Importing release summary for 13.1-cert1 release.

git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/tags/13.1-cert1@431520 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.version

@@ -1 +1 @@
-13.1.0
+13.1-cert1

ChangeLog

@@ -1,3 +1,809 @@
+2015-01-30  Asterisk Development Team <asteriskteam@digium.com>
+
+	* Certified Asterisk 13.1-cert1 Released.
+
+2015-01-30 17:53 +0000 [r431494]  Richard Mudgett <rmudgett@digium.com>
+
+	* apps/app_agent_pool.c, /: app_agent_pool: Fix initial module load
+	  agent device state reporting. When the app_agent_pool module
+	  initially loads there is a race condition between the thread
+	  loading agents.conf and the device state internal processing
+	  thread. If the device state internal processing thread handles
+	  the agent creation state updates before the thread that loaded
+	  agents.conf registers the device state provider callback then the
+	  cached agent state is "Invalid". When a consumer module like
+	  app_queue asks for the agent state it gets the cached "Invalid"
+	  state instead of the real state from the provider. * Moved
+	  loading the agents.conf configuration to the last thing setup by
+	  app_agent_pool in load_module(). Now the device state provider
+	  callback is registered before the config is loaded so the agent
+	  creation state updates are guaranteed to get the initial device
+	  state. * Removed some now redundant config cleanup on error in
+	  load_config(). * Added lock protection when accessing the device
+	  state in agent_pvt_devstate_get() and eliminated the RAII_VAR()
+	  usage. ASTERISK-24737 #close Reported by: Steve Pitts Review:
+	  https://reviewboard.asterisk.org/r/4390/ ........ Merged
+	  revisions 431492 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-30 16:50 +0000 [r431470]  Mark Michelson <mmichelson@digium.com>
+
+	* main/stasis_channels.c, channels/chan_pjsip.c, main/xmldoc.c,
+	  res/res_pjsip_refer.c, main/pbx.c, main/manager.c,
+	  pbx/pbx_spool.c, /, main/bridge_after.c: Fix some memory leaks.
+	  These memory leaks were found and fixed by John Hardin. I'm just
+	  committing them for him. ASTERISK-24736 #close Reported by Mark
+	  Michelson Review: https://reviewboard.asterisk.org/r/4389
+	  ........ Merged revisions 431468 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-30 16:41 +0000 [r431467]  Jonathan Rose <jrose@digium.com>
+
+	* main/manager.c, /: Merge r431153 from asterisk/branches/13
+	  r431153 | jrose | 2015-01-27 11:22:52 -0600 (Tue, 27 Jan 2015) |
+	  9 lines Manager: Fix Manager Action ModuleLoad to give correct
+	  response when reloading Prior to this patch, ModuleLoad would
+	  respond with an error indicating that the requested module wasn't
+	  found in spite of finding and reloading the module. Review:
+	  https://reviewboard.asterisk.org/r/4373/ ASTERISK-24721 #close
+
+2015-01-28 21:53 +0000 [r431326-431334]  Mark Michelson <mmichelson@digium.com>
+
+	* funcs/func_curl.c, /: Multiple revisions 431297-431298 ........
+	  r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan
+	  2015) | 17 lines Mitigate possible HTTP injection attacks using
+	  CURL() function in Asterisk. CVE-2014-8150 disclosed a
+	  vulnerability in libcURL where HTTP request injection can be
+	  performed given properly-crafted URLs. Since Asterisk makes use
+	  of libcURL, and it is possible that users of Asterisk may get
+	  cURL URLs from user input or remote sources, we have made a patch
+	  to Asterisk to prevent such HTTP injection attacks from
+	  originating from Asterisk. ASTERISK-24676 #close Reported by Matt
+	  Jordan Review: https://reviewboard.asterisk.org/r/4364
+	  AST-2015-002 ........ r431298 | mmichelson | 2015-01-28 11:12:49
+	  -0600 (Wed, 28 Jan 2015) | 3 lines Fix compilation error from
+	  previous patch. ........ Merged revisions 431297-431298 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 431299 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+	  revisions 431301 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* res/res_pjsip_t38.c, res/res_pjsip_session.c, /,
+	  res/res_pjsip_sdp_rtp.c: Fix file descriptor leak in RTP code.
+	  SIP requests that offered codecs incompatible with configured
+	  values could result in the allocation of RTP and RTCP ports that
+	  would not get reclaimed later. ASTERISK-24666 #close Reported by
+	  Y Ateya Review: https://reviewboard.asterisk.org/r/4323
+	  AST-2015-001 ........ Merged revisions 431300 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+	  revisions 431303 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-28 04:11 +0000 [r431244]  Richard Mudgett <rmudgett@digium.com>
+
+	* /, res/res_pjsip_outbound_registration.c, res/res_pjsip.c,
+	  main/sorcery.c: res_pjsip_outbound_registration: Fix reload race
+	  condition. Performing a CLI "module reload" command when there
+	  are new pjsip.conf registration objects defined frequently failed
+	  to load them correctly. What happens is a race condition between
+	  res_pjsip pushing its reload into an asynchronous task processor
+	  task and the thread that does the rest of the reloads when it
+	  gets to reloading the res_pjsip_outbound_registration module. A
+	  similar race condition happens between a reload and the CLI/AMI
+	  show registrations commands. The reload updates the
+	  current_states container and the CLI/AMI commands call
+	  get_registrations() which builds a new current_states container.
+	  * Made res_pjsip.c reload_module() use
+	  ast_sip_push_task_synchronous() instead of ast_sip_push_task() to
+	  eliminate two threads processing config reloads at the same time.
+	  * Made get_registrations() not replace the global current_states
+	  container so the CLI/AMI show registrations command cannot
+	  interfere with reloading. You could never add/remove objects in
+	  the container without the possibility of the container being
+	  replaced out from under you by get_registrations(). * Added a
+	  registration loaded sorcery instance observer to purge any dead
+	  registration objects since get_registrations() cannot do this job
+	  anymore. The struct ast_sorcery_instance_observer callbacks must
+	  be used because the callback happens inline with the load
+	  process. The struct ast_sorcery_observer callbacks are pushed to
+	  a different thread. * Added some global current_states NULL
+	  pointer checks in case the container disappears because of
+	  unload_module(). * Made sorcery's struct
+	  ast_sorcery_instance_observer.object_type_loaded callbacks
+	  guaranteed to be called before any struct
+	  ast_sorcery_observer.loaded callbacks will be called. * Moved the
+	  check for non-reloadable objects to before the sorcery instance
+	  loading callbacks happen to short circuit unnecessary work.
+	  Previously with non-reloadable objects, the sorcery instance
+	  loading/loaded callbacks would always happen, the individual
+	  wizard loading/loaded would be prevented, and the non-reloadable
+	  type logging message would be logged for each associated wizard.
+	  ASTERISK-24729 #close Review:
+	  https://reviewboard.asterisk.org/r/4381/ ........ Merged
+	  revisions 431243 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 23:02 +0000 [r431200-431221]  Kevin Harwell <kharwell@digium.com>
+
+	* main/tcptls.c, /: tcptls: Bad file descriptor error when
+	  reloading chan_sip While running through some scenarios using
+	  chan_sip and tcp a problem would occur that resulted in a flood
+	  of bad file descriptor messages on the cli: tcptls.c:712
+	  ast_tcptls_server_root: Accept failed: Bad file descriptor The
+	  message is received because the underlying socket has been
+	  closed, so is valid. This is probably happening because unloading
+	  of chan_sip is not atomic. That however is outside the scope of
+	  this patch. This patch simply stops the logging of multiple
+	  occurrences of that message. ASTERISK-24728 #close Reported by:
+	  Thomas Thompson Review: https://reviewboard.asterisk.org/r/4380/
+	  ........ Merged revisions 431218 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 431219 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* /, channels/chan_sip.c: chan_sip: stale nonce causes failure When
+	  refreshing (with a small expiration) a registration that was sent
+	  to chan_sip the nonce would be considered stale and reject the
+	  registration. What was happening was that the initial
+	  registration's "dialog" still existed in the dialogs container
+	  and upon refresh the dialog match algorithm would choose that as
+	  the "dialog" instead of the newly created one. This occurred
+	  because the algorithm did not check to see if the from tag
+	  matched if authentication info was available after the 401. So,
+	  it ended up assuming the original "dialog" was a match and
+	  stopped the search. The old "dialog" of course had an old nonce,
+	  thus the stale nonce message. This fix attempts to leave the
+	  original functionality alone except in the case of a REGISTER. If
+	  a REGISTER is received if searches for an existing "dialog"
+	  matching only on the callid. If the expires value is low enough
+	  it will reuse dialog that is there, otherwise it will create a
+	  new one. ASTERISK-24715 #close Reported by: John Bigelow Review:
+	  https://reviewboard.asterisk.org/r/4367/ ........ Merged
+	  revisions 431187 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 431194 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 17:52 +0000 [r431162]  Richard Mudgett <rmudgett@digium.com>
+
+	* /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c:
+	  app_confbridge: Repeatedly starting and stopping recording ref
+	  leaks the recording channel. Starting and stopping conference
+	  recording more than once causes the recording channels to be
+	  leaked. For v13 the channels also show up in the CLI "core show
+	  channels" output. * Reworked and simplified the recording channel
+	  code to use ast_bridge_impart() instead of managing the recording
+	  thread in the ConfBridge code. The recording channel's ref
+	  handling easily falls into place and other off nominal code paths
+	  get handled better as a result. ASTERISK-24719 #close Reported
+	  by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/
+	  Review: https://reviewboard.asterisk.org/r/4369/ ........ Merged
+	  revisions 431135 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 431160 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 17:35 +0000 [r431159]  Joshua Colp <jcolp@digium.com>
+
+	* res/res_pjsip_sdp_rtp.c, main/bridge_channel.c, /: bridge /
+	  res_pjsip_sdp_rtp: Fix issues with media not being reinvited
+	  during direct media. This change fixes two issues: 1. During a
+	  swap operation bridging added the new channel before having the
+	  swap channel leave. This was not handled in bridge_native_rtp and
+	  could result in a channel not getting reinvited back to Asterisk.
+	  After this change the swap channel will leave first and the new
+	  channel will then join. 2. If a re-invite was received after a
+	  session had been established any upstream elements (such as
+	  bridge_native_rtp) were not notified that they may want to
