[Pkg-voip-commits] [asterisk] 03/05: Merge tag 'upstream/13.0.0_dfsg'
tzafrir at debian.org
tzafrir at debian.org
Sun Oct 26 08:15:59 UTC 2014
This is an automated email from the git hooks/post-receive script.
tzafrir pushed a commit to branch master
in repository asterisk.
commit e9e591cabc3ebc65bce1b20a54bc63ccc2c8ddb3
Merge: 0fbb123 8dc0777
Author: Tzafrir Cohen <tzafrir at debian.org>
Date: Sun Oct 26 09:03:00 2014 +0200
Merge tag 'upstream/13.0.0_dfsg'
Upstream version 13.0.0~dfsg
.version | 2 +-
CHANGES | 1969 +-
CREDITS | 398 +-
ChangeLog | 45490 +++++++------------
LICENSE | 18 +-
Makefile | 170 +-
Makefile.moddir_rules | 2 +-
Makefile.rules | 2 +-
README | 6 +-
README-SERIOUSLY.bestpractices.txt | 3 +-
UPGRADE-11.txt | 280 +
UPGRADE-12.txt | 478 +
UPGRADE.txt | 707 +-
addons/Makefile | 3 +-
addons/app_mysql.c | 76 +-
addons/app_saycountpl.c | 138 -
addons/cdr_mysql.c | 25 +-
addons/chan_mobile.c | 264 +-
addons/chan_ooh323.c | 501 +-
addons/chan_ooh323.h | 5 +-
addons/format_mp3.c | 11 +-
addons/ooh323cDriver.c | 159 +-
addons/ooh323cDriver.h | 6 +-
addons/res_config_mysql.c | 154 +-
agi/Makefile | 2 +-
apps/Makefile | 3 +-
apps/app_adsiprog.c | 22 +-
apps/app_agent_pool.c | 2678 ++
apps/app_alarmreceiver.c | 1016 +-
apps/app_amd.c | 44 +-
apps/app_authenticate.c | 10 +-
apps/app_bridgewait.c | 514 +
apps/app_cdr.c | 212 +-
apps/app_celgenuserevent.c | 12 +-
apps/app_chanisavail.c | 7 +-
apps/app_channelredirect.c | 6 +-
apps/app_chanspy.c | 204 +-
apps/app_confbridge.c | 2208 +-
apps/app_controlplayback.c | 129 +-
apps/app_dahdibarge.c | 311 -
apps/app_dahdiras.c | 9 +-
apps/app_db.c | 2 +-
apps/app_dial.c | 410 +-
apps/app_dictate.c | 19 +-
apps/app_directed_pickup.c | 208 +-
apps/app_directory.c | 142 +-
apps/app_disa.c | 18 +-
apps/app_dumpchan.c | 39 +-
apps/app_echo.c | 7 +-
apps/app_externalivr.c | 5 +-
apps/app_fax.c | 104 +-
apps/app_festival.c | 50 +-
apps/app_followme.c | 101 +-
apps/app_forkcdr.c | 288 +-
apps/app_getcpeid.c | 3 +-
apps/app_ices.c | 21 +-
apps/app_image.c | 5 +-
apps/app_ivrdemo.c | 5 +-
apps/app_jack.c | 70 +-
apps/app_macro.c | 9 +-
apps/app_meetme.c | 630 +-
apps/app_milliwatt.c | 17 +-
apps/app_minivm.c | 69 +-
apps/app_mixmonitor.c | 320 +-
apps/app_morsecode.c | 5 +-
apps/app_mp3.c | 36 +-
apps/app_nbscat.c | 37 +-
apps/app_originate.c | 35 +-
apps/app_osplookup.c | 114 +-
apps/app_page.c | 44 +-
apps/app_parkandannounce.c | 249 -
apps/app_playback.c | 20 +-
apps/app_queue.c | 3025 +-
apps/app_readfile.c | 134 -
apps/app_record.c | 66 +-
apps/app_saycounted.c | 5 +-
apps/app_sayunixtime.c | 16 +-
apps/app_senddtmf.c | 17 +-
apps/app_setcallerid.c | 5 +-
apps/app_skel.c | 72 +-
apps/app_sms.c | 22 +-
apps/app_speech_utils.c | 31 +-
apps/app_stack.c | 90 +-
apps/app_stasis.c | 114 +
apps/app_talkdetect.c | 21 +-
apps/app_test.c | 22 +-
apps/app_transfer.c | 2 +-
apps/app_url.c | 5 +-
apps/app_userevent.c | 77 +-
apps/app_verbose.c | 2 +-
apps/app_voicemail.c | 865 +-
apps/app_waitforring.c | 5 +-
apps/app_waitforsilence.c | 18 +-
apps/app_waituntil.c | 2 +-
apps/app_zapateller.c | 5 +-
apps/confbridge/conf_chan_announce.c | 210 +
apps/confbridge/conf_chan_record.c | 104 +
apps/confbridge/conf_config_parser.c | 892 +-
apps/confbridge/conf_state.c | 40 +-
apps/confbridge/conf_state_empty.c | 44 +-
apps/confbridge/conf_state_inactive.c | 41 +-
apps/confbridge/conf_state_multi.c | 39 +-
apps/confbridge/conf_state_multi_marked.c | 144 +-
apps/confbridge/conf_state_single.c | 50 +-
apps/confbridge/conf_state_single_marked.c | 46 +-
apps/confbridge/confbridge_manager.c | 549 +
apps/confbridge/include/conf_state.h | 36 +-
apps/confbridge/include/confbridge.h | 301 +-
asterisk-11.13.1-summary.html | 65 -
asterisk-11.13.1-summary.txt | 95 -
asterisk-13.0.0-summary.html | 6034 +++
asterisk-13.0.0-summary.txt | 8206 ++++
autoconf/ast_ext_tool_check.m4 | 1 +
bridges/Makefile | 2 +-
bridges/bridge_builtin_features.c | 533 +-
bridges/bridge_builtin_interval_features.c | 217 +
bridges/bridge_holding.c | 449 +
bridges/bridge_multiplexed.c | 432 -
bridges/bridge_native_rtp.c | 491 +
bridges/bridge_simple.c | 62 +-
bridges/bridge_softmix.c | 755 +-
build_tools/cflags-devmode.xml | 3 -
build_tools/cflags.xml | 18 +-
build_tools/install_subst | 43 +
build_tools/make_buildopts_h | 3 +
build_tools/make_linker_version_script | 3 +-
build_tools/make_version | 8 +-
build_tools/menuselect-deps.in | 5 +-
build_tools/mkpkgconfig | 1 +
build_tools/post_process_documentation.py | 9 +-
cdr/Makefile | 2 +-
cdr/cdr_adaptive_odbc.c | 31 +-
cdr/cdr_csv.c | 17 +-
cdr/cdr_custom.c | 28 +-
cdr/cdr_manager.c | 38 +-
cdr/cdr_odbc.c | 38 +-
cdr/cdr_pgsql.c | 106 +-
cdr/cdr_radius.c | 22 +-
cdr/cdr_sqlite.c | 10 +-
cdr/cdr_sqlite3_custom.c | 9 +-
cdr/cdr_syslog.c | 30 +-
cdr/cdr_tds.c | 17 +-
cel/Makefile | 2 +-
cel/cel_custom.c | 18 +-
cel/cel_manager.c | 22 +-
cel/cel_odbc.c | 31 +-
cel/cel_pgsql.c | 92 +-
cel/cel_radius.c | 19 +-
cel/cel_sqlite3_custom.c | 19 +-
cel/cel_tds.c | 18 +-
channels/Makefile | 86 +-
channels/chan_agent.c | 2593 --
channels/chan_alsa.c | 70 +-
channels/chan_bridge.c | 235 -
channels/chan_bridge_media.c | 221 +
channels/chan_console.c | 76 +-
channels/chan_dahdi.c | 3920 +-
channels/chan_dahdi.h | 825 +
channels/chan_gtalk.c | 2415 -
channels/chan_h323.c | 3505 --
channels/chan_iax2.c | 3633 +-
channels/chan_jingle.c | 2051 -
channels/chan_local.c | 1484 -
channels/chan_mgcp.c | 688 +-
channels/chan_misdn.c | 331 +-
channels/chan_motif.c | 403 +-
channels/chan_multicast_rtp.c | 49 +-
channels/chan_nbs.c | 72 +-
channels/chan_oss.c | 83 +-
channels/chan_phone.c | 241 +-
channels/chan_pjsip.c | 2328 +
channels/chan_sip.c | 5605 ++-
channels/chan_skinny.c | 3773 +-
channels/chan_unistim.c | 809 +-
channels/chan_vpb.cc | 192 +-
channels/dahdi/bridge_native_dahdi.c | 925 +
channels/dahdi/bridge_native_dahdi.h | 47 +
channels/h323/ChangeLog | 43 -
channels/h323/INSTALL.openh323 | 18 -
channels/h323/Makefile.in | 53 -
channels/h323/README | 144 -
channels/h323/TODO | 9 -
channels/h323/ast_h323.cxx | 2678 --
channels/h323/ast_h323.h | 187 -
channels/h323/ast_ptlib.h | 34 -
channels/h323/caps_h323.cxx | 383 -
channels/h323/caps_h323.h | 172 -
channels/h323/chan_h323.h | 275 -
channels/h323/cisco-h225.asn | 74 -
channels/h323/cisco-h225.cxx | 853 -
channels/h323/cisco-h225.h | 300 -
channels/h323/compat_h323.cxx | 139 -
channels/h323/compat_h323.h | 96 -
channels/h323/noexport.map | 5 -
channels/iax2-parser.c | 1306 -
channels/iax2-parser.h | 177 -
channels/iax2-provision.c | 565 -
channels/iax2-provision.h | 53 -
channels/iax2.h | 301 -
channels/iax2/codec_pref.c | 534 +
channels/iax2/firmware.c | 340 +
channels/iax2/format_compatibility.c | 136 +
channels/iax2/include/codec_pref.h | 150 +
channels/iax2/include/firmware.h | 105 +
channels/iax2/include/format_compatibility.h | 65 +
channels/iax2/include/iax2.h | 301 +
channels/iax2/include/parser.h | 179 +
channels/iax2/include/provision.h | 58 +
channels/iax2/parser.c | 1337 +
channels/iax2/provision.c | 569 +
channels/pjsip/dialplan_functions.c | 912 +
channels/pjsip/include/chan_pjsip.h | 58 +
channels/pjsip/include/dialplan_functions.h | 76 +
channels/sig_analog.c | 322 +-
channels/sig_pri.c | 813 +-
channels/sig_pri.h | 52 +-
channels/sig_ss7.c | 1987 +-
channels/sig_ss7.h | 108 +-
channels/sip/config_parser.c | 62 +-
channels/sip/dialplan_functions.c | 8 +-
channels/sip/include/config_parser.h | 2 +-
channels/sip/include/reqresp_parser.h | 24 +
channels/sip/include/route.h | 120 +
channels/sip/include/sdp_crypto.h | 84 -
channels/sip/include/sip.h | 140 +-
channels/sip/include/srtp.h | 59 -
channels/sip/reqresp_parser.c | 109 +-
channels/sip/route.c | 205 +
channels/sip/sdp_crypto.c | 318 -
channels/sip/security_events.c | 21 +-
channels/sip/srtp.c | 55 -
channels/sip/utils.c | 2 +-
codecs/Makefile | 2 +-
codecs/codec_a_mu.c | 42 +-
codecs/codec_adpcm.c | 53 +-
codecs/codec_alaw.c | 52 +-
codecs/codec_dahdi.c | 433 +-
codecs/codec_g722.c | 65 +-
codecs/codec_g726.c | 65 +-
codecs/codec_gsm.c | 56 +-
codecs/codec_ilbc.c | 46 +-
codecs/codec_lpc10.c | 53 +-
codecs/codec_resample.c | 78 +-
codecs/codec_speex.c | 124 +-
codecs/codec_ulaw.c | 80 +-
codecs/ex_adpcm.h | 3 +-
codecs/ex_alaw.h | 2 +-
codecs/ex_g722.h | 2 +-
codecs/ex_g726.h | 2 +-
codecs/ex_gsm.h | 3 +-
codecs/ex_ilbc.h | 3 +-
codecs/ex_lpc10.h | 2 +-
codecs/ex_speex.h | 5 +-
codecs/ex_ulaw.h | 3 +-
codecs/log2comp.h | 2 +-
codecs/speex/speex_resampler.h | 20 +-
config.guess | 329 +-
config.sub | 85 +-
configs/agents.conf.sample | 97 -
configs/alarmreceiver.conf.sample | 80 -
configs/asterisk.conf.sample | 102 -
configs/cdr_pgsql.conf.sample | 15 -
configs/cel.conf.sample | 126 -
configs/cel_pgsql.conf.sample | 67 -
configs/chan_dahdi.conf.sample | 1618 -
configs/cli_aliases.conf.sample | 201 -
configs/confbridge.conf.sample | 368 -
configs/extconfig.conf.sample | 102 -
configs/extensions.conf.sample | 858 -
configs/extensions_minivm.conf.sample | 159 -
configs/features.conf.sample | 236 -
configs/gtalk.conf.sample | 27 -
configs/h323.conf.sample | 210 -
configs/http.conf.sample | 86 -
configs/iax.conf.sample | 668 -
configs/jabber.conf.sample | 39 -
configs/jingle.conf.sample | 20 -
configs/logger.conf.sample | 127 -
configs/manager.conf.sample | 155 -
configs/motif.conf.sample | 92 -
configs/ooh323.conf.sample | 204 -
configs/phoneprov.conf.sample | 137 -
configs/queuerules.conf.sample | 20 -
configs/queues.conf.sample | 574 -
configs/res_ldap.conf.sample | 198 -
configs/res_odbc.conf.sample | 131 -
configs/res_pgsql.conf.sample | 29 -
configs/{ => samples}/acl.conf.sample | 0
configs/{ => samples}/adsi.conf.sample | 0
configs/samples/agents.conf.sample | 70 +
configs/samples/alarmreceiver.conf.sample | 91 +
configs/{ => samples}/alsa.conf.sample | 0
configs/{ => samples}/amd.conf.sample | 0
configs/{ => samples}/app_mysql.conf.sample | 0
configs/{ => samples}/app_skel.conf.sample | 0
configs/samples/ari.conf.sample | 31 +
configs/{ => samples}/asterisk.adsi | 0
configs/samples/asterisk.conf.sample | 97 +
configs/{ => samples}/calendar.conf.sample | 0
configs/{ => samples}/ccss.conf.sample | 0
configs/{ => samples}/cdr.conf.sample | 0
.../{ => samples}/cdr_adaptive_odbc.conf.sample | 0
configs/{ => samples}/cdr_custom.conf.sample | 0
configs/{ => samples}/cdr_manager.conf.sample | 0
configs/{ => samples}/cdr_mysql.conf.sample | 0
configs/{ => samples}/cdr_odbc.conf.sample | 0
configs/samples/cdr_pgsql.conf.sample | 16 +
.../{ => samples}/cdr_sqlite3_custom.conf.sample | 0
configs/{ => samples}/cdr_syslog.conf.sample | 0
configs/{ => samples}/cdr_tds.conf.sample | 0
configs/samples/cel.conf.sample | 116 +
configs/{ => samples}/cel_custom.conf.sample | 0
configs/{ => samples}/cel_odbc.conf.sample | 0
configs/samples/cel_pgsql.conf.sample | 68 +
.../{ => samples}/cel_sqlite3_custom.conf.sample | 0
configs/{ => samples}/cel_tds.conf.sample | 0
configs/samples/chan_dahdi.conf.sample | 1695 +
