported by: jmhunter Patches: + chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by: + jmhunter, DEA ........ ................ + +2010-09-20 21:25 +0000 [r287640] Brett Bryant + + * /, main/logger.c: Merged revisions 287639 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287639 | bbryant | 2010-09-20 17:19:12 -0400 (Mon, 20 Sep 2010) + | 8 lines Fixes an error with the logger that caused verbose + messages to be spammed to the screen if syslog was configured in + logger.conf (closes issue #17974) Reported by: lmadsen Review: + https://reviewboard.asterisk.org/r/915/ ........ + +2010-09-20 15:57 +0000 [r287560] Matthew Nicholson + + * main/pbx.c, /: Merged revisions 287559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287559 | mnicholson | 2010-09-20 10:57:14 -0500 + (Mon, 20 Sep 2010) | 21 lines Merged revisions 287558 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500 + (Mon, 20 Sep 2010) | 14 lines Use ast_str when processing hint + state changes Merged revisions 287555 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep + 2010) | 5 lines Use ast_dynamic_str when processing hint state + changes (related to issue #17928) Reported by: mdu113 ........ + ................ ................ + +2010-09-19 16:12 +0000 [r287472] Olle Johansson + + * main/manager.c, /: Merged revisions 287471 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287471 | oej | 2010-09-19 18:09:28 +0200 (Sön, + 19 Sep 2010) | 21 lines Merged revisions 287470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön, + 19 Sep 2010) | 14 lines Merged revisions 287469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 + lines Make sure we always free variables properly in manager + originate. (closes issue #17891) reported, solved and tested by + oej Review: https://reviewboard.asterisk.org/r/869/ ........ + ................ ................ + +2010-09-17 21:10 +0000 [r287389] Tilghman Lesher + + * /, apps/app_queue.c: Merged revisions 287388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287388 | tilghman | 2010-09-17 16:08:54 -0500 + (Fri, 17 Sep 2010) | 21 lines Merged revisions 287387 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500 + (Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) + | 7 lines Blank columns should get set on reload, not ignored. + (closes issue #16893) Reported by: haakon Patches: + 20100818__issue16893.diff.txt uploaded by tilghman (license 14) + ........ ................ ................ + +2010-09-17 13:38 +0000 [r287310] Matthew Nicholson + + * main/pbx.c, /: Merged revisions 287309 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287309 | mnicholson | 2010-09-17 08:37:10 -0500 + (Fri, 17 Sep 2010) | 19 lines Merged revisions 287308 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500 + (Fri, 17 Sep 2010) | 12 lines Merged revisions 287307 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep + 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while + processing in ast_hint_state_changed(). (related to issue #17928) + Reported by: mdu113 ........ ................ ................ + +2010-09-17 08:46 +0000 [r287272] Jan Kalab + + * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, /, + res/res_calendar_caldav.c, res/res_calendar_ews.c: Merged + revisions 287269-287271 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287269 | pitel | 2010-09-17 10:37:49 +0200 (Pá, 17 zář 2010) | 8 + lines Support for HTTP redirects in calendar's URL libneon does + not support HTTP redirects (3xx responses) by default. You must + tell it to follow them. Also, another little unsigned int fix. + (closes issue #17776) Review: + https://reviewboard.asterisk.org/r/921/ ........ r287270 | pitel + | 2010-09-17 10:42:37 +0200 (Pá, 17 zář 2010) | 6 lines Asterisk + crashing because of double free when EWS request fails The free + is done later in code. I think ast_free() should have built in + checks for double free. (closes issue #17782) ........ r287271 | + pitel | 2010-09-17 10:44:28 +0200 (Pá, 17 zář 2010) | 6 lines + Events are visible after they were removed from EWS calendar + Because we must merge calendar even when it's empty. (closes + issue #17786) ........ + +2010-09-16 22:05 +0000 [r287196] Jason Parker + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 287195 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287195 | qwell | 2010-09-16 17:04:38 -0500 (Thu, 16 Sep 2010) | + 7 lines Don't fail when running the Debian init script directly + (as one would normally do). readlink apparently returns 1 when + the arg isn't a symlink, which caused the script to exit. (closes + issue #17910) Reported by: wurstsalat ........ + +2010-09-16 22:00 +0000 [r287194] Russell Bryant + + * /, configs/queues.conf.sample, apps/app_queue.c, UPGRADE-1.8.txt: + Merged revisions 287193 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) + | 4 lines Set the default for "autofill" and "shared_lastcall" to + "yes" in queues.conf. Review: + https://reviewboard.asterisk.org/r/922/ ........ + +2010-09-16 20:08 +0000 [r287117-287121] Matthew Nicholson + + * main/pbx.c, /: Merged revisions 287120 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287120 | mnicholson | 2010-09-16 15:07:38 -0500 + (Thu, 16 Sep 2010) | 22 lines Merged revisions 287119 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500 + (Thu, 16 Sep 2010) | 15 lines Merged revisions 287118 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep + 2010) | 8 lines Don't limit hint processing in + ast_hint_state_changed() to AST_MAX_EXTENSION length strings. + (closes issue #17928) Reported by: mdu113 Patches: + 20100831__issue17928.diff.txt uploaded by tilghman (license 14) + Tested by: mdu113 ........ ................ ................ + + * main/cdr.c, /: Merged revisions 287116 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287116 | mnicholson | 2010-09-16 14:54:48 -0500 + (Thu, 16 Sep 2010) | 22 lines Merged revisions 287115 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500 + (Thu, 16 Sep 2010) | 15 lines Merged revisions 287114 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep + 2010) | 8 lines Don't stop printing cdr variables if we encounter + one with a blank name or value. (closes issue #17900) Reported + by: under Patches: core-show-channel-cdr-fix1.diff uploaded by + mnicholson (license 96) Tested by: mnicholson ........ + ................ ................ + +2010-09-16 16:49 +0000 [r287086-287087] Olle Johansson + + * channels/chan_sip.c: We do not handle AST_CAUSE_INTERWORKING + which we set on a lot of incoming SIP messages. Adding error + based on RFC 3398 recommendations. + + * main/indications.c: Add doxygen docs for indications.c + +2010-09-15 22:28 +0000 [r287057] Terry Wilson + + * /, res/res_srtp.c: Merged revisions 287056 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) + | 10 lines Don't hang up a call on an SRTP unprotect failure Also + make it more obvious when there is an issue en/decrypting. + (closes issue #17563) Reported by: Alexcr Patches: + res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by: + twilson ........ + +2010-09-15 21:00 +0000 [r287021] Jeff Peeler + + * /, main/features.c: Merged revisions 287020 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287020 | jpeeler | 2010-09-15 15:58:39 -0500 (Wed, 15 Sep 2010) + | 1 line fix uninintialized variable ........ + +2010-09-15 20:56 +0000 [r287018] Richard Mudgett + + * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 287017 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 + (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, + 15 Sep 2010) | 58 lines The handling of call transfer signaling + for mISDN PTMP is not fully implemented. The handling of call + transfer signaling for mISDN PTMP is not fully implemented. The + signaling of number updates with ISDN/DSS1 ECT supplementary + services (ETS 300 369-1) comes along with a notification + indicator IE and redirection number IE for PTMP. The + implementation in the current Asterisk mISDN channel + unfortunately can handle these information elements only in a + NOTIFY message. These information elements are also signaled in a + FACILTY message with a RequestSubaddress facility, when the + subscriber is already in the active state (see 9.2.4 and 9.2.5 of + ETS 300 369-1). ********** abe_2526_ast.patch * Added support to + handle the notification indicator IE and redirection number IE + with the RequestSubaddress facility. * Made + misdn_update_connected_line() send a NOTIFY message if Asterisk + originated the call and it is not connected yet. * Made + misdn_update_connected_line() send a FACILITY message if the call + is already connected. This patch requires the presence of the + associated mISDN patches to compile. I had to enhance mISDN to + allow the notification indicator IE and the redirection number IE + to be used with a FACILITY message. Earlier versions of the + Digium enhanced mISDN are no longer going to work. ********** + abe_2526_misdn.patch * Made an incoming FACILITY message allow + the presence of the notification indicator IE and the redirection + number IE. ********** abe_2526_misdnuser_v3.patch * Added support + to send and receive a FACILITY message with the notification + indicator IE and the redirection number IE. * Added the ability + to send a NOTIFY message in PTMP/NT mode to all responding + subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: + abe_2526_ast.patch uploaded by rmudgett (license 664) + abe_2526_misdn.patch uploaded by rmudgett (license 664) + abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) + Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 + .......... ................ + +2010-09-15 20:36 +0000 [r286939-287016] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 287015 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287015 | jpeeler | 2010-09-15 15:32:52 -0500 + (Wed, 15 Sep 2010) | 21 lines Merged revisions 286998 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500 + (Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) + | 7 lines Ensure mailbox is not filled to capacity before doing + message forwarding. Specifically, before prompting to record a + prepended message the capacity is checked first. If the mailbox + is full the extension will be reprompted. ABE-2517 ........ + ................ ................ + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c, main/features.c, CHANGES, + channels/chan_iax2.c, channels/sip/include/sip.h, + configs/features.conf.sample, channels/chan_mgcp.c, + include/asterisk/features.h: Merged revisions 286931 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 + Sep 2010) | 16 lines Add parking extension for non-default + parking lots. This is a new feature that allows for parking to + custom parking lots to be accessed directly, rather than with + channel variables or by changing the default parking lot. The + extension is set with the parkext option just as the default + parking lot is done. Also, the manager action has been updated to + optionally allow a specified parking lot. (closes issue #14882) + Reported by: vmikhnevych Patches: patch_14882.txt uploaded by + mnick (license 874) modified by me Review: + https://reviewboard.asterisk.org/r/884/ ........ + +2010-09-15 18:30 +0000 [r286906] Richard Mudgett + + * channels/sig_analog.c, /: Merged revisions 286904-286905 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286904 | rmudgett | 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) + | 12 lines Unable to originate calls using E&M over T1. When + originating a call from Unit Under Test to Reference Unit using + E&M RBS signaling mode, I get the following warning message: + "Ring/Off-hook in strange state 3 on channel 1". Fixed the + sig_analog outgoing flag. It was never set when sig_analog was + extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408 ........ + r286905 | rmudgett | 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) + | 1 line Simplify some code in sig_analog. ........ + +2010-09-15 13:10 +0000 [r286869] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 286868 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep + 2010) | 16 lines Set tohost to the domain specified in the + configuration file instead of the IP address of the host we are + calling. This fixes a regression introduced in r274783. (closes + issue #17960) Reported by: adriavidal Patches: + sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested + by: mich, mnicholson, adriavidal (closes issue #17676) Reported + by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson + (license 96) Tested by: mnicholson ........ + +2010-09-14 22:02 +0000 [r286835] David Vossel + + * /, channels/chan_sip.c: Merged revisions 286834 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286834 | dvossel | 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010) + | 2 lines Sets subscribed type for outgoing MWI subscriptions so + correct Event header is used. ........ + +2010-09-14 19:29 +0000 [r286683-286759] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 286758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286758 | mnicholson | 2010-09-14 14:28:38 -0500 + (Tue, 14 Sep 2010) | 27 lines Merged revisions 286757 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500 + (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep + 2010) | 13 lines Don't clear the username from a realtime + database when a registration expires. Non-realtime chan_sip does + not clear the username from memory when a registration expiries + so realtime probably shouldn't either. (closes issue #17551) + Reported by: ricardolandim Patches: + reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license + 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson + (license 96) reg-expiry-username-1.8-fix1.diff uploaded by + mnicholson (license 96) reg-expiry-username-trunk-fix1.diff + uploaded by mnicholson (license 96) Tested by: ricardolandim, + mnicholson ........ ................ ................ + + * main/channel.c, /: Merged revisions 286682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286682 | mnicholson | 2010-09-14 13:04:21 -0500 + (Tue, 14 Sep 2010) | 21 lines Merged revisions 286681 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500 + (Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep + 2010) | 7 lines Only drop duplicate answer frames if the channel + is bridged. Back in r3710 ast_read() was modified to drop answer + frames on channels that were in the UP state. This modification + prevented bridges that were up before the answer from being + broken and reestablished by an ANSWER control frame. That change + also prevents pickup of channels called from the ast_dial + framework from working properly. The ast_dial framework expects + to see an ANSWER frame after dialing and the pickup code queues + one but ast_read() drops it. This new change only drops ANSWER + frames when the channel is bridged, allowing the answer queued by + the pickup code to properly pass through ast_read() on to the + ast_dial framework. ABE-2473 (related to issue #2342) ........ + ................ ................ + +2010-09-14 15:31 +0000 [r286648] Richard Mudgett + + * /, doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Merged + revisions 286647 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286647 | rmudgett | 2010-09-14 10:30:49 -0500 (Tue, 14 Sep 2010) + | 1 line Corrected documented CONNECTED_LINE and REDIRECTING + party manipulation macro names. ........ + +2010-09-14 06:58 +0000 [r286618] Jan Kalab + + * /, res/res_calendar_ews.c: Merged revisions 286617 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r286617 | pitel | 2010-09-14 08:55:44 +0200 (Út, 14 zář + 2010) | 7 lines Merging events for Exchange web service doesn't + work as expected, resulting in only one event in calendar The + solution is to use "global" counter of events, since we do new + requests for every event and calendar sync after every request. + So now we do sync only after last request. (closes issue #17877) + Review: https://reviewboard.asterisk.org/r/916/ ........ + +2010-09-14 05:08 +0000 [r286529-286589] Tilghman Lesher + + * /, contrib/realtime/mysql/voicemail_messages.sql (added), + contrib/realtime/mysql/voicemail_data.sql (added): Merged + revisions 286588 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286588 | tilghman | 2010-09-14 00:07:16 -0500 + (Tue, 14 Sep 2010) | 9 lines Merged revisions 286587 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 + Sep 2010) | 2 lines Add documentation on missing backend tables + for Voicemail ........ ................ + + * /, main/features.c: Merged revisions 286558 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286558 | tilghman | 2010-09-13 18:50:34 -0500 + (Mon, 13 Sep 2010) | 9 lines Merged revisions 286557 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 + Sep 2010) | 2 lines C precedence got me ........ ................ + + * /, main/features.c: Merged revisions 286528 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286528 | tilghman | 2010-09-13 18:12:21 -0500 + (Mon, 13 Sep 2010) | 9 lines Merged revisions 286527 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 + Sep 2010) | 2 lines Refactor conversion to ast_poll() to fix + callparking regression. ........ ................ + +2010-09-13 22:13 +0000 [r286498] Russell Bryant + + * /, main/db.c: Merged revisions 286112 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286112 | russell | 2010-09-10 15:31:58 -0500 (Fri, 10 Sep 2010) + | 9 lines Rate limit calls to fsync() to 1 per second after astdb + updates. Astdb was determined to be one of the most significant + bottlenecks in SIP registration processing. This patch improved + the speed of an astdb load test by 50000% (yes, Fifty-Thousand + Percent). On this particular load test setup, this doubled the + number of SIP registrations the server could handle. Review: + https://reviewboard.asterisk.org/r/825/ ........ + +2010-09-13 19:40 +0000 [r286458] Jason Parker + + * /, channels/chan_sip.c: Merged revisions 286457 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286457 | qwell | 2010-09-13 14:40:05 -0500 + (Mon, 13 Sep 2010) | 12 lines Merged revisions 286456 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | + 5 lines Remove "Internal IP" from sip show settings, as it's not + at all useful to display. (closes issue #17840) Reported by: oej + ........ ................ + +2010-09-13 15:53 +0000 [r286427] Richard Mudgett + + * configs/chan_dahdi.conf.sample, /: Merged revisions 286426 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286426 | rmudgett | 2010-09-13 10:52:14 -0500 (Mon, 13 Sep 2010) + | 1 line Update chan_dahdi.conf.sample to reflect new libpri T309 + default value. ........ + +2010-09-11 17:35 +0000 [r286271-286342] Olle Johansson + + * main/say.c, main/app.c: Whitespace cleanup + + * main/features.c: Whitespace cleanup and reformatting with { and } + + * /, main/file.c: Merged revisions 286270 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286270 | oej | 2010-09-11 19:09:22 +0200 (Lör, + 11 Sep 2010) | 18 lines Merged revisions 286268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör, + 11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 + lines Handle error response when we can't make file compatible + Review: https://reviewboard.asterisk.org/r/911/ ........ + ................ ................ + + * channels/chan_sip.c: Formatting changes. + +2010-09-10 22:15 +0000 [r286190] Terry Wilson + + * channels/chan_local.c, /, funcs/func_channel.c, + include/asterisk/channel.h, include/asterisk/pbx.h, + include/asterisk/frame.h: Merged revisions 286189 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286189 | twilson | 2010-09-10 17:04:53 -0500 + (Fri, 10 Sep 2010) | 30 lines Merged revisions 286115 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286115 | twilson | 2010-09-10 15:35:25 -0500 + (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) + | 16 lines Inherit CHANNEL() writes to both sides of a Local + channel Having Local (/n) channels as queue members and setting + the language in the extension with Set(CHANNEL(language)=fr) sets + the language on the Local/...,2 channel. Hold time report + playbacks happen on the Local/...,1 channel and therefor do not + play in the specified language. This patch modifies + func_channel_write to call the setoption callback and pass the + CHANNEL() write info to the callback. chan_local uses this + information to look up the other side of the channel and apply + the same changes to it. (closes issue #17673) Reported by: + Guggemand Review: https://reviewboard.asterisk.org/r/903/ + ........ ................ ................ + +2010-09-10 21:13 +0000 [r286121] Paul Belanger + + * /, channels/chan_iax2.c: Merged revisions 286120 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286120 | pabelanger | 2010-09-10 17:11:08 -0400 + (Fri, 10 Sep 2010) | 18 lines Merged revisions 286117 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400 + (Fri, 10 Sep 2010) | 11 lines Merged revisions 286114 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep + 2010) | 4 lines Load iax.conf before registering any + functions/applications/actions. Review: + https://reviewboard.asterisk.org/r/914/ ........ ................ + ................ + +2010-09-10 21:03 +0000 [r286119] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 286118 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286118 | rmudgett | 2010-09-10 15:55:37 -0500 + (Fri, 10 Sep 2010) | 25 lines Merged revisions 286116 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500 + (Fri, 10 Sep 2010) | 18 lines Merged revisions 286113 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) + | 11 lines An outgoing call may not get hung up if a pre-connect + incoming ISDN call is disconnected. If the ISDN link a + pre-connect incoming call is using fails or is reset, the + outgoing leg may not hang up or be delayed in hanging up. + (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, + PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and + PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the + incoming call leg hangs up before connecting for any reason. It + makes no sense to send a BUSY or CONGESTION control frame to the + outgoing call leg under these circumstances. ........ + ................ ................ + +2010-09-10 13:20 +0000 [r285993] David Ruggles + + * doc/externalivr.txt, CHANGES: Merged revisions 285992 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285992 | diruggles | 2010-09-10 09:13:16 -0400 (Fri, 10 Sep + 2010) | 1 line Added missing documentation for ExternalIVR + feature added in January 2010 ........ + +2010-09-10 05:33 +0000 [r285932-285963] Tilghman Lesher + + * include/asterisk/select.h, /: Merged revisions 285962 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285962 | tilghman | 2010-09-10 00:32:18 -0500 + (Fri, 10 Sep 2010) | 13 lines Merged revisions 285961 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) + | 6 lines Another fix for Mac OS X. While trying to fix this the + "right" way, I wandered into dependency hell. Two hours later, I + backed out, and just removed the offending code. ast_inline_api + only goes one level deep and then it breaks. Ouch. ........ + ................ + + * include/asterisk/select.h, /, configure, + include/asterisk/autoconfig.h.in, configure.ac, + tests/test_poll.c: Merged revisions 285931 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285931 | tilghman | 2010-09-09 20:25:50 -0500 + (Thu, 09 Sep 2010) | 21 lines Merged revisions 285930 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500 + (Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) + | 7 lines Fix Mac OS X build. This also fixes a rather grievous + calculation error for the offset of ast_fdset, which was masked + on Linux and FreeBSD, because these platforms check the first 256 + FDs regardless of the bitmask setting (due to backwards + compatibility). ........ ................ ................ + +2010-09-09 22:53 +0000 [r285820] Paul Belanger + + * /, codecs/gsm/Makefile: Merged revisions 285819 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285819 | pabelanger | 2010-09-09 18:52:31 -0400 + (Thu, 09 Sep 2010) | 22 lines Merged revisions 285818 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285818 | pabelanger | 2010-09-09 18:49:19 -0400 + (Thu, 09 Sep 2010) | 15 lines Merged revisions 285817 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep + 2010) | 8 lines GCC 4.2.x optimizations result in improper + behavior of GSM codec (closes issue #17688) Reported by: + pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by + pprindeville (license 347) Tested by: mkeuter, pprindeville + ........ ................ ................ + +2010-09-09 20:13 +0000 [r285746] Jason Parker + + * main/channel.c, /: Merged revisions 285745 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285745 | qwell | 2010-09-09 15:11:06 -0500 + (Thu, 09 Sep 2010) | 23 lines Merged revisions 285744 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285744 | qwell | 2010-09-09 15:09:23 -0500 + (Thu, 09 Sep 2010) | 16 lines Merged revisions 285742 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | + 9 lines Transmit silence when reading DTMF in ast_readstring. + Otherwise, you could get issues with DTMF timeouts causing + hangups. (closes issue #17370) Reported by: makoto Patches: + channel-readstring-silence-generator.patch uploaded by makoto + (license 38) ........ ................ ................ + +2010-09-09 18:53 +0000 [r285641-285712] Brett Bryant + + * main/pbx.c, /: Merged revisions 285711 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285711 | bbryant | 2010-09-09 14:51:52 -0400 + (Thu, 09 Sep 2010) | 15 lines Merged revisions 285710 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) + | 8 lines Fixes an issue with dialplan pattern matching where the + specificity for pattern ranges and pattern special characters was + inconsistent. (closes issue #16903) Reported by: Nick_Lewis + Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license + 657) Tested by: Nick_Lewis ........ ................ + + * /, res/res_musiconhold.c: Merged revisions 285640 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285640 | bbryant | 2010-09-09 13:23:28 -0400 + (Thu, 09 Sep 2010) | 21 lines Merged revisions 285639 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285639 | bbryant | 2010-09-09 13:22:25 -0400 + (Thu, 09 Sep 2010) | 14 lines Merged revisions 285638 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010) + | 7 lines Fixes an issue with MOH where it doesn't recover + cleanly when it can't play a file and would just stop, instead of + continuing to find the next playable file in the MOH class. + (closes issue #17807) Reported by: kshumard Review: + https://reviewboard.asterisk.org/r/910/ ........ ................ + ................ + +2010-09-08 22:15 +0000 [r285565-285569] David Vossel + + * /, channels/chan_sip.c: Merged revisions 285568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285568 | dvossel | 2010-09-08 17:14:19 -0500 + (Wed, 08 Sep 2010) | 16 lines Merged revisions 285567 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285567 | dvossel | 2010-09-08 17:11:28 -0500 + (Wed, 08 Sep 2010) | 9 lines Merged revisions 285566 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 + Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the + end of the function on a transmit failure. ........ + ................ ................ + + * /, channels/chan_sip.c: Merged revisions 285564 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285564 | dvossel | 2010-09-08 16:48:37 -0500 + (Wed, 08 Sep 2010) | 60 lines Merged revisions 285563 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) + | 54 lines Fixes interoperability problems with session timer + behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require" + header. This is not to our benefit and RFC 4028 section 7.1 even + warns against it. It is possible for one endpoint to perform + session-timer refreshes while the other endpoint does not support + them. If in this case the end point performing the refreshing + puts "timer" in the Require field during a refresh, the dialog + will likely get terminated by the other end. 2. Change the + behavior of 'session-timer=accept' in sip.conf (which is the + default behavior of Asterisk with no session timer configuration + specified) to only run session-timers as result of an incoming + INVITE request if the INVITE contains an "Session-Expires" + header... Asterisk is currently treating having the "timer" + option in the "Supported" header as a request for session timers + by the UAC. I do not agree with this. Session timers should only + be negotiated in "accept" mode when the incoming INVITE supplies + a "Session-Expires" header, otherwise RFC 4028 says we should + treat a request containing no "Session-Expires" header as a + session with no expiration. Below I have outlined some situations + and what Asterisk's behavior is. The table reflects the behavior + changes implemented by this patch. SITUATIONS: -Asterisk as UAS + 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS + "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO + "Session-Expires". 200 Ok Response HAS "Session-Expires" header + 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO + "Session-Expires" header 5. Outgoing INVITE: HAS + "Session-Expires". Active - Asterisk will have an active refresh + timer regardless if the other endpoint does. Inactive - Asterisk + does not have an active refresh timer regardless if the other + endpoint does. XXXXXXX - Not possible for mode. + ______________________________________ |SITUATIONS | + 'session-timer' MODES | |___________|________________________| | + | originate | accept | |-----------|------------|-----------| |1. + | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX | + Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX | + -------------------------------------- (closes issue #17005) + Reported by: alexrecarey ........ ................ + +2010-09-08 21:00 +0000 [r285534] Brett Bryant + + * /, apps/app_meetme.c: Merged revisions 285533 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285533 | bbryant | 2010-09-08 16:58:43 -0400 + (Wed, 08 Sep 2010) | 15 lines Merged revisions 285532 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) + | 8 lines Fixes a bug with MeetMe where after announcing the + amount of time left in a conference, if music on hold was + playing, it doesn't restart. (closes issue #17408) Reported by: + sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by + sysreq (license 1009) Tested by: sysreq ........ ................ + +2010-09-08 20:43 +0000 [r285528-285531] Jason Parker + + * /, include/asterisk/astobj2.h, res/res_musiconhold.c: Merged + revisions 285530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285530 | qwell | 2010-09-08 15:43:10 -0500 + (Wed, 08 Sep 2010) | 9 lines Merged revisions 285529 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep + 2010) | 1 line Follow coding guidelines in moh rescan fix. Also + fix the documentation that got me in trouble. ........ + ................ + + * /, res/res_musiconhold.c: Merged revisions 285527 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285527 | qwell | 2010-09-08 15:32:13 -0500 + (Wed, 08 Sep 2010) | 15 lines Merged revisions 285526 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep 2010) | + 8 lines Fixes issue where moh files were no longer rescanned + during a reload. (closes issue #16744) Reported by: pj Patches: + 16744-reload.diff uploaded by qwell (license 4) Tested by: qwell + ........ ................ + +2010-09-08 07:15 +0000 [r285485] Tilghman Lesher + + * /, funcs/func_channel.c: Merged revisions 285484 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r285484 | tilghman | 2010-09-08 02:14:17 -0500 (Wed, 08 + Sep 2010) | 2 lines Documentation only ........ + +2010-09-07 22:23 +0000 [r285394-285456] Jason Parker + + * /, channels/chan_sip.c: Merged revisions 285455 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285455 | qwell | 2010-09-07 17:22:14 -0500 (Tue, 07 Sep 2010) | + 8 lines Don't automatically add domains for wildcard bindaddrs. + (closes issue #17832) Reported by: oej Patches: + 17832-wildcard.diff uploaded by qwell (license 4) Tested by: + qwell ........ + + * /, channels/chan_sip.c: Merged revisions 285369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285369 | qwell | 2010-09-07 15:58:34 -0500 (Tue, 07 Sep 2010) | + 7 lines Add note to 'sip show settings' regarding dual-stack + support, and a :: bindaddress. (closes issue #17831) Reported by: + oej Patches: 17831-v6wildcardbind.diff uploaded by qwell (license + 4) ........ + +2010-09-07 21:21 +0000 [r285374-285390] Tilghman Lesher + + * pbx/pbx_spool.c, /: Merged revisions 285386 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285386 | tilghman | 2010-09-07 16:20:16 -0500 (Tue, 07 Sep 2010) + | 13 lines Don't notify on attribute changes, and change how the + queuing mechanism works. Fixes call spools in 1.8. (closes issue + #17337) Reported by: loloski Patches: + 20100827__issue17337.diff.txt uploaded by tilghman (license 14) + (closes issue #17924) Reported by: mkeuter Tested by: mkeuter + ........ + + * /, funcs/func_channel.c: Merged revisions 285373 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r285373 | tilghman | 2010-09-07 16:14:03 -0500 (Tue, 07 + Sep 2010) | 7 lines Add CHANNEL(checkhangup) to check whether a + channel is in the process of being hanged up. (closes issue + #17652) Reported by: kobaz Patches: func_channel.patch uploaded + by kobaz (license 834) ........ + +2010-09-07 21:12 +0000 [r285372] Richard Mudgett + + * /, main/features.c: Merged revisions 285371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285371 | rmudgett | 2010-09-07 16:08:35 -0500 (Tue, 07 Sep 2010) + | 1 line Fix cut-n-paste error. ........ + +2010-09-07 20:56 +0000 [r285269-285368] Tilghman Lesher + + * /, pbx/pbx_config.c: Merged revisions 285367 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285367 | tilghman | 2010-09-07 15:56:07 -0500 + (Tue, 07 Sep 2010) | 23 lines Merged revisions 285366 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285366 | tilghman | 2010-09-07 15:31:41 -0500 + (Tue, 07 Sep 2010) | 16 lines Merged revisions 285365 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010) + | 9 lines Catch invalid extensions at the parser, instead of + making the core deal with them. (closes issue #17794) Reported + by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded + by tilghman (license 14) 20100820__issue17794__1.4.diff.txt + uploaded by tilghman (license 14) Tested by: PavelL ........ + ................ ................ + + * /, include/asterisk/compiler.h, addons/ooh323c/src/ooSocket.h: + Merged revisions 285336 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285336 | tilghman | 2010-09-07 14:38:12 -0500 (Tue, 07 Sep 2010) + | 2 lines Fix build on FreeBSD 8.0, take 2. ........ + + * /, main/poll.c: Merged revisions 285268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285268 | tilghman | 2010-09-07 14:08:09 -0500 + (Tue, 07 Sep 2010) | 18 lines Merged revisions 285267 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285267 | tilghman | 2010-09-07 14:07:17 -0500 + (Tue, 07 Sep 2010) | 11 lines Merged revisions 285266 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010) + | 4 lines Use poll, if indicated to do so, in the ast_poll2 + implementation. This fixes the unit tests on FreeBSD 8.0. + ........ ................ ................ + +2010-09-07 17:57 +0000 [r285199] Brett Bryant + + * /, apps/app_voicemail.c: Merged revisions 285197 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285197 | bbryant | 2010-09-07 13:54:21 -0400 + (Tue, 07 Sep 2010) | 24 lines Merged revisions 285196 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285196 | bbryant | 2010-09-07 13:49:07 -0400 + (Tue, 07 Sep 2010) | 17 lines Merged revisions 285194 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) + | 10 lines Fixes voicemail.conf issues where mailboxes with + passwords that don't precede a comma would throw unnecessary + error messages. (closes issue #15726) Reported by: 298 Patches: + M15726.diff uploaded by junky (license 177) Tested by: junky + Review: [full review board URL with trailing slash] ........ + ................ ................ + +2010-09-07 17:55 +0000 [r285198] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 285195 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285195 | rmudgett | 2010-09-07 12:47:34 -0500 + (Tue, 07 Sep 2010) | 20 lines Merged revisions 285193 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + ........ Merged revisions 285192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........ + r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010) + | 8 lines COLP/CONP and chan_misdn missing update chan_misdn does + not update the caller id of the channel if a new connected number + or ECT-INFORM (w/ new peer number on call transfer) is received. + JIRA ABE-2502 JIRA SWP-2058 ........ ........ ................ + +2010-09-06 20:10 +0000 [r285163] Russell Bryant + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 285161-285162 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285161 | russell | 2010-09-06 15:10:03 -0500 (Mon, 06 Sep 2010) + | 4 lines Fix libsrtp -fPIC check for when non-standard prefix is + used. Thanks to loompek in #asterisk for reporting the issue and + testing this patch. ........ r285162 | russell | 2010-09-06 + 15:10:24 -0500 (Mon, 06 Sep 2010) | 1 line regenerate configure + script. ........ + +2010-09-06 06:57 +0000 [r285091] Tilghman Lesher + + * /, BSDmakefile (added), makeopts.in: Merged revisions 285090 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285090 | tilghman | 2010-09-06 01:56:07 -0500 + (Mon, 06 Sep 2010) | 16 lines Merged revisions 285089 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285089 | tilghman | 2010-09-06 01:55:17 -0500 + (Mon, 06 Sep 2010) | 9 lines Merged revisions 285088 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 + Sep 2010) | 2 lines Silly convenience script for BSD platforms. + ........ ................ ................ + +2010-09-04 18:10 +0000 [r285058] Russell Bryant + + * /, include/asterisk/cli.h: Merged revisions 285057 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r285057 | russell | 2010-09-04 13:08:19 -0500 (Sat, 04 + Sep 2010) | 2 lines Add a C++ compatible version of + AST_CLI_DEFINE(). ........ + +2010-09-03 23:23 +0000 [r285029] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 285017 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285017 | twilson | 2010-09-03 18:19:54 -0500 (Fri, 03 Sep 2010) + | 4 lines Call correct lock function as transferer is a sip_pvt + not a channel Both functions are #defined to ao2_lock, but + still... ........ + +2010-09-03 22:23 +0000 [r285007] David Vossel + + * /, channels/chan_sip.c, configs/sip.conf.sample, + channels/sip/include/sip.h: Merged revisions 285006 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r285006 | dvossel | 2010-09-03 17:21:50 -0500 (Fri, 03 + Sep 2010) | 9 lines Disables auth_options_request option by + default. The auth_options_request option was created to do + authentication on OPTIONS request just like INVITES are done. + Since it has been noted that some endpoints use OPTIONS requests + as a way of qualifying a peer and that a 401 authentication + response could result in interoperability issues, this option has + been disabled by default. ........ + +2010-09-03 18:21 +0000 [r284973] Brett Bryant + + * /, channels/chan_iax2.c: Merged revisions 284967 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284967 | bbryant | 2010-09-03 14:19:53 -0400 + (Fri, 03 Sep 2010) | 15 lines Merged revisions 284958 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03 Sep 2010) + | 8 lines This is a patch provided for issue #17935 to add the + ActionID to the IAXregistry AMI response. (closes issue #17935) + Reported by: alexkuklin Patches: iaxshowreg uploaded by + alexkuklin (license 1115) Tested by: alexkuklin ........ + ................ + +2010-09-03 18:04 +0000 [r284951-284953] David Vossel + + * /, channels/chan_sip.c: Merged revisions 284952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284952 | dvossel | 2010-09-03 13:03:23 -0500 (Fri, 03 Sep 2010) + | 2 lines During OPTIONS authentication, the authpeer does not + need to be returned for any reason. ........ + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: Merged revisions 284950 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 + Sep 2010) | 14 lines authenticate OPTIONS requests just like we + would an INVITE OPTIONS requests should be treated the same as an + INVITE This includes authentication. This patch adds the ability + for incoming out of dialog OPTION requests to be authenticated + before providing a response indicating whether an extension is + available or not. The authentication routine works the exact same + way as it does for incoming INVITEs. This means that if a peer + has 'insecure=invite' in their peer definition, the same will be + true for the processing of the OPTIONS request. Review: + https://reviewboard.asterisk.org/r/881/ ........ + +2010-09-03 16:42 +0000 [r284922] Terry Wilson + + * /, apps/app_chanspy.c: Merged revisions 284921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284921 | twilson | 2010-09-03 11:28:18 -0500 + (Fri, 03 Sep 2010) | 19 lines Merged revisions 284897 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284897 | twilson | 2010-09-03 11:20:45 -0500 + (Fri, 03 Sep 2010) | 12 lines Merged revisions 284881 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) + | 5 lines Properly detect when a sound file doesn't exist + ast_fileexists returns -1 for error and 0 for a non-existant + file. The existing code treated missing files as though they + existed. ........ ................ ................ + +2010-09-03 13:09 +0000 [r284851-284853] Jan Kalab + + * /, res/res_calendar_ews.c: Merge of strdupa() fix for calendars + categories priorities + + * res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c, + include/asterisk/calendar.h, res/res_calendar_ews.c, + res/res_calendar.c: Support for calendar events priorities and + categories (with ISO C90 fix) See RFC 5545 ch. 3.8.1.2 and 9. + (closes issue #17837) Review: + https://reviewboard.asterisk.org/r/880/ + +2010-09-02 21:08 +0000 [r284782] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: + Merged revisions 284779-284780 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284779 | rmudgett | 2010-09-02 15:59:12 -0500 (Thu, 02 Sep 2010) + | 8 lines Made output libpri event names if pri debugging is + enabled when sig_pri processes them. * Simplified CLI "pri debug + xx span xx" command code and removed redundant debugging enabled + messages. * Made CLI "pri debug xx span xx" command only close + the debugging log file if it was opened. ........ r284780 | + rmudgett | 2010-09-02 16:02:54 -0500 (Thu, 02 Sep 2010) | 2 lines + Simplified pri_dchannel() poll timeout duration code. ........ + +2010-09-02 16:57 +0000 [r284706] David Vossel + + * /, channels/chan_sip.c: Merged revisions 284705 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284705 | dvossel | 2010-09-02 11:56:43 -0500 + (Thu, 02 Sep 2010) | 20 lines Merged revisions 284704 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284704 | dvossel | 2010-09-02 11:48:51 -0500 + (Thu, 02 Sep 2010) | 13 lines Merged revisions 284703 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) + | 7 lines Removed relatedpeer code from sip_autodestruct Handling + of the relatedpeer structure associated with a sip_pvt should be + done during the final sip_destruction function, not in + sip_autodestruct. ........ ................ ................ + +2010-09-02 16:44 +0000 [r284702] Jason Parker + + * /, formats/format_wav.c: Merged revisions 284701 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r284701 | qwell | 2010-09-02 11:43:09 -0500 (Thu, 02 Sep + 2010) | 8 lines Add slin16 support for format_wav (new wav16 file + extension) (closes issue #15029) Reported by: andrew Patches: + wav16.patch uploaded by andrew (license 240) Tested by: qwell, + andrew ........ + +2010-09-02 16:36 +0000 [r284700] Tilghman Lesher + + * /, addons/ooh323c/src/oochannels.c: Merged revisions 284696 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284696 | tilghman | 2010-09-02 11:27:52 -0500 (Thu, 02 Sep 2010) + | 2 lines Fixing build ........ + +2010-09-02 16:35 +0000 [r284699] Richard Mudgett + + * /, doc/tex/channelvariables.tex, doc/tex/partymanip.tex (added), + doc/tex/asterisk.tex: Merged revisions 284698 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284698 | rmudgett | 2010-09-02 11:34:32 -0500 (Thu, 02 Sep 2010) + | 5 lines Added documentation for CONNECTEDLINE and REDIRECTING + functions. (closes issue #17808) Reported by: jtodd Review: + https://reviewboard.asterisk.org/r/875/ ........ + +2010-09-02 16:12 +0000 [r284598-284667] Tilghman Lesher + + * channels/chan_usbradio.c, /: Merged revisions 284666 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284666 | tilghman | 2010-09-02 11:11:15 -0500 + (Thu, 02 Sep 2010) | 9 lines Merged revisions 284665 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02 + Sep 2010) | 2 lines Fixing build. ........ ................ + + * /, apps/app_queue.c: Merged revisions 284632 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284632 | tilghman | 2010-09-02 00:31:02 -0500 + (Thu, 02 Sep 2010) | 14 lines Merged revisions 284631 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) + | 7 lines Don't reset queue stats on a module reload. (closes + issue #17535) Reported by: raarts Patches: + 20100819__issue17535.diff.txt uploaded by tilghman (license 14) + ........ ................ + + * /, channels/chan_sip.c, channels/chan_agent.c, + channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c, + apps/app_followme.c, apps/app_speech_utils.c, main/loader.c, + pbx/pbx_loopback.c, channels/chan_dahdi.c, funcs/func_aes.c, + include/asterisk/module.h, pbx/pbx_realtime.c, pbx/pbx_dundi.c, + apps/app_stack.c, channels/chan_mgcp.c, apps/app_adsiprog.c, + apps/app_voicemail.c: Merged revisions 284610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) + | 10 lines When optional_api is non-optional, force dependent + modules to be loaded. (closes issue #17707) Reported by: ira + Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by + tilghman (license 14) Tested by: tilghman Review: + https://reviewboard.asterisk.org/r/876/ ........ + + * main/stun.c, res/res_ais.c, /, include/asterisk/autoconfig.h.in, + configure.ac, channels/console_video.c, + include/asterisk/channel.h, res/res_jabber.c, res/res_pktccops.c, + main/poll.c, channels/chan_usbradio.c, include/asterisk/select.h + (added), channels/chan_phone.c, channels/chan_misdn.c, configure, + main/features.c, include/asterisk/poll-compat.h, + tests/test_poll.c (added), addons/ooh323c/src/oochannels.c, + main/asterisk.c, addons/ooh323c/src/ooSocket.h: Merged revisions + 284597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284597 | tilghman | 2010-09-02 00:00:34 -0500 + (Thu, 02 Sep 2010) | 29 lines Merged revisions 284593,284595 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500 + (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) + | 11 lines Ensure that all areas that previously used select(2) + now use poll(2), with implementations that need poll(2) + implemented with select(2) safe against 1024-bit overflows. This + is a followup to the fix for the pthread timer in 1.6.2 and + beyond, fixing a potential crash bug in all supported releases. + (closes issue #17678) Reported by: russell Branch: + https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select + Review: https://reviewboard.asterisk.org/r/824/ ........ + ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500 + (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after + last commit ................ ................ + +2010-09-01 21:48 +0000 [r284562] David Vossel + + * /, channels/chan_sip.c: Merged revisions 284561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) + | 9 lines During request to dialog matching, verify init_ruri is + present before comparing. During request to dialog matching, we + attempt a best effort routine for fork detection which requires + several elements to be in place. The dialog's initial request uri + is one of those elements. Since it is best effort, if the + init_ruri is not present for some reason we can not proceed with + that routine. ........ + +2010-09-01 18:52 +0000 [r284479] Terry Wilson + + * res/res_rtp_asterisk.c, include/asterisk/res_srtp.h, + main/rtp_engine.c, /, channels/chan_sip.c, res/res_srtp.c: Merged + revisions 284477 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) + | 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP + lines Adding code to Asterisk that changed the SSRC during + bridges and masquerades broke SRTP functionality. Also broken was + handling the situation where an incoming INVITE had more than one + crypto offer. This patch caches the SRTP policies the we use so + that we can change the ssrc and inform libsrtp of the new + streams. It also uses the first acceptable a=crypto line from the + incoming INVITE. (closes issue #17563) Reported by: Alexcr + Patches: srtp.diff uploaded by twilson (license 396) Tested by: + twilson Review: https://reviewboard.asterisk.org/r/878/ ........ + +2010-09-01 18:19 +0000 [r284440-284474] Tilghman Lesher + + * res/res_config_pgsql.c, /: Merged revisions 284473 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284473 | tilghman | 2010-09-01 13:16:37 -0500 + (Wed, 01 Sep 2010) | 12 lines Merged revisions 284472 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01 Sep 2010) + | 5 lines Don't warn on floats and timestamps (closes issue + #17082) Reported by: coolmig ........ ................ + + * /, channels/chan_sip.c: Merged revisions 284415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284415 | tilghman | 2010-08-31 15:22:10 -0500 + (Tue, 31 Aug 2010) | 21 lines Merged revisions 284399 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500 + (Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) + | 7 lines Don't send a devstate change on poke_noanswer if the + state did not change. (closes issue #17741) Reported by: schmidts + Patches: chan_sip.c.patch uploaded by schmidts (license 1077) + ........ ................ ................ + +2010-08-31 19:01 +0000 [r284315-284319] Leif Madsen + + * /, configs/say.conf.sample: Merged revisions 284318 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284318 | lmadsen | 2010-08-31 14:00:15 -0500 + (Tue, 31 Aug 2010) | 22 lines Merged revisions 284317 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284317 | lmadsen | 2010-08-31 13:59:31 -0500 + (Tue, 31 Aug 2010) | 15 lines Merged revisions 284316 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) + | 7 lines Update say.conf.sample to match the rules in say.c + (closes issue #17835) Reported by: RoadKill Patches: + say.conf.sample.patch.rules uploaded by RoadKill (license 933) + Tested by: RoadKill ........ ................ ................ + + * channels/chan_sip.c: Add trustrpid and sendrpid global values to + 'sip show settings' (closes issue #17860) Reported by: jtodd + Patches: __20100816-chan_sip-sip-show-settings.txt uploaded by + lmadsen (license 10) Tested by: lmadsen, russell + +2010-08-30 22:30 +0000 [r284282] Tilghman Lesher + + * /, apps/app_festival.c: Merged revisions 284281 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284281 | tilghman | 2010-08-30 17:28:47 -0500 + (Mon, 30 Aug 2010) | 18 lines Merged revisions 284280 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) + | 11 lines Fix 3 coding errors: 1) After we close FD, we should + not be trying to write to it. 2) Call _exit(0), not exit(0), to + avoid running shutdown routines in a child. 3) Use endian, not + processor, detection to ensure bytes are written in the correct + order. (closes issue #15706) Reported by: modelnine Patches: + asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine + (license 865) Tested by: gmartinez ........ ................ + +2010-08-30 09:32 +0000 [r284189-284248] Olle Johansson + + * main/file.c: Small doxygen fix and doc addition + + * main/say.c: Clean upp doxygen documentation + + * include/asterisk/say.h: Doxygen formatting You can't write "same + as above" in hypertext documentation. Above doesn't make sense in + hyperspace. + + * apps/app_playback.c: Add doxygen documentation + +2010-08-29 07:06 +0000 [r284097-284159] Tilghman Lesher + + * /, configs/res_curl.conf.sample (added): Merged revisions 284158 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284158 | tilghman | 2010-08-29 02:05:27 -0500 (Sun, 29 Aug 2010) + | 2 lines Missed adding this file ........ + + * /, sounds: Merged revisions 284127 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284127 | tilghman | 2010-08-29 00:17:37 -0500 (Sun, 29 Aug 2010) + | 2 lines Also ignore the checksums ........ + + * cel/cel_odbc.c (added), /, configs/cel_adaptive_odbc.conf.sample + (removed), configs/cel_odbc.conf.sample (added), + cel/cel_adaptive_odbc.c (removed): Merged revisions 284096 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284096 | tilghman | 2010-08-28 21:51:14 -0500 (Sat, 28 Aug 2010) + | 3 lines Rename CEL adaptive driver to plain driver, since there + isn't another ODBC driver (and the other CEL drivers have + adaptive capabilities, anyway). ........ + +2010-08-28 21:30 +0000 [r284066] Russell Bryant + + * main/manager.c, /: Merged revisions 284065 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010) + | 13 lines Be more flexible with whitespace on AMI action + headers. Previously, this code required exactly one space to be + after the ':' in headers for an AMI action. This now makes + whitespace optional, and allows whitespace that is there to vary + in amount. (closes issue #17862) Reported by: cmoye Patches: + manager.c.patch_trunk uploaded by cmoye (license 858) + manager.c.patch_1.8 uploaded by cmoye (license 858) Tested by: + cmoye ........ + +2010-08-27 22:39 +0000 [r284033] David Vossel + + * /, channels/chan_sip.c: Merged revisions 284032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284032 | dvossel | 2010-08-27 17:37:11 -0500 + (Fri, 27 Aug 2010) | 21 lines Merged revisions 284002 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500 + (Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) + | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests. + (closes issue #17758) Reported by: ibc Patches: + multiple_accept_headers_1.4.diff uploaded by dvossel (license + 671) ........ ................ ................ + +2010-08-27 21:50 +0000 [r283958] Russell Bryant + + * /, pbx/pbx_realtime.c: Merged revisions 283951 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283951 | russell | 2010-08-27 16:33:55 -0500 (Fri, 27 Aug 2010) + | 2 lines Print exten@context:priority in verbose messages from + pbx_realtime. ........ + +2010-08-27 20:32 +0000 [r283883] Jason Parker + + * res/res_config_odbc.c, /, main/config.c, + addons/res_config_mysql.c: Merged revisions 283882 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283882 | qwell | 2010-08-27 15:31:55 -0500 + (Fri, 27 Aug 2010) | 22 lines Merged revisions 283881 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r283881 | qwell | 2010-08-27 15:30:27 -0500 + (Fri, 27 Aug 2010) | 15 lines Merged revisions 283880 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | + 8 lines Fix issue with decoding ^-escaped characters in realtime. + (closes issue #17790) Reported by: denzs Patches: + 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell, + denzs ........ ................ ................ + +2010-08-27 14:01 +0000 [r283803] Olle Johansson + + * main/manager.c: Doxygen formatting changes + +2010-08-26 23:51 +0000 [r283771] Tilghman Lesher + + * /, res/res_musiconhold.c: Merged revisions 283770 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r283770 | tilghman | 2010-08-26 18:47:02 -0500 (Thu, 26 + Aug 2010) | 8 lines Convert MOH to use generic timers. (closes + issue #17726) Reported by: lmadsen Patches: + 20100825__issue17726__2.diff.txt uploaded by tilghman (license + 14) Tested by: tilghman ........ + +2010-08-26 15:28 +0000 [r283693] David Vossel + + * /, channels/chan_sip.c: Merged revisions 283692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283692 | dvossel | 2010-08-26 10:26:37 -0500 + (Thu, 26 Aug 2010) | 32 lines Merged revisions 283691 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500 + (Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) + | 19 lines Fixed how Asterisk destroys a dialog on channel hangup + before invite receives a response. If an ast_channel with a SIP + tech pvt hangs up before the sip dialog gets a response to its + outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is + not rfc compliant and results in confusion at the other endpoint. + sip_pretend_ack will ack and remove all the packets in the + retransmit queue. This means that the INVITE will stop + retransmitting, and that any response to that INVITE that comes + after the pretend_ack occurs will be ignored. Instead of faking + any sort of acknowledgement for an outgoing INVITE during an + internal hangup, we should let the protocol stack process the + INVITE transaction and terminate the dialog properly. This is + achieved by setting the PENDING_BYE flag. When this flag is used, + once the dialog proceeds to an escapable state the transaction + will either be canceled with a SIP_CANCEL or completed followed + immediately by a BYE. Attempting to do this any other way is + incorrect. If the endpoint is not responding to the INVITE + request, the INVITE must continue to be retransmitted until it + times out which will result in the dialog being destroyed. + ........ ................ ................ + +2010-08-26 13:28 +0000 [r283628-283660] Russell Bryant + + * /, res/res_odbc.c: Merged revisions 283659 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283659 | russell | 2010-08-26 08:26:14 -0500 (Thu, 26 Aug 2010) + | 2 lines Slight improvement to a debug message. ........ + + * Makefile, /, keys/iaxtel.pub (removed), keys/freeworlddialup.pub + (removed): Merged revisions 283629 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283629 | russell | 2010-08-26 07:48:45 -0500 (Thu, 26 Aug 2010) + | 2 lines Remove public keys that are no longer useful. ........ + + * /, configs/manager.conf.sample: Merged revisions 283627 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283627 | russell | 2010-08-26 07:26:22 -0500 (Thu, 26 Aug 2010) + | 2 lines Move httptimeout out from in between port and bindaddr. + ........ + +2010-08-25 22:59 +0000 [r283596] David Vossel + + * /, channels/chan_sip.c: Merged revisions 283595 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283595 | dvossel | 2010-08-25 17:57:56 -0500 + (Wed, 25 Aug 2010) | 14 lines Merged revisions 283594 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) + | 7 lines Add to and from tags to NOTIFY dialog-info xml body so + pickup can occur. When pedantic mode is used, the dialog-info xml + generated during a ringing event must contain the to and from tag + values. Otherwise if a pickup occurs using INVITE with replaces, + Astrisk will not be able to locate the subscription. ........ + ................ + +2010-08-25 16:14 +0000 [r283562] Tilghman Lesher + + * /, res/res_odbc.c: Merged revisions 283561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283561 | tilghman | 2010-08-25 11:12:43 -0500 (Wed, 25 Aug 2010) + | 5 lines Initialize connect timeout on each time through the + loop. (closes issue #17911) Reported by: wurstsalat ........ + +2010-08-25 15:56 +0000 [r283560] David Vossel + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged + revisions 283559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283559 | dvossel | 2010-08-25 10:54:11 -0500 + (Wed, 25 Aug 2010) | 16 lines Merged revisions 283558 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) + | 10 lines Asterisk will not advertise session timers are + supported when 'session-timers=refuse' is used. Asterisk now + dynamically builds the "Supported" header depending on what is + enabled/disabled in sip.conf. Session timers used to always be + advertised as being supported even when they were disabled in the + configuration. This caused problems with some end points. (issue + #17005) ........ ................ + +2010-08-25 14:55 +0000 [r283528] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 283527 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) + | 2 lines Convert ast_log(LOG_DEBUG, ...) to ast_debug(...) + ........ + +2010-08-24 20:42 +0000 [r283495] Damien Wedhorn + + * channels/chan_skinny.c: Ignore redial hard button when no + previous number. (closes issue #17887) Reported by: salecha + Patches: skinny.redial.diff uploaded by wedhorn (license 30) + Tested by: wedhorn, salecha + +2010-08-24 20:36 +0000 [r283494] David Vossel + + * /, configs/sip.conf.sample, UPGRADE-1.8.txt, + channels/sip/include/sip.h: Merged revisions 283493 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 + Aug 2010) | 2 lines Changes the default behavior for sip.conf's + pedantic option from "no" to "yes". ........ + +2010-08-24 18:58 +0000 [r283458] Leif Madsen + + * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions + 283457 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) + | 9 lines Fix issue where TOS is no longer set on RTP packets. + Fix issue where the tos is no longer being set on RTP packets + through res_rtp_asterisk. (closes issue #17890) Reported by: + elguero Patches: qos_18.diff uploaded by elguero (license 37) + Review: https://reviewboard.asterisk.org/r/868 ........ + +2010-08-24 18:45 +0000 [r283383-283456] David Vossel + + * res/res_stun_monitor.c: This fix downgrades the ERROR message + indicating no res_stun_monitor.conf to a WARNING message. + + * /, channels/chan_sip.c: Merged revisions 283382 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283382 | dvossel | 2010-08-24 11:11:18 -0500 + (Tue, 24 Aug 2010) | 25 lines Merged revisions 283381 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500 + (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) + | 11 lines This fix makes sure the ast_channel hangs up correctly + when the dialog's PENDING_BYE flag is set. When the pending bye + flag is used, it is possible that the dialog will terminate and + leave the sip_pvt->owner channel up. This is because we never + hangup the ast_channel after sending the SIP_BYE request. When we + receive the response for the SIP_BYE we set need_destroy which we + would expect to destroy the dialog on the next do_monitor loop, + but this is not the case. The dialog will only be destroyed once + the owner is hungup even with the need_destroy flag set. This + patch sets the softhangup flag on the ast_channel when a SIP_BYE + request is sent as a result of the pending bye flag. ........ + ................ ................ + +2010-08-24 12:51 +0000 [r283351] Russell Bryant + + * /, funcs/func_odbc.c: Merged revisions 283350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283350 | russell | 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010) + | 2 lines Don't attempt to release a NULL ODBC handle. ........ + +2010-08-23 21:35 +0000 [r283320] Tilghman Lesher + + * /, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, + cel/cel_adaptive_odbc.c: Merged revisions 283319 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283319 | tilghman | 2010-08-23 16:33:47 -0500 + (Mon, 23 Aug 2010) | 9 lines Merged revisions 283318 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 + Aug 2010) | 2 lines CDR drivers depend upon res_odbc, not + directly on the ODBC libraries ........ ................ + +2010-08-23 20:50 +0000 [r283287-283289] Damien Wedhorn + + * channels/chan_skinny.c: Hack to allow easy debugging of skinny in + trunk. + + * channels/chan_skinny.c: Add additional AST_CONTROL_ states to + control2str. + + * channels/chan_skinny.c: Fixes display issues on 7910 and older + phones. Also correct the callinfo provided in skinny_answer. + (closes issue #17876) Reported by: salecha Patches: + skinny_cnd3.diff uploaded by wedhorn (license 30) Tested by: + salecha, wedhorn Review: NA + +2010-08-23 13:35 +0000 [r283178-283242] Russell Bryant + + * configs/cel.conf.sample, /: Merged revisions 283241 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r283241 | russell | 2010-08-23 08:35:35 -0500 (Mon, 23 + Aug 2010) | 2 lines Add sample configuration for cel_radius. + ........ + + * /, include/asterisk/cel.h, main/cel.c: Merged revisions 283230 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010) + | 7 lines Make the AST_CEL_AMA enum match up with the AST_CDR_ + ama flag values. Really, having 2 enums for this is silly and + error prone, demonstrated by the crash that I hit because there + was an assumption in the code that the values in each matched up. + However, this is a quick fix to get them to match up so it will + work. ........ + + * /, main/cel.c: Merged revisions 283209 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283209 | russell | 2010-08-23 08:06:57 -0500 (Mon, 23 Aug 2010) + | 2 lines Don't blow up on an invalid AMA flag. ........ + + * /, configs/cel_custom.conf.sample: Merged revisions 283207 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283207 | russell | 2010-08-23 07:31:20 -0500 (Mon, 23 Aug 2010) + | 2 lines Tack on ${eventextra} to the sample cel_custom.conf. + ........ + + * /, configs/cel_custom.conf.sample: Merged revisions 283177 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283177 | russell | 2010-08-23 07:12:53 -0500 (Mon, 23 Aug 2010) + | 2 lines Cut down on excessive quotation. ........ + +2010-08-23 12:09 +0000 [r283176] Tilghman Lesher + + * /, res/res_stun_monitor.c: Merged revisions 283175 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r283175 | tilghman | 2010-08-23 07:06:26 -0500 (Mon, 23 + Aug 2010) | 2 lines Don't fail to start if the config file is + missing. ........ + +2010-08-23 11:59 +0000 [r283174] Russell Bryant + + * /, configs/cel_custom.conf.sample: Merged revisions 283173 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283173 | russell | 2010-08-23 06:58:34 -0500 (Mon, 23 Aug 2010) + | 5 lines Expand cel_custom.conf.sample. Include the usage of + CSV_QUOTE() to ensure data has valid CSV formatting. Also list + the special CEL variables that are available for use in the + mapping. ........ + +2010-08-20 15:39 +0000 [r283051] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 283050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283050 | rmudgett | 2010-08-20 10:35:38 -0500 + (Fri, 20 Aug 2010) | 36 lines Merged revisions 283049 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500 + (Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) + | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a + protocol error The PRI layer in chan_dadhi will check if a + PROGRESS message has already been sent, and not allow sending + another (although that is technically allowed by the Q931 spec), + however it does not protect against sending an ALERTING and then + sending a PROGRESS message, which is a violation of the + specification. Most switches don't seem to care too deeply about + this, but some do, and will disconnect the call when receiving + this invalid sequence. Protocol specification reference: + T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview + protocol control (network side) point-point (sheet 3 of 8)" + (closes issue #17874) Reported by: nic_bellamy Patches: + asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by + nic bellamy (license 299) + asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded + by nic bellamy (license 299) + asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded + by nic bellamy (license 299) ........ ................ + ................ + +2010-08-20 12:45 +0000 [r282980-283015] Russell Bryant + + * /, configs/cel_adaptive_odbc.conf.sample: Merged revisions 283013 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283013 | russell | 2010-08-20 07:45:12 -0500 (Fri, 20 Aug 2010) + | 2 lines Fix a typo in a column name. ........ + + * /, apps/app_celgenuserevent.c: Merged revisions 282979 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282979 | russell | 2010-08-20 06:52:37 -0500 (Fri, 20 Aug 2010) + | 2 lines Add an argument missing from the CELGenUserEvent + documentation. ........ + + * channels/chan_multicast_rtp.c, /: Merged revisions 282638 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282638 | russell | 2010-08-18 07:30:40 -0500 (Wed, 18 Aug 2010) + | 4 lines Split _all_ arguments before parsing them. This fixes + multicast RTP paging using linksys mode. ........ + +2010-08-19 21:08 +0000 [r282892-282896] David Vossel + + * /, channels/chan_sip.c: Merged revisions 282895 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282895 | dvossel | 2010-08-19 16:07:20 -0500 + (Thu, 19 Aug 2010) | 25 lines Merged revisions 282894 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500 + (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) + | 11 lines tos_sip option was not being set correctly When + tos_sip is used, the tos of the sip socket is only set correctly + if the socket binding changes on a reload. If the binding stays + the same but the TOS changes, the new tos value would not take + into effect. This patch fixes that. (closes issue #17712) + Reported by: nickb ........ ................ ................ + + * /, channels/chan_sip.c: Merged revisions 282891 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282891 | dvossel | 2010-08-19 15:34:41 -0500 + (Thu, 19 Aug 2010) | 11 lines Merged revisions 282890 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) + | 5 lines fixes sip peer memory leaks in the peer_by_ip table + (issue #17798) ........ ................ + +2010-08-19 20:02 +0000 [r282861] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 282860 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282860 | mnicholson | 2010-08-19 15:01:11 -0500 + (Thu, 19 Aug 2010) | 30 lines Merged revisions 282859 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500 + (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul + 2010) | 16 lines Regression with T.38 negotiation Prior to + 1.4.26.3 T.38 negotiation worked properly, in the case of the + reporter. (issue #16852) Reported by: cfc (closes issue #16705) + Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded + by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, + samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ + ................ ................ + +2010-08-19 14:46 +0000 [r282827] Tilghman Lesher + + * main/netsock2.c, /: Merged revisions 282826 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282826 | tilghman | 2010-08-19 09:44:51 -0500 (Thu, 19 Aug 2010) + | 2 lines Only output debugging if the debug level is on. + ........ + +2010-08-19 12:13 +0000 [r282798] Russell Bryant + + * include/asterisk/cel.h: Add a todo item for CEL. + +2010-08-19 02:20 +0000 [r282751] Terry Wilson + + * /, configs/sip.conf.sample: Merged revisions 282740 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282740 | twilson | 2010-08-18 21:18:50 -0500 + (Wed, 18 Aug 2010) | 16 lines Merged revisions 282730 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282730 | twilson | 2010-08-18 21:14:28 -0500 + (Wed, 18 Aug 2010) | 9 lines Merged revisions 282729 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 + Aug 2010) | 2 lines Add some documentation about codec + negotiation to sip.conf ........ ................ + ................ + +2010-08-18 21:34 +0000 [r282701] Damien Wedhorn + + * channels/chan_skinny.c: Cleanup: consolidate offhook (new call). + Consolidates all offhook (new call with dialtone) to + setsubstate_offhook. This should be roughly equivalent to + existing code, although a couple of calls now run through the + full offhook sequence rather than an abbreviated one. (closes + issue #17812) Reported by: wedhorn Patches: + cleanup.stateoffhook.diff uploaded by wedhorn (license 30) Tested + by: salecha, wedhorn Review: NA + +2010-08-18 15:35 +0000 [r282673] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /: Merged revisions + 282671-282672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282671 | rmudgett | 2010-08-18 10:27:51 -0500 (Wed, 18 Aug 2010) + | 1 line Use the correct operator when calculating the PRI span + devstate. ........ r282672 | rmudgett | 2010-08-18 10:28:27 -0500 + (Wed, 18 Aug 2010) | 1 line Use the correct type for + aoce_delayhangup bit field. ........ + +2010-08-18 13:11 +0000 [r282640] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 282639 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug + 2010) | 13 lines Properly handle 200 and unknown responses + conatined in NOTIFY requests received in response to REFER + requests. This patch fixes the way asterisk handles NOTIFY + requests received in response to REFER requests. These changes to + NOTIFY handler were first introduced in r217482. This new change + properly handles the 200 response by queueing an + AST_TRANSFER_SUCCESS control frame and also prevents that control + frame from being queued when provisional and unknown responses + are received. (issue #17486) Reported by: davidw Tested by: + mnicholson (issue #12713) Reported by: davidw Review: + https://reviewboard.asterisk.org/r/860/ ........ + +2010-08-18 07:50 +0000 [r282609] Tilghman Lesher + + * /, channels/sig_pri.c: Merged revisions 282608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282608 | tilghman | 2010-08-18 02:49:04 -0500 + (Wed, 18 Aug 2010) | 16 lines Merged revisions 282607 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) + | 9 lines Don't warn on callerid when completely text, instead of + numeric with localdialplan prefixes. (closes issue #16770) + Reported by: jamicque Patches: 20100413__issue16770.diff.txt + uploaded by tilghman (license 14) 20100811__issue16770.diff.txt + uploaded by tilghman (license 14) Tested by: jamicque ........ + ................ + +2010-08-17 21:37 +0000 [r282544-282578] David Vossel + + * /, channels/chan_sip.c: Merged revisions 282577 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282577 | dvossel | 2010-08-17 16:36:57 -0500 + (Tue, 17 Aug 2010) | 16 lines Merged revisions 282576 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010) + | 9 lines fixes no default transport for temp peer creation in + chan_sip (closes issue #17829) Reported by: falves11 Patches: + issue_17829.rev1.txt uploaded by russell (license 2) + issue_17829.diff uploaded by dvossel (license 671) Tested by: + falves11 ........ ................ + + * /, channels/chan_iax2.c: Merged revisions 282545 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r282545 | dvossel | 2010-08-17 15:08:56 -0500 (Tue, 17 + Aug 2010) | 6 lines ACCEPT message should respond with the new + FORMAT2 ie (closes issue #17804) Reported by: tpanton ........ + + * /, include/asterisk/unaligned.h: Merged revisions 282543 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282543 | dvossel | 2010-08-17 14:34:06 -0500 (Tue, 17 Aug 2010) + | 4 lines fixes truncated uint64_t value in + put_unaligned_uint64_t() function (issue #17804) ........ + +2010-08-16 20:40 +0000 [r282502] Terry Wilson + + * main/channel.c, /: Merged revisions 282468 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282468 | twilson | 2010-08-16 12:53:44 -0500 + (Mon, 16 Aug 2010) | 30 lines Merged revisions 282467 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282467 | twilson | 2010-08-16 12:32:01 -0500 + (Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) + | 16 lines Send a SRCCHANGE indication when we masquerade + Masquerading a channel means that the src of the audio is + potentially changing, so send a SRCCHANGE so that RTP-based media + streams can get a new SSRC generated to reflect the change. + Original patch by addix (along with lots of testing--thanks!). + (closes issue #17007) Reported by: addix Patches: + 1001-reset-SSRC-original-channel.diff uploaded by addix (license + 1006) srcchange.diff uploaded by twilson (license 396) Tested by: + addix, twilson Review: https://reviewboard.asterisk.org/r/862/ + ........ ................ ................ + +2010-08-16 18:02 +0000 [r282471] Leif Madsen + + * doc/tex/sounds.tex (added), /, doc/tex/asterisk.tex: Merged + revisions 282470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282470 | lmadsen | 2010-08-16 13:01:00 -0500 + (Mon, 16 Aug 2010) | 15 lines Merged revisions 282469 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) + | 7 lines Add information about creating sounds files using the + sounds tools publically available so that others can create their + own sounds prompts using the same tools we use to generate sounds + releases. This allows people creating their own prompts to sound + consistent with the prompts available from the open source + project. SWP-595 ........ ................ + +2010-08-15 13:08 +0000 [r282397] Tzafrir Cohen + + * utils/muted.c, configure, main/Makefile, configure.ac, + main/acl.c, channels/chan_oss.c, main/netsock.c: Support for + GNU/kFreeBSD kFreeBSD is GNU (with glibc) on to of a FreeBSD + kernel. See http://glibc-bsd.alioth.debian.org/porting/PORTING + This patch gets Asterisk close to building on Debian kFreeBSD + i386, mainly by adding an extra test for __GLIBC__ in one or two + (or more) places. OSARCH is set to 'kfreebsd-gnu' DAHDI support + (and support for chan_vpb) was not tested. Review: + https://reviewboard.asterisk.org/r/858/ + +2010-08-14 04:58 +0000 [r282367] Tilghman Lesher + + * /, include/asterisk/sched.h, channels/chan_iax2.c: Merged + revisions 282366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) + | 4 lines Fix our FRACKing issue with chan_iax2 a different way. + Review: https://reviewboard.asterisk.org/r/861/ ........ + +2010-08-13 23:57 +0000 [r282335] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 282334 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13 + Aug 2010) | 6 lines PRI CCSS may use a stale dial string for the + recall dial string. If an outgoing call negotiates a different B + channel than initially requested, the saved original dial string + was not transferred to the new B channel. CCSS uses that dial + string to generate the recall dial string. ........ + +2010-08-13 22:27 +0000 [r282237-282304] David Vossel + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + UPGRADE-1.8.txt: Merged revisions 282302 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) + | 10 lines remove current STUN support from chan_sip.c This patch + removes the current broken/useless stun support from chan_sip. + (closes issue #17622) Reported by: philipp2 Review: + https://reviewboard.asterisk.org/r/855/ ........ + + * /, CHANGES: Merged revisions 282271 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) + | 2 lines res_stun_monitor and corresponding options CHANGES + documentation ........ + + * configs/iax.conf.sample, /, channels/chan_sip.c, + include/asterisk/event_defs.h, res/res_stun_monitor.c (added), + configs/res_stun_monitor.conf.sample (added), + configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions + 282269 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) + | 4 lines res_stun_monitor for monitoring network changes behind + a NAT device Review: https://reviewboard.asterisk.org/r/854 + ........ + + * /, channels/chan_sip.c: Merged revisions 282236 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282236 | dvossel | 2010-08-13 13:58:10 -0500 + (Fri, 13 Aug 2010) | 23 lines Merged revisions 282235 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) + | 16 lines only do magic pickup when notifycid is enabled A new + way of doing BLF pickup was introduced into 1.6.2. This feature + adds a call-id value into the XML of a SIP_NOTIFY message sent to + alert a subscriber that a device is ringing. This option should + only be enabled when the new 'notifycid' option is set... but + this was not the case. Instead the call-id value was included for + every RINGING Notify message, which caused a regression for + people who used other methods for call pickup. (closes issue + #17633) Reported by: urosh Patches: chan_sip.txt uploaded by + urosh (license ) blf_cid_issue.diff uploaded by dvossel (license + 671) Tested by: dvossel, urosh, okrief, alecdavis ........ + ................ + +2010-08-13 16:08 +0000 [r282202] Terry Wilson + + * /, configure, configure.ac: Merged revisions 282200-282201 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010) + | 10 lines Detect when libsrtp cannot be linked in a shared + library The libsrtp build system currently does not produce a + shared library or a static library compiled with -fPIC, so on + 64-bit systems it is possible that we will get a compile error if + libsrtp is installed and res_srtp is selected in menuselect. This + patch attempts to detect this situation and provide the user with + instructions to work around the problem. ........ r282201 | + twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines + Whitespace fix :-/ ........ + +2010-08-12 22:52 +0000 [r282132] Jason Parker + + * /, pbx/pbx_config.c: Merged revisions 282131 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282131 | qwell | 2010-08-12 17:51:44 -0500 + (Thu, 12 Aug 2010) | 16 lines Merged revisions 282130 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282130 | qwell | 2010-08-12 17:50:54 -0500 + (Thu, 12 Aug 2010) | 9 lines Merged revisions 282129 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug + 2010) | 1 line Register CLI commands before parsing config, in + case there is a config error. ........ ................ + ................ + +2010-08-12 22:10 +0000 [r282099] Richard Mudgett + + * /, main/ccss.c, include/asterisk/ccss.h: Merged revisions 282098 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) + | 7 lines Separate call completion config parameter allocation + and default initialization. If you ever have a need to reset the + call completion config parameters to defaults, now you can. And + no Virginia, C++ idioms do not always work in C. ........ + +2010-08-12 20:44 +0000 [r282067] Russell Bryant + + * /, CHANGES, main/cli.c: Merged revisions 282066 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) + | 4 lines Add a "core reload" CLI command. Review: + https://reviewboard.asterisk.org/r/859/ ........ + +2010-08-12 20:17 +0000 [r282048] David Vossel + + * /, main/translate.c, CHANGES, include/asterisk/translate.h, + main/cli.c: Merged revisions 282047 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) + | 35 lines improved translation paths for wideband codecs The + problem I'm addressing is that Asterisk's current method of + building the least cost translation paths between codecs does not + take into account sample rate. For instance, it was possible for + siren14 (a 32khz codec), to contain the a translation path to + siren7 (a 16khz audio codec) that goes through slin at 8khz. In + this case Asterisk takes a 32khz codec, down samples it to 8khz + and then up samples it to 16khz which is terrible regardless if + it is computationally less expensive. This patch now builds + translation paths that give priority to maintaining the best + possible sample rate before taking into consideration + computational cost. This patch also adds cli commands to expose + what translation paths are actually being used. Changes: 1. + Translation paths will never contain a step that changes the + sample rate unless absolutely necessary. 2. When choosing the + best codec to make two channels compatible. Shared codecs with + the highest sample rate are given priority. 3. A new cli command + to show all translation paths available for a specific codec + 'core show translation paths [codec name]' has been added. 4. + 'core show translation' which displays the translation matrix now + includes the new higher bit audio codecs in the table. 5. 'core + show channel [channel name]' now displays the translation paths + if translation is used. (closes issue #16841) Reported by: + dvossel Review: https://reviewboard.asterisk.org/r/842/ ........ + +2010-08-12 18:04 +0000 [r281983-282016] Russell Bryant + + * main/pbx.c, /: Merged revisions 282015 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282015 | russell | 2010-08-12 13:03:56 -0500 (Thu, 12 Aug 2010) + | 2 lines Put back pointer value output for ast_debug(), such + that it is only removed for verbose output. ........ + + * main/pbx.c, /: Merged revisions 281982 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) + | 5 lines Remove debugging output from verbose messages. Pointer + values to internal objects is not terribly useful to users in the + verbose messages about adding extensions and contexts. ........ + +2010-08-12 03:08 +0000 [r281914] Jeff Peeler + + * main/channel.c, /: Merged revisions 281913 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281913 | jpeeler | 2010-08-11 22:03:37 -0500 + (Wed, 11 Aug 2010) | 34 lines Merged revisions 281912 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500 + (Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) + | 20 lines Ensure SSRC is changed when media source is changed to + resolve audio delay. This change causes the SSRC to change right + before the channels are bridged, which is what used to happen. It + seems that fixes were made to attempt limiting SSRC changes, + targeted mainly at sending DTMF. DTMF is not affecting the SSRC + with this change. There are two other control frames sent in + ast_channel_bridge that probably should also be changed to + AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change + up to the discretion of resolving issue #17007. For reference - + old review implementing new control frame SRCCHANGE: + https://reviewboard.asterisk.org/r/540 (closes issue #17404) + Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler + (license 325) Tested by: sdolloff ........ ................ + ................ + +2010-08-11 21:13 +0000 [r281877] Leif Madsen + + * /, configs/say.conf.sample: Merged revisions 281875 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281875 | lmadsen | 2010-08-11 16:12:13 -0500 + (Wed, 11 Aug 2010) | 21 lines Merged revisions 281873 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281873 | lmadsen | 2010-08-11 16:09:47 -0500 + (Wed, 11 Aug 2010) | 14 lines Merged revisions 281819 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010) + | 6 lines Add Danish support to say.conf.sample (closes issue + #17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk + uploaded by RoadKill (license 933) ........ ................ + ................ + +2010-08-11 21:12 +0000 [r281876] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 281874 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281874 | mnicholson | 2010-08-11 16:11:54 -0500 (Wed, 11 Aug + 2010) | 10 lines handle all possible responses to REFER requests + (closes issue #17486) Reported by: davidw Patches: + Issue17486-counterbid.diff.txt uploaded by davidw (license 780) + Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/ + ........ + +2010-08-11 20:38 +0000 [r281871] Richard Mudgett + + * channels/sig_analog.c, /, channels/sig_analog.h: Merged revisions + 281870 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281870 | rmudgett | 2010-08-11 15:30:29 -0500 (Wed, 11 Aug 2010) + | 4 lines Fix a call to analog_set_pulsedial() not setting 0 or 1 + only. * Also a couple minor tweaks. ........ + +2010-08-11 17:55 +0000 [r281765] Leif Madsen + + * /, configs/say.conf.sample: Merged revisions 281764 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281764 | lmadsen | 2010-08-11 12:54:56 -0500 + (Wed, 11 Aug 2010) | 21 lines Merged revisions 281763 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281763 | lmadsen | 2010-08-11 12:54:09 -0500 + (Wed, 11 Aug 2010) | 14 lines Merged revisions 281762 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010) + | 6 lines Allow say.conf to handle large numbers ending with + multiple zeros. (closes issue #17833) Reported by: RoadKill + Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill + (license 933) ........ ................ ................ + +2010-08-11 17:29 +0000 [r281761] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 281760 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281760 | mnicholson | 2010-08-11 12:27:59 -0500 (Wed, 11 Aug + 2010) | 4 lines Avoid a deadlock in add_header_max_forwards(). + Related to r276951 ........ + +2010-08-11 15:20 +0000 [r281726] Tilghman Lesher + + * /, apps/app_readexten.c: Merged revisions 281723 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281723 | tilghman | 2010-08-11 10:18:40 -0500 + (Wed, 11 Aug 2010) | 14 lines Merged revisions 281722 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11 Aug 2010) + | 7 lines Only set status TIMEOUT, if we have no digits. (closes + issue #15188) Reported by: jcovert Patches: + app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license + 551) ........ ................ + +2010-08-11 13:31 +0000 [r281688] + + * main/netsock2.c, /, include/asterisk/netsock2.h, + configs/sip.conf.sample, channels/sip/config_parser.c: Merged + revisions 281687 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281687 | simon.perreault | 2010-08-11 09:30:59 -0400 (Wed, 11 + Aug 2010) | 9 lines Fix parsing of IPv6 address literals in + outboundproxy (closes issue #17757) Reported by: oej Patches: + 17757.diff uploaded by sperreault (license 252) sip.conf.diff + uploaded by sperreault (license 252) Tested by: oej ........ + +2010-08-10 21:50 +0000 [r281530-281651] Russell Bryant + + * /, configs/sip.conf.sample, UPGRADE-1.8.txt, + channels/sip/include/sip.h: Merged revisions 281650 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10 + Aug 2010) | 5 lines Change the default value for alwaysauthreject + in sip.conf to "yes". (closes issue #17756) Reported by: oej + ........ + + * /, main/sched.c: Merged revisions 281575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281575 | russell | 2010-08-10 13:05:07 -0500 + (Tue, 10 Aug 2010) | 16 lines Merged revisions 281574 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010) + | 9 lines Don't move the time threshold for running scheduled + events on every iteration. Instead, only calculate the time + threshold each time ast_sched_runq() is called. (closes issue + #17742) Reported by: schmidts Patches: sched.c.patch uploaded by + schmidts (license 1077) ........ ................ + + * apps/app_dial.c, /: Merged revisions 281568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281568 | russell | 2010-08-10 12:48:42 -0500 + (Tue, 10 Aug 2010) | 22 lines Merged revisions 281567 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281567 | russell | 2010-08-10 12:47:13 -0500 + (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) + | 8 lines Reset visible indication after answer. (closes issue + #17641) Reported by: klaus3000 Patches: + ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by + klaus3000 (license 65) Tested by: schmidts ........ + ................ ................ + + * /, channels/chan_sip.c: Merged revisions 281532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281532 | russell | 2010-08-10 11:54:20 -0500 (Tue, 10 Aug 2010) + | 8 lines Ensure that the proper external address is used for the + RTP destination. (closes issue #17044) Reported by: ebroad Tested + by: ebroad Review: https://reviewboard.asterisk.org/r/566/ + ........ + + * /, main/cli.c: Merged revisions 281529 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281529 | russell | 2010-08-10 11:21:58 -0500 (Tue, 10 Aug 2010) + | 8 lines Resolve a problem with channel name tab completion. + Hitting tab without typing any part of a channel name resulted in + no results. This now results in getting a full list of active + channels, just as it did in previous versions of Asterisk. + Review: https://reviewboard.asterisk.org/r/818/ ........ + +2010-08-10 07:26 +0000 [r281498] TransNexus OSP Development + + * apps/app_osplookup.c: Fixed the issue caused by EXTEN including + user parameters. + +2010-08-09 23:04 +0000 [r281467] Jeff Peeler + + * channels/chan_local.c, /: Merged revisions 281466 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281466 | jpeeler | 2010-08-09 18:04:02 -0500 (Mon, 09 + Aug 2010) | 2 lines Add some more stuff to copy from 281429. + ........ + +2010-08-09 20:49 +0000 [r281433] David Vossel + + * /, channels/chan_sip.c: Merged revisions 281432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281432 | dvossel | 2010-08-09 15:47:53 -0500 + (Mon, 09 Aug 2010) | 20 lines Merged revisions 281430 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) + | 13 lines fixes SIP peers memory leak We zeroed out the peer's + addr before it was removed from the peers_by_ip container. This + made it impossible to be removed from the container as the addr + is the key used by the container to find the peer. (closes issue + #17774) Reported by: kkm Patches: + 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888) + 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888) + ........ ................ + +2010-08-09 20:46 +0000 [r281431] Jeff Peeler + + * channels/chan_local.c, /: Merged revisions 281429 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281429 | jpeeler | 2010-08-09 15:43:54 -0500 + (Mon, 09 Aug 2010) | 27 lines Merged revisions 281391 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500 + (Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) + | 13 lines Prevent loss of Caller ID information set on local + channel after masquerade. Caller ID set on the channel before a + masquerade occurs when using a local channel would cause the + information to be lost. The problem was that the information was + set on a channel destined to be hung up. The somewhat confusing + fix is to detect if any Caller ID has been set on the channel and + if so preswap the Caller ID data so that basically the masquerade + puts the data back. (closes issue #17138) Reported by: kobaz + Review: https://reviewboard.asterisk.org/r/847/ ........ + ................ ................ + +2010-08-09 14:52 +0000 [r281359] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 281358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281358 | mnicholson | 2010-08-09 09:49:38 -0500 (Mon, 09 Aug + 2010) | 4 lines Validate minrate, maxrate, and modem settings + before attempting a fax session. FAX-224 ........ + +2010-08-09 14:32 +0000 [r281357] + + * /, configs/sip.conf.sample: Merged revisions 281356 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281356 | simon.perreault | 2010-08-09 10:31:40 -0400 + (Mon, 09 Aug 2010) | 2 lines Added comment about IPv4-mapped IPv6 + addresses and the output of netstat. ........ + +2010-08-09 12:52 +0000 [r281295-281326] Russell Bryant + + * /, configs/cdr.conf.sample: Merged revisions 281325 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281325 | russell | 2010-08-09 07:51:43 -0500 (Mon, 09 + Aug 2010) | 2 lines Add a couple of default values to the + documentation of cdr.conf. ........ + + * /, configs/cdr.conf.sample: Merged revisions 281294 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281294 | russell | 2010-08-09 07:14:34 -0500 (Mon, 09 + Aug 2010) | 5 lines Reorder some options in cdr.conf.sample. Put + all of the options that affect the contents of CDRs together, + instead of having the batch mode options in the middle of them. + ........ + +2010-08-07 22:36 +0000 [r281226-281257] Damien Wedhorn + + * channels/chan_skinny.c: Fix up handling and indications during + transfer. Cleaned up handling of onhook indications and added + indications if more than one sub on device. Also fixes issue in + 12324 so that the phone can call itself without locking up. + (closes issue #17692) Reported by: jmhunter Patches: + chan_skinny-transfer-v4.txt uploaded by DEA (license 3) + skinnytransfver.v8.diff uploaded by wedhorn (license 30) Tested + by: jmhunter, salecha, wedhorn Review: NA + + * channels/chan_skinny.c: Move call answering stuff into new + setsubstate_connected. Move call answering stuff into new + setsubstate_connected. Also add sub->substate var and set it to + SUBSTATE_CONNECTED in setsubstate_connected. (closes issue + #17772) Reported by: wedhorn Patches: + cleanup.stateconnected2.diff uploaded by wedhorn (license 30) + Tested by: wedhorn, salecha Review: NA + + * channels/chan_skinny.c: Start rtp on answer before the answer is + queued (closes issue #17770) Reported by: salecha Patches: + skinny.answercrash.diff uploaded by wedhorn (license 30) Tested + by: salecha Review: NA + +2010-08-06 18:58 +0000 [r281086] Tilghman Lesher + + * /, main/utils.c: Merged revisions 281085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281085 | tilghman | 2010-08-06 13:57:10 -0500 (Fri, 06 Aug 2010) + | 8 lines Fix alignment of stringfields on the SPARC architecture + (closes issue #17789) Reported by: Ian Mason Patches: + 20100806__issue17789__2.diff.txt uploaded by tilghman (license + 14) Tested by: Ian_Mason ........ + +2010-08-05 13:19 +0000 [r281054] Russell Bryant + + * main/cdr.c, /: Merged revisions 281052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281052 | russell | 2010-08-05 08:16:11 -0500 + (Thu, 05 Aug 2010) | 16 lines Merged revisions 281051 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010) + | 9 lines Cleanup default option value handling for cdr.conf + [general]. The default values would differ depending on whether + or not cdr.conf exists. That is no longer the case. Apply a + default value to the unanswered option. Define all default values + as named constants. ........ ................ + +2010-08-05 07:47 +0000 [r280985] Tilghman Lesher + + * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 280984 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280984 | tilghman | 2010-08-05 02:46:36 -0500 + (Thu, 05 Aug 2010) | 22 lines Merged revisions 280983 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500 + (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) + | 8 lines Change context lock back to a mutex, because + functionality depends upon the lock being recursive. (closes + issue #17643) Reported by: zerohalo Patches: + 20100726__issue17643.diff.txt uploaded by tilghman (license 14) + Tested by: zerohalo ........ ................ ................ + +2010-08-04 15:22 +0000 [r280910] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 280909 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280909 | mnicholson | 2010-08-04 10:11:13 -0500 (Wed, 04 Aug + 2010) | 2 lines Initialize FAXOPT() status variables in sendfax + and receivefax instead of when the details structure is created. + ........ + +2010-08-04 14:05 +0000 [r280810-280880] Tilghman Lesher + + * /, channels/chan_mgcp.c: Merged revisions 280879 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280879 | tilghman | 2010-08-04 09:04:07 -0500 (Wed, 04 + Aug 2010) | 14 lines Check cur value before attempting a deref. + (closes issue #17775) Reported by: svinson Patches: + 20100804__issue17775.diff.txt uploaded by tilghman (license 14) + Tested by: svinson (closes issue #17743) Reported by: tgruenberg + Patches: 20100804__issue17775.diff.txt uploaded by tilghman + (license 14) Tested by: tgruenberg ........ + + * /, funcs/func_strings.c, CHANGES: Merged revisions 280809 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280809 | tilghman | 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010) + | 12 lines Sneak FIELDNUM() into 1.8. Returns a 1-based index + into a list of a specified item. Matches up with FIELDQTY() and + CUT(). (closes issue #17713) Reported by: gareth Patches: + svn-279754.diff uploaded by gareth (license 208) Tested by: + gareth, tilghman Review: https://reviewboard.asterisk.org/r/810/ + ........ + +2010-08-03 19:59 +0000 [r280745-280780] + + * /, channels/chan_sip.c: Merged revisions 280778 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280778 | simon.perreault | 2010-08-03 15:54:03 -0400 (Tue, 03 + Aug 2010) | 9 lines Fixed IPv6-related SIP parsing bugs. (closes + issue #17663) Reported by: oej Patches: diff uploaded by + sperreault (license 252) diff2 uploaded by sperreault (license + 252) get_domain.diff uploaded by sperreault (license 252) + ........ + + * /, configs/sip.conf.sample: Merged revisions 280777 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280777 | simon.perreault | 2010-08-03 15:53:07 -0400 + (Tue, 03 Aug 2010) | 8 lines Better documentation related to + IPv6. (closes issue #17737) Reported by: oej Patches: doc.diff + uploaded by sperreault (license 252) Tested by: mmichelson + ........ + + * contrib/realtime/mysql/voicemail.sql, channels/chan_sip.c, + contrib/realtime/mysql/iaxfriends.sql, + contrib/realtime/mysql/meetme.sql, + contrib/realtime/postgresql/realtime.sql, + configs/sip.conf.sample, contrib/realtime/mysql/sipfriends.sql: + Reverted r280706 and r280707. Will commit in branch 1.8 and merge + to trunk properly. + +2010-08-03 18:50 +0000 [r280743] Russell Bryant + + * contrib/scripts/get_mp3_source.sh (added), /, addons/Makefile, + addons/mp3 (removed): Merged revisions 280742 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280742 | russell | 2010-08-03 13:48:45 -0500 (Tue, 03 Aug 2010) + | 4 lines Remove the MP3 decoder source code and replace it with + a small shell script. Review: + https://reviewboard.asterisk.org/r/836/ ........ + +2010-08-03 18:43 +0000 [r280741] Tilghman Lesher + + * /, doc/asterisk.8, doc/Makefile (added), doc/asterisk.sgml: + Merged revisions 280740 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280740 | tilghman | 2010-08-03 13:42:24 -0500 + (Tue, 03 Aug 2010) | 9 lines Merged revisions 280739 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 + Aug 2010) | 2 lines Document -B and -W flags and regenerate + manpage from sgml ........ ................ + +2010-08-03 16:52 +0000 [r280706-280707] + + * channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes + issue #17663) Reported by: oej Patches: diff uploaded by + sperreault (license 252) diff2 uploaded by sperreault (license + 252) get_domain.diff uploaded by sperreault (license 252) + + * configs/sip.conf.sample: Better documentation related to IPv6. + +2010-08-02 21:28 +0000 [r280629-280673] Tilghman Lesher + + * /, apps/app_voicemail.c: Merged revisions 280672 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280672 | tilghman | 2010-08-02 16:27:25 -0500 + (Mon, 02 Aug 2010) | 9 lines Merged revisions 280671 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02 + Aug 2010) | 2 lines Allow the pipe, but also allow the comma + ........ ................ + + * /, main/Makefile: Merged revisions 280628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280628 | tilghman | 2010-08-02 09:41:46 -0500 (Mon, 02 Aug 2010) + | 2 lines Make this a little more deterministic... we want the + latest value, not just a 1 somewhere. ........ + +2010-08-02 14:30 +0000 [r280627] David Vossel + + * channels/chan_sip.c: if totag is not present for an ACK request, + do not send an error response + +2010-08-02 14:28 +0000 [r280626] Tilghman Lesher + + * /, main/Makefile: Merged revisions 280624 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280624 | tilghman | 2010-08-02 09:27:20 -0500 (Mon, 02 Aug 2010) + | 2 lines Apparently, the values in makeopts are sometimes 1:1 + and sometimes 1. Compensate for this. ........ + +2010-07-30 09:12 +0000 [r280589] Damien Wedhorn + + * channels/chan_skinny.c: Cleanup transmit_ for handle_register and + keepalives Moved inline packet sending to transmit_ subs. Removed + handle_keep_alive and handle_register_message to inline in + handle_message. Also moved transmit_response(d) to + transmit_response_bysessions(s) and created a wrapper + transmit_response(d) that calls + transmit_response_bysession(d->session). (closes issue #16980) + Reported by: wedhorn Patches: skinny-clean06b.diff uploaded by + wedhorn (license 30) Tested by: wedhorn, DEA Review: NA + +2010-07-29 21:08 +0000 [r280559] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 280557 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280557 | mnicholson | 2010-07-29 16:07:21 -0500 (Thu, 29 Jul + 2010) | 4 lines Fix regression introduced in r1664. Give the fax + stack time to shutdown and populate the FAXOPT output variables. + FAX-222 ........ + +2010-07-29 21:06 +0000 [r280555] Paul Belanger + + * CHANGES, channels/chan_iax2.c: PeerStatus now includes Address + and Port (closes issue #17730) Reported by: jkroon Patches: + iax2-peerstate-address.patch uploaded by jkroon (license 714) + Tested by: lmadsen + +2010-07-29 20:44 +0000 [r280553] David Vossel + + * /, channels/chan_sip.c: Merged revisions 280552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280552 | dvossel | 2010-07-29 15:43:47 -0500 + (Thu, 29 Jul 2010) | 17 lines Merged revisions 280551 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010) + | 11 lines fixes wrong SRV query for TLS connection (closes issue + #17612) Reported by: marcelloceschia Patches: + chan-sip_srvQuery.patch uploaded by marcelloceschia (license + 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) + chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia + (license 1079) Tested by: marcelloceschia, st, pabelanger + ........ ................ + +2010-07-29 20:36 +0000 [r280550] Russell Bryant + + * /, configs/ccss.conf.sample: Merged revisions 280549 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280549 | russell | 2010-07-29 15:35:30 -0500 (Thu, 29 + Jul 2010) | 5 lines Add header to ccss.conf to appease oej. + (closes issue #17755) Reported by: oej ........ + +2010-07-29 19:48 +0000 [r280520] Sean Bright + + * /, channels/sig_pri.c: Merged revisions 280519 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280519 | seanbright | 2010-07-29 15:47:16 -0400 (Thu, 29 Jul + 2010) | 7 lines Fix compilation error in chan_dahdi (strdupa -> + ast_strdupa). (closes issue #17751) Reported by: b11d Patches: + strdupa_oops.diff uploaded by malcolmd (license 924) ........ + +2010-07-29 19:35 +0000 [r280459-280518] David Vossel + + * channels/chan_sip.c: respond with 481 when request requiring + totag has no totag to match against + + * main/channel.c, /: Merged revisions 280450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280450 | dvossel | 2010-07-29 14:13:27 -0500 + (Thu, 29 Jul 2010) | 25 lines Merged revisions 280449 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500 + (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) + | 12 lines fixes issue with translator frame not getting freed A + translator frame even if it local storage so the translation path + can be freed. This issue prevented g729 licenses from being freed + up. (closes issue #17630) Reported by: manvirr Patches: + encoder_fix.diff uploaded by dvossel (license 671) Tested by: + manvirr, dvossel ........ ................ ................ + +2010-07-29 18:51 +0000 [r280447] Paul Belanger + + * /, tests/test_utils.c: Merged revisions 280446 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280446 | pabelanger | 2010-07-29 14:37:32 -0400 (Thu, 29 Jul + 2010) | 2 lines Remove res_crypto dependency. ........ + +2010-07-29 16:47 +0000 [r280416] Jean Galarneau + + * /, apps/app_meetme.c: Merged revisions 280346 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280346 | jeang | 2010-07-29 11:07:16 -0500 + (Thu, 29 Jul 2010) | 17 lines Merged revisions 280345 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280345 | jeang | 2010-07-29 11:01:35 -0500 + (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | + 2 lines Fix a dsp structure leak occuring when a local channel is + put into a meetme conference, then masquaraded away. ABE-2422 + ........ ................ ................ + +2010-07-29 16:45 +0000 [r280415] Paul Belanger + + * /, tests/test_utils.c: Merged revisions 280414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280414 | pabelanger | 2010-07-29 12:44:22 -0400 (Thu, 29 Jul + 2010) | 2 lines crypto_loaded_test depends on res_crypto, else + test will fail. ........ + +2010-07-29 16:26 +0000 [r280395] Russell Bryant + + * main/rtp_engine.c, /: Merged revisions 280391 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280391 | russell | 2010-07-29 11:25:43 -0500 (Thu, 29 Jul 2010) + | 2 lines Don't blow up if get_codec() was not provided in the + RTP glue. ........ + +2010-07-29 15:58 +0000 [r280308-280344] Matthew Nicholson + + * channels/chan_usbradio.c, /: Merged revisions 280343 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280343 | mnicholson | 2010-07-29 10:57:57 -0500 (Thu, + 29 Jul 2010) | 4 lines Use PRIx64 instead of PRId64 in format + string. related to r280302 ........ + + * channels/chan_usbradio.c, /: Merged revisions 280302 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280302 | pabelanger | 2010-07-28 19:45:34 -0500 (Wed, + 28 Jul 2010) | 2 lines Use PRId64 with format_t ........ + + * channels/chan_usbradio.c: Make chan_usbradio.c build on 64bit + platforms. + + * main/channel.c, channels/chan_local.c, /: Merged revisions 280307 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280307 | mnicholson | 2010-07-29 08:56:35 -0500 + (Thu, 29 Jul 2010) | 11 lines Merged revisions 280306 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul + 2010) | 2 lines Implement support for ast_channel_queryoption on + local channels. Currently only AST_OPTION_T38_STATE is supported. + ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ + Additionally, pass AST_CONTROL_T38_PARAMETERS control frames + through generic bridges. This change appears to have been + unintentionally left out of rev 203699. ................ + +2010-07-28 20:50 +0000 [r280270] Jeff Peeler + + * /, channels/sip/reqresp_parser.c: Merged revisions 280269 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280269 | jpeeler | 2010-07-28 15:49:26 -0500 (Wed, 28 Jul 2010) + | 2 lines Give test category missing leading slash ........ + +2010-07-28 20:19 +0000 [r280247] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 280235 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280235 | rmudgett | 2010-07-28 15:12:16 -0500 + (Wed, 28 Jul 2010) | 9 lines Merged revisions 280229 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28 + Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7 + called_nai and calling_nai config options. ........ + ................ + +2010-07-28 20:04 +0000 [r280234] Jason Parker + + * /, sounds/Makefile: Merged revisions 280233 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280233 | qwell | 2010-07-28 15:03:22 -0500 + (Wed, 28 Jul 2010) | 13 lines Merged revisions 280231 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) | + 6 lines Work around some silly behavior on BSD. A non-zero exit + from a subshell should make the build fail. (closes issue #17621) + ........ ................ + +2010-07-28 19:37 +0000 [r280226] Terry Wilson + + * res/res_rtp_asterisk.c, /: Merged revisions 280225 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280225 | twilson | 2010-07-28 12:34:42 -0700 (Wed, 28 + Jul 2010) | 3 lines Do rtp/rtcp debugging when it is turned on + w/o filtering ........ + +2010-07-28 18:25 +0000 [r280196] Jason Parker + + * /, sounds/Makefile: Merged revisions 280195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280195 | qwell | 2010-07-28 13:24:29 -0500 + (Wed, 28 Jul 2010) | 16 lines Merged revisions 280193 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) | + 9 lines Remove unnecessary subshells. Attempt to make + checksumming work. Also improves readability. (issue #17621) + Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/ + ........ ................ + +2010-07-28 16:53 +0000 [r280162] Sean Bright + + * /, apps/app_queue.c: Merged revisions 280161 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280161 | seanbright | 2010-07-28 12:52:12 -0400 + (Wed, 28 Jul 2010) | 15 lines Merged revisions 280160 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul + 2010) | 8 lines Plug a reference leak in app_queue when adding + members dynamically. (closes issue #17738) Reported by: + bobwienholt Patches: issue17738.patch uploaded by bobwienholt + (license 950) Tested by: bobwienholt, seanbright ........ + ................ + +2010-07-28 14:14 +0000 [r280093] Olle Johansson + + * channels/chan_sip.c: Formatting changes + +2010-07-28 13:53 +0000 [r280091] Leif Madsen + + * contrib/scripts/live_ast, /: Merged revisions 280090 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280090 | lmadsen | 2010-07-28 08:52:50 -0500 + (Wed, 28 Jul 2010) | 16 lines Merged revisions 280089 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500 + (Wed, 28 Jul 2010) | 9 lines Merged revisions 280088 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 + Jul 2010) | 1 line Update help text to be less confusing. + ........ ................ ................ + +2010-07-28 13:02 +0000 [r280059] Russell Bryant + + * /, res/res_crypto.c: Merged revisions 280058 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280058 | russell | 2010-07-28 08:01:15 -0500 (Wed, 28 Jul 2010) + | 2 lines s/init keys/keys init/ ........ + +2010-07-28 01:39 +0000 [r280024] Paul Belanger + + * channels/chan_usbradio.c, /: Merged revisions 280023 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280023 | pabelanger | 2010-07-27 21:37:10 -0400 (Tue, + 27 Jul 2010) | 5 lines Resolve compiler warning about formatting + (closes issue #17732) Reported by: pabelanger ........ + +2010-07-27 21:16 +0000 [r279954] Russell Bryant + + * /, utils, codecs, main/db1-ast/mpool, Makefile.rules, cdr, + formats, codecs/gsm/src, bridges, codecs/lpc10, configure, + main/editline, channels/sip, pbx, res/ael, channels, + main/stdtime, main/editline/np, main/db1-ast/hash, cel, apps, + configure.ac, main/db1-ast/db, res/ais, res/snmp, funcs, + main/db1-ast/btree, codecs/g722, main, main/db1-ast/recno, + makeopts.in, res: Merged revisions 279953 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279953 | russell | 2010-07-27 16:16:05 -0500 (Tue, 27 Jul 2010) + | 5 lines Add --enable-coverage option to configure script. This + option enables the proper compiler flags for tracking code + coverage, which is useful along side automated testing. ........ + +2010-07-27 20:59 +0000 [r279951] David Vossel + + * main/channel.c, /, include/asterisk/audiohook.h, + main/audiohook.c: Merged revisions 279949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279949 | dvossel | 2010-07-27 15:57:00 -0500 + (Tue, 27 Jul 2010) | 31 lines Merged revisions 279946 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500 + (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) + | 19 lines remove empty audiohook write list on channel If a + channel has an audiohook write list created on it, that list + stays on the channel until the channel is destroyed. There is no + reason to keep that list on the channel if it becomes empty. If + it is empty that just means we are doing needless translating for + every ast_read and ast_write. This patch removes the audiohook + list from the channel once it is detected to be empty on either a + read or write. If a audiohook is added back to the channel after + this list is destroyed, the list just gets recreated as if it + never existed to begin with. (closes issue #17630) Reported by: + manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ + ................ ................ + +2010-07-27 19:55 +0000 [r279917] Russell Bryant + + * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions + 279916 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010) + | 12 lines Fix inband DTMF detection on outgoing ISDN calls. This + is a regression from the sig_pri split from chan_dahdi. When a + call is first initiated, the inband DTMF detector is not enabled + if it's an outgoing ISDN call. However, it needs to be turned on + once the media path starts up. This handling was put back in the + open_media() callback of chan_dahdi. In sig_pri, open_media() + calls were added to a few places where it was needed, including + handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and + PRI_EVENT_PROCEEDING. Thanks to rmudgett for helping me with the + patch! ........ + +2010-07-27 18:55 +0000 [r279888] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 279887 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279887 | mmichelson | 2010-07-27 13:54:07 -0500 (Tue, 27 Jul + 2010) | 16 lines Fix parsing error in sip_sipredirect(). The code + was written in a way that did a bad job of parsing the port out + of a URI. Specifically, it would do badly when dealing with an + IPv6 address. In this particular scenario, there was no value + from parsing the port out, so I just removed that logic. And + while I was messing around in the function, I changed some + variable names to be more descriptive. (closes issue #17661) + Reported by: oej Patches: 17661.diff uploaded by mmichelson + (license 60) ........ + +2010-07-27 16:41 +0000 [r279851] Jason Parker + + * /, sounds/Makefile: Merged revisions 279850 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279850 | qwell | 2010-07-27 11:40:05 -0500 + (Tue, 27 Jul 2010) | 9 lines Merged revisions 279849 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul + 2010) | 1 line Simply sounds/Makefile some more. ........ + ................ + +2010-07-27 16:11 +0000 [r279818] David Vossel + + * main/netsock2.c, /, channels/chan_sip.c: Merged revisions 279817 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279817 | dvossel | 2010-07-27 11:09:15 -0500 (Tue, 27 Jul 2010) + | 2 lines fix sip transaction match with authentication, fix + confusing log message when using getaddrinfo ........ + +2010-07-27 16:08 +0000 [r279816] Russell Bryant + + * main/channel.c, channels/chan_dahdi.c, /: Merged revisions + 279636,279815 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279636 | russell | 2010-07-26 16:53:30 -0500 (Mon, 26 Jul 2010) + | 2 lines Ignore a control subclass of -1 in + ast_waitfordigit_full(). ........ r279815 | russell | 2010-07-27 + 11:06:58 -0500 (Tue, 27 Jul 2010) | 4 lines Support "channels" in + addition to "channel" in chan_dahdi.conf. Review: + https://reviewboard.asterisk.org/r/804 ........ + +2010-07-27 15:16 +0000 [r279786] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 279785 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279785 | mmichelson | 2010-07-27 10:15:22 -0500 + (Tue, 27 Jul 2010) | 20 lines Merged revisions 279784 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul + 2010) | 14 lines Fix bad behavior of dynamic_exclude_static + option in sip.conf. We were attempting to create a contactdeny + rule based on the peer's IP address before the peer's IP address + had been set. By moving the processing further down in the + function, we can ensure stuff works as we expect for it to. + (closes issue #17717) Reported by: mmichelson Patches: + 17717.patch uploaded by mmichelson (license 60) Tested by: + DennisD ........ ................ + +2010-07-27 03:02 +0000 [r279727-279756] Paul Belanger + + * channels/chan_dahdi.c, /: Merged revisions 279755 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r279755 | pabelanger | 2010-07-26 22:57:33 -0400 (Mon, + 26 Jul 2010) | 10 lines If dringXcontext is null, fallback to + default context value. (closes issue #17693) Reported by: + iasgoscouk Patches: issue17693.patch uploaded by pabelanger + (license 224) Tested by: iasgoscouk Review: + https://reviewboard.asterisk.org/r/803/ ........ + + * /, main/http.c: Merged revisions 279726 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279726 | pabelanger | 2010-07-26 21:53:38 -0400 (Mon, 26 Jul + 2010) | 9 lines Use ast_sockaddr_setnull() when http is not + enabled. Otherwise, ast_tcptls_server_start() will still start + http. (closes issue #17708) Reported by: pabelanger Patches: + http.patch uploaded by pabelanger (license 224) ........ + +2010-07-27 01:39 +0000 [r279725] Russell Bryant + + * CHANGES: Make a formatting change. (Demonstrating the commit IRC + bot to pabelanger) + +2010-07-26 23:35 +0000 [r279692] Paul Belanger + + * /, CHANGES, UPGRADE-1.8.txt: Merged revisions 279689 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon, + 26 Jul 2010) | 2 lines Updated documentation for FAX logger + level. ........ + +2010-07-26 23:06 +0000 [r279659] Jason Parker + + * /, sounds/Makefile.380 (removed), configure, + include/asterisk/autoconfig.h.in, sounds/Makefile.381 (removed), + configure.ac, sounds/Makefile (added): Merged revisions 279658 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279658 | qwell | 2010-07-26 18:03:38 -0500 + (Mon, 26 Jul 2010) | 12 lines Merged revisions 279657 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) | + 5 lines Really fix sounds Makefile (and make it readableish). + There was a rather large syntax error that should have caused ALL + versions of GNU make to fail. I don't know how it worked. + ........ ................ + +2010-07-26 21:21 +0000 [r279602-279624] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 279619 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279619 | tilghman | 2010-07-26 16:20:12 -0500 + (Mon, 26 Jul 2010) | 9 lines Merged revisions 279609 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26 + Jul 2010) | 2 lines Dunno why this worked on my machine, but it + works better this way. ........ ................ + + * /, res/res_config_ldap.c: Merged revisions 279601 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279601 | tilghman | 2010-07-26 16:07:45 -0500 + (Mon, 26 Jul 2010) | 19 lines Merged revisions 279597 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26 Jul 2010) | + 13 lines Apply all patches in: + https://issues.asterisk.org/view.php?id=13573 (closes issue + #13573) Reported by: navkumar Patches: + res_config_ldap-category.diff uploaded by navkumar (license 580) + res_config_ldap.patch uploaded by bencer (license 961) + res_config_ldap uploaded by bencer (license 961) Tested by: + suretec ........ ................ + +2010-07-26 21:07 +0000 [r279600] Gavin Henry + + * /: Merged revisions 279598 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279598 | ghenry | 2010-07-26 21:58:12 +0100 (Mon, 26 Jul 2010) | + 21 lines Merged revisions 279597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/1.6.2 + ----------------------------------------------------------------------- + r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) | + 13 lines Apply all patches in: + https://issues.asterisk.org/view.php?id=13573 [^] (closes issue + 0013573) Reported by: navkumar Patches: + res_config_ldap-category.diff uploaded by navkumar (license 580) + res_config_ldap.patch uploaded by bencer (license 961) + res_config_ldap uploaded by bencer (license 961) Tested by: + suretec + ------------------------------------------------------------------------ + ........ + +2010-07-26 20:00 +0000 [r279569] David Vossel + + * /, channels/chan_sip.c, channels/sip/reqresp_parser.c, + channels/sip/include/sip.h, + channels/sip/include/reqresp_parser.h: Merged revisions 279568 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279568 | dvossel | 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) + | 21 lines transaction matching using top most Via header This + patch modifies the way chan_sip.c does transaction to dialog + matching. Asterisk now stores information in the top most Via + header of the initial incoming request and compares that against + other Requests that have the same call-id. This results in + Asterisk being able to detect a forked call in which it has + received multiple legs of the fork. I completely stripped out the + previous matching code and made the comparisons a little more + explicit and easier to understand. My comments in the code should + offer all the details involving this patch. This patch also fixes + a bug with the usage of the OBJ-MULTIPLE flag to find multiple + dialogs with the same call-id. Since the callback function was + returning (CMP_MATCH | CMP_STOP) only the first item found was + being returned. I fixed this by making a new callback function + for finding multiple dialogs that only returns (CMP_MATCH) on a + match allowing for multiple items to be returned. Review: + https://reviewboard.asterisk.org/r/776/ ........ + +2010-07-26 19:58 +0000 [r279567] Paul Belanger + + * /, CHANGES, UPGRADE-1.8.txt, configs/logger.conf.sample: Merged + revisions 279566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279566 | pabelanger | 2010-07-26 15:51:39 -0400 (Mon, 26 Jul + 2010) | 8 lines Add documentation for FAX logger level. (closes + issue #17715) Reported by: vrban Patches: 17715.patch uploaded by + pabelanger (license 224) Tested by: vrban ........ + +2010-07-26 19:20 +0000 [r279564] Tilghman Lesher + + * /, sounds/Makefile.380 (added), configure, + include/asterisk/autoconfig.h.in, sounds/Makefile.381 (added), + configure.ac, sounds/Makefile (removed): Merged revisions 279562 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279562 | tilghman | 2010-07-26 14:18:26 -0500 + (Mon, 26 Jul 2010) | 9 lines Merged revisions 279561 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 + Jul 2010) | 2 lines Use a special Makefile for noobs who still + have GNU Make 3.80. ........ ................ + +2010-07-26 16:44 +0000 [r279533] Mark Michelson + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + channels/sip/reqresp_parser.c: Merged revisions 279504 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279504 | mmichelson | 2010-07-26 11:04:09 -0500 (Mon, 26 Jul + 2010) | 14 lines Allow for systems without locale support to be + usable. A recent change to SIP URI comparison code added a + locale-specific string comparison to the mix, and certain systems + do not support such functions. This fix allows for those systems + to still use Asterisk 1.8 (closes issue #17697) Reported by: + pprindeville Patches: asterisk-trunk-bugid17697.patch uploaded by + pprindeville (license 347) Tested by: mmichelson ........ + +2010-07-26 15:44 +0000 [r279503] Sean Bright + + * /, autoconf/ast_ext_lib.m4: Merged revisions 279502 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279502 | seanbright | 2010-07-26 11:43:54 -0400 + (Mon, 26 Jul 2010) | 12 lines Merged revisions 279501 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon, 26 Jul + 2010) | 5 lines Expand the correct value within AST_OPTION_ONLY. + (closes issue #17703) Reported by: stuarth ........ + ................ + +2010-07-26 03:28 +0000 [r279473] Tilghman Lesher + + * /, formats/format_sln.c, formats/format_wav.c, + formats/format_ogg_vorbis.c, formats/format_wav_gsm.c, + formats/format_sln16.c, formats/format_siren7.c, + formats/format_ilbc.c, formats/format_vox.c, + formats/format_pcm.c, formats/format_h263.c, + formats/format_g723.c, formats/format_h264.c, + formats/format_siren14.c, formats/format_jpeg.c, + formats/format_g726.c, formats/format_gsm.c, + formats/format_g719.c, formats/format_g729.c: Merged revisions + 279472 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279472 | tilghman | 2010-07-25 22:27:06 -0500 (Sun, 25 Jul 2010) + | 2 lines Formats need to load before apps, because some apps + call ast_format_str_reduce() at load time. ........ + +2010-07-25 21:28 +0000 [r279443] Paul Belanger + + * /, tests/test_func_file.c: Merged revisions 279442 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r279442 | pabelanger | 2010-07-25 17:26:42 -0400 (Sun, + 25 Jul 2010) | 2 lines Add trailing backslash to silence warning + message. ........ + +2010-07-25 18:22 +0000 [r279391-279413] Tilghman Lesher + + * /, cdr/cdr_odbc.c: Merged revisions 279410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279410 | tilghman | 2010-07-25 13:21:27 -0500 (Sun, 25 Jul 2010) + | 8 lines Don't re-register CDR module on reload. (closes issue + #17304) Reported by: jnemeth Patches: + 20100507__issue17304.diff.txt uploaded by tilghman (license 14) + Tested by: jnemeth ........ + + * /, main/logger.c: Merged revisions 279390 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279390 | tilghman | 2010-07-25 12:32:21 -0500 (Sun, 25 Jul 2010) + | 8 lines Don't assume qlog is open. (closes issue #17704) + Reported by: vrban Patches: issue17704.patch uploaded by + pabelanger (license 224) Tested by: vrban ........ + +2010-07-24 20:49 +0000 [r279274-279315] Paul Belanger + + * Makefile, /: Merged revisions 279314 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279314 | pabelanger | 2010-07-24 16:47:52 -0400 (Sat, 24 Jul + 2010) | 7 lines Remove duplicate -c flag when using $(INSTALL) + (closes issue #17695) Reported by: pabelanger Patches: + Makefile.diff uploaded by pabelanger (license 224) ........ + + * /, include/asterisk/netsock2.h: Merged revisions 279280 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279280 | pabelanger | 2010-07-24 14:18:43 -0400 (Sat, 24 Jul + 2010) | 8 lines Check if ast_sockaddr is NULL then return. + (closes issue #17677) Reported by: outcast Patches: + issue0017677.patch uploaded by pabelanger (license 224) Tested + by: elguero ........ + + * main/manager.c, /: Merged revisions 279273 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279273 | pabelanger | 2010-07-24 13:36:42 -0400 (Sat, 24 Jul + 2010) | 6 lines Default sin_family to AF_INET for TCP / TLS + Bindaddress. Otherwise, 'manager show settings' will generate + errors if manager is not enabled. ........ + +2010-07-23 22:24 +0000 [r279156-279245] Richard Mudgett + + * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 279227 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279227 | rmudgett | 2010-07-23 17:20:47 -0500 + (Fri, 23 Jul 2010) | 21 lines Merged revisions 279207 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500 + (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) + | 7 lines SIP promiscuous redirect could fail to dial the + redirect. The ast_channel was created with one variable to + ast_request() but the call to ast_call() that initiates the + outgoing call was using a different variable. The two variables + are not equivalent if the call_forward string included a channel + technology specifier. e.g., SIP/200 ........ ................ + ................ + + * channels/chan_dahdi.c: Make "dahdi show channels" show an + outgoing called number. The "dahdi show channels" extension + column previously only showed the called number of an incoming + call. It now shows the called number for an outgoing call as + well. (closes issue #17653) Reported by: amazinzay Patches: + issue17653_trunk.txt uploaded by rmudgett (license 664) + +2010-07-23 19:17 +0000 [r279116-279118] Russell Bryant + + * UPGRADE.txt, UPGRADE-1.8.txt (added): Shuffle UPGRADE.txt files + for 1.10. + + * CHANGES: Start a new section in CHANGES for 1.10. + +2010-07-23 18:56 +0000 [r279115] Tilghman Lesher + + * /, res/res_odbc.c: Merged revisions 279113 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279113 | tilghman | 2010-07-23 13:56:04 -0500 (Fri, 23 Jul 2010) + | 2 lines Silly 64-bit compilers (who uses 64-bit anyway?) + ........ + +2010-07-23 18:22 +0000 [r279063-279084] Russell Bryant + + * /: Remove old properties. + + * /: Add branch-1.8-merged and branch-1.8-blocked properties to + trunk. +