+	  re-evaluate things. After this change an UPDATE_RTP_PEER control
+	  frame is queued when this situation occurs and upstream can
+	  react. AST-1524 #close Review:
+	  https://reviewboard.asterisk.org/r/4378/ ........ Merged
+	  revisions 431157 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 17:18 +0000 [r431140]  Matthew Jordan <mjordan@digium.com>
+
+	* /, apps/confbridge/include/confbridge.h,
+	  apps/confbridge/conf_config_parser.c: app_confbridge: Restore
+	  user's menu name to CLI output of 'confbridge list' When issuing
+	  a 'confbridge list XXXX' CLI command, the resulting output no
+	  longer displays the menu associated with a ConfBridge
+	  participant. The issue was caused by ASTERISK-22760. When that
+	  patch was done, it removed the copying of the menu name
+	  associated with the user from the actual user profile. This patch
+	  fixes the issue by copying the menu name over to the user profile
+	  when the menu hooks are applied to the user. Since that function
+	  now does a little bit more than just apply the hooks, the name of
+	  the function has been changed to cover the copying of the menu
+	  name over as well. In addition, there is a disparity between the
+	  menu name length as it is stored on the conf_menu structure and
+	  the confbridge_user structure; this patch makes the lengths match
+	  so that a strcpy can be used. Review:
+	  https://reviewboard.asterisk.org/r/4372/ ASTERISK-24723 #close
+	  Reported by: Steve Pitts ........ Merged revisions 431134 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-27 11:48 +0000 [r431116]  Joshua Colp <jcolp@digium.com>
+
+	* res/parking/parking_manager.c, /: res_parking: Fix crash due to
+	  race condition when unloading. There is currently a race
+	  condition when unloading the res_parking module. Depending on the
+	  will of the universe the subscription invocation may occur AFTER
+	  the module is unloaded. This is because the module does NOT use
+	  stasis_unsubscribe_and_join when terminating the subscription. It
+	  merely uses stasis_unsubscribe. This change makes it use
+	  stasis_unsubscribe_and_join which is documented for usage in this
+	  exact scenario. AST-1520 #close Review:
+	  https://reviewboard.asterisk.org/r/4375/ ........ Merged
+	  revisions 431114 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-23 15:24 +0000 [r431016]  Kevin Harwell <kharwell@digium.com>
+
+	* res/res_ari_events.c, include/asterisk/stasis_app.h,
+	  res/res_pjsip_mwi.c, res/parking/parking_applications.c,
+	  channels/chan_iax2.c, res/res_pjsip/pjsip_global_headers.c,
+	  res/res_pjsip_pubsub.c, res/res_ari_channels.c, res/res_stasis.c,
+	  rest-api-templates/param_parsing.mustache, /,
+	  res/res_ari_endpoints.c: Investigate and fix memory leaks in
+	  Asterisk Fixed memory leaks that were found in Asterisk.
+	  ASTERISK-24693 #close Reported by: Kevin Harwell Review:
+	  https://reviewboard.asterisk.org/r/4347/ ........ Merged
+	  revisions 430999 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-21 19:47 +0000 [r430898]  Richard Mudgett <rmudgett@digium.com>
+
+	* CHANGES, /, res/res_pjsip_outbound_registration.c: Multiple
+	  revisions 430223,430373,430395 ........ r430223 | gtjoseph |
+	  2015-01-06 11:35:21 -0600 (Tue, 06 Jan 2015) | 24 lines
+	  outbound_registration: Add 'pjsip send register' and update 'send
+	  unregister' The current behavior of 'pjsip send unregister' is to
+	  send the unregister (REGISTER with 0 exp) but let the next
+	  scheduled register proceed normally. I don't think that's a good
+	  idea. If you unregister, it should stay unregistered until you
+	  decide to start registrations again. So this patch just adds a
+	  cancel_registration call to the current unregister_task to cancel
+	  the timer. Of course, now you need a way to start registration
+	  again so I've added a 'pjsip send register' command that
+	  unregisters and cancels any existing registration (the same as
+	  send unregister), then sends an immediate registration and starts
+	  the timer back up again. Both changes also ripple to AMI. There's
+	  a new PJSIPRegister command. There's no harm in calling either
+	  command repeatedly. They don't care about the actual state.
+	  Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/4301/ ........ r430373 |
+	  gtjoseph | 2015-01-08 11:48:29 -0600 (Thu, 08 Jan 2015) | 25
+	  lines res_pjsip_outbound_registration: Fix several reload issues
+	  There are 2 issues with reloading registrations... 1. The
+	  'can_reuse_registration' test wasn't considering the intervals or
+	  expiration in its determination of whether a registration changed
+	  or not so if you changed any of the intervals or the expiration
+	  and reloaded, the object would get reloaded but the actual timers
+	  wouldn't change. can_reuse_registration now does a sorcery diff
+	  on the old and new objects instead of discretely testing certain
+	  fields. Now if you change expiration for instance, and reload,
+	  the timer is updated and re-registration will occur on the new
+	  value. 2. If you mung up your password on an outbound
+	  registration you get a permanent failure. If you fix the password
+	  (on the outbound_auth object) and reload, nothing tells
+	  outbound_registration to try again because the registration
+	  itself didn't change. This patch adds an observer on the "auth"
+	  object type and if any auth changes, existing registration states
+	  are searched and those in a REJECTED_PERMANENT state are retried.
+	  Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/4304/ ........ r430395 |
+	  gtjoseph | 2015-01-08 15:37:42 -0600 (Thu, 08 Jan 2015) | 14
+	  lines res_pjsip_outbound_registration: Fix reference leak. Every
+	  time a registration started,
+	  sip_outbound_registration_response_cb bumps the ref count on
+	  client_state then pushes a handle_registration_response task.
+	  handle_registration_response never unreffed it though. So every
+	  time a registration goes out, the ref count goes up by one. This
+	  patch adds the unreffs to handle_registration_response.
+	  Tested-by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/4303/ ........ Merged
+	  revisions 430223,430373,430395 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-21 13:36 +0000 [r430843-430865]  Matthew Jordan <mjordan@digium.com>
+
+	* /, channels/chan_sip.c: channels/chan_sip: Fix registration leak
+	  during reload When the SIP registrations were migrated to using
+	  ao2 in what was then trunk, the explicit destruction of the
+	  registrations on module reload was removed and not replaced with
+	  an ao2 equivalent. Debugging done by Stefan Engström, the issue
+	  reporter, on ASTERISK-24673 confirmed that the reference in the
+	  registry_list container was being leaked. Since the purpose of
+	  cleanup_all_regs is to prep a registration for destruction, this
+	  function now calls an ao2_callback function callback with the
+	  OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the
+	  registrations. This cleans up each registration, and also removes
+	  it from the registration container registry_list. Review:
+	  https://reviewboard.asterisk.org/r/4355/ ASTERISK-24640 #close
+	  Reported by: Max Man ASTERISK-24673 #close Reported by: Stefan
+	  Engström Tested by: Stefan Engström ........ Merged revisions
+	  430864 from http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* apps/app_dial.c, /: apps/app_dial: Don't publish DialEnd twice on
+	  unexpected GoSub/Macro values The Dial application has some
+	  interesting options with the mid-call Macro (M) and GoSub (U)
+	  options. If the MACRO_RESULT/GOSUB_RESULT returns specific
+	  values, the Dial application will take some action upon the
+	  channels involved in the dial operation (such as hanging up a
+	  particular party, etc.) The Dial application ensures that a
+	  Stasis message is published in the event that
+	  MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial
+	  operation, so that there is a corresponding DialEnd event
+	  published in AMI/ARI for the DialBegin event that preceeded it. A
+	  bug exists where that same DialEnd event will be published on
+	  Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is
+	  not one that the Dial application cares about. This causes two
+	  DialEnd events to be published - one with the
+	  MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is
+	  all sorts of wrong. This patch fixes the bug by ensuring that we
+	  only publish a DialEnd message to Stasis if the Dial
+	  application's mid-call Macro/GoSub returns something that Dial
+	  cares about. Review: https://reviewboard.asterisk.org/r/4336
+	  ASTERISK-24682 #close Reported by: Matt Jordan ........ Merged
+	  revisions 430842 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-19 18:18 +0000 [r430782]  Mark Michelson <mmichelson@digium.com>
+
+	* main/pbx.c, /: Call extension state callbacks at hint creation.
+	  When a hint gets created, any subsequent device or presence state
+	  changes result in extension status events getting sent out to
+	  interested parties. However, at the time of hint creation, no
+	  such event gets sent out, so watchers of extension state are
+	  potentially left in the dark until the first state change after
+	  hint creation. Patch contributed by John Hardin (License #6512)
+	  ........ Merged revisions 430776 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-15 12:11 +0000 [r430666]  Joshua Colp <jcolp@digium.com>
+
+	* /, res/res_pjsip_outbound_registration.c:
+	  res_pjsip_outbound_registration: Fix race condition when
+	  reloading and listing registrations. Due to the split of outbound
+	  registration state from configuration it is possible during a
+	  reload for a "pjsip show registrations" CLI command to be
+	  executed which gets an older snapshot of the configuration. This
+	  configuration may include outbound registrations which have been
+	  removed due to a reload operation occurring at the same time. The
+	  code for printing the outbound registration did not take this
+	  into account but now it does. AST-1506 #close Review:
+	  https://reviewboard.asterisk.org/r/4338/ ........ Merged
+	  revisions 430664 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-07 03:29 +0000 [r430253-430293]  Matthew Jordan <mjordan@digium.com>
+
+	* utils/conf2ael.c, apps/app_waitforring.c, formats/format_vox.c,
+	  res/res_timing_pthread.c, pbx/pbx_ael.c,
+	  cel/cel_sqlite3_custom.c, res/res_hep_rtcp.c,
+	  formats/format_jpeg.c, apps/app_jack.c, apps/app_adsiprog.c,
+	  cdr/cdr_sqlite3_custom.c, res/res_snmp.c, channels/chan_sip.c,
+	  cel/cel_tds.c, apps/app_dictate.c, apps/app_festival.c,
+	  agi/eagi-test.c, res/res_hep_pjsip.c, apps/app_alarmreceiver.c,
+	  apps/app_image.c, channels/chan_console.c, apps/app_getcpeid.c,
+	  apps/app_talkdetect.c, channels/chan_oss.c,
+	  channels/chan_misdn.c, apps/app_mp3.c, channels/chan_alsa.c,
+	  pbx/pbx_dundi.c, channels/chan_nbs.c, utils/extconf.c,
+	  apps/app_zapateller.c, cel/cel_pgsql.c, res/res_config_pgsql.c,
+	  utils/muted.c, apps/app_test.c, utils/smsq.c,
+	  apps/app_morsecode.c, apps/app_ices.c, cdr/cdr_csv.c,