configs/{ => samples}/chan_mobile.conf.sample | 0
configs/{ => samples}/cli.conf.sample | 0
configs/samples/cli_aliases.conf.sample | 203 +
configs/{ => samples}/cli_permissions.conf.sample | 0
configs/{ => samples}/codecs.conf.sample | 0
configs/samples/confbridge.conf.sample | 373 +
configs/{ => samples}/config_test.conf.sample | 0
configs/{ => samples}/console.conf.sample | 0
configs/{ => samples}/dbsep.conf.sample | 0
configs/{ => samples}/dnsmgr.conf.sample | 0
configs/{ => samples}/dsp.conf.sample | 0
configs/{ => samples}/dundi.conf.sample | 0
configs/{ => samples}/enum.conf.sample | 0
configs/samples/extconfig.conf.sample | 109 +
configs/{ => samples}/extensions.ael.sample | 0
configs/samples/extensions.conf.sample | 857 +
configs/{ => samples}/extensions.lua.sample | 0
configs/samples/extensions_minivm.conf.sample | 159 +
configs/samples/features.conf.sample | 115 +
configs/{ => samples}/festival.conf.sample | 0
configs/{ => samples}/followme.conf.sample | 0
configs/{ => samples}/func_odbc.conf.sample | 0
configs/samples/hep.conf.sample | 16 +
configs/samples/http.conf.sample | 96 +
configs/samples/iax.conf.sample | 672 +
configs/{ => samples}/iaxprov.conf.sample | 0
configs/{ => samples}/indications.conf.sample | 0
configs/samples/logger.conf.sample | 134 +
configs/samples/manager.conf.sample | 157 +
configs/{ => samples}/meetme.conf.sample | 0
configs/{ => samples}/mgcp.conf.sample | 0
configs/{ => samples}/minivm.conf.sample | 0
configs/{ => samples}/misdn.conf.sample | 0
configs/{ => samples}/modules.conf.sample | 0
configs/samples/motif.conf.sample | 99 +
configs/{ => samples}/musiconhold.conf.sample | 0
configs/{ => samples}/muted.conf.sample | 0
configs/samples/ooh323.conf.sample | 208 +
configs/{ => samples}/osp.conf.sample | 0
configs/{ => samples}/oss.conf.sample | 0
configs/{ => samples}/phone.conf.sample | 0
configs/samples/phoneprov.conf.sample | 143 +
configs/samples/pjsip.conf.sample | 916 +
.../pjsip_notify.conf.sample} | 0
configs/samples/queuerules.conf.sample | 31 +
configs/samples/queues.conf.sample | 549 +
configs/{ => samples}/res_config_mysql.conf.sample | 0
.../{ => samples}/res_config_sqlite.conf.sample | 0
.../{ => samples}/res_config_sqlite3.conf.sample | 0
configs/{ => samples}/res_corosync.conf.sample | 0
configs/{ => samples}/res_curl.conf.sample | 0
configs/{ => samples}/res_fax.conf.sample | 0
configs/samples/res_ldap.conf.sample | 199 +
configs/samples/res_odbc.conf.sample | 121 +
configs/samples/res_parking.conf.sample | 121 +
configs/samples/res_pgsql.conf.sample | 30 +
configs/{ => samples}/res_pktccops.conf.sample | 0
configs/{ => samples}/res_snmp.conf.sample | 0
configs/{ => samples}/res_stun_monitor.conf.sample | 0
configs/{ => samples}/rtp.conf.sample | 0
configs/{ => samples}/say.conf.sample | 0
configs/samples/sip.conf.sample | 1573 +
configs/{ => samples}/sip_notify.conf.sample | 0
configs/samples/skinny.conf.sample | 208 +
configs/{ => samples}/sla.conf.sample | 0
configs/{ => samples}/smdi.conf.sample | 0
configs/samples/sorcery.conf.sample | 67 +
configs/samples/ss7.timers.sample | 65 +
configs/samples/stasis.conf.sample | 122 +
configs/samples/statsd.conf.sample | 8 +
configs/{ => samples}/telcordia-1.adsi | 0
configs/samples/test_sorcery.conf.sample | 14 +
configs/{ => samples}/udptl.conf.sample | 0
configs/samples/unistim.conf.sample | 88 +
configs/{ => samples}/users.conf.sample | 0
configs/samples/voicemail.conf.sample | 469 +
configs/{ => samples}/vpb.conf.sample | 0
configs/samples/xmpp.conf.sample | 42 +
configs/sip.conf.sample | 1555 -
configs/skinny.conf.sample | 191 -
configs/unistim.conf.sample | 86 -
configs/voicemail.conf.sample | 456 -
configs/xmpp.conf.sample | 39 -
configure | 2916 +-
configure.ac | 225 +-
contrib/ast-db-manage/README.md | 63 +
contrib/ast-db-manage/cdr.ini.sample | 57 +
contrib/ast-db-manage/cdr/env.py | 74 +
contrib/ast-db-manage/cdr/script.py.mako | 22 +
.../cdr/versions/210693f3123d_create_cdr_table.py | 64 +
contrib/ast-db-manage/config.ini.sample | 57 +
contrib/ast-db-manage/config/env.py | 74 +
contrib/ast-db-manage/config/script.py.mako | 22 +
.../versions/10aedae86a32_add_outgoing_enum_va.py | 83 +
.../1758e8bbf6b_increase_useragent_column_size.py | 41 +
.../versions/1d50859ed02e_create_accountcode.py | 20 +
.../21e526ad3040_add_pjsip_debug_option.py | 21 +
.../versions/28887f25a46f_create_queue_tables.py | 141 +
...930b41b3_add_pjsip_endpoint_options_for_12_1.py | 176 +
.../3855ee4e5f85_add_missing_pjsip_options.py | 24 +
.../versions/43956d550a44_add_tables_for_pjsip.py | 189 +
.../versions/4c573e7135bd_fix_tos_field_types.py | 61 +
.../config/versions/4da0c5f79a9c_create_tables.py | 330 +
.../5139253c0423_make_q_member_uniqueid_autoinc.py | 60 +
.../51f8cb66540e_add_further_dtls_options.py | 32 +
.../versions/581a4264e537_adding_extensions.py | 50 +
.../5950038a6ead_fix_pjsip_verifiy_typo.py | 29 +
...b23a8_create_pjsip_subscription_persistence_.py | 36 +
.../versions/d39508cb8d8_create_queue_rules.py | 31 +
.../e96a0b8071c_increase_pjsip_column_size.py | 39 +
contrib/ast-db-manage/voicemail.ini.sample | 57 +
contrib/ast-db-manage/voicemail/env.py | 74 +
contrib/ast-db-manage/voicemail/script.py.mako | 22 +
.../39428242f7f5_increase_recording_column_size.py | 44 +
.../versions/a2e9769475e_create_tables.py | 58 +
contrib/asterisk-ng-doxygen | 1610 +-
contrib/init.d/rc.debian.asterisk | 2 +-
contrib/init.d/rc.gentoo.asterisk | 2 +-
contrib/init.d/rc.mandriva.asterisk | 2 +-
contrib/init.d/rc.redhat.asterisk | 2 +-
contrib/init.d/rc.slackware.asterisk | 2 +-
contrib/realtime/mysql/iaxfriends.sql | 56 -
contrib/realtime/mysql/meetme.sql | 21 -
contrib/realtime/mysql/musiconhold.sql | 19 -
contrib/realtime/mysql/mysql_cdr.sql | 32 +
contrib/realtime/mysql/mysql_config.sql | 699 +
contrib/realtime/mysql/mysql_voicemail.sql | 34 +
contrib/realtime/mysql/queue_log.sql | 24 -
contrib/realtime/mysql/sippeers.sql | 97 -
contrib/realtime/mysql/voicemail.sql | 70 -
contrib/realtime/mysql/voicemail_data.sql | 29 -
contrib/realtime/mysql/voicemail_messages.sql | 31 -
contrib/realtime/oracle/oracle_cdr.sql | 46 +
contrib/realtime/oracle/oracle_config.sql | 984 +
contrib/realtime/oracle/oracle_voicemail.sql | 52 +
contrib/realtime/postgresql/postgresql_cdr.sql | 36 +
contrib/realtime/postgresql/postgresql_config.sql | 735 +
.../realtime/postgresql/postgresql_voicemail.sql | 36 +
contrib/realtime/postgresql/realtime.sql | 164 -
contrib/realtime/sqlserver/mssql_cdr.sql | 42 +
contrib/realtime/sqlserver/mssql_config.sql | 980 +
contrib/realtime/sqlserver/mssql_voicemail.sql | 48 +
contrib/scripts/ast_tls_cert | 8 +-
contrib/scripts/asterisk.ldap-schema | 12 +-
contrib/scripts/asterisk.ldif | 11 +-
contrib/scripts/autosupport | 107 +-
contrib/scripts/dahdi_span_config_hook | 32 +
contrib/scripts/get_swagger_ui.sh | 36 +
contrib/scripts/install_prereq | 136 +-
contrib/scripts/live_ast | 6 +-
contrib/scripts/safe_asterisk | 24 +-
contrib/scripts/sip_to_pjsip/astconfigparser.py | 467 +
contrib/scripts/sip_to_pjsip/astdicts.py | 298 +
contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 1159 +
doc/Asterisk-11-Reference.pdf | Bin 1790999 -> 0 bytes
doc/README.txt | 6 +-
doc/appdocsxml.dtd | 59 +-
doc/appdocsxml.xslt | 140 +
doc/asterisk.8 | 2 +-
formats/Makefile | 2 +-
formats/format_g719.c | 18 +-
formats/format_g723.c | 17 +-
formats/format_g726.c | 18 +-
formats/format_g729.c | 18 +-
formats/format_gsm.c | 17 +-
formats/format_h263.c | 29 +-
formats/format_h264.c | 29 +-
formats/format_ilbc.c | 17 +-
formats/format_jpeg.c | 16 +-
formats/format_ogg_vorbis.c | 25 +-
formats/format_pcm.c | 34 +-
formats/format_siren14.c | 17 +-
formats/format_siren7.c | 17 +-
formats/format_sln.c | 81 +-
formats/format_vox.c | 17 +-
formats/format_wav.c | 29 +-
formats/format_wav_gsm.c | 31 +-
funcs/Makefile | 2 +-
funcs/func_aes.c | 3 +-
funcs/func_audiohookinherit.c | 252 +-
funcs/func_blacklist.c | 2 +-
funcs/func_callcompletion.c | 2 +-
funcs/func_callerid.c | 53 +-
funcs/func_cdr.c | 592 +-
funcs/func_channel.c | 95 +-
funcs/func_config.c | 2 +-
funcs/func_curl.c | 3 +-
funcs/func_db.c | 2 +-
funcs/func_devstate.c | 3 +-
funcs/func_dialgroup.c | 2 +-
funcs/func_dialplan.c | 3 +-
funcs/func_env.c | 2 +-
funcs/func_frame_trace.c | 56 +-
funcs/func_global.c | 45 +-
funcs/func_groupcount.c | 2 +-
funcs/func_hangupcause.c | 2 +-
funcs/func_iconv.c | 2 +-
funcs/func_jitterbuffer.c | 349 +-
funcs/func_lock.c | 2 +-
funcs/func_math.c | 2 +-
funcs/func_odbc.c | 3 +-
funcs/func_periodic_hook.c | 527 +
funcs/func_periodic_hook.exports.in | 7 +
funcs/func_pitchshift.c | 11 +-
funcs/func_pjsip_endpoint.c | 161 +
funcs/func_presencestate.c | 192 +-
funcs/func_rand.c | 2 +-
funcs/func_realtime.c | 2 +-
funcs/func_shell.c | 2 +-
funcs/func_sorcery.c | 221 +
funcs/func_speex.c | 8 +-
funcs/func_srv.c | 2 +-
funcs/func_strings.c | 8 +-
funcs/func_sysinfo.c | 2 +-
funcs/func_talkdetect.c | 404 +
funcs/func_timeout.c | 4 +-
funcs/func_uri.c | 2 +-
funcs/func_version.c | 2 +-
funcs/func_vmcount.c | 23 +-
funcs/func_volume.c | 2 +-
include/asterisk.h | 13 +-
include/asterisk/_private.h | 71 +-
include/asterisk/abstract_jb.h | 30 +-
include/asterisk/acl.h | 30 +-
include/asterisk/aoc.h | 5 +-
include/asterisk/app.h | 722 +-
include/asterisk/ari.h | 243 +
include/asterisk/astdb.h | 27 +-
include/asterisk/astobj.h | 6 +-
include/asterisk/astobj2.h | 814 +-
include/asterisk/audiohook.h | 34 +-
include/asterisk/autochan.h | 5 +-
include/asterisk/autoconfig.h.in | 69 +-
include/asterisk/backtrace.h | 97 +
include/asterisk/beep.h | 45 +
include/asterisk/bridge.h | 1075 +
include/asterisk/bridge_after.h | 244 +
include/asterisk/bridge_basic.h | 150 +
include/asterisk/bridge_channel.h | 654 +
include/asterisk/bridge_channel_internal.h | 211 +
include/asterisk/bridge_features.h | 866 +
include/asterisk/bridge_internal.h | 216 +
include/asterisk/bridge_roles.h | 173 +
include/asterisk/bridge_technology.h | 246 +
include/asterisk/bridging.h | 590 -
include/asterisk/bridging_features.h | 354 -
include/asterisk/bridging_technology.h | 196 -
include/asterisk/bucket.h | 397 +
include/asterisk/callerid.h | 10 +-
include/asterisk/ccss.h | 18 +-
include/asterisk/cdr.h | 694 +-
include/asterisk/cel.h | 230 +-
include/asterisk/channel.h | 1112 +-
include/asterisk/channel_internal.h | 5 +-
include/asterisk/channelstate.h | 5 +-
include/asterisk/chanvars.h | 18 +
include/asterisk/cli.h | 16 +-
include/asterisk/codec.h | 186 +
include/asterisk/compat.h | 10 +-
include/asterisk/compiler.h | 6 +
include/asterisk/config.h | 331 +-
include/asterisk/config_options.h | 150 +-
include/asterisk/core_local.h | 137 +
include/asterisk/core_unreal.h | 246 +
include/asterisk/crypto.h | 6 +-
include/asterisk/data.h | 1 +
include/asterisk/datastore.h | 17 +-
include/asterisk/devicestate.h | 120 +-
include/asterisk/dial.h | 35 +-
include/asterisk/dns.h | 3 +
include/asterisk/doxygen/architecture.h | 24 +-
include/asterisk/doxygen/asterisk-git-howto.h | 16 +-
include/asterisk/doxygen/commits.h | 46 +-
include/asterisk/doxygen/licensing.h | 2 +-
include/asterisk/doxygen/mantisworkflow.h | 206 -
include/asterisk/doxygen/releases.h | 18 +-
include/asterisk/doxygen/reviewboard.h | 50 +-
include/asterisk/doxyref.h | 451 +-
include/asterisk/endpoints.h | 204 +
include/asterisk/event.h | 450 +-
include/asterisk/event_defs.h | 75 +-
include/asterisk/features.h | 248 +-
include/asterisk/features_config.h | 238 +
include/asterisk/file.h | 60 +-
include/asterisk/format.h | 558 +-
include/asterisk/format_cache.h | 296 +
include/asterisk/format_cap.h | 389 +-
include/asterisk/format_compatibility.h | 129 +
include/asterisk/format_pref.h | 114 -
include/asterisk/frame.h | 171 +-
include/asterisk/frame_defs.h | 0
include/asterisk/framehook.h | 304 +-
include/asterisk/heap.h | 3 +
include/asterisk/http.h | 189 +-
include/asterisk/http_websocket.h | 165 +-
include/asterisk/image.h | 2 +-
include/asterisk/inline_api.h | 2 +-
include/asterisk/jabber.h | 224 -
include/asterisk/jingle.h | 66 -
include/asterisk/json.h | 1036 +
include/asterisk/localtime.h | 7 +-
include/asterisk/lock.h | 124 +-
include/asterisk/logger.h | 99 +-
include/asterisk/manager.h | 285 +-
include/asterisk/md5.h | 3 +-
include/asterisk/media_index.h | 108 +
include/asterisk/message.h | 147 +-
include/asterisk/mixmonitor.h | 105 +
include/asterisk/mod_format.h | 10 +-
include/asterisk/module.h | 139 +-
include/asterisk/monitor.h | 4 +-
include/asterisk/musiconhold.h | 7 +-
include/asterisk/netsock.h | 2 +
include/asterisk/netsock2.h | 78 +-
include/asterisk/optional_api.h | 287 +-
include/asterisk/options.h | 21 +-
include/asterisk/opus.h | 41 +
include/asterisk/parking.h | 289 +
include/asterisk/paths.h | 3 +-
include/asterisk/pbx.h | 100 +-
include/asterisk/phoneprov.h | 124 +
include/asterisk/pickup.h | 91 +
include/asterisk/presencestate.h | 53 +-
include/asterisk/res_fax.h | 11 +
include/asterisk/res_hep.h | 111 +
include/asterisk/res_mwi_external.h | 226 +
include/asterisk/res_odbc.h | 14 +-
include/asterisk/res_pjsip.h | 1956 +
include/asterisk/res_pjsip_body_generator_types.h | 70 +
include/asterisk/res_pjsip_cli.h | 110 +
include/asterisk/res_pjsip_outbound_publish.h | 165 +
include/asterisk/res_pjsip_presence_xml.h | 115 +
include/asterisk/res_pjsip_pubsub.h | 686 +
include/asterisk/res_pjsip_session.h | 648 +
include/asterisk/rtp_engine.h | 349 +-
include/asterisk/say.h | 14 +-
include/asterisk/sched.h | 2 +-
include/asterisk/sdp_srtp.h | 127 +
include/asterisk/security_events.h | 30 +
include/asterisk/security_events_defs.h | 17 +-
include/asterisk/sem.h | 157 +
include/asterisk/sip_api.h | 3 +-
include/asterisk/slin.h | 6 +-
include/asterisk/slinfactory.h | 6 +-
include/asterisk/smdi.h | 56 +-
include/asterisk/smoother.h | 89 +
include/asterisk/sorcery.h | 993 +
include/asterisk/sounds_index.h | 55 +
include/asterisk/speech.h | 6 +-
include/asterisk/spinlock.h | 488 +
include/asterisk/srv.h | 51 +-
include/asterisk/stasis.h | 1277 +
include/asterisk/stasis_app.h | 847 +
include/asterisk/stasis_app_device_state.h | 95 +
include/asterisk/stasis_app_impl.h | 108 +
include/asterisk/stasis_app_mailbox.h | 91 +
include/asterisk/stasis_app_playback.h | 157 +
include/asterisk/stasis_app_recording.h | 297 +
include/asterisk/stasis_app_snoop.h | 60 +
include/asterisk/stasis_bridges.h | 547 +
include/asterisk/stasis_cache_pattern.h | 153 +
include/asterisk/stasis_channels.h | 620 +
include/asterisk/stasis_endpoints.h | 229 +
include/asterisk/stasis_internal.h | 67 +
include/asterisk/stasis_message_router.h | 211 +
include/asterisk/stasis_system.h | 131 +
include/asterisk/stasis_test.h | 142 +
include/asterisk/statsd.h | 85 +
include/asterisk/stringfields.h | 80 +-
include/asterisk/strings.h | 264 +-
include/asterisk/taskprocessor.h | 198 +-
include/asterisk/tcptls.h | 3 +-
include/asterisk/term.h | 73 +-
include/asterisk/test.h | 194 +-
include/asterisk/threadpool.h | 226 +
include/asterisk/threadstorage.h | 10 +-
include/asterisk/time.h | 11 +
include/asterisk/timing.h | 61 +-
include/asterisk/translate.h | 39 +-
include/asterisk/udptl.h | 18 +-
include/asterisk/uri.h | 181 +
include/asterisk/utils.h | 120 +-
include/asterisk/uuid.h | 118 +
include/asterisk/vector.h | 333 +
include/asterisk/xml.h | 39 +
include/asterisk/xmldoc.h | 65 +-
include/asterisk/xmpp.h | 24 +-
main/Makefile | 30 +-
main/abstract_jb.c | 329 +-
main/acl.c | 91 +-
main/aoc.c | 430 +-
main/app.c | 1249 +-
main/ast_expr2f.c | 6 +-
main/asterisk.c | 506 +-
main/asterisk.dynamics | 1 +
main/asterisk.exports.in | 3 +
main/astfd.c | 4 +-
main/astmm.c | 68 +-
main/astobj2.c | 986 +-
main/astobj2_container.c | 1219 +
main/astobj2_container_private.h | 345 +
main/astobj2_hash.c | 1153 +
main/astobj2_private.h | 49 +
main/astobj2_rbtree.c | 2096 +
main/audiohook.c | 260 +-
main/autoservice.c | 50 +-
main/backtrace.c | 225 +
main/bridge.c | 5451 +++
main/bridge_after.c | 648 +
main/bridge_basic.c | 3454 ++
main/bridge_channel.c | 2638 ++
main/bridge_roles.c | 499 +
main/bridging.c | 1676 -
main/bucket.c | 965 +
main/callerid.c | 28 +-
main/ccss.c | 331 +-
main/cdr.c | 4689 +-
main/cel.c | 1834 +-
main/channel.c | 4213 +-
main/channel_internal_api.c | 360 +-
main/chanvars.c | 68 +-
main/cli.c | 340 +-
main/codec.c | 381 +
main/codec_builtin.c | 845 +
main/config.c | 1198 +-
main/config_options.c | 638 +-
main/core_local.c | 1071 +
main/core_unreal.c | 1042 +
main/crypt.c | 202 +
main/data.c | 105 +-
main/datastore.c | 18 +-
main/db.c | 7 +-
main/devicestate.c | 664 +-
main/dial.c | 265 +-
main/dns.c | 53 +-
main/dnsmgr.c | 14 +-
main/dsp.c | 168 +-
main/editline/readline.c | 1 -
main/endpoints.c | 523 +
main/enum.c | 14 +-
main/event.c | 1752 +-
main/features.c | 9253 +---
main/features_config.c | 1946 +
main/file.c | 351 +-
main/format.c | 1468 +-
main/format_cache.c | 515 +
main/format_cap.c | 925 +-
main/format_compatibility.c | 274 +
main/format_pref.c | 344 -
main/frame.c | 604 +-
main/framehook.c | 155 +-
main/hashtab.c | 6 +-
main/heap.c | 2 +-
main/http.c | 1721 +-
main/image.c | 6 +-
main/indications.c | 24 +-
main/io.c | 2 +-
main/jitterbuf.c | 2 +-
main/json.c | 913 +
main/libasteriskssl.c | 2 +-
main/loader.c | 436 +-
main/lock.c | 2 +-
main/logger.c | 404 +-
main/manager.c | 1966 +-
main/manager_bridges.c | 650 +
main/manager_channels.c | 1180 +
main/manager_endpoints.c | 89 +
main/manager_mwi.c | 200 +
main/manager_system.c | 81 +
main/media_index.c | 597 +
main/message.c | 551 +-
main/mixmonitor.c | 98 +
main/named_acl.c | 121 +-
main/netsock.c | 113 +-
main/netsock2.c | 85 +-
main/optional_api.c | 350 +
main/parking.c | 249 +
main/pbx.c | 2417 +-
main/pickup.c | 409 +
main/presencestate.c | 223 +-
main/rtp_engine.c | 1848 +-
main/say.c | 584 +-
main/sched.c | 155 +-
main/sdp_srtp.c | 387 +
main/security_events.c | 680 +-
main/sem.c | 116 +
main/sha1.c | 4 +-
main/slinfactory.c | 36 +-
main/smoother.c | 227 +
main/sorcery.c | 1944 +
main/sounds_index.c | 333 +
main/srv.c | 4 +-
main/stasis.c | 1529 +
main/stasis_bridges.c | 1318 +
main/stasis_cache.c | 933 +
main/stasis_cache_pattern.c | 201 +
main/stasis_channels.c | 1605 +
main/stasis_endpoints.c | 317 +
main/stasis_message.c | 201 +
main/stasis_message_router.c | 352 +
main/stasis_system.c | 426 +
main/stdtime/localtime.c | 3 +-
main/strings.c | 51 +-
main/stun.c | 2 +-
main/taskprocessor.c | 588 +-
main/tcptls.c | 61 +-
main/tdd.c | 7 +-
main/term.c | 60 +-
main/test.c | 224 +-
main/threadpool.c | 1218 +
main/threadstorage.c | 2 +-
main/timing.c | 58 +-
main/translate.c | 795 +-
main/udptl.c | 215 +-
main/uri.c | 323 +
main/utils.c | 410 +-
main/uuid.c | 231 +
main/xml.c | 76 +-
main/xmldoc.c | 1217 +-
makeopts.in | 18 +
menuselect/Makefile | 22 +-
menuselect/README | 7 +-
menuselect/acinclude.m4 | 199 -
menuselect/aclocal.m4 | 13 +-
menuselect/autoconfig.h.in | 15 +-
menuselect/bootstrap.sh | 4 +-
menuselect/config.guess | 559 +-
menuselect/config.sub | 291 +-
menuselect/configure | 5646 +--
menuselect/configure.ac | 12 +-
menuselect/example_menuselect-tree | 2 +-
menuselect/makeopts.in | 4 +
menuselect/menuselect.c | 641 +-
menuselect/menuselect.h | 13 +-
menuselect/mxml/ANNOUNCEMENT | 5 -
menuselect/mxml/CHANGES | 213 -
menuselect/mxml/COPYING | 482 -
menuselect/mxml/Makefile.in | 342 -
menuselect/mxml/README | 204 -
menuselect/mxml/config.h.in | 69 -
menuselect/mxml/configure | 5702 ---
menuselect/mxml/configure.in | 343 -
menuselect/mxml/install-sh | 251 -
menuselect/mxml/mxml-attr.c | 181 -
menuselect/mxml/mxml-entity.c | 472 -
menuselect/mxml/mxml-file.c | 2844 --
menuselect/mxml/mxml-index.c | 649 -
menuselect/mxml/mxml-node.c | 664 -
menuselect/mxml/mxml-private.c | 128 -
menuselect/mxml/mxml-search.c | 199 -
menuselect/mxml/mxml-set.c | 257 -
menuselect/mxml/mxml-string.c | 377 -
menuselect/mxml/mxml.h | 254 -
menuselect/mxml/mxml.list.in | 115 -
menuselect/mxml/mxml.pc.in | 10 -
pbx/Makefile | 2 +-
pbx/dundi-parser.c | 2 +-
pbx/pbx_ael.c | 15 +-
pbx/pbx_config.c | 203 +-
pbx/pbx_dundi.c | 12 +-
pbx/pbx_loopback.c | 2 +-
pbx/pbx_lua.c | 4 +-
pbx/pbx_realtime.c | 68 +-
pbx/pbx_spool.c | 17 +-
res/Makefile | 46 +-
res/ael/ael.flex | 4 +-
res/ael/ael.tab.c | 2 +-
res/ael/ael_lex.c | 4 +-
res/ael/pval.c | 45 +-
res/ari.make | 63 +
res/ari/ari_model_validators.c | 5113 +++
res/ari/ari_model_validators.h | 1490 +
res/ari/ari_websockets.c | 190 +
res/ari/cli.c | 267 +
res/ari/config.c | 351 +
res/ari/internal.h | 167 +
res/ari/resource_applications.c | 172 +
res/ari/resource_applications.h | 131 +
res/ari/resource_asterisk.c | 193 +
res/ari/resource_asterisk.h | 121 +
res/ari/resource_bridges.c | 990 +
res/ari/resource_bridges.h | 357 +
res/ari/resource_channels.c | 1123 +
res/ari/resource_channels.h | 673 +
res/ari/resource_device_states.c | 111 +
res/ari/resource_device_states.h | 106 +
res/ari/resource_endpoints.c | 281 +
res/ari/resource_endpoints.h | 142 +
res/ari/resource_events.c | 275 +
res/ari/resource_events.h | 94 +
res/ari/resource_mailboxes.c | 93 +
res/ari/resource_mailboxes.h | 108 +
res/ari/resource_playbacks.c | 139 +
res/ari/resource_playbacks.h | 95 +
res/ari/resource_recordings.c | 313 +
res/ari/resource_recordings.h | 201 +
res/ari/resource_sounds.c | 226 +
res/ari/resource_sounds.h | 82 +
res/parking/parking_applications.c | 889 +
res/parking/parking_bridge.c | 464 +
res/parking/parking_bridge_features.c | 740 +
res/parking/parking_controller.c | 281 +
res/parking/parking_devicestate.c | 124 +
res/parking/parking_manager.c | 697 +
res/parking/parking_tests.c | 877 +
res/parking/parking_ui.c | 208 +
res/parking/res_parking.h | 571 +
res/res_adsi.c | 57 +-
res/res_ael_share.c | 3 +-
res/res_agi.c | 554 +-
res/res_ari.c | 1109 +
res/res_ari.exports.in | 6 +
res/res_ari_applications.c | 548 +
res/res_ari_asterisk.c | 449 +
res/res_ari_bridges.c | 1419 +
res/res_ari_channels.c | 2597 ++
res/res_ari_device_states.c | 364 +
res/res_ari_endpoints.c | 503 +