+	  channels/chan_phone.c, funcs/func_pitchshift.c,
+	  funcs/func_audiohookinherit.c,
+	  res/res_pjsip_phoneprov_provider.c, apps/app_minivm.c,
+	  res/res_statsd.c, apps/app_sms.c, res/res_config_ldap.c,
+	  utils/streamplayer.c, utils/check_expr.c, cel/cel_radius.c,
+	  apps/app_nbscat.c, res/res_hep.c, apps/app_waitforsilence.c,
+	  apps/app_dahdiras.c, pbx/pbx_lua.c, res/res_ael_share.c,
+	  cdr/cdr_radius.c, cdr/cdr_tds.c, utils/stereorize.c,
+	  apps/app_osplookup.c, channels/chan_skinny.c,
+	  funcs/func_frame_trace.c, apps/app_amd.c, pbx/pbx_realtime.c,
+	  apps/app_url.c, apps/app_externalivr.c, cdr/cdr_odbc.c,
+	  res/res_timing_kqueue.c, channels/chan_mgcp.c,
+	  channels/chan_unistim.c, res/res_phoneprov.c, utils/astman.c,
+	  cdr/cdr_pgsql.c, res/res_config_sqlite.c: Disable extended
+	  support modules
+
+	* /,
+	  contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py:
+	  contrib/ast-db-manage: Correct down_revision path for
+	  user_eq_phone When the user_eq_phone patch was backported to 13,
+	  it referenced the downward revision that the PJSIP optimistic
+	  encryption option also references. This creates a multi-path
+	  upgrade Exception when generating the SQL files. This patch
+	  corrects this in the 13 branch. Note that trunk, which already
+	  contained both of these features, is unaffected by this problem.
+	  ........ Merged revisions 430252 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2015-01-06 19:53 +0000 [r430245]  Scott Griepentrog <sgriepentrog@digium.com>
+
+	* main/bridge_basic.c, /: bridge: avoid leaking channel during
+	  blond transfer pt2 A blond transfer to a failed destination, when
+	  followed by a recall attempt, lead to a leak of the reference to
+	  the destination channel. In addition to correcting the regression
+	  on the previous attempt (r429826) this fixes the leak and two
+	  additional reference leaks on failures of bridge_import.
+	  ASTERISK-24513 #close Review:
+	  https://reviewboard.asterisk.org/r/4302/ ........ Merged
+	  revisions 430199 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+	  revisions 430200 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-24 15:27 +0000 [r430085-430094]  Matthew Jordan <mjordan@digium.com>
+
+	* res/res_agi.c, /: res/res_agi: Make Verbose message for 'stream
+	  file' match other playbacks The Verbose message displayed when a
+	  file is played back via 'stream file' was formatted differently
+	  than other playbacks: * It didn't include the channel name * It
+	  didn't include the channel language It does, however, include the
+	  playback offset as well as any escape digits. That information
+	  was kept; however, this patch updates the formatting to more
+	  closely match the Verbose messages displayed when a file is
+	  played back by 'control stream file', Playback, ControlPlayback,
+	  or any other file playback operation. ........ Merged revisions
+	  429519 from http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py
+	  (added), /, res/res_pjsip.c: res_pjsip: Backport missing commits
+	  for user_eq_phone This backports the following from trunk, which
+	  were missed: r427257 | file | 2014-11-04 16:31:16 -0600 (Tue, 04
+	  Nov 2014) | 2 lines res_pjsip: Allow + at the beginning of a
+	  phone number when user_eq_phone is enabled. r427259 | file |
+	  2014-11-04 16:51:32 -0600 (Tue, 04 Nov 2014) | 2 lines res_pjsip:
+	  Apply the 'user_eq_phone' setting to the To header as well. It
+	  also adds the Alembic script for the option. ASTERISK-24643
+	  ........ Merged revisions 430092 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* /, tests/test_stasis_channels.c: Stasis: Update unittest for
+	  channel snapshots This adjusts the unit test for channel
+	  snapshots to take the new language key into account. ........
+	  Merged revisions 429352 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* CHANGES, res/res_pjsip.c, include/asterisk/res_pjsip.h,
+	  res/res_pjsip_keepalive.c (added), res/res_pjsip/config_global.c,
+	  /, configs/samples/pjsip.conf.sample: res_pjsip_keepalive: Add
+	  runtime configurable keepalive module for connection-oriented
+	  transports. Note that this is backport from trunk of r425825.
+	  This change adds a module which is configurable using the
+	  keep_alive_interval setting in the global section that will send
+	  a CRLF keep alive to all active connection-oriented transports at
+	  the provided interval. This is useful because it can help keep
+	  connections open through NATs. This functionality also exists
+	  within PJSIP but can not be controlled at runtime and requires
+	  recompiling it. Review: https://reviewboard.asterisk.org/r/4084/
+	  ASTERISK-24644 #close ........ Merged revisions 430084 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* /, res/res_pjsip/pjsip_configuration.c,
+	  res/res_pjsip_caller_id.c, CHANGES, res/res_pjsip.c,
+	  include/asterisk/res_pjsip.h: res_pjsip: Add 'user_eq_phone'
+	  option to add a 'user=phone' parameter when applicable. Note that
+	  this is a backport of r425804 from trunk. This change adds a
+	  configuration option which adds a 'user=phone' parameter if the
+	  user portion of the request URI or the From URI is determined to
+	  be a number. Review: https://reviewboard.asterisk.org/r/4073/
+	  ASTERISK-24643 #close ........ Merged revisions 430083 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-22 21:22 +0000 [r430030-430046]  Richard Mudgett <rmudgett@digium.com>
+
+	* main/bridge_basic.c, /: DTMF atxfer: Setup recall channels as if
+	  the transferee initiated the call. After the initial DTMF atxfer
+	  call attempt to the transfer target fails to answer during a
+	  blonde transfer, the recall callback channels do not get setup
+	  with information from the initial transferrer channel. As a
+	  result, the recall callback to the transferrer does not have
+	  callid, channel variables, datastores, accountcode, peeraccount,
+	  COLP, and CLID setup. A similar situation happens with the recall
+	  callback to the transfer target but it is less visible. The
+	  recall callback to the transfer target does not have callid,
+	  channel variables, datastores, accountcode, peeraccount, and COLP
+	  setup. * Added missing information to the recall callback
+	  channels before initiating the call. callid, channel variables,
+	  datastores, accountcode, peeraccount, COLP, and CLID * Set callid
+	  of the transferrer channel on the DTMF atxfer controller thread
+	  attended_transfer_monitor_thread(). * Added missing channel
+	  unlocks and props unref to off nominal paths in
+	  attended_transfer_properties_alloc(). ASTERISK-23841 #close
+	  Reported by: Richard Mudgett Review:
+	  https://reviewboard.asterisk.org/r/4259/ ........ Merged
+	  revisions 430034 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* include/asterisk/_private.h, main/asterisk.c, /, main/logger.c:
+	  queue_log: Post QUEUESTART entry when Asterisk fully boots. The
+	  QUEUESTART log entry has historically acted like a fully booted
+	  event for the queue_log file. When the QUEUESTART entry was
+	  posted to the log was broken by the change made by
+	  ASTERISK-15863. * Made post the QUEUESTART queue_log entry when
+	  Asterisk fully boots. This restores the intent of that log entry
+	  and happens after realtime has had a chance to load. AST-1444
+	  #close Reported by: Denis Martinez Review:
+	  https://reviewboard.asterisk.org/r/4282/ ........ Merged
+	  revisions 430009 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 430010 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-22 18:35 +0000 [r430007-430008]  bebuild <bebuild@localhost>:
+
+	* /, res/res_pjsip/pjsip_options.c: Multiple revisions
+	  429128,429246 ........ r429128 | kmoore | 2014-12-09 08:00:50
+	  -0600 (Tue, 09 Dec 2014) | 12 lines PJSIP: Stagger outbound
+	  qualifies This change staggers initiation of outbound qualify
+	  (OPTIONS) attempts to reduce instantaneous server load and
+	  prevent network congestion. Review:
+	  https://reviewboard.asterisk.org/r/4246/ ASTERISK-24342 #close
+	  Reported by: Richard Mudgett ........ Merged revisions 429127
+	  from http://svn.asterisk.org/svn/asterisk/branches/12 ........
+	  r429246 | kmoore | 2014-12-10 07:14:56 -0600 (Wed, 10 Dec 2014) |
+	  8 lines PJSIP: Fix assert on initial mass qualify This fixes the
+	  MWI test regressions caused by r429127 and ensures that contacts
+	  have non-zero qualify_frequency before attempting scheduling.
+	  ........ Merged revisions 429245 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+	  revisions 429128,429246 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* main/manager.c, /: Prevent possible race condition on dual
+	  redirect of channels in the same bridge. The
+	  AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent
+	  bridges from prematurely acting on orphaned channels in bridges.
+	  The problem with the AMI redirect action was that it was setting
+	  this flag on channels based on the presence of a PBX, not whether
+	  the channel was in a bridge. Whether a channel has a PBX is
+	  irrelevant, so the condition has been altered to check if the
+	  channel is in a bridge. ASTERISK-24536 #close Reported by Niklas
+	  Larsson Review: https://reviewboard.asterisk.org/r/4268 ........
+	  Merged revisions 429741 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+2014-12-19 21:52 +0000 [r429855-429892]  bebuild <bebuild@localhost>:
+
+	* CHANGES, res/res_ari_channels.c, res/ari/resource_channels.h, /,
+	  rest-api/api-docs/channels.json, res/ari/resource_channels.c:
+	  ari: Add support for specifying an originator channel when
+	  originating. If an originator channel is specified when
+	  originating a channel the linked ID of it will be applied to the
+	  newly originated outgoing channel. This allows an association to
+	  be made between the two so it is known that the originator has
+	  dialed the originated channel. ASTERISK-24552 #close Reported by:
+	  Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/
+	  ........ Merged revisions 429153 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* res/ari/ari_model_validators.c, main/manager_channels.c,
+	  res/ari/ari_model_validators.h, /, main/stasis_channels.c,
+	  rest-api/api-docs/channels.json: ARI/AMI: Include language in
+	  standard channel snapshot output The channel "language" was
+	  already part of a channel snapshot, however is was not sent out
+	  over AMI or ARI. This patch makes it so the channel "language" is
+	  included in the appropriate AMI or ARI events. ASTERISK-24553
+	  #close Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/4245/ ........ Merged
+	  revisions 429204 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+	  revisions 429206 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* res/res_pjsip_session.c, /: res_pjsip_session: Fix issue where a
+	  declined media stream in a re-INVITE would fail SDP negotiation.
+	  In the past the SDP negotiation within res_pjsip_session was made
+	  more tolerant of certain situations. The only case where SDP
+	  negotiation will fail is when a major error occurs during
+	  negotiation. Receiving an already declined media stream is not
+	  considered a major error. When producing the local SDP the logic
+	  took this into account so on the initial INVITE the declined
+	  media stream did not cause an SDP negotiation failure.
+	  Unfortunately the logic for handling media streams with a handler
+	  did not mirror this logic and considered an already declined
+	  media stream an error and thus failed the SDP negotiation. This
+	  change makes the logic between both situations match so only
+	  under major errors will the SDP negotiation fail. ASTERISK-24607
+	  #close Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/4254/ ........ Merged
+	  revisions 429407 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* include/asterisk/format.h, main/format.c, /, main/codec.c: media:
+	  Fix crash when determining sample count of a frame during
+	  shutdown. When shutting down Asterisk the codecs are cleaned up.
+	  As a result anything attempting to get a codec based on ID or
+	  details will find that no codec exists. This currently occurs
+	  when determining the sample count of a frame. This code did not
+	  take this situation into account. This change fixes this by
+	  getting the codec directly from the format and eliminates the
+	  lookup. This is both faster and also provides a guarantee that
+	  the codec will exist and will be valid. ASTERISK-24604 #close
+	  Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/4260/ ........ Merged
+	  revisions 429497 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* /, res/res_pjsip_outbound_registration.c: Prevent potential
+	  infinite outbound authentication loops in registration. Prior to
+	  this patch, Asterisk would always respond to 401 responses to
+	  registration attempts by trying to provide a registration with
+	  authentication credentials. Even if subsequent attempts were
+	  rejected with 401 responses, Asterisk would continue this
+	  behavior. If authentication credentials were incorrect, this
+	  could continue forever. With this patch, we keep track of whether
+	  we have attempted authentication on an outbound registration
+	  attempt. If we already have, we don not try again until the next
+	  attempt. This prevents the infinite loop scenario. Review:
+	  https://reviewboard.asterisk.org/r/4273 ........ Merged revisions
+	  429761 from http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* res/res_pjsip_outbound_publish.c, /: res_pjsip_outbound_publish:
+	  stack overflow when using non-default sorcery wizard When using a
+	  non-default sorcery wizard (in this instance realtime) for
+	  outbound publishes Asterisk will crash after a stack overflow
+	  occurs due to the code infinitely recursing. The fix entails
+	  removing the outbound publish state dependency from the outbound
+	  publish sorcery object and instead keeping an in memory container
+	  that can be used to lookup the state when needed. ASTERISK-24514
+	  #close Reported by: Mark Michelson Review:
+	  https://reviewboard.asterisk.org/r/4178/ ........ Merged
+	  revisions 429175 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* /, res/res_pjsip_sdp_rtp.c: PJSIP: Allow use of 'inactive'
+	  streams for hold This allows use of the 'inactive' stream
+	  direction identifier to be used for hold where 'sendonly' is
+	  normally used. Some Seimens phones use 'inactive' and this change
+	  allows music on hold to operate properly. Review:
+	  https://reviewboard.asterisk.org/r/4252/ Reported by: Steve Pitts
+	  ........ Merged revisions 429432 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+	  revisions 429433 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* channels/chan_pjsip.c, res/res_pjsip_session.c,
+	  include/asterisk/res_pjsip_session.h, /,
+	  res/res_pjsip_session.exports.in: res_pjsip_session: Delay
+	  sending BYE if a re-INVITE transaction is in progress. Given the
+	  scenario where a PJSIP channel is in a native RTP bridge with
+	  direct media and the channel is then hung up the code will
+	  currently re-INVITE the channel back to Asterisk and send a BYE
+	  at the same time. Many SIP implementations dislike this greatly.
+	  This change makes it so that if a re-INVITE transaction is in
+	  progress the BYE is queued to occur after the completion of the
+	  transaction (be it through normal means or a timeout). Review:
+	  https://reviewboard.asterisk.org/r/4248/ ........ Merged
+	  revisions 429409 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* /, channels/chan_pjsip.c: chan_pjsip: Race between channel answer
+	  and bridge setup when using direct media When direct media is
+	  enabled and a pjsip channel is answered a race would occur
+	  between the handling of the answer and bridge setup. Sometimes
+	  the media negotiation would take place after the native bridge
+	  was setup. This resulted in a NULL media address, which in turn
+	  resulted in Asterisk using its address as the remote media
+	  address when sending a reinvite. This patch makes the chan_pjsip
+	  answer handler synchronous thus alleviating the race condition
+	  (the bridge won't start setting things up until after it
+	  returns). ASTERISK-24563 #close Reported by: Steve Pitts Review:
+	  https://reviewboard.asterisk.org/r/4257/ ........ Merged
+	  revisions 429477 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* main/rtp_engine.c, /, channels/chan_sip.c,
+	  include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c: Direct
+	  Media calls within private network sometimes get one way audio
+	  When endpoints with direct_media enabled, behind a firewall
+	  (Asterisk on a separate network) and were bridged sometimes
+	  Asterisk would send the ip address of the firewall in the sdp to
+	  one of the phones in the reinvite resulting in one way audio.
+	  When sending the reinvite Asterisk will retrieve the media
+	  address from the associated rtp instance, but if frames were
+	  being read this can be overwritten with another address (in this
+	  case the firewall's). This patch ensures that Asterisk uses the
+	  original device address when using direct media. ASTERISK-24563
+	  Reported by: Steve Pitts Review:
+	  https://reviewboard.asterisk.org/r/4216/ ........ Merged
+	  revisions 429195 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
+	  revisions 429196 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* channels/pjsip/dialplan_functions.c, /: Ensure the correct value
+	  is returned for CHANNEL(pjsip, secure) Prior to this patch, we
+	  were using the PJSIP dialog's secure flag to determine if a
+	  secure transport was being used. Unfortunately, the dialog's
+	  secure flag was only set if a SIPS URI were in use, as required
+	  by RFC 3261 sections 12.1.1 and 12.1.2. What we're interested in
+	  is not dialog security, but transport security. This code change
+	  switches to a model where we use the dialog's target URI to
+	  determine what transport would be used to communicate, and then
+	  check if that transport is secure. AST-1450 #close Reported by
+	  John Bigelow Review: https://reviewboard.asterisk.org/r/4277
+	  ........ Merged revisions 429739 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* channels/chan_dahdi.c, /: chan_dahdi: Don't ignore setvar when
+	  using configuration section scheme. When the configuration
+	  section scheme of chan_dahdi.conf is used (keyword dahdichan
+	  instead of channel) all setvar= options are completely ignored.
+	  No variable defined this way appears in the created DAHDI
+	  channels. * Move the clearing of setvar values to after the
+	  deferred processing of dahdichan. AST-1378 #close Reported by:
+	  Guenther Kelleter Patch by: Guenther Kelleter ........ Merged
+	  revisions 429825 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 429829 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* /, include/asterisk/lock.h, main/lock.c: DEBUG_THREADS: Fix
+	  regression and lock tracking initialization problems. This patch
+	  started with David Lee's patch at
+	  https://reviewboard.asterisk.org/r/2826/ and includes a
+	  regression fix introduced by the ASTERISK-22455 patch. The
+	  initialization of a mutex's lock tracking structure was not
+	  protected in a critical section. This is fine for any mutex that
+	  is explicitly initialized, but a static mutex may have its lock
+	  tracking double initialized if multiple threads attempt the first
+	  lock simultaneously. * Added a global mutex to properly serialize
+	  initialization of the lock tracking structure. The painful global
+	  lock can be mitigated by adding a double checked lock flag as
+	  discussed on the original review request. * Defer lock tracking
+	  initialization until first use. * Don't be "helpful" and
+	  initialize an uninitialized lock when DEBUG_THREADS is enabled.
+	  Debug code is not supposed to fix or change normal code behavior.
+	  We don't need a lock initialization race that would force a
+	  re-setup of lock tracking. Lock tracking already handles
+	  initialization on first use. * Properly handle allocation
+	  failures of the lock tracking structure. * No need to initialize
+	  tracking data in __ast_pthread_mutex_destroy() just to turn
+	  around and destroy it. The regression introduced by
+	  ASTERISK-22455 is the result of manipulating a pthread_mutex_t
+	  struct outside of the pthread library code. The pthread_mutex_t
+	  struct seems to have a global linked list pointer member that can
+	  get changed by other threads. Therefore, saving and restoring the
+	  contents of a pthread_mutex_t struct is a bad thing. Thanks to
+	  Thomas Airmont for finding this obscure regression. * Don't
+	  overwrite the struct ast_lock_track.reentr_mutex member to
+	  restore tracking data in __ast_cond_wait() and
+	  __ast_cond_timedwait(). The pthread_mutex_t struct must be
+	  treated as a read-only opaque variable. Miscellaneous other items
+	  fixed by this patch: * Match ast_suspend_lock_info() with
+	  ast_restore_lock_info() in __ast_cond_timedwait(). * Made some
+	  uninitialized lock sanity checks return EINVAL and try a
+	  DO_THREAD_CRASH. * Fix bad canlog initialization expressions.
+	  ASTERISK-24614 #close Reported by: Thomas Airmont Review:
+	  https://reviewboard.asterisk.org/r/4247/ Review:
+	  https://reviewboard.asterisk.org/r/2826/ ........ Merged
+	  revisions 429539 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 429540 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* /, res/res_pjsip_pubsub.c: Activate persistent subscriptions when
+	  they are recreated. Prior to this change, recreating persistent
+	  subscriptions would create the subscription but would not
+	  activate it. This led to subscriptions being listed in the "NULL"
+	  state by diagnostics and not sending NOTIFYs when expected.
+	  Review: https://reviewboard.asterisk.org/r/4261 ........ Merged
+	  revisions 429571 from
+	  http://svn.asterisk.org/svn/asterisk/branches/13
+
+	* /, asterisk-13.1.0-summary.html (removed),
+	  asterisk-13.1.0-summary.txt (removed): Update properties; remove
+	  old summaries
+
+	* / (added): Create Certified Asterisk 13.1 branch
+
 2014-12-15  Asterisk Development Team <asteriskteam@digium.com>
 