res/res_ari_events.c | 378 +
res/res_ari_mailboxes.c | 370 +
res/res_ari_model.c | 211 +
res/res_ari_model.exports.in | 6 +
res/res_ari_playbacks.c | 321 +
res/res_ari_recordings.c | 845 +
res/res_ari_sounds.c | 250 +
res/res_calendar.c | 55 +-
res/res_calendar_caldav.c | 4 +-
res/res_calendar_ews.c | 17 +-
res/res_calendar_exchange.c | 107 +-
res/res_calendar_icalendar.c | 4 +-
res/res_chan_stats.c | 187 +
res/res_clialiases.c | 23 +-
res/res_clioriginate.c | 21 +-
res/res_config_curl.c | 141 +-
res/res_config_ldap.c | 310 +-
res/res_config_odbc.c | 215 +-
res/res_config_pgsql.c | 195 +-
res/res_config_sqlite.c | 324 +-
res/res_config_sqlite3.c | 104 +-
res/res_corosync.c | 394 +-
res/res_crypto.c | 5 +-
res/res_curl.c | 44 +-
res/res_fax.c | 964 +-
res/res_fax.exports.in | 1 +
res/res_fax_spandsp.c | 122 +-
res/res_format_attr_celt.c | 208 +-
res/res_format_attr_h263.c | 352 +-
res/res_format_attr_h264.c | 403 +-
res/res_format_attr_opus.c | 260 +
res/res_format_attr_silk.c | 231 +-
res/res_hep.c | 627 +
res/res_hep.exports.in | 7 +
res/res_hep_pjsip.c | 179 +
res/res_hep_rtcp.c | 144 +
res/res_http_post.c | 136 +-
res/res_http_websocket.c | 686 +-
res/res_http_websocket.exports.in | 14 +-
res/res_jabber.c | 4830 --
res/res_limit.c | 4 +-
res/res_manager_devicestate.c | 154 +
res/res_manager_presencestate.c | 153 +
res/res_monitor.c | 152 +-
res/res_musiconhold.c | 296 +-
res/res_mutestream.c | 192 +-
res/res_mwi_external.c | 959 +
res/res_mwi_external.exports.in | 6 +
res/res_mwi_external_ami.c | 380 +
res/res_odbc.c | 36 +-
res/res_odbc.exports.in | 1 -
res/res_parking.c | 1271 +
res/res_phoneprov.c | 1501 +-
res/res_phoneprov.exports.in | 6 +
res/res_pjsip.c | 3184 ++
res/res_pjsip.exports.in | 8 +
res/res_pjsip/config_auth.c | 342 +
res/res_pjsip/config_domain_aliases.c | 67 +
res/res_pjsip/config_global.c | 140 +
res/res_pjsip/config_system.c | 243 +
res/res_pjsip/config_transport.c | 790 +
res/res_pjsip/include/res_pjsip_private.h | 115 +
res/res_pjsip/location.c | 912 +
res/res_pjsip/pjsip_cli.c | 342 +
res/res_pjsip/pjsip_configuration.c | 1983 +
res/res_pjsip/pjsip_distributor.c | 399 +
res/res_pjsip/pjsip_global_headers.c | 171 +
res/res_pjsip/pjsip_options.c | 1107 +
res/res_pjsip/pjsip_outbound_auth.c | 96 +
res/res_pjsip/presence_xml.c | 175 +
res/res_pjsip/security_events.c | 286 +
res/res_pjsip_acl.c | 305 +
res/res_pjsip_authenticator_digest.c | 494 +
res/res_pjsip_caller_id.c | 750 +
res/res_pjsip_dialog_info_body_generator.c | 214 +
res/res_pjsip_diversion.c | 350 +
res/res_pjsip_dtmf_info.c | 170 +
res/res_pjsip_endpoint_identifier_anonymous.c | 128 +
res/res_pjsip_endpoint_identifier_ip.c | 470 +
res/res_pjsip_endpoint_identifier_user.c | 134 +
res/res_pjsip_exten_state.c | 498 +
res/res_pjsip_exten_state.exports.in | 7 +
res/res_pjsip_header_funcs.c | 626 +
res/res_pjsip_log_forwarder.c | 125 +
res/res_pjsip_logger.c | 267 +
res/res_pjsip_messaging.c | 762 +
res/res_pjsip_multihomed.c | 227 +
res/res_pjsip_mwi.c | 924 +
res/res_pjsip_mwi_body_generator.c | 116 +
res/res_pjsip_nat.c | 305 +
res/res_pjsip_notify.c | 1034 +
res/res_pjsip_one_touch_record_info.c | 131 +
res/res_pjsip_outbound_authenticator_digest.c | 168 +
res/res_pjsip_outbound_publish.c | 1015 +
res/res_pjsip_outbound_publish.exports.in | 6 +
res/res_pjsip_outbound_registration.c | 1311 +
res/res_pjsip_path.c | 253 +
res/res_pjsip_phoneprov_provider.c | 424 +
res/res_pjsip_pidf_body_generator.c | 139 +
res/res_pjsip_pidf_digium_body_supplement.c | 117 +
res/res_pjsip_pidf_eyebeam_body_supplement.c | 116 +
res/res_pjsip_publish_asterisk.c | 929 +
res/res_pjsip_pubsub.c | 4297 ++
res/res_pjsip_pubsub.exports.in | 42 +
res/res_pjsip_refer.c | 1012 +
res/res_pjsip_registrar.c | 832 +
res/res_pjsip_registrar_expire.c | 230 +
res/res_pjsip_rfc3326.c | 150 +
res/res_pjsip_sdp_rtp.c | 1283 +
res/res_pjsip_send_to_voicemail.c | 231 +
res/res_pjsip_session.c | 2334 +
res/res_pjsip_session.exports.in | 25 +
res/res_pjsip_t38.c | 880 +
res/res_pjsip_transport_websocket.c | 401 +
res/res_pjsip_xpidf_body_generator.c | 181 +
res/res_pktccops.c | 3 +-
res/res_rtp_asterisk.c | 1252 +-
res/res_rtp_multicast.c | 11 +-
res/res_security_log.c | 102 +-
res/res_smdi.c | 340 +-
res/res_snmp.c | 24 +-
res/res_sorcery_astdb.c | 392 +
res/res_sorcery_config.c | 387 +
res/res_sorcery_memory.c | 242 +
res/res_sorcery_realtime.c | 289 +
res/res_speech.c | 45 +-
res/res_speech.exports.in | 17 +-
res/res_srtp.c | 3 +-
res/res_stasis.c | 2043 +
res/res_stasis.exports.in | 6 +
res/res_stasis_answer.c | 78 +
res/res_stasis_answer.exports.in | 6 +
res/res_stasis_device_state.c | 417 +
res/res_stasis_device_state.exports.in | 6 +
res/res_stasis_mailbox.c | 166 +
res/res_stasis_mailbox.exports.in | 6 +
res/res_stasis_playback.c | 688 +
res/res_stasis_playback.exports.in | 6 +
res/res_stasis_recording.c | 660 +
res/res_stasis_recording.exports.in | 6 +
res/res_stasis_snoop.c | 414 +
res/res_stasis_snoop.exports.in | 6 +
res/res_stasis_test.c | 287 +
res/res_stasis_test.exports.in | 6 +
res/res_statsd.c | 325 +
res/res_statsd.exports.in | 8 +
res/res_stun_monitor.c | 46 +-
res/res_timing_dahdi.c | 89 +-
res/res_timing_kqueue.c | 203 +-
res/res_timing_pthread.c | 117 +-
res/res_timing_timerfd.c | 247 +-
res/res_xmpp.c | 343 +-
res/snmp/agent.c | 10 +-
res/stasis/app.c | 1318 +
res/stasis/app.h | 273 +
res/stasis/command.c | 166 +
res/stasis/command.h | 76 +
res/stasis/control.c | 1054 +
res/stasis/control.h | 112 +
res/stasis/messaging.c | 531 +
res/stasis/messaging.h | 83 +
res/stasis/stasis_bridge.c | 235 +
res/stasis/stasis_bridge.h | 74 +
res/stasis_recording/stored.c | 528 +
rest-api-templates/README.txt | 15 +
rest-api-templates/api.wiki.mustache | 63 +
rest-api-templates/ari.make.mustache | 26 +
rest-api-templates/ari_model_validators.c.mustache | 122 +
rest-api-templates/ari_model_validators.h.mustache | 191 +
rest-api-templates/ari_resource.c.mustache | 53 +
rest-api-templates/ari_resource.h.mustache | 109 +
rest-api-templates/asterisk_processor.py | 245 +
rest-api-templates/body_parsing.mustache | 71 +
rest-api-templates/do-not-edit.mustache | 4 +
rest-api-templates/make_ari_stubs.py | 95 +
rest-api-templates/models.wiki.mustache | 22 +
rest-api-templates/odict.py | 261 +
rest-api-templates/param_cleanup.mustache | 26 +
rest-api-templates/param_parsing.mustache | 113 +
rest-api-templates/res_ari_resource.c.mustache | 250 +
rest-api-templates/rest_handler.mustache | 40 +
rest-api-templates/swagger_model.py | 756 +
rest-api-templates/transform.py | 62 +
rest-api/README.txt | 9 +
rest-api/api-docs/applications.json | 172 +
rest-api/api-docs/asterisk.json | 259 +
rest-api/api-docs/bridges.json | 656 +
rest-api/api-docs/channels.json | 1456 +
rest-api/api-docs/deviceStates.json | 151 +
rest-api/api-docs/endpoints.json | 275 +
rest-api/api-docs/events.json | 717 +
rest-api/api-docs/mailboxes.json | 134 +
rest-api/api-docs/playbacks.json | 155 +
rest-api/api-docs/recordings.json | 378 +
rest-api/api-docs/sounds.json | 99 +
rest-api/resources.json | 54 +
sounds/Makefile | 2 +-
static-http/ajamdemo.html | 17 +
static-http/astman.css | 18 +
static-http/mantest.html | 20 +-
tests/Makefile | 2 +-
tests/test_abstract_jb.c | 77 +-
tests/test_aoc.c | 2 +-
tests/test_app.c | 30 +-
tests/test_ari.c | 570 +
tests/test_ari_model.c | 457 +
tests/test_astobj2.c | 1656 +-
tests/test_astobj2_thrash.c | 2 +-
tests/test_bucket.c | 873 +
tests/test_callerid.c | 6 +-
tests/test_cdr.c | 2630 ++
tests/test_cel.c | 2216 +
tests/test_channel_feature_hooks.c | 324 +
tests/test_config.c | 614 +-
tests/test_core_codec.c | 369 +
tests/test_core_format.c | 975 +
tests/test_devicestate.c | 252 +-
tests/test_dlinklists.c | 2 +-
tests/test_endpoints.c | 157 +
tests/test_event.c | 801 +-
tests/test_expr.c | 2 +-
tests/test_format_api.c | 859 -
tests/test_format_cache.c | 281 +
tests/test_format_cap.c | 1479 +
tests/test_func_file.c | 2 +-
tests/test_gosub.c | 4 +-
tests/test_hashtab_thrash.c | 15 +-
tests/test_jitterbuf.c | 52 +-
tests/test_json.c | 1810 +
tests/test_linkedlists.c | 2 +-
tests/test_locale.c | 2 +-
tests/test_logger.c | 2 +-
tests/test_message.c | 888 +
tests/test_optional_api.c | 187 +
tests/test_poll.c | 2 +-
tests/test_res_stasis.c | 198 +
tests/test_scoped_lock.c | 281 +
tests/test_security_events.c | 64 +-
tests/test_sorcery.c | 3096 ++
tests/test_sorcery_astdb.c | 638 +
tests/test_sorcery_realtime.c | 898 +
tests/test_stasis.c | 1799 +
tests/test_stasis_channels.c | 331 +
tests/test_stasis_endpoints.c | 310 +
tests/test_stringfields.c | 108 +
tests/test_strings.c | 145 +-
tests/test_substitution.c | 5 +-
tests/test_taskprocessor.c | 750 +
tests/test_threadpool.c | 1646 +
tests/test_time.c | 2 +-
tests/test_uri.c | 154 +
tests/test_utils.c | 133 +-
tests/test_uuid.c | 152 +
tests/test_voicemail_api.c | 271 +-
tests/test_websocket_client.c | 161 +
tests/test_xml_escape.c | 2 +-
utils/Makefile | 38 +-
utils/ael_main.c | 9 +-
utils/astman.c | 2 +-
utils/check_expr.c | 10 +-
utils/clicompat.c | 2 +-
utils/conf2ael.c | 7 +-
utils/extconf.c | 31 +-
utils/muted.c | 9 +
utils/refcounter.c | 312 -
utils/utils.xml | 4 -
1201 files changed, 312319 insertions(+), 142468 deletions(-)
diff --cc ChangeLog
index fd58c5f,d690832..426b1f0
--- a/ChangeLog
+++ b/ChangeLog
@@@ -1,40 -1,1425 +1,1426 @@@
- 2014-10-20 Asterisk Development Team <asteriskteam at digium.com>
+ 2014-10-24 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 13.0.0 Released.
+
+ 2014-10-22 21:27 +0000 [r426097] Shaun Ruffell <sruffell at digium.com>
+
+ * codecs/codec_dahdi.c: codec_dahdi: Cannot use struct
+ ast_translator.core_{src,src}_codec. This fixes a Segmentation
+ fault introduced in r419044 "media formats: re-architect handling
+ of media for performance improvements". The problem is that
+ codec_dahdi was using core_src_codec and core_dst_codec in the
+ ast_translator structure when these fields were never set. Now
+ instead of trying to map the new core codec descriptions to the
+ way DAHDI defines different codecs, we will store the DAHDI
+ specific formats in 'struct translator' directly so we can refer
+ to them without mapping. This also allows us to remove the
+ "global_format_map" structure, since we can now query the list of
+ translators directly to make sure we do not ever register a DAHDI
+ based translator for a specific path more than once and eliminate
+ the need to keep the list and the map in sync. ASTERISK-24435
+ #close Reported by: Marian Koniuszko Review:
+ https://reviewboard.asterisk.org/r/4105/
+
+ 2014-10-21 17:47 +0000 [r426079] Richard Mudgett <rmudgett at digium.com>
+
+ * main/translate.c: translage.c: Fix regression when generating
+ translation path strings. Fix the AMI Status action read and
+ write translation path strings from growing for each channel in
+ the status event list by reseting the ast string given to
+ ast_translate_path_to_str() to fill in the given translation
+ path.