 	* Asterisk 13.1.0 Released.

certified-asterisk-13.1-cert1-summary.html

@@ -0,0 +1,410 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
+<html xmlns="http://www.w3.org/1999/xhtml">
+<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - certified-asterisk-13.1-cert1</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">certified-asterisk-13.1-cert1</h3>
+<h3 align="center">Date: 2015-01-30</h3>
+<h3 align="center">&lt;asteriskteam@digium.com&gt;</h3>
+<hr/>
+<h2 align="center">Table of Contents</h2>
+<ol>
+   <li><a href="#summary">Summary</a></li>
+   <li><a href="#contributors">Contributors</a></li>
+   <li><a href="#issues">Closed Issues</a></li>
+   <li><a href="#commits">Other Changes</a></li>
+   <li><a href="#diffstat">Diffstat</a></li>
+</ol>
+<hr/>
+<a name="summary"><h2 align="center">Summary</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This release includes new features.  For a list of new features that have been included with this release, please see the CHANGES file inside the source package.  Since this is new major release, users are encouraged to do extended testing before upgrading to this version in a production environment.</p>
+<p>The data in this summary reflects changes that have been made since the previous release, certified-asterisk-13.1.0.</p>
+<hr/>
+<a name="contributors"><h2 align="center">Contributors</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release.  For coders, the number is how many of their patches (of any size) were committed into this release.  For testers, the number is the number of times their name was listed as assisting with testing a patch.  Finally, for reporters, the number is the number of issues that they reported that were closed by commits that went into this release.</p>
+<table width="100%" border="0">
+<tr>
+<td width="33%"><h3>Coders</h3></td>
+<td width="33%"><h3>Testers</h3></td>
+<td width="33%"><h3>Reporters</h3></td>
+</tr>
+<tr valign="top">
+<td>
+16 bebuild<br/>
+11 mjordan<br/>
+5 rmudgett<br/>
+4 mmichelson<br/>
+3 kharwell<br/>
+1 jrose<br/>
+1 sgriepentrog<br/>
+</td>
+<td>
+2 Stefan Engström<br/>
+</td>
+<td>
+10 mjordan<br/>
+6 rmudgett<br/>
+5 kharwell<br/>
+2 mmichelson<br/>
+1 maxman<br/>
+1 pnlarsson<br/>
+1 StefanEng86<br/>
+1 yateya<br/>
+</td>
+</tr>
+</table>
+<hr/>
+<a name="issues"><h2 align="center">Closed Issues</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p>
+<h3>Category: Applications/app_agent_pool</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24737">ASTERISK-24737</a>: When agent not logged in, agent status shows unavailable, queue status shows agent invalid<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431494">431494</a><br/>
+Reporter: rmudgett<br/>
+Coders: rmudgett<br/>
+<br/>
+<h3>Category: Applications/app_confbridge</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24719">ASTERISK-24719</a>: ConfBridge recording channels get stuck when recording started/stopped more than once<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431162">431162</a><br/>
+Reporter: rmudgett<br/>
+Coders: rmudgett<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24723">ASTERISK-24723</a>: confbridge: CLI command 'confbridge list XXXX' no longer displays user menus<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431140">431140</a><br/>
+Reporter: mjordan<br/>
+Coders: mjordan<br/>
+<br/>
+<h3>Category: Applications/app_dial</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24682">ASTERISK-24682</a>: app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430843">430843</a><br/>
+Reporter: mjordan<br/>
+Coders: mjordan<br/>
+<br/>
+<h3>Category: Channels/chan_pjsip</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24536">ASTERISK-24536</a>: AMI redirect with PJSIP fails to move extra channel<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430007">430007</a><br/>
+Reporter: pnlarsson<br/>
+Coders: bebuild<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24666">ASTERISK-24666</a>: Security Vulnerability: RTP not closed after sip call using unsupported codec<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431326">431326</a><br/>
+Reporter: yateya<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Registration</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24640">ASTERISK-24640</a>: Registration pending stays forever after sip reload<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430865">430865</a><br/>
+Reporter: maxman<br/>
+Testers: Stefan Engström<br/>
+Coders: mjordan<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24673">ASTERISK-24673</a>: outgoing sip registers cannot be removed or modified without doing restart (or doing module unload chan_sip.so)<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430865">430865</a><br/>
+Reporter: StefanEng86<br/>
+Testers: Stefan Engström<br/>
+Coders: mjordan<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24715">ASTERISK-24715</a>: chan_sip: stale nonce causes failure<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431200">431200</a><br/>
+Reporter: kharwell<br/>
+Coders: kharwell<br/>
+<br/>
+<h3>Category: Core/Bridging</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24536">ASTERISK-24536</a>: AMI redirect with PJSIP fails to move extra channel<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430007">430007</a><br/>
+Reporter: pnlarsson<br/>
+Coders: bebuild<br/>
+<br/>
+<h3>Category: Core/Bridging/bridge_basic</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24513">ASTERISK-24513</a>: Local channel apparently leaked in off-nominal DTMF attended transfer<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430245">430245</a><br/>
+Reporter: mmichelson<br/>
+Coders: sgriepentrog<br/>
+<br/>
+<h3>Category: Core/CodecInterface</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24604">ASTERISK-24604</a>: res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429871">429871</a><br/>
+Reporter: mjordan<br/>
+Coders: bebuild<br/>
+<br/>
+<h3>Category: Core/General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24614">ASTERISK-24614</a>: Deadlock when DEBUG_THREADS compiler flag enabled<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429859">429859</a><br/>
+Reporter: rmudgett<br/>
+Coders: bebuild<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24736">ASTERISK-24736</a>: Memory Leak Fixes<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431470">431470</a><br/>
+Reporter: mmichelson<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: Core/ManagerInterface</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24536">ASTERISK-24536</a>: AMI redirect with PJSIP fails to move extra channel<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430007">430007</a><br/>
+Reporter: pnlarsson<br/>
+Coders: bebuild<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24553">ASTERISK-24553</a>: ARI/AMI: Include language in standard channel snapshot output<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429891">429891</a><br/>
+Reporter: mjordan<br/>
+Coders: bebuild<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24721">ASTERISK-24721</a>: manager: ModuleLoad action incorrectly reports 'module not found' during a Reload operation<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431467">431467</a><br/>
+Reporter: mjordan<br/>
+Coders: jrose<br/>
+<br/>
+<h3>Category: Core/Stasis</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24682">ASTERISK-24682</a>: app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430843">430843</a><br/>
+Reporter: mjordan<br/>
+Coders: mjordan<br/>
+<br/>
+<h3>Category: Features</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23841">ASTERISK-23841</a>: DTMF atxfer doesn't set CallerID for the recall calls to the transferrer.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430046">430046</a><br/>
+Reporter: rmudgett<br/>
+Coders: rmudgett<br/>
+<br/>
+<h3>Category: Functions/func_curl</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24676">ASTERISK-24676</a>: Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431334">431334</a><br/>
+Reporter: mjordan<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: Resources/res_ari</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24552">ASTERISK-24552</a>: ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429892">429892</a><br/>
+Reporter: mjordan<br/>
+Coders: bebuild<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24553">ASTERISK-24553</a>: ARI/AMI: Include language in standard channel snapshot output<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429891">429891</a><br/>
+Reporter: mjordan<br/>
+Coders: bebuild<br/>
+<br/>
+<h3>Category: Resources/res_ari_channels</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24552">ASTERISK-24552</a>: ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429892">429892</a><br/>
+Reporter: mjordan<br/>
+Coders: bebuild<br/>
+<br/>
+<h3>Category: Resources/res_config_curl</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24676">ASTERISK-24676</a>: Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431334">431334</a><br/>
+Reporter: mjordan<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: Resources/res_pjsip</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24342">ASTERISK-24342</a>: PJSIP: Qualifying endpoints attempts to do them all at the same time.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430008">430008</a><br/>
+Reporter: rmudgett<br/>
+Coders: bebuild<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24643">ASTERISK-24643</a>: res_pjsip: Add user=phone option<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430085">430085</a><br/>
+Reporter: mjordan<br/>
+Coders: mjordan<br/>
+<br/>
+<h3>Category: Resources/res_pjsip_keepalive</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24644">ASTERISK-24644</a>: res_pjsip_keepalive: Add keepalive module for connection-oriented transports.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430086">430086</a><br/>
+Reporter: mjordan<br/>
+Coders: mjordan<br/>
+<br/>
+<h3>Category: Resources/res_pjsip_outbound_registration</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24514">ASTERISK-24514</a>: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429866">429866</a><br/>
+Reporter: kharwell<br/>
+Coders: bebuild<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24729">ASTERISK-24729</a>: Outbound registration not occuring on new registrations after reload.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=431244">431244</a><br/>
+Reporter: rmudgett<br/>
+Coders: rmudgett<br/>
+<br/>
+<h3>Category: Resources/res_pjsip_session</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24607">ASTERISK-24607</a>: res_pjsip_session: re-INVITE with declined media streams results in 488<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429890">429890</a><br/>
+Reporter: mjordan<br/>
+Coders: bebuild<br/>
+<br/>
+<h3>Category: Resources/res_rtp_asterisk</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24604">ASTERISK-24604</a>: res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429871">429871</a><br/>
+Reporter: mjordan<br/>
+Coders: bebuild<br/>
+<br/>
+<hr/>
+<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker.  The commits may have been marked as being related to an issue.  If that is the case, the issue numbers are listed here, as well.</p>
+<table width="100%" border="1">
+<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429855">429855</a></td><td>bebuild</td><td>Create Certified Asterisk 13.1 branch</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429856">429856</a></td><td>bebuild</td><td>Update properties; remove old summaries</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429858">429858</a></td><td>bebuild</td><td>Activate persistent subscriptions when they are recreated.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429862">429862</a></td><td>bebuild</td><td>Direct Media calls within private network sometimes get one way audio</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429864">429864</a></td><td>bebuild</td><td>res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429865">429865</a></td><td>bebuild</td><td>PJSIP: Allow use of 'inactive' streams for hold</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429869">429869</a></td><td>bebuild</td><td>Prevent potential infinite outbound authentication loops in registration.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430090">430090</a></td><td>mjordan</td><td>Stasis: Update unittest for channel snapshots</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430093">430093</a></td><td>mjordan</td><td>res_pjsip: Backport missing commits for user_eq_phone</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430094">430094</a></td><td>mjordan</td><td>res/res_agi: Make Verbose message for 'stream file' match other playbacks</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430253">430253</a></td><td>mjordan</td><td>contrib/ast-db-manage: Correct down_revision path for user_eq_phone</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430293">430293</a></td><td>mjordan</td><td>Disable extended support modules</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430782">430782</a></td><td>mmichelson</td><td>Call extension state callbacks at hint creation.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430898">430898</a></td><td>rmudgett</td><td>Multiple revisions 430223,430373,430395</td>
+<td></td></tr></table>
+<hr/>
+<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
+<pre>
+.version                                                                                |    2
+CHANGES                                                                                 |   29
+ChangeLog                                                                               |    4
+agi/eagi-test.c                                                                         |    1
+apps/app_adsiprog.c                                                                     |    1
+apps/app_agent_pool.c                                                                   |   32
+apps/app_alarmreceiver.c                                                                |    1
+apps/app_amd.c                                                                          |    1
+apps/app_confbridge.c                                                                   |  228 +---
+apps/app_dahdiras.c                                                                     |    1
+apps/app_dial.c                                                                         |   51
+apps/app_dictate.c                                                                      |    1
+apps/app_externalivr.c                                                                  |    1
+apps/app_festival.c                                                                     |    1
+apps/app_getcpeid.c                                                                     |    1
+apps/app_ices.c                                                                         |    1
+apps/app_image.c                                                                        |    1
+apps/app_jack.c                                                                         |    1
+apps/app_minivm.c                                                                       |    1
+apps/app_morsecode.c                                                                    |    1
+apps/app_mp3.c                                                                          |    1
+apps/app_nbscat.c                                                                       |    1
+apps/app_osplookup.c                                                                    |    1
+apps/app_sms.c                                                                          |    1
+apps/app_talkdetect.c                                                                   |    1
+apps/app_test.c                                                                         |    1
+apps/app_url.c                                                                          |    1
+apps/app_waitforring.c                                                                  |    1
+apps/app_waitforsilence.c                                                               |    1
+apps/app_zapateller.c                                                                   |    1
+apps/confbridge/conf_config_parser.c                                                    |    7
+apps/confbridge/include/confbridge.h                                                    |   16
+asterisk-13.1.0-rc2-summary.html                                                        |   64 -
+asterisk-13.1.0-rc2-summary.txt                                                         |   95 -
+cdr/cdr_csv.c                                                                           |    1
+cdr/cdr_odbc.c                                                                          |    1
+cdr/cdr_pgsql.c                                                                         |    1
+cdr/cdr_radius.c                                                                        |    1
+cdr/cdr_sqlite3_custom.c                                                                |    1
+cdr/cdr_tds.c                                                                           |    1
+cel/cel_pgsql.c                                                                         |    1
+cel/cel_radius.c                                                                        |    1
+cel/cel_sqlite3_custom.c                                                                |    1
+cel/cel_tds.c                                                                           |    1
+channels/chan_alsa.c                                                                    |    1
+channels/chan_console.c                                                                 |    1
+channels/chan_dahdi.c                                                                   |   15
+channels/chan_iax2.c                                                                    |    6
+channels/chan_mgcp.c                                                                    |    1
+channels/chan_misdn.c                                                                   |    1
+channels/chan_nbs.c                                                                     |    1
+channels/chan_oss.c                                                                     |    1
+channels/chan_phone.c                                                                   |    1
+channels/chan_pjsip.c                                                                   |   37
+channels/chan_sip.c                                                                     |   65 -
+channels/chan_skinny.c                                                                  |    1
+channels/chan_unistim.c                                                                 |    1
+channels/pjsip/dialplan_functions.c                                                     |    6
+configs/samples/pjsip.conf.sample                                                       |    3
+contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py |   30
+formats/format_jpeg.c                                                                   |    1
+formats/format_vox.c                                                                    |    1
+funcs/func_audiohookinherit.c                                                           |    1
+funcs/func_curl.c                                                                       |   83 +
+funcs/func_frame_trace.c                                                                |    1
+funcs/func_pitchshift.c                                                                 |    1
+include/asterisk/_private.h                                                             |    1
+include/asterisk/format.h                                                               |   11
+include/asterisk/lock.h                                                                 |   47
+include/asterisk/res_pjsip.h                                                            |   18
+include/asterisk/res_pjsip_session.h                                                    |    8
+include/asterisk/rtp_engine.h                                                           |   82 +
+include/asterisk/stasis_app.h                                                           |    3
+main/asterisk.c                                                                         |    2
+main/bridge_after.c                                                                     |    1
+main/bridge_basic.c                                                                     |  118 ++
+main/bridge_channel.c                                                                   |   17
+main/codec.c                                                                            |    5
+main/format.c                                                                           |    5
+main/lock.c                                                                             |  570 ++++------
+main/logger.c                                                                           |   42
+main/manager.c                                                                          |   55
+main/manager_channels.c                                                                 |    2
+main/pbx.c                                                                              |   20
+main/rtp_engine.c                                                                       |   38
+main/sorcery.c                                                                          |   25
+main/stasis_channels.c                                                                  |    8
+main/tcptls.c                                                                           |    3
+main/xmldoc.c                                                                           |   13
+pbx/pbx_ael.c                                                                           |    1
+pbx/pbx_dundi.c                                                                         |    1
+pbx/pbx_lua.c                                                                           |    1
+pbx/pbx_realtime.c                                                                      |    1
+pbx/pbx_spool.c                                                                         |    2
+res/ari/ari_model_validators.c                                                          |   16
+res/ari/ari_model_validators.h                                                          |    1
+res/ari/resource_channels.c                                                             |  214 +++
+res/ari/resource_channels.h                                                             |    4
+res/parking/parking_applications.c                                                      |    1
+res/parking/parking_manager.c                                                           |    2
+res/res_ael_share.c                                                                     |    1
+res/res_agi.c                                                                           |    5
+res/res_ari_channels.c                                                                  |   18
+res/res_ari_endpoints.c                                                                 |    4
+res/res_ari_events.c                                                                    |    2
+res/res_config_ldap.c                                                                   |    1
+res/res_config_pgsql.c                                                                  |    1
+res/res_config_sqlite.c                                                                 |    1
+res/res_hep.c                                                                           |    1
+res/res_hep_pjsip.c                                                                     |    1
+res/res_hep_rtcp.c                                                                      |    1
+res/res_phoneprov.c                                                                     |    1
+res/res_pjsip.c                                                                         |   62 +
+res/res_pjsip/config_global.c                                                           |   19
+res/res_pjsip/pjsip_configuration.c                                                     |    1
+res/res_pjsip/pjsip_global_headers.c                                                    |    1
+res/res_pjsip/pjsip_options.c                                                           |   19
+res/res_pjsip_caller_id.c                                                               |   18
+res/res_pjsip_keepalive.c                                                               |  267 ++++
+res/res_pjsip_mwi.c                                                                     |    2
+res/res_pjsip_outbound_publish.c                                                        |  563 ++++++---
+res/res_pjsip_outbound_registration.c                                                   |  335 ++++-
+res/res_pjsip_phoneprov_provider.c                                                      |    1
+res/res_pjsip_pubsub.c                                                                  |   11
+res/res_pjsip_refer.c                                                                   |    9
+res/res_pjsip_sdp_rtp.c                                                                 |   11
+res/res_pjsip_session.c                                                                 |   98 +
+res/res_pjsip_session.exports.in                                                        |    1
+res/res_pjsip_t38.c                                                                     |    1
+res/res_rtp_asterisk.c                                                                  |    3
+res/res_snmp.c                                                                          |    1
+res/res_stasis.c                                                                        |    2
+res/res_statsd.c                                                                        |    1
+res/res_timing_kqueue.c                                                                 |    1
+res/res_timing_pthread.c                                                                |    1
+rest-api-templates/param_parsing.mustache                                               |    2
+rest-api/api-docs/channels.json                                                         |   21
+tests/test_stasis_channels.c                                                            |    2
+utils/astman.c                                                                          |    1
+utils/check_expr.c                                                                      |    1
+utils/conf2ael.c                                                                        |    1
+utils/extconf.c                                                                         |    1
+utils/muted.c                                                                           |    1
+utils/smsq.c                                                                            |    1
+utils/stereorize.c                                                                      |    1
+utils/streamplayer.c                                                                    |    1
+146 files changed, 2455 insertions(+), 1202 deletions(-)
+</pre><br/>
+<hr/>
+</body>
+</html>