+
+ 2014-10-20 14:15 +0000 [r425991] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_xmpp.c, main/tcptls.c, /: AST-2014-011: Fix POODLE
+ security issues There are two aspects to the vulnerability: (1)
+ res_jabber/res_xmpp use SSLv3 only. This patch updates the module
+ to use TLSv1+. At this time, it does not refactor
+ res_jabber/res_xmpp to use the TCP/TLS core, which should be done
+ as an improvement at a latter date. (2) The TCP/TLS core, when
+ tlsclientmethod/sslclientmethod is left unspecified, will default
+ to the OpenSSL SSLv23_method. This method allows for all
+ encryption methods, including SSLv2/SSLv3. A MITM can exploit
+ this by forcing a fallback to SSLv3, which leaves the server
+ vulnerable to POODLE. This patch adds WARNINGS if a user uses
+ SSLv2/SSLv3 in their configuration, and explicitly disables
+ SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk
+ will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly
+ chosen. For TLS servers, Asterisk will no longer support SSLv2 or
+ SSLv3. Much thanks to abelbeck for reporting the vulnerability
+ and providing a patch for the res_jabber/res_xmpp modules.
+ Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425
+ #close Reported by: abelbeck Tested by: abelbeck, opsmonitor,
+ gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by
+ abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch
+ uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff
+ uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded
+ by mjordan (License 6283) ........ Merged revisions 425987 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-19 17:07 +0000 [r425965] George Joseph <george.joseph at fairview5.com>
- * Asterisk 11.13.1 Released.
-
- * AST-2014-011: Fix POODLE security issues
-
- There are two aspects to the vulnerability:
- (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module
- to use TLSv1+. At this time, it does not refactor res_jabber/
- res_xmpp to use the TCP/TLS core, which should be done as an
- improvement at a latter date.
- (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left
- unspecified, will default to the OpenSSL SSLv23_method. This
- method allows for all encryption methods, including SSLv2/SSLv3.
- A MITM can exploit this by forcing a fallback to SSLv3, which
- leaves the server vulnerable to POODLE. This patch adds WARNINGS
- if a user uses SSLv2/SSLv3 in their configuration, and explicitly
- disables SSLv2/SSLv3 if using SSLv23_method.
-
- For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or
- SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer
- support SSLv2 or SSLv3.
-
- Much thanks to abelbeck for reporting the vulnerability and providing
- a patch for the res_jabber/res_xmpp modules.
-
- 2014-09-24 Asterisk Development Team <asteriskteam at digium.com>
-
- * Asterisk 11.13.0 Released.
-
- 2014-09-19 Asterisk Development Team <asteriskteam at digium.com>
-
- * Asterisk 11.13.0-rc1 Released.
-
- 2014-09-18 16:30 +0000 [r423400] Richard Mudgett <rmudgett at digium.com>
-
- * /, main/astobj2.c, contrib/scripts/refcounter.py:
+ * Makefile, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, makeopts.in: build: Force -fsigned-char on
+ platforms where the default for char is unsigned gcc on the ARM
+ platform defaults 'char' to 'unsigned char' whereas Intel and
+ SPARC default to 'signed char'. This is only an issue in the rare
+ cases where negative values are assigned to a 'char' but this
+ this patch insures compatibility by detecting platforms that
+ default to 'unsigned' and adding an '-fsigned-char' flag to
+ _ASTCFLAGS. If compiling for ARM (native or cross-compile) be
+ sure to run ./bootstrap.sh and ./configure to regenerate the
+ build files. You shouldn't have to do this for Intel or SPARC.
+ Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4091/ ........ Merged
+ revisions 425964 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-19 04:01 +0000 [r425923-425944] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Revert 425922
+ This patch for r425922 introduced a bug, wherein sending an
+ INVITE request with no SDP would cause Asterisk to not send an
+ SDP Offer in the 200 OK. The current structure of
+ res_pjsip_sdp_rtp is a bit hard to deal with to fix this, as
+ create_outgoing_sdp has no knowledge of whether or not it is
+ creating an SDP as a new Offer or an Answer. This is something of
+ an oversight in the callback definition, as the caller of it does
+ have this information.
+
+ * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Remove left over
+ reference to override_prefs The usage of the local override_prefs
+ variable in create_outgoing_sdp_stream was previously to track an
+ override format preference set by PJSIP_MEDIA_OFFER. Now,
+ however, that function simply sets the joint capabilities
+ structure, session->req_caps. During the media format rework, the
+ override_prefs was instead used to check if there were any
+ formats in session->req_caps. However, this usage isn't useful in
+ create_outgoing_sdp_stream. session->req_caps contains the
+ negotiated formats for *all* streams, not just the current one
+ being created. Thus, so long as any stream of any type has
+ provided a format, override_prefs will be non-zero. Hence, its
+ usage in checking whether or not we should look at the formats on
+ the endpoint or the joint capabilities is generally useless.
+ There's only two things useful to check: (1) Does the endpoint
+ have a format for the media type? (2) Did we negotiate a format
+ for the media type? If either of those is a 'no', then we must
+ kill the media stream.
+
+ 2014-10-17 22:43 +0000 [r425905] Jonathan Rose <jrose at digium.com>
+
+ * configs/samples/cli_aliases.conf.sample: Sample Configurations:
+ make 'pjsip reload' reload all reloadable pjsip modules AST-1432
+ #close Reported by: John Bigelow
+
+ 2014-10-17 13:35 +0000 [r425821-425879] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_pjsip_sdp_rtp.c, res/res_pjsip.c,
+ res/res_pjsip_session.c, /: res_pjsip_session/res_pjsip_sdp_rtp:
+ Be more tolerant of offers When an inbound SDP offer is received,
+ Asterisk currently makes a few incorrection assumptions: (1) If
+ the offer contains more than a single audio/video stream,
+ Asterisk will reject the entire stream with a 488. This is an
+ overly strict response; generally, Asterisk should accept the
+ media streams that it can accept and decline the others. (2) If
+ the offer contains a declined media stream, Asterisk will attempt
+ to process it anyway. This can result in attempting to match
+ format capabilities on a declined media stream, leading to a 488.
+ Asterisk should simply ignore declined media streams. (3)
+ Asterisk will currently attempt to handle offers with AVPF with
+ use_avpf=No/AVP with use_avpf=Yes. This mismatch results in
+ invalid SDP answers being sent in response. If there is a
+ mismatch between the media type being offered and the
+ configuration, Asterisk must reject the offer with a 488. This
+ patch does the following: * Asterisk will accept SDP offers with
+ at least one media stream that it can use. Some WARNING messages
+ have been dropped to NOTICEs as a result. * Asterisk will not
+ accept an offer with a media type that doesn't match its
+ configuration. * Asterisk will ignore declined media streams
+ properly. #SIPit31 Review:
+ https://reviewboard.asterisk.org/r/4063/ ASTERISK-24122 #close
+ Reported by: James Van Vleet ASTERISK-24381 #close Reported by:
+ Matt Jordan ........ Merged revisions 425868 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, channels/chan_sip.c: channels/chan_sip: Respect outboundproxy
+ setting when sending qualify requests The outboundproxy setting
+ is currently ignored when sending OPTIONS requests as a result of
+ the qualify setting. This means that if an Asterisk server is
+ unable to send the packet directly to a peer, it is unable to
+ qualify any non-inbound registered peer (e.g. a peer SIP Trunk).
+ This patch grabs the outboundproxy information for a peer when a
+ qualify attempt is being constructed and, if it finds the
+ information, uses it when sending the OPTIONS request. Review:
+ https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close
+ Reported by: Damian Ivereigh patches: outboundproxy-dai.patch
+ uploaded by Damian Ivereigh (License 6632) ........ Merged
+ revisions 425818 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425819 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425820 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-17 02:41 +0000 [r425783] Richard Mudgett <rmudgett at digium.com>
+
+ * main/core_unreal.c, main/channel.c, /: AMI: Add missing VarSet
+ events when a channel inherits variables. There should be AMI
+ VarSet events when channel variables are inherited by an outgoing
+ channel. Also local;2 should generate VarSet events when it gets
+ all of its channel variables from channel local;1. ASTERISK-24415
+ #close Reported by: Richard Mudgett Patches:
+ jira_asterisk_24415_v12.patch (license #5621) patch uploaded by
+ Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/
+ ........ Merged revisions 425782 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-17 01:57 +0000 [r425736-425761] Matthew Jordan <mjordan at digium.com>
+
+ * /, bridges/bridge_native_rtp.c: bridge_native_rtp: Fix audio
+ issues when moving from remote bridge to softmix When a native
+ RTP bridge that is remotely bridging its participants switches to
+ a softmix bridge, it may not properly re-INVITE the media for one
+ or both participants back to Asterisk. This is due to the current
+ bridge_native_rtp code only re-INVITEs if it believes the channel
+ will survive the bridge operation. Currently, that code is
+ failing, as it expects the channels to have a soft hangup flag
+ set on it indicating that a redirect has occurred or that the
+ channel is going to leave the bridge. (The code did not take into
+ account a smart bridge operation). This patch also renames a few
+ things to be more reflective of the underlying types. Review:
+ https://reviewboard.asterisk.org/r/3997/ ASTERISK-24327 #close
+ ........ Merged revisions 425760 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, tests/test_cel.c: test_cel: Update pickup test to expect
+ CANCEL instead of ANSWSER The CEL pickup test previously looked
+ for a disposition of ANSWER between the original caller/peer when
+ the call is picked up. This is actually incorrect: the
+ disposition should, at the very least, not be ANSWER as the call
+ was never ANSWERed. The disposition is now CANCEL; this patch
+ updates the test accordingly. ........ Merged revisions 425757
+ from http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * main/cdr.c, /: main/cdr: Use 'time' when rescheduling batched
+ CDRs as opposed to 'size' When refactoring CDRs to use the
+ configuration framework, a 'whoops' was introduced where the CDR
+ batch size was used when rescheduling a batch, as opposed to the
+ time duration. This patch corrects that obvious mistake.
+ ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged
+ revisions 425735 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-16 17:30 +0000 [r425714] George Joseph <george.joseph at fairview5.com>
+
+ * include/asterisk/config.h, tests/test_config.c, main/config.c, /:
+ config: Fix inf loop using ast_category_browse and
+ ast_variable_retrieve Fix infinite loop when calling
+ ast_variable_retrieve inside an ast_category_browse loop when
+ there is more than 1 category with the same name. Tested-by:
+ George Joseph Review: https://reviewboard.asterisk.org/r/4089/
+ ........ Merged revisions 425713 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-16 14:35 +0000 [r425691] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_pjsip_t38.c, res/res_pjsip_registrar_expire.c,
+ res/res_pjsip_mwi_body_generator.c,
+ res/res_pjsip_endpoint_identifier_user.c,
+ res/res_pjsip_send_to_voicemail.c,
+ include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_outbound_authenticator_digest.c,
+ res/res_pjsip_outbound_registration.c,
+ res/res_pjsip_endpoint_identifier_anonymous.c,
+ res/res_pjsip_path.c, res/res_pjsip_one_touch_record_info.c,
+ res/res_pjsip_acl.c, res/res_pjsip_pubsub.c,
+ res/res_pjsip_diversion.c, res/res_pjsip_refer.c,
+ include/asterisk/res_pjsip.h,
+ res/res_pjsip_pidf_body_generator.c, res/res_pjsip_dtmf_info.c,
+ res/res_pjsip_multihomed.c, res/res_pjsip_authenticator_digest.c,
+ res/res_pjsip_sdp_rtp.c, res/res_hep_pjsip.c,
+ res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c,
+ res/res_pjsip_logger.c, res/res_pjsip_nat.c,
+ res/res_pjsip_session.c, res/res_pjsip_exten_state.c,
+ res/res_pjsip_header_funcs.c, res/res_pjsip_rfc3326.c,
+ res/res_pjsip_phoneprov_provider.c, res/res_pjsip_mwi.c,
+ res/res_pjsip_dialog_info_body_generator.c,
+ res/res_pjsip_xpidf_body_generator.c, res/res_pjsip_registrar.c,
+ channels/chan_pjsip.c, res/res_pjsip_transport_websocket.c,
+ res/res_pjsip_pidf_eyebeam_body_supplement.c,
+ include/asterisk/res_pjsip_session.h, /, res/res_pjsip_notify.c,
+ res/res_pjsip_pidf_digium_body_supplement.c,
+ res/res_pjsip_endpoint_identifier_ip.c,
+ res/res_pjsip_publish_asterisk.c: PJSIP: Enforce module load
+ dependencies This enforces that res_pjsip, res_pjsip_session, and
+ res_pjsip_pubsub have loaded properly before attempting to load
+ any modules that depend on them since the module loader system is
+ not currently capable of resolving module dependencies on its
+ own. ASTERISK-24312 #close Reported by: Dafi Ni Review:
+ https://reviewboard.asterisk.org/r/4062/ ........ Merged
+ revisions 425690 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-16 06:11 +0000 [r425669] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c, /: Fix loss of voice after second call
+ drops (on a second line) in case using multiple lines on unistim
+ phones. There is regression was introduced in r391379. Reported
+ by: Rustam Khankishyiev (closes issue ASTERISK-23846) ........
+ Merged revisions 425667 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425668 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-16 01:25 +0000 [r425646] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix a bug where ICE
+ state would get reset when it shouldn't. In the case where the
+ ICE negotiation had not yet started current state would get wiped
+ when it shouldn't. This also removes channel binding as in
+ practice this does not work well with other implementations.
+ ........ Merged revisions 425644 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425645 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-15 19:31 +0000 [r425627] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_motif.c: chan_motif: Cleanup
+ jingle_tech.capabilities only once.
+
+ 2014-10-15 19:05 +0000 [r425611] Jonathan Rose <jrose at digium.com>
+
+ * res/parking/parking_tests.c: parking_tests: Fix assertions and
+ possibly crashes in res_parking unit tests Assertions were caused
+ by attempting to play music on hold to a channel with no formats.
+ Parking unit test channels were given formats and a technology so
+ that they would be able to pretend to read/write frames.
+ ASTERISK-24413 #close Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/4075/
+
+ 2014-10-15 09:59 +0000 [r425590] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: chan_ooh323: fix rtptimeout general
+ value checking correct condition to check rtptimeout in [general]
+ config section ASTERISK-24393 #close Reported by: Dmitry Melekhov
+ Tested by: Dmitry Melekhov Patches: ASTERISK-24393.patch ........
+ Merged revisions 425547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425548 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425589 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-14 20:46 +0000 [r425526] George Joseph <george.joseph at fairview5.com>
+
+ * /, include/asterisk/config.h, tests/test_config.c, main/config.c:
+ config: Fix SEGV in unit test with MALLOC_DEBUG With MALLOC_DEBUG
+ the /main/config config_basic_ops test was causing a SEGV while
+ doing an ast_category_delete in an ast_category_browse loop.
+ Apparently this never worked but was also never tested. I removed
+ the test, added 2 notes to config.h indicating that it's not
+ supported and added a few lines of code to ast_category_delete to
+ prevent the SEGV should someone attempt it in the future.
+ Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4078/ ........ Merged
+ revisions 425525 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-14 19:00 +0000 [r425504] Jonathan Rose <jrose at digium.com>
+
+ * main/sched.c, /: Scheduler: Fix a nasty scheduler caching bug
+ which makes new tasks not execute Tasks that were marked for
+ pending deletion in the scheduler would be moved to the cache for
+ later reuse, but after being recycled the deleted mark wouldn't
+ be removed resulting in fresh tasks being deleted without
+ reason... and immediately moved back into the cache where they
+ could be reused again. This could cause horrendous things to
+ happen in just about anything that used a scheduler.