certified-asterisk-13.1-cert1-summary.txt

@@ -0,0 +1,509 @@
+                                Release Summary
+
+                         certified-asterisk-13.1-cert1
+
+                                Date: 2015-01-30
+
+                           <asteriskteam@digium.com>
+
+     ----------------------------------------------------------------------
+
+                               Table of Contents
+
+    1. Summary
+    2. Contributors
+    3. Closed Issues
+    4. Other Changes
+    5. Diffstat
+
+     ----------------------------------------------------------------------
+
+                                    Summary
+
+                                 [Back to Top]
+
+   This release includes new features. For a list of new features that have
+   been included with this release, please see the CHANGES file inside the
+   source package. Since this is new major release, users are encouraged to
+   do extended testing before upgrading to this version in a production
+   environment.
+
+   The data in this summary reflects changes that have been made since the
+   previous release, certified-asterisk-13.1.0.
+
+     ----------------------------------------------------------------------
+
+                                  Contributors
+
+                                 [Back to Top]
+
+   This table lists the people who have submitted code, those that have
+   tested patches, as well as those that reported issues on the issue tracker
+   that were resolved in this release. For coders, the number is how many of
+   their patches (of any size) were committed into this release. For testers,
+   the number is the number of times their name was listed as assisting with
+   testing a patch. Finally, for reporters, the number is the number of
+   issues that they reported that were closed by commits that went into this
+   release.
+
+     Coders                   Testers                  Reporters              
+   16 bebuild               2 Stefan EngstrAP:m      10 mjordan               
+   11 mjordan                                        6 rmudgett               
+   5 rmudgett                                        5 kharwell               
+   4 mmichelson                                      2 mmichelson             
+   3 kharwell                                        1 maxman                 
+   1 jrose                                           1 pnlarsson              
+   1 sgriepentrog                                    1 StefanEng86            
+                                                     1 yateya                 
+
+     ----------------------------------------------------------------------
+
+                                 Closed Issues
+
+                                 [Back to Top]
+
+   This is a list of all issues from the issue tracker that were closed by
+   changes that went into this release.
+
+  Category: Applications/app_agent_pool
+
+   ASTERISK-24737: When agent not logged in, agent status shows unavailable,
+   queue status shows agent invalid
+   Revision: 431494
+   Reporter: rmudgett
+   Coders: rmudgett
+
+  Category: Applications/app_confbridge
+
+   ASTERISK-24719: ConfBridge recording channels get stuck when recording
+   started/stopped more than once
+   Revision: 431162
+   Reporter: rmudgett
+   Coders: rmudgett
+
+   ASTERISK-24723: confbridge: CLI command 'confbridge list XXXX' no longer
+   displays user menus
+   Revision: 431140
+   Reporter: mjordan
+   Coders: mjordan
+
+  Category: Applications/app_dial
+
+   ASTERISK-24682: app_dial: Multiple DialEnd events emitted when
+   MACRO_RESULT or GOSUB_RESULT are an unexpected value
+   Revision: 430843
+   Reporter: mjordan
+   Coders: mjordan
+
+  Category: Channels/chan_pjsip
+
+   ASTERISK-24536: AMI redirect with PJSIP fails to move extra channel
+   Revision: 430007
+   Reporter: pnlarsson
+   Coders: bebuild
+
+   ASTERISK-24666: Security Vulnerability: RTP not closed after sip call
+   using unsupported codec
+   Revision: 431326
+   Reporter: yateya
+   Coders: mmichelson
+
+  Category: Channels/chan_sip/Registration
+
+   ASTERISK-24640: Registration pending stays forever after sip reload
+   Revision: 430865
+   Reporter: maxman
+   Testers: Stefan EngstrAP:m
+   Coders: mjordan
+
+   ASTERISK-24673: outgoing sip registers cannot be removed or modified
+   without doing restart (or doing module unload chan_sip.so)
+   Revision: 430865
+   Reporter: StefanEng86
+   Testers: Stefan EngstrAP:m
+   Coders: mjordan
+
+   ASTERISK-24715: chan_sip: stale nonce causes failure
+   Revision: 431200
+   Reporter: kharwell
+   Coders: kharwell
+
+  Category: Core/Bridging
+
+   ASTERISK-24536: AMI redirect with PJSIP fails to move extra channel
+   Revision: 430007
+   Reporter: pnlarsson
+   Coders: bebuild
+
+  Category: Core/Bridging/bridge_basic
+
+   ASTERISK-24513: Local channel apparently leaked in off-nominal DTMF
+   attended transfer
+   Revision: 430245
+   Reporter: mmichelson
+   Coders: sgriepentrog
+
+  Category: Core/CodecInterface
+
+   ASTERISK-24604: res_rtp_asterisk: Crash during restart due to race
+   condition in accessing codec in stored ast_frame and codec core
+   Revision: 429871
+   Reporter: mjordan
+   Coders: bebuild
+
+  Category: Core/General
+
+   ASTERISK-24614: Deadlock when DEBUG_THREADS compiler flag enabled
+   Revision: 429859
+   Reporter: rmudgett
+   Coders: bebuild
+
+   ASTERISK-24736: Memory Leak Fixes
+   Revision: 431470
+   Reporter: mmichelson
+   Coders: mmichelson
+
+  Category: Core/ManagerInterface
+
+   ASTERISK-24536: AMI redirect with PJSIP fails to move extra channel
+   Revision: 430007
+   Reporter: pnlarsson
+   Coders: bebuild
+
+   ASTERISK-24553: ARI/AMI: Include language in standard channel snapshot
+   output
+   Revision: 429891
+   Reporter: mjordan
+   Coders: bebuild
+
+   ASTERISK-24721: manager: ModuleLoad action incorrectly reports 'module not
+   found' during a Reload operation
+   Revision: 431467
+   Reporter: mjordan
+   Coders: jrose
+
+  Category: Core/Stasis
+
+   ASTERISK-24682: app_dial: Multiple DialEnd events emitted when
+   MACRO_RESULT or GOSUB_RESULT are an unexpected value
+   Revision: 430843
+   Reporter: mjordan
+   Coders: mjordan
+
+  Category: Features
+
+   ASTERISK-23841: DTMF atxfer doesn't set CallerID for the recall calls to
+   the transferrer.
+   Revision: 430046
+   Reporter: rmudgett
+   Coders: rmudgett
+
+  Category: Functions/func_curl
+
+   ASTERISK-24676: Security Vulnerability: URL request injection in libCURL
+   (CVE-2014-8150)
+   Revision: 431334
+   Reporter: mjordan
+   Coders: mmichelson
+
+  Category: Resources/res_ari
+
+   ASTERISK-24552: ARI: Allow associating a channel as an initiator of an
+   Origination for record keeping purposes
+   Revision: 429892
+   Reporter: mjordan
+   Coders: bebuild
+
+   ASTERISK-24553: ARI/AMI: Include language in standard channel snapshot
+   output
+   Revision: 429891
+   Reporter: mjordan
+   Coders: bebuild
+
+  Category: Resources/res_ari_channels
+
+   ASTERISK-24552: ARI: Allow associating a channel as an initiator of an
+   Origination for record keeping purposes
+   Revision: 429892
+   Reporter: mjordan
+   Coders: bebuild
+
+  Category: Resources/res_config_curl
+
+   ASTERISK-24676: Security Vulnerability: URL request injection in libCURL
+   (CVE-2014-8150)
+   Revision: 431334
+   Reporter: mjordan
+   Coders: mmichelson
+
+  Category: Resources/res_pjsip
+
+   ASTERISK-24342: PJSIP: Qualifying endpoints attempts to do them all at the
+   same time.
+   Revision: 430008
+   Reporter: rmudgett
+   Coders: bebuild
+
+   ASTERISK-24643: res_pjsip: Add user=phone option
+   Revision: 430085
+   Reporter: mjordan
+   Coders: mjordan
+
+  Category: Resources/res_pjsip_keepalive
+
+   ASTERISK-24644: res_pjsip_keepalive: Add keepalive module for
+   connection-oriented transports.
+   Revision: 430086
+   Reporter: mjordan
+   Coders: mjordan
+
+  Category: Resources/res_pjsip_outbound_registration
+
+   ASTERISK-24514: res_pjsip_outbound_registration: stack overflow when using
+   non-default sorcery wizard
+   Revision: 429866
+   Reporter: kharwell
+   Coders: bebuild
+
+   ASTERISK-24729: Outbound registration not occuring on new registrations
+   after reload.
+   Revision: 431244
+   Reporter: rmudgett
+   Coders: rmudgett
+
+  Category: Resources/res_pjsip_session
+
+   ASTERISK-24607: res_pjsip_session: re-INVITE with declined media streams
+   results in 488
+   Revision: 429890
+   Reporter: mjordan
+   Coders: bebuild
+
+  Category: Resources/res_rtp_asterisk
+
+   ASTERISK-24604: res_rtp_asterisk: Crash during restart due to race
+   condition in accessing codec in stored ast_frame and codec core
+   Revision: 429871
+   Reporter: mjordan
+   Coders: bebuild
+
+     ----------------------------------------------------------------------
+
+                      Commits Not Associated with an Issue
+
+                                 [Back to Top]
+
+   This is a list of all changes that went into this release that did not
+   directly close an issue from the issue tracker. The commits may have been
+   marked as being related to an issue. If that is the case, the issue
+   numbers are listed here, as well.
+
+   +------------------------------------------------------------------------+
+   | Revision | Author     | Summary                           | Issues     |
+   |          |            |                                   | Referenced |
+   |----------+------------+-----------------------------------+------------|
+   | 429855   | bebuild    | Create Certified Asterisk 13.1    |            |
+   |          |            | branch                            |            |
+   |----------+------------+-----------------------------------+------------|
+   | 429856   | bebuild    | Update properties; remove old     |            |
+   |          |            | summaries                         |            |
+   |----------+------------+-----------------------------------+------------|
+   | 429858   | bebuild    | Activate persistent subscriptions |            |
+   |          |            | when they are recreated.          |            |
+   |----------+------------+-----------------------------------+------------|
+   |          |            | Direct Media calls within private |            |
+   | 429862   | bebuild    | network sometimes get one way     |            |
+   |          |            | audio                             |            |
+   |----------+------------+-----------------------------------+------------|
+   |          |            | res_pjsip_session: Delay sending  |            |
+   | 429864   | bebuild    | BYE if a re-INVITE transaction is |            |
+   |          |            | in progress.                      |            |
+   |----------+------------+-----------------------------------+------------|
+   | 429865   | bebuild    | PJSIP: Allow use of 'inactive'    |            |
+   |          |            | streams for hold                  |            |
+   |----------+------------+-----------------------------------+------------|
+   |          |            | Prevent potential infinite        |            |
+   | 429869   | bebuild    | outbound authentication loops in  |            |
+   |          |            | registration.                     |            |
+   |----------+------------+-----------------------------------+------------|
+   | 430090   | mjordan    | Stasis: Update unittest for       |            |
+   |          |            | channel snapshots                 |            |
+   |----------+------------+-----------------------------------+------------|
+   | 430093   | mjordan    | res_pjsip: Backport missing       |            |
+   |          |            | commits for user_eq_phone         |            |
+   |----------+------------+-----------------------------------+------------|
+   |          |            | res/res_agi: Make Verbose message |            |
+   | 430094   | mjordan    | for 'stream file' match other     |            |
+   |          |            | playbacks                         |            |
+   |----------+------------+-----------------------------------+------------|
+   |          |            | contrib/ast-db-manage: Correct    |            |
+   | 430253   | mjordan    | down_revision path for            |            |
+   |          |            | user_eq_phone                     |            |
+   |----------+------------+-----------------------------------+------------|
+   | 430293   | mjordan    | Disable extended support modules  |            |
+   |----------+------------+-----------------------------------+------------|
+   | 430782   | mmichelson | Call extension state callbacks at |            |
+   |          |            | hint creation.                    |            |
+   |----------+------------+-----------------------------------+------------|
+   | 430898   | rmudgett   | Multiple revisions                |            |
+   |          |            | 430223,430373,430395              |            |
+   +------------------------------------------------------------------------+
+
+     ----------------------------------------------------------------------
+
+                                Diffstat Results
+
+                                 [Back to Top]
+
+   This is a summary of the changes to the source code that went into this
+   release that was generated using the diffstat utility.
+
+ .version                                                                                |    2
+ CHANGES                                                                                 |   29
+ ChangeLog                                                                               |    4
+ agi/eagi-test.c                                                                         |    1
+ apps/app_adsiprog.c                                                                     |    1
+ apps/app_agent_pool.c                                                                   |   32
+ apps/app_alarmreceiver.c                                                                |    1
+ apps/app_amd.c                                                                          |    1
+ apps/app_confbridge.c                                                                   |  228 +---
+ apps/app_dahdiras.c                                                                     |    1
+ apps/app_dial.c                                                                         |   51
+ apps/app_dictate.c                                                                      |    1
+ apps/app_externalivr.c                                                                  |    1
+ apps/app_festival.c                                                                     |    1
+ apps/app_getcpeid.c                                                                     |    1
+ apps/app_ices.c                                                                         |    1
+ apps/app_image.c                                                                        |    1
+ apps/app_jack.c                                                                         |    1
+ apps/app_minivm.c                                                                       |    1
+ apps/app_morsecode.c                                                                    |    1
+ apps/app_mp3.c                                                                          |    1
+ apps/app_nbscat.c                                                                       |    1
+ apps/app_osplookup.c                                                                    |    1
+ apps/app_sms.c                                                                          |    1
+ apps/app_talkdetect.c                                                                   |    1
+ apps/app_test.c                                                                         |    1
+ apps/app_url.c                                                                          |    1
+ apps/app_waitforring.c                                                                  |    1
+ apps/app_waitforsilence.c                                                               |    1
+ apps/app_zapateller.c                                                                   |    1
+ apps/confbridge/conf_config_parser.c                                                    |    7
+ apps/confbridge/include/confbridge.h                                                    |   16
+ asterisk-13.1.0-rc2-summary.html                                                        |   64 -
+ asterisk-13.1.0-rc2-summary.txt                                                         |   95 -
+ cdr/cdr_csv.c                                                                           |    1
+ cdr/cdr_odbc.c                                                                          |    1
+ cdr/cdr_pgsql.c                                                                         |    1
+ cdr/cdr_radius.c                                                                        |    1
+ cdr/cdr_sqlite3_custom.c                                                                |    1
+ cdr/cdr_tds.c                                                                           |    1
+ cel/cel_pgsql.c                                                                         |    1
+ cel/cel_radius.c                                                                        |    1
+ cel/cel_sqlite3_custom.c                                                                |    1
+ cel/cel_tds.c                                                                           |    1
+ channels/chan_alsa.c                                                                    |    1
+ channels/chan_console.c                                                                 |    1
+ channels/chan_dahdi.c                                                                   |   15
+ channels/chan_iax2.c                                                                    |    6
+ channels/chan_mgcp.c                                                                    |    1
+ channels/chan_misdn.c                                                                   |    1
+ channels/chan_nbs.c                                                                     |    1
+ channels/chan_oss.c                                                                     |    1
+ channels/chan_phone.c                                                                   |    1
+ channels/chan_pjsip.c                                                                   |   37
+ channels/chan_sip.c                                                                     |   65 -
+ channels/chan_skinny.c                                                                  |    1
+ channels/chan_unistim.c                                                                 |    1
+ channels/pjsip/dialplan_functions.c                                                     |    6
+ configs/samples/pjsip.conf.sample                                                       |    3
+ contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py |   30
+ formats/format_jpeg.c                                                                   |    1
+ formats/format_vox.c                                                                    |    1
+ funcs/func_audiohookinherit.c                                                           |    1
+ funcs/func_curl.c                                                                       |   83 +
+ funcs/func_frame_trace.c                                                                |    1
+ funcs/func_pitchshift.c                                                                 |    1
+ include/asterisk/_private.h                                                             |    1
+ include/asterisk/format.h                                                               |   11
+ include/asterisk/lock.h                                                                 |   47
+ include/asterisk/res_pjsip.h                                                            |   18
+ include/asterisk/res_pjsip_session.h                                                    |    8
+ include/asterisk/rtp_engine.h                                                           |   82 +
+ include/asterisk/stasis_app.h                                                           |    3
+ main/asterisk.c                                                                         |    2
+ main/bridge_after.c                                                                     |    1
+ main/bridge_basic.c                                                                     |  118 ++
+ main/bridge_channel.c                                                                   |   17
+ main/codec.c                                                                            |    5
+ main/format.c                                                                           |    5
+ main/lock.c                                                                             |  570 ++++------
+ main/logger.c                                                                           |   42
+ main/manager.c                                                                          |   55
+ main/manager_channels.c                                                                 |    2
+ main/pbx.c                                                                              |   20
+ main/rtp_engine.c                                                                       |   38
+ main/sorcery.c                                                                          |   25
+ main/stasis_channels.c                                                                  |    8
+ main/tcptls.c                                                                           |    3
+ main/xmldoc.c                                                                           |   13
+ pbx/pbx_ael.c                                                                           |    1
+ pbx/pbx_dundi.c                                                                         |    1
+ pbx/pbx_lua.c                                                                           |    1
+ pbx/pbx_realtime.c                                                                      |    1
+ pbx/pbx_spool.c                                                                         |    2
+ res/ari/ari_model_validators.c                                                          |   16
+ res/ari/ari_model_validators.h                                                          |    1
+ res/ari/resource_channels.c                                                             |  214 +++
+ res/ari/resource_channels.h                                                             |    4
+ res/parking/parking_applications.c                                                      |    1
+ res/parking/parking_manager.c                                                           |    2
+ res/res_ael_share.c                                                                     |    1
+ res/res_agi.c                                                                           |    5
+ res/res_ari_channels.c                                                                  |   18
+ res/res_ari_endpoints.c                                                                 |    4
+ res/res_ari_events.c                                                                    |    2
+ res/res_config_ldap.c                                                                   |    1
+ res/res_config_pgsql.c                                                                  |    1
+ res/res_config_sqlite.c                                                                 |    1
+ res/res_hep.c                                                                           |    1
+ res/res_hep_pjsip.c                                                                     |    1
+ res/res_hep_rtcp.c                                                                      |    1
+ res/res_phoneprov.c                                                                     |    1
+ res/res_pjsip.c                                                                         |   62 +
+ res/res_pjsip/config_global.c                                                           |   19
+ res/res_pjsip/pjsip_configuration.c                                                     |    1
+ res/res_pjsip/pjsip_global_headers.c                                                    |    1
+ res/res_pjsip/pjsip_options.c                                                           |   19
+ res/res_pjsip_caller_id.c                                                               |   18
+ res/res_pjsip_keepalive.c                                                               |  267 ++++
+ res/res_pjsip_mwi.c                                                                     |    2
+ res/res_pjsip_outbound_publish.c                                                        |  563 ++++++---
+ res/res_pjsip_outbound_registration.c                                                   |  335 ++++-
+ res/res_pjsip_phoneprov_provider.c                                                      |    1
+ res/res_pjsip_pubsub.c                                                                  |   11
+ res/res_pjsip_refer.c                                                                   |    9
+ res/res_pjsip_sdp_rtp.c                                                                 |   11
+ res/res_pjsip_session.c                                                                 |   98 +
+ res/res_pjsip_session.exports.in                                                        |    1
+ res/res_pjsip_t38.c                                                                     |    1
+ res/res_rtp_asterisk.c                                                                  |    3
+ res/res_snmp.c                                                                          |    1
+ res/res_stasis.c                                                                        |    2
+ res/res_statsd.c                                                                        |    1
+ res/res_timing_kqueue.c                                                                 |    1
+ res/res_timing_pthread.c                                                                |    1
+ rest-api-templates/param_parsing.mustache                                               |    2
+ rest-api/api-docs/channels.json                                                         |   21
+ tests/test_stasis_channels.c                                                            |    2
+ utils/astman.c                                                                          |    1
+ utils/check_expr.c                                                                      |    1
+ utils/conf2ael.c                                                                        |    1
+ utils/extconf.c                                                                         |    1
+ utils/muted.c                                                                           |    1
+ utils/smsq.c                                                                            |    1
+ utils/stereorize.c                                                                      |    1
+ utils/streamplayer.c                                                                    |    1
+ 146 files changed, 2455 insertions(+), 1202 deletions(-)
+
+     ----------------------------------------------------------------------