+ ASTERISK-24321 #close Reported by: Steve Pitts Review:
+ https://reviewboard.asterisk.org/r/4071/ ........ Merged
+ revisions 425503 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-14 18:12 +0000 [r425481] George Joseph <george.joseph at fairview5.com>
+
+ * res/res_phoneprov.c, include/asterisk/phoneprov.h, /,
+ res/res_pjsip_phoneprov_provider.c: res_phoneprov: Create
+ accessor for ast_phoneprov_std_variable_lookup Based on feedback
+ from Richard, I created an accessor for
+ res_phoneprov/ast_phoneprov_std_variable_lookup and added load
+ priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by:
+ Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/
+ ........ Merged revisions 425480 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-14 16:46 +0000 [r425459] Corey Farrell <git at cfware.com>
+
+ * /, res/res_fax.c: res_fax: Fix reference leak caused by gateway
+ sessions Fax gateway session objects can be re-used, causing the
+ same gateway session to be added to faxregistry.container more
+ than once. This change causes fax_session_new to remove the
+ reserved session from the container before it's id is changed,
+ ensuring it's possible for the session to be freed.
+ ASTERISK-24392 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4049/ ........ Merged
+ revisions 425457 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425458 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-14 16:35 +0000 [r425455] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/stasis_channels.c: stasis_channels.c: Resolve unfinished
+ Dials when doing masquerades (Part 2) Masquerades into and out of
+ channels that are involved in a dial operation don't create the
+ expected dial end event. The missing dial end event goes against
+ the model for things like CDRs and generating Dial end manager
+ actions and such. There are four cases: 1) A channel masquerades
+ into the caller channel. The case happens when performing a
+ blonde transfer using the channel driver's protocol. 2) A channel
+ masquerades into a callee channel. The case happens when
+ performing a directed call pickup. 3) The caller channel
+ masquerades out of dial. The case happens when using the Bridge
+ application on the caller channel. 4) A callee channel
+ masquerades out of dial. The case happens when using the Bridge
+ application on a peer channel. As it turned out, all four cases
+ need to be handled instead of just the first one. ASTERISK-24237
+ Reported by: Richard Mudgett ASTERISK-24394 #close Reported by:
+ Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/
+ ........ Merged revisions 425430 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-14 16:19 +0000 [r425415] Corey Farrell <git at cfware.com>
+
+ * /, res/res_fax.c: res_fax: Resolve module reference leak caused
+ by reserved sessions Remove reference to module providing
+ reserved session after adding a reference to the final module.
+ This re-reference is done to ensure that module references are
+ correct even if the final session selects a different module than
+ the reserved session. ASTERISK-18923 #close Reported by: Grigoriy
+ Puzankin Review: https://reviewboard.asterisk.org/r/4048/
+ ........ Merged revisions 425405 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425407 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425411 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-13 16:10 +0000 [r425384] George Joseph <george.joseph at fairview5.com>
+
+ * apps/app_directory.c, tests/test_sorcery.c, main/config.c,
+ tests/test_sorcery_realtime.c, res/res_sorcery_realtime.c,
+ apps/app_voicemail.c, res/res_sorcery_config.c, main/manager.c,
+ /, include/asterisk/config.h, pbx/pbx_realtime.c,
+ tests/test_config.c: manager/config: Support templates and
+ non-unique category names via AMI This patch provides the
+ capability to manipulate templates and categories with non-unique
+ names via AMI. Summary of changes: GetConfig and GetConfigJSON:
+ Added "Filter" parameter: A comma separated list of
+ name_regex=value_regex expressions which will cause only
+ categories whose variables match all expressions to be
+ considered. The special variable name TEMPLATES can be used to
+ control whether templates are included. Passing 'include' as the
+ value will include templates along with normal categories.
+ Passing 'restrict' as the value will restrict the operation to
+ ONLY templates. Not specifying a TEMPLATES expression results in
+ the current default behavior which is to not include templates.
+ UpdateConfig: NewCat now includes options for allowing duplicate
+ category names, indicating if the category should be created as a
+ template, and specifying templates the category should inherit
+ from. The rest of the actions now accept a filter string as
+ defined above. If there are non-unique category names, you can
+ now update specific ones based on variable values. To facilitate
+ the new capabilities in manager, corresponding changes had to be
+ made to config, most notably the addition of filter criteria to
+ many of the APIs. In some cases it was easy to change the
+ references to use the new prototype but others would have
+ required touching too many files for this patch so a wrapper with
+ the original prototype was created. Macros couldn't be used in
+ this case because it would break binary compatibility with
+ modules such as res_digium_phone that are linked to real symbols.
+ Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4033/ ........ Merged
+ revisions 425383 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-12 21:09 +0000 [r425362] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Make the ICE
+ transport check case insensitive as some implementations use
+ 'udp'. ........ Merged revisions 425360 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425361 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-12 08:15 +0000 [r425289-425299] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: chan_sip: Fix so asterisk won't send
+ reINVITE after a BYE. After a reINVITE glare situation, Asterisk
+ would re-send the reINVITE even though the call had been hung up
+ in the mean time. This patch unschedules the reinvite when
+ handling the BYE. ASTERISK-22791 #close Reported by: Paolo
+ Compagnini Tested by: Paolo Compagnini Review:
+ https://reviewboard.asterisk.org/r/4056/ (testcase is in review
+ r4055) ........ Merged revisions 425296 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425297 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425298 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, Makefile: build: Relax badshell tilde test to allow for ~ in
+ middle of DESTDIR. The main Makefile has a target test called
+ 'badshell' that tests if DESTDIR does not happen to have an
+ an-expanded tilde (~). This might be the case if you run: make
+ install DESTDIR=~/somewhere/ That test also disallowed valid
+ tildes in directory names. The test is now changed to only
+ trigger on a tilde at the start of the path. ASTERISK-13797
+ #close Reported by: Tzafrir Cohen Review:
+ https://reviewboard.asterisk.org/r/4064/ ........ Merged
+ revisions 425291 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425292 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425293 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, res/res_calendar_ews.c: res_calendar_ews: Relax neon version
+ check to work with 0.30 too. Allow res_calendar_ews to work not
+ only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close
+ Reported by: Tzafrir Cohen Review:
+ https://reviewboard.asterisk.org/r/4068/ ........ Merged
+ revisions 425286 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425287 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425288 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-11 21:08 +0000 [r425265] George Joseph <george.joseph at fairview5.com>
+
+ * /, res/res_phoneprov.c: res_phoneprov: Cleanup module load error
+ handling Tested module load/reload interaction between
+ res_phoneprov and res_pjsip_phoneprov_provider in cases where
+ res_phoneprov didn't load correctly (usually misconfiguration or
+ missing phoneprov.conf) Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4069/ ........ Merged
+ revisions 425264 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-10 20:48 +0000 [r425243] Joshua Colp <jcolp at digium.com>
+
+ * /, main/bridge.c, bridges/bridge_native_rtp.c: bridge: During a
+ smart bridge operation provide a more complete bridge to the old
+ technology. When a smart bridge operation occurs and a bridge
+ transitions from one technology to another the old technology is
+ provided the channels formerly in it and told that they are
+ leaving. Unfortunately the bridge provided along with them is
+ incomplete. The bridge, despite there being channels in it,
+ contains none. This forces technology implementations to have
+ additional logic when channels are leaving or to store their own
+ duplicated state. This change makes the bridge more complete so
+ it contains the expected channels. Now that the bridge is
+ complete special logic within bridge_native_rtp is no longer
+ needed and has been removed. Review:
+ https://reviewboard.asterisk.org/r/4057/ ........ Merged
+ revisions 425242 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-10 14:31 +0000 [r425221] Matthew Jordan <mjordan at digium.com>
+
+ * /, res/res_phoneprov.c: res/res_phoneprov: Bail on registration
+ if res_phoneprov didn't load If res_phoneprov failed to fully
+ load (due to not being configured), the providers container will
+ be NULL. If a module attempts to register a phone provisioning
+ provider, it should check for the presence of the container. If
+ there is no providers container, it should return an error. This
+ patch makes the ast_phoneprov_provider_register function do
+ that... otherwise this would be a silly commit message. ........
+ Merged revisions 425220 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-10 14:23 +0000 [r425217] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_pjsip_phoneprov_provider.c:
+ res_pjsip_phoneprov_provider: Add missing dependency on
+ pjproject. ........ Merged revisions 425216 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-10 13:01 +0000 [r425155] Kinsey Moore <kmoore at digium.com>
+
+ * /, tests/test_callerid.c, main/callerid.c: CallerID: Fix parsing
+ regression This fixes a regression in callerid parsing introduced
+ when another bug was fixed. This bug occurred when the name was
+ composed entirely of DTMF keys and quoted without a number
+ section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard
+ Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by
+ Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/
+ ........ Merged revisions 425152 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425153 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425154 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-10 12:10 +0000 [r425132] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_nat.c, /: res_pjsip_nat: Place source port into
+ rport of responses if 'force_rport' is on. When the 'force_rport'
+ option is enabled the behavior should be the same as if the
+ remote side placed rport into the message themselves. Therefore
+ any responses we send should include the source port of the
+ request in the rport of the Via header. #SIPit31 ASTERISK-24387
+ #close Reported by: Matt Jordan ........ Merged revisions 425131
+ from http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-10 07:32 +0000 [r425071] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: chan_sip: Fix dialog leak resulting from
+ missing ACK to re-INVITE. If a device re-INVITEs at the same time
+ as the dialog is hung up, and if then the ACK to the re-INVITE
+ never reaches Asterisk, chan_sip would fail to destroy the dialog
+ after a while. This resulted in (most prominently) file handle
+ leaks. (Patch reindented by me.) ASTERISK-20784 #close
+ ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal
+ Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle
+ (License #5334) patch_asterisk_20784.txt uploaded by Nitesh
+ Bansal (License #6418) Reviewboard:
+ https://reviewboard.asterisk.org/r/4052/ (testcase can be found
+ at r4051) ........ Merged revisions 425068 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425069 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 425070 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-09 23:35 +0000 [r425052] George Joseph <george.joseph at fairview5.com>
+
+ * res/res_pjsip_phoneprov_provider.c: res_pjsip_phoneprov_provider:
+ fix compile breakage on AST_VECTOR endpoint->inbound_auths was
+ changed to a vector in 13 and I committed the 12 patch instead of
+ the 13 patch. Tested-by: George Joseph
+
+ 2014-10-09 21:38 +0000 [r425031] Kevin Harwell <kharwell at digium.com>
+
+ * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Crash if no
+ candidates received for component When starting ice if there is
+ not at least one remote ice candidate with an RTP component
+ asterisk will crash. This is due to an assertion in pjnath as it
+ expects at least one candidate with an RTP component. Added a
+ check to make sure at least one candidate contains an RTP
+ component and at least one candidate has an RTCP component.
+ ASTERISK-24383 #close Review:
+ https://reviewboard.asterisk.org/r/4039/ ........ Merged
+ revisions 425030 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-09 20:54 +0000 [r425008] George Joseph <george.joseph at fairview5.com>
+
+ * /, res/res_pjsip_phoneprov_provider.c (added),
+ configs/samples/pjsip.conf.sample: res_pjsip_phoneprov_provider:
+ Provides pjsip integration with res_phoneprov This module allows
+ res_pjsip to integrate with res_phoneprov. It handles the pjsip
+ 'phoneprov' object type. Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3976/ ........ Merged
+ revisions 425007 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-09 18:37 +0000 [r424986] Matthew Jordan <mjordan at digium.com>
+
+ * /, res/res_phoneprov.c: res/res_phoneprov: Don't cancel Asterisk
+ load on module load failure ........ Merged revisions 424985 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-09 17:45 +0000 [r424964] George Joseph <george.joseph at fairview5.com>
+
+ * include/asterisk/phoneprov.h (added), /,
+ configs/samples/phoneprov.conf.sample,
+ include/asterisk/chanvars.h, res/res_phoneprov.c,
+ res/res_phoneprov.exports.in (added), main/chanvars.c:
+ res_phoneprov: Refactor phoneprov to allow pluggable config
+ providers This patch makes res_phoneprov more modular so other
+ modules (like pjsip) can provide configuration information
+ instead of res_phoneprov relying solely on users.conf and
+ sip.conf. To accomplish this a new ast_phoneprov public API is
+ now exposed which allows config providers to register themselves,
+ set defaults (server profile, etc) and add user extensions. *
+ ast_phoneprov_provider_register registers the provider and
+ provides callbacks for loading default settings and loading
+ users. * ast_phoneprov_provider_unregister clears the defaults
+ and users. * ast_phoneprov_add_extension should be called once
+ for each user/extension by the provider's load_users callback to
+ add them. * ast_phoneprov_delete_extension deletes one extension.
+ * ast_phoneprov_delete_extensions deletes all extensions for the
+ provider. Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3970/ ........ Merged
+ revisions 424963 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-09 16:36 +0000 [r424942] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/cdr.c: cdr.c: Make turning on CDR debug a one step
+ process instead of two. Now "cdr set debug on" doesn't also
+ require "core set verbose 1" to see CDR debug output. ........
+ Merged revisions 424941 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-09 08:08 +0000 [r424880] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, contrib/scripts/safe_asterisk: safe_asterisk: Don't
+ automatically exceed MAXFILES value of 2^20. On systems with lots
+ of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can
+ exceed the per-process file limit of 2^20. This patch ensures the
+ value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close
+ Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff
+ uploaded by Michael Myles (License #6626) ........ Merged
+ revisions 424875 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 424878 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 424879 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-08 18:46 +0000 [r424854] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Allow only UDP ICE
+ candidates. The underlying library, pjnath, that res_rtp_asterisk
+ uses for ICE support does not have support for ICE-TCP. As
+ candidates are passed through directly to it this can cause error
+ messages to occur when it receives something unexpected (such as
+ a TCP candidate). This change merely ignores all non-UDP
+ candidates so they never reach pjnath. ASTERISK-24326 #close
+ Reported by: Joshua Colp ........ Merged revisions 424852 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 424853 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-08 18:24 +0000 [r424769-424850] Kinsey Moore <kmoore at digium.com>
+
+ * main/stasis.c: Stasis: Relegate log message to dev-mode This
+ error message primarily applies to development tasks and will now
+ only show up when dev-mode is enabled via configure.
+
+ * main/sounds_index.c: Indexer: Format message types may not exist
+ In Asterisk 13+, any given message type is not guaranteed to
+ exist even if Asterisk comes up correctly since creation of the
+ message type could be declined. The indexer should not prevent
+ Asterisk from starting under these conditions.