contrib/realtime/mysql/mysql_config.sql

@@ -703,3 +703,9 @@ ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic ENUM('yes','no')
 
 UPDATE alembic_version SET version_num='eb88a14f2a';
 
+-- Running upgrade eb88a14f2a -> 371a3bf4143e
+
+ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone ENUM('yes','no');
+
+UPDATE alembic_version SET version_num='371a3bf4143e';
+

contrib/realtime/oracle/oracle_config.sql

@@ -984,7 +984,17 @@ ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (media_encryption_opt
 
 /
 
-INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a')
+-- Running upgrade eb88a14f2a -> 371a3bf4143e
+
+ALTER TABLE ps_endpoints ADD user_eq_phone VARCHAR(3 CHAR)
+
+/
+
+ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (user_eq_phone IN ('yes', 'no'))
+
+/
+
+INSERT INTO alembic_version (version_num) VALUES ('371a3bf4143e')
 
 /
 

contrib/realtime/postgresql/postgresql_config.sql

@@ -733,7 +733,11 @@ DROP TYPE sip_directmedia_values;
 
 ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic yesno_values;
 
-INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a');
+-- Running upgrade eb88a14f2a -> 371a3bf4143e
+
+ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone yesno_values;
+
+INSERT INTO alembic_version (version_num) VALUES ('371a3bf4143e');
 
 COMMIT;
 

contrib/realtime/sqlserver/mssql_config.sql

@@ -982,7 +982,17 @@ ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (media_encryption_opt
 
 GO
 
-INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a');
+-- Running upgrade eb88a14f2a -> 371a3bf4143e
+
+ALTER TABLE ps_endpoints ADD user_eq_phone VARCHAR(3) NULL;
+
+GO
+
+ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (user_eq_phone IN ('yes', 'no'));
+
+GO
+
+INSERT INTO alembic_version (version_num) VALUES ('371a3bf4143e');
 
 GO