+
+ * main/stasis.c: Stasis: Only log errors for non-declined types
+ When message type creation is declined via stasis.conf, certain
+ operations log errors assuming that the declined type is being
+ used before initialization or after destruction. These error
+ messages get quite spammy for oft used message types and should
+ not be logged in the first place since the message type is
+ validly NULL. Reported by: Matt DiMeo
+
+ 2014-10-07 18:33 +0000 [r424752] Joshua Colp <jcolp at digium.com>
+
+ * main/data.c: data: Properly access formats in capabilities
+ structure when adding codecs. Formats within a capabilities
+ structure are addressed starting at 0, not 1. Assuming 1 causes
+ it to exceed an array. ASTERISK-24389 #close Reported by: Kevin
+ Harwell
+
+ 2014-10-07 17:41 +0000 [r424692-424731] Matthew Jordan <mjordan at digium.com>
+
+ * /, res/res_pjsip_outbound_registration.c:
+ res/res_pjsip_outbound_registration: Initialize
+ auth_reject_permanent parameter Prior to this patch, the
+ auth_reject_permanent parameter was not initialized on the
+ registration client state, leading to the parameter being
+ disabled regardless of the value specified in pjsip.conf. This
+ patch initialized the setting on the registration client state to
+ the provided configuration value. ASTERISK-24398 #close ........
+ Merged revisions 424730 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Fix typo in WARNING
+ message
+
+ * main/message.c, /: message: Don't close an AMI connection on
+ SendMessage action error If SendMessage encounters an error (such
+ as incorrect input provided to the action), it will currently
+ return -1. Actions should only return -1 if the connection to the
+ AMI client should be closed. In this case, SendMessage causing
+ the client to disconnect is inappropriate. This patch causes the
+ action to return 0, which simply causes the action to fail.
+ Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354
+ #close Reported by: Peter Katzmann patches: sendMessage.patch
+ uploaded by Peter Katzmann (License 5968) ........ Merged
+ revisions 424690 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 424691 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-06 15:38 +0000 [r424669] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c, /: features.c: Fix lingering channel ref while
+ Bridge() application is active. Using the Bridge application to
+ bridge a channel that is executing an applicaiton such as Wait
+ results in a lingering Surrogate channel in the CLI "core show
+ channels" output even though it has already hungup. * Fix
+ bridge_exec() to not hold onto the current_dest_chan ref once it
+ has been put into the bridge. * Eliminated bridge_exec()'s use of
+ RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson
+ Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged
+ revisions 424668 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-06 12:38 +0000 [r424601-424647] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/sdp_srtp.c: sdp_srtp: Add new lines to some WARNING
+ messages ........ Merged revisions 424646 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, res/res_pjsip/pjsip_options.c: res_pjsip/pjsip_options: Do not
+ 404 an OPTIONS request not sent to an endpoint An OPTIONS request
+ that is sent to Asterisk but not to a specific endpoint is
+ currently sent a 404 in response. This is because, not
+ surprisingly, an empty extension is never going to be found in
+ the dialplan. This patch makes it so that we only attempt to look
+ up the endpoint in the dialplan if it is specified in the OPTIONS
+ request URI. #SIPit31 ASTERISK-24370 #close Reported by: Matt
+ Jordan ........ Merged revisions 424624 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * channels/pjsip/dialplan_functions.c, /: pjsip/dialplan_functions:
+ Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels Calling
+ PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your
+ health. It will treat the channels as a PJSIP channel, eventually
+ hitting an ao2 error, FRACKing on assertion error, and quite
+ likely crashing. This patch adds checks to the read/write
+ callbacks that ensure that the channel technology is of type
+ 'PJSIP' before attempting to operate on the channel. #SIPit31
+ ASTERISK-24382 #close Reported by: Matt Jordan ........ Merged
+ revisions 424621 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, res/res_hep_pjsip.c, res/res_pjsip/pjsip_distributor.c,
+ res/res_pjsip_logger.c: res_pjsip: Prevent crashes when PJPROJECT
+ presents an rdata with no message When a message that exceeds the
+ PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible
+ (although it shouldn't occur) for pjproject to pass up an rdata
+ object with a NULL msg in the msg_info. Needless to say, things
+ that attempt to dereference this are in for a rough ride. In
+ particular, this caused crashes in three different locations, all
+ of which are 'low level' enough to intercept an rdata object
+ early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3)
+ res_pjsip/distributor Anything that can intercept an rdata object
+ before res_pjsip/distributor should be defensive when looking at
+ the received packet. #SIPit31 ASTERISK-24369 #close Reported by:
+ Matt Jordan ........ Merged revisions 424618 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Gracefully handle
+ errors when re-creating subscriptions A subscription that has
+ been persisted can - for various reasons - fail to be re-created
+ on startup. This patch resolves a number of crashes that occurred
+ when a subscription cannot be re-created on several off-nominal
+ paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan
+
+ 2014-10-05 00:48 +0000 [r424552-424580] Corey Farrell <git at cfware.com>
+
+ * main/manager.c, /: Release AMI connections on shutdown.
+ ASTERISK-24378 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4037/ ........ Merged
+ revisions 424578 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 424579 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * channels/chan_motif.c: chan_motif: Correct last commit to use
+ ao2_cleanup to free format cap This fix applies to 13 and trunk.
+ ASTERISK-24384 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4043/
+
+ * /, channels/chan_motif.c: chan_motif: Release format capabilities
+ and config on module load error ASTERISK-24384 #close Reported
+ by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4043/ ........ Merged
+ revisions 424550 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 424551 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-03 21:56 +0000 [r424472-424529] Richard Mudgett <rmudgett at digium.com>
+
+ * /, CHANGES, res/res_pjsip.c: res_pjsip: Fix XML typo and update
+ CHANGES. ASTERISK-24199 ........ Merged revisions 424528 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c, /,
+ main/framehook.c: audiohooks: Reevaluate the bridge technology
+ when an audiohook is added or removed. Adding a mixmonitor to a
+ channel causes the bridge to change technologies from native to
+ simple_bridge so the call can be recorded. However, when the
+ mixmonitor is stopped the bridge does not switch back to the
+ native technology. * Added unbridge requests to reevaluate the
+ bridge when a channel audiohook is removed. * Moved the unbridge
+ request into ast_audiohook_attach() ensure that the bridge
+ reevaluates whenever an audiohook is attached. This simplified
+ the mixmonitor and chan_spy start code as well. * Added defensive
+ code to stop_mixmonitor_full() in case additional arguments are
+ ever added to the StopMixMonitor application. * Made
+ ast_framehook_detach() not do an unbridge request if the
+ framehook does not exist. * Made ast_framehook_list_fixup() do an
+ unbridge request if there are any framehooks. Also simplified the
+ loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/4046/ ........ Merged
+ revisions 424506 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * main/core_unreal.c, main/taskprocessor.c, channels/chan_iax2.c,
+ res/res_pjsip_session.c, main/channel.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, funcs/func_frame_trace.c,
+ channels/chan_motif.c, include/asterisk/frame.h,
+ main/bridge_channel.c, channels/chan_pjsip.c,
+ channels/chan_unistim.c, include/asterisk/res_pjsip_session.h,
+ addons/chan_ooh323.c, /, include/asterisk/taskprocessor.h,
+ channels/chan_sip.c, res/res_pjsip_session.exports.in:
+ chan_pjsip: Fix deadlock when masquerading PJSIP channels.
+ Performing a directed call pickup resulted in a deadlock when
+ PJSIP channels were involved. A masquerade needs to hold onto the
+ channel locks while it swaps channel information between the two
+ channels involved in the masquerade. With PJSIP channels, the
+ fixup routine needed to push a fixup task onto the PJSIP
+ channel's serializer. Unfortunately, if the serializer was also
+ processing a task that needed to lock the channel, you get
+ deadlock. * Added a new control frame that is used to notify the
+ channels that a masquerade is about to start and when it has
+ completed. * Added the ability to query taskprocessors if the
+ current thread is the taskprocessor thread. * Added the ability
+ to suspend/unsuspend the PJSIP serializer thread so a masquerade
+ could fixup the PJSIP channel without using the serializer.
+ ASTERISK-24356 #close Reported by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/4034/ ........ Merged
+ revisions 424471 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-03 15:54 +0000 [r424448] George Joseph <george.joseph at fairview5.com>
+
+ * /, main/sorcery.c: sorcery: Prevent SEGV in sorcery_wizard_create
+ when there's no create function When you call
+ ast_sorcery_create() you don't necessarily know which wizard is
+ going to be invoked. If it happens to be a wizard like 'config'
+ that doesn't have a 'create' virtual function you get a segfault
+ in the sorcery_wizard_create callback. This patch catches the
+ null function pointer, does an ast_assert, and logs an error.
+ Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged
+ revisions 424447 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-03 13:58 +0000 [r424424-424427] Kinsey Moore <kmoore at digium.com>
+
+ * configs/samples/pjsip.conf.sample, /,
+ res/res_pjsip/pjsip_configuration.c: PJSIP: Restore functional
+ default for callerid_privacy The pjsip config option default
+ fixups from r424263 altered the functional default from
+ "allowed_not_screened" to "allowed". This change restores the
+ functional default value when none is provided. ........ Merged
+ revisions 424426 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * main/manager.c, /: Manager: Add missing fields and documentation
+ for CoreShowChannels This corrects some issues introduced in the
+ responses to the CoreShowChannels AMI command as well as adding
+ documentation for the responses. The command in Asterisk 12 was
+ missing the following fields: Duration, Application,
+ ApplicationData, and BridgedChannel and BridgedUniqueID (replaced
+ with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn
+ Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged
+ revisions 424423 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-03 07:54 +0000 [r424415] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_session.c, /: res_pjsip_session: Reduce SDP size by
+ removing duplicate connection lines. Due to the architecture of
+ how media streams are handled each individual handler adds
+ connection details (IP address) for it. The first media stream is
+ then used as the top level SDP connection line. In practice each
+ line ends up being the same so to reduce the SDP size
+ stream-level connection information is also added to the SDP if
+ it differs from the top level SDP connection line. ........
+ Merged revisions 424414 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-02 21:52 +0000 [r424394] Richard Mudgett <rmudgett at digium.com>
+
+ * /, configs/samples/pjsip.conf.sample, res/res_pjsip.c,
+ res/res_pjsip/config_transport.c: res_pjsip: Make transport
+ cipher option accept a comma separated list of cipher names.
+ Improvements to the res_pjsip transport cipher option. * Made the
+ cipher option accept a comma separated list of OpenSSL cipher
+ names. Users of realtime will be glad if they have more than one
+ name to list. * Added the CLI command 'pjsip list ciphers' so a
+ user can know what OpenSSL names are available for the cipher
+ option. * Updated the cipher option online XML documentation to
+ specify what is expected for the value. * Updated
+ pjsip.conf.sample to not indicate that ALL is acceptable since
+ ALL does not imply a preference order for the ciphers and PJSIP
+ does not simply pass the string to OpenSSL for interpretation.
+ ASTERISK-24199 #close Reported by: Joshua Colp Review:
+ https://reviewboard.asterisk.org/r/4018/ ........ Merged
+ revisions 424393 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-02 20:15 +0000 [r424373] Jonathan Rose <jrose at digium.com>
+
+ * /,
+ contrib/ast-db-manage/config/versions/10aedae86a32_add_outgoing_enum_va.py
+ (added): Alembic: Add enumerator value to sippeers -> directmedia
+ - 'outgoing' The 'outgoing' value was left off of the enumerator
+ when first creating the column. This patch adds it, and should
+ gracefully upgrade keeping the existing data in tact.
+ ASTERISK-23781 #close Reported by: Stephen More Review:
+ https://reviewboard.asterisk.org/r/4013/ ........ Merged
+ revisions 424372 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-02 13:35 +0000 [r424338] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * /, configs/samples/pjsip.conf.sample: res_pjsip: document use of
+ rewrite_contact in sample conf Without setting rewrite_contact,
+ an invite to an endpoint behind NAT will not reach it - unless
+ the endpoint itself uses STUN or TURN to discover it's public
+ URI. Thus, the use of this should be in the sample documentation.
+ Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged
+ revisions 424337 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-01 22:52 +0000 [r424333] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_pjsip.c: chan_pjsip: Fix an assertion for channels
+ that lack formats on creation ASTERISK-24222 #close Reported by:
+ Mark Michelson Review: https://reviewboard.asterisk.org/r/4017/
+
+ 2014-10-01 20:36 +0000 [r424313] Corey Farrell <git at cfware.com>
+
+ * res/res_hep.c, /: res_hep: Release allocation reference to
+ configuration. ASTERISK-24362 #close Reported by: Corey Farrell
+ Review: https://reviewboard.asterisk.org/r/4026/ ........ Merged
+ revisions 424312 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-01 16:37 +0000 [r424288-424291] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_pjsip/pjsip_configuration.c,
+ configs/samples/pjsip.conf.sample, res/res_pjsip.c: res_pjsip:
+ Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.
+ During the latest update to DTLS-SRTP support the ability to
+ configure the hash used for fingerprints was added. This gave us
+ two supported ones: SHA-1 and SHA-256. The default was
+ accordingly updated to SHA-256. Unfortunately this configuration
+ ability was not exposed within res_pjsip. This change adds a
+ dtls_fingerprint option that controls it. #SIPit31 ........
+ Merged revisions 424290 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Accept DTLS
+ attributes in top level, not just media session. #SIPit31
+ ........ Merged revisions 424287 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-01 12:27 +0000 [r424245-424266] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_pjsip/config_transport.c, /, res/res_pjsip/location.c,
+ res/res_pjsip_endpoint_identifier_ip.c,
+ res/res_pjsip/pjsip_configuration.c,
+ configs/samples/pjsip.conf.sample: PJSIP: Handle defaults
+ properly This updates the code behind PJSIP configuration options
+ with custom handlers to deal with the assigned default values
+ properly where it makes sense and adjusting the default value
+ where it doesn't. Before applying this patch, there were several
+ cases where the default value for an option would prevent that
+ config section from loading properly. Reported by: Thomas
+ Thompson Review: https://reviewboard.asterisk.org/r/4019/
+ ........ Merged revisions 424263 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, res/res_pjsip_nat.c: PJSIP: Force transport on contact rewrite
+ If contact rewriting is enabled but the contact differs in
+ transport from what is actually being used, messages after the
+ initial INVITE transaction can be sent to an incorrect
+ transport/port combination. In the case where this bug occurred
+ the remote party never received a BYE since it was sent to the
+ remote party's TCP port over UDP. Review:
+ https://reviewboard.asterisk.org/r/4032/ ........ Merged
+ revisions 424244 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-10-01 10:09 +0000 [r424179-424184] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: chan_sip: Simplify some unref code by
+ removing unlink_peer_from_tables. ASTERISK-22945 #related
+ Reported by: ibercom Patches:
+ asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License
+ #6599) ........ Merged revisions 424181 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 424182 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 424183 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, channels/chan_sip.c: chan_sip: Remove excess ref of realtime
+ peer before sip_poke_peer. The peer is referenced at the end of
+ sip_poke_peer, it should not get an extra ref before the call to
+ sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close
+ Reported by: ibercom Tested by: Yuriy Gorlichenko Patches:
+ asterisk11.patch uploaded by ibercom (License #6599) Review:
+ https://reviewboard.asterisk.org/r/4031/ ........ Merged
+ revisions 424176 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 424177 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 424178 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-30 11:40 +0000 [r424153-424156] Joshua Colp <jcolp at digium.com>
+
+ * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Don't place an
+ extra whitespace before 'rport' and don't put IPv6 addresses in
+ brackets. #SIPit31 ........ Merged revisions 424155 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the base
+ and mapped address for candidates is present in SDP. This change
+ fixes an issue where ICE candidates put into the SDP did not
+ contain the 'raddr' and 'rport' information for server reflexive
+ and relay candidates. #SIPit31 ........ Merged revisions 424151
+ from http://svn.asterisk.org/svn/asterisk/branches/11 ........
+ Merged revisions 424152 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-29 21:59 +0000 [r424129] George Joseph <george.joseph at fairview5.com>
+
+ * /, res/res_pjsip/pjsip_cli.c: pjsip_cli: Suppress header print on
+ error or no objects If there's an error on the pjsip command line
+ or there are no objects, don't print the column headers.
+ ASTERISK-24350 #close Reported-by: Brad Latus Tested-by: George
+ Joseph Tested-by: Brad Latus Review:
+ https://reviewboard.asterisk.org/r/4025/ ........ Merged
+ revisions 424128 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-29 21:26 +0000 [r424126] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, contrib/scripts/autosupport: autosupport: Fix bashism. '==' is
+ bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
+ 'case' works better there. Originally committed in r375059 and
+ r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported
+ by: Tzafrir Cohen ........ Merged revisions 424117 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 424125 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-29 21:17 +0000 [r424097-424105] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c,
+ /, res/res_pjsip_authenticator_digest.c: Simplify UUID generation
+ in several places. Replace code using ast_uuid_generate() with
+ simpler and faster code using ast_uuid_generate_str(). The new
+ code avoids a malloc(), free(), and copy. ........ Merged
+ revisions 424103 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, main/threadpool.c: threadpool.c: Minor cleanup fixes. * Fix
+ threadpool_alloc() prototype. * Add missing off-nominal NULL
+ check of pool in threadpool_alloc(). * searializer_create() does
+ not need to create the object with a lock as the lock is not
+ used. ........ Merged revisions 424096 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-27 12:43 +0000 [r424057] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_pjsip.c, res/res_pjsip_session.c, /:
+ res_pjsip_session: Add additional checks for delaying session
+ refreshes. There are certain situations which no checks existed
+ for which need to prevent session refreshes. This includes
+ sending a session refresh with SDP before SDP negotiation has
+ completed and sending a session refresh before the dialog itself
+ has been established. Checks for these have been added.
+ Additionally COLP related UPDATEs were including SDP when it is
+ not needed. Review: https://reviewboard.asterisk.org/r/4008/
+ ........ Merged revisions 424056 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-26 15:21 +0000 [r423992] Richard Mudgett <rmudgett at digium.com>
+
+ * /, res/res_fax.c: res_fax: Fix out of bounds error in
+ update_modem_bits(). ASTERISK-24357 #close Reported by: Jeremy
+ Laine Patches: res_fax_bounds.patch (license #6561) patch
+ uploaded by Jeremy Laine Modified patch to not use magic numbers.
+ ........ Merged revisions 423979 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 423983 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 423987 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-26 08:25 +0000 [r423918] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, doc/asterisk.8: docs: Escape unescaped minus sign in
+ asterisk.8 manpage. ASTERISK-23768 #close Reported by: Jeremy
+ Lainé Patches: escape_manpage_hyphen.patch uploaded by Jeremy
+ Lainé (License #6561) ........ Merged revisions 423915 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 423916 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 423917 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-25 21:01 +0000 [r423895] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip.c, /: res_pjsip.c: Add missing off nominal cleanup
+ in ast_sip_push_task_synchronous(). * Made memset the std struct
+ in ast_sip_push_task_synchronous() because if DEBUG_THREADS is
+ enabled then uninitialized lock tracking data is used. ........
+ Merged revisions 423894 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-24 18:32 +0000 [r423867] Richard Mudgett <rmudgett at digium.com>
+
+ * /, res/res_pjsip/pjsip_options.c, res/res_pjsip.c:
+ pjsip_options.c: Fix race condition stopping periodic out of
+ dialog OPTIONS request. The crash on the issues is a result of an
+ invalid transport configuration change when asterisk is
+ restarted. The attempt to send the qualify request fails and we
+ cleaned up. However, the callback is also called which results in
+ a double unref of the objects involved. * Put a wrapper around
+ pjsip_endpt_send_request() to detect when the passed in callback
+ is called because of an error so callers can know to not cleanup.
+ * Made send_request_cb() able to handle repeated challenges (Up
+ to 10). * Fix periodic endpoint qualify OPTIONS sched deletion
+ race by avoiding it. The sched entry will no longer self stop and
+ must be externally stopped. * Added REF_DEBUG description tags to
+ struct sched_data in pjsip_options.c. * Fix some off-nominal ref
+ leaks in schedule_qualify(), qualify_and_schedule(). * Reordered
+ pjsip_options.c module start/stop code to cleanup better on
+ error. ASTERISK-24295 #close Reported by: Rogger Padilla Review:
+ https://reviewboard.asterisk.org/r/3954/ ........ Merged
+ revisions 423866 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-24 08:53 +0000 [r423803] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: chan_sip: Unref outbound proxy structure
+ on dialog/pvt destruction. Make sure outbound proxy refs are
+ always unreffed on dialog destruction. Review:
+ https://reviewboard.asterisk.org/r/4016/ ........ Merged
+ revisions 423800 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 423801 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 423802 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-23 14:29 +0000 [r423783] Mark Michelson <mmichelson at digium.com>
+
+ * tests/test_cel.c, tests/test_cdr.c: Make CDR and CEL unit tests
+ less FRACKy. Prior to this commit, CDR and CEL tests were
+ expected to trigger FRACKs (i.e. assertions) due to the fact that
+ the channels they create have no formats on them. Some code was
+ independently added recently that attempts to prevent FRACKs from
+ occurring by failing early when attempting to set up translation
+ paths if one or both channels support no formats. Unfortunately,
+ this attempt to be helpful made the CDR and CEL tests go from
+ simply FRACKing to outright failing and in some cases, failing so
+ badly as to crash Asterisk. This commit seeks to correct past
+ mistakes by adding the ulaw format to channels created by the CDR
+ and CEL unit tests. This makes setting up translation paths
+ succeed, eliminates previously-seen FRACKs, and ultimately causes
+ the unit tests to succeed again. Review:
+ https://reviewboard.asterisk.org/r/4014
+
+ 2014-09-22 19:48 +0000 [r423660-423723] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: chan_sip: On INVITE retransmission, don't
+ add an extra 503 response. INVITE arrives to asterisk, asterisk
+ responds Busy(). If the INVITE is retransmitted, asterisk would
+ generate a 503 in addition to the 486. Thanks Torrey Searle for
+ providing a working regression test. ASTERISK-24335 #close
+ Review: https://reviewboard.asterisk.org/r/4003/ Patches:
+ retrans_486_invite.patch uploaded by Torrey Searle (License
+ #5334) ........ Merged revisions 423720 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 423721 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 423722 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * /, main/editline/readline.c: cli.c: Fix tab completion "module
+ load" when MALLOC_DEBUG is enabled. r421600 conflicted with
+ r155763. ASTERISK-24348 #close ........ Merged revisions 423657
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 423658 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 423659 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-21 01:15 +0000 [r423618-423641] Matthew Jordan <mjordan at digium.com>
+
+ * main/channel.c: main/channel: Unlock channel in off-nominal path
+ In r423414 (13) / r423415 (trunk), an API call that determines if
+ a format capability structure is empty was added. This returns
+ true if the format capability structure is completely empty or
+ "none". A check for this was added in channel.c's set_format
+ call. Unfortunately, when this check was true, it returned from
+ the function while still holding the channel lock. This caused
+ the CDR unit tests - which have a tendency to create channels
+ with no formats - to deadlock. Whoops. This patch unlocks the
+ channel on the off-nominal path.
+
+ * rest-api/api-docs/events.json, /: rest-api/api-docs/events.json:
+ Remove non-compliant 'extends' attribute Prior to the release of
+ Swagger 1.2, the attribute 'extends' was being promoted as a
+ possible way to show that a particular object extends an existing
+ object. Instead, the Swagger specification went with the
+ 'subTypes' attribute in the base object. This patch removes the
+ unsupported attribute; the object that the offending objects
+ proposed to extend already lists them in its 'subTypes'
+ attribute. ASTERISK-24300 #close Reported by: Bradley Watkins
+ ........ Merged revisions 423620 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
+ rest-api/api-docs/bridges.json,
+ rest-api/api-docs/recordings.json,
+ rest-api/api-docs/deviceStates.json,
+ rest-api/api-docs/endpoints.json,
+ rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
+ /, rest-api/api-docs/asterisk.json,
+ rest-api/api-docs/applications.json,
+ rest-api/api-docs/playbacks.json: rest-api/api-docs: Correct
+ basePath in resources to match top resources file The
+ resources.json file that defines the resource JSON files used
+ with ARI references a basePath of 'http://localhost:8088/ari'.
+ This does not match what is defined in the resource files
+ themselves, 'http://localhost:8088/stasis'. The correct base path
+ is the one that includes 'ari' in the URL; this patch updates the
+ various resource JSON files to have the correct basePath.
+ ASTERISK-24339 #close Reported by: Bradley Watkins ........
+ Merged revisions 423617 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-19 19:51 +0000 [r423580] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_pjsip_notify.c: res_pjsip_notify: Fix crash on
+ unload/load and don't say the module doesn't exist on reload.
+ When unloading the module did not unregister the CLI commands
+ causing a crash upon load when they were registered again. When
+ reloading the module the return value from the config options
+ framework was not checked to determine if an error occurred or
+ not. This caused a message to be output saying the module did not
+ exist when reloading if no changes were present. AST-1433 #close
+ AST-1434 #close ........ Merged revisions 423579 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-19 17:08 +0000 [r423561] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_pjsip.c, res/res_pjsip_sdp_rtp.c:
+ res_pjsip_sdp_rtp.c: Fix native formats containing formats that
+ were not negotiated. Outgoing PJSIP calls can result in
+ non-negotiated formats listed in the channel's native formats if
+ video formats are listed in the endpoint's configuration. The
+ resulting call could then use a non-negotiated format resulting
+ in one way audio. * Simplified the update of session->req_caps in
+ set_caps(). Why do something in five steps when only one is
+ needed? AFS-162 #close Review:
+ https://reviewboard.asterisk.org/r/4000/
+
+ 2014-09-19 15:18 +0000 [r423524-423530] Jonathan Rose <jrose at digium.com>
+
+ * /, main/stasis_channels.c: Stasis_channels: Resolve unfinished
+ Dials when doing masquerades Masquerades into channels that are
+ in the dialing state don't end their dial and this goes against
+ the model for things like CDRs and generating Dial end manager
+ actions and such. ASTERISK-24237 #close Reported by: Richard
+ Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........
+ Merged revisions 423525 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * channels/chan_iax2.c: chan_iax2: Fix a crash when using chan_iax2
+ jitterbuffer settings Caused by format changes in Asterisk 13
+ ASTERISK-24265 #close Reported by: Dafi Ni Review:
+ https://reviewboard.asterisk.org/r/3999/
+
+ 2014-09-19 12:45 +0000 [r423504] Kinsey Moore <kmoore at digium.com>
+
+ * include/asterisk/framehook.h, /, main/framehook.c,
+ res/res_pjsip_t38.c: PJSIP: Prevent T38 framehook being put on
+ wrong channel This change gives framehooks a reverse-direction
+ masquerade callback in addition to chan_fixup_cb similar to the
+ callback added to datastores to handle the same situation. The
+ new callback provides the same parameters as the fixup callback,
+ but is called on the new channel's framehooks before moving
+ framehooks from the old channel to the new channel. This gives
+ the framehooks an oppurtunity to decide whether they should
+ remain on the new channel or be removed. This new callback is
+ used to prevent the PJSIP T.38 framehook from remaining on a
+ masqueraded channel if the new channel is not also a PJSIP
+ channel. This was causing a crash when a local channel was
+ masqueraded into a PJSIP channel and the framehook was executed
+ on the local channel since the channel's tech private data was
+ not structured as expected. Review:
+ https://reviewboard.asterisk.org/r/4001/ ........ Merged
+ revisions 423503 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-18 19:30 +0000 [r423482] Sean Bright <sean at malleable.com>
+
+ * res/res_pjsip/config_auth.c, /: res_pjsip: Don't require a
+ password when doing userpass authentication. An empty password is
+ valid for username/password authentication so we should allow
+ password to be empty/not supplied. Review:
+ https://reviewboard.asterisk.org/r/3988 ........ Merged revisions
+ 423481 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-18 19:22 +0000 [r423478] George Joseph <george.joseph at fairview5.com>
+
+ * tests/test_strings.c, /, main/utils.c,
+ include/asterisk/strings.h: utils: Create ast_strsep function
+ that ignores separators inside quotes This function acts like
+ strsep with three exceptions... * The separator is a single
+ character instead of a string. * Separators inside quotes are
+ treated literally instead of like separators. * You can elect to
+ have leading and trailing whitespace and quotes stripped from the
+ result and have '\' sequences unescaped. Like strsep, ast_strsep
+ maintains no internal state and you can call it recursively using
+ different separators on the same storage. Also like strsep, for
+ consistent results, consecutive separators are not collapsed so
+ you may get an empty string as a valid result. Tested by: George
+ Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........
+ Merged revisions 423476 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-18 18:31 +0000 [r423462] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_pjsip_pubsub.c: Add subscription state test events. These
+ are needed for a set of batched notification RLS tests that are
+ about to be committed to the testsuite. Review:
+ https://reviewboard.asterisk.org/r/3967
+
+ 2014-09-18 17:11 +0000 [r423425] Jonathan Rose <jrose at digium.com>
+
+ * res/res_pjsip_endpoint_identifier_ip.c, /:
+ res_pjsip_endpoint_identifier_ip: Fix parsing of match value with
+ CIDR Also fixes comma separates match lists ASTERISK-24290 #close
+ Reported by: Ray Crumrine Review:
+ https://reviewboard.asterisk.org/r/3995/ ........ Merged
+ revisions 423417 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ 2014-09-18 17:09 +0000 [r423418-423423] Richard Mudgett <rmudgett at digium.com>
+
+ * bridges/bridge_softmix.c: bridge_softmix.c: Made use
+ ao2_replace() instead of the inline equivalent. * Clarified some
+ read/write format comments. * Fixed a doxygen tag typo.
+
+ * main/astobj2.c, contrib/scripts/refcounter.py, /:
++>>>>>>> upstream/13.0.0_dfsg
astobj2.c/refcounter.py: Fix to deal with invalid object refs. *
Make astob2 REF_DEBUG output an invalid object line when an
invalid ao2 object ref/unref is attempted. This is similar to the
--
Alioth's /usr/local/bin/git-commit-notice on /srv/git.debian.org/git/pkg-voip/asterisk.git
More information about the Pkg-voip-commits
mailing list