[asterisk-commits] bebuild: tag 12.4.0-rc1 r418185 - in /tags/12.4.0-rc1: ./ contrib/realtime/my...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jul 8 09:50:17 CDT 2014


Author: bebuild
Date: Tue Jul  8 09:50:14 2014
New Revision: 418185

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=418185
Log:
Importing files for 12.4.0-rc1 release.

Added:
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    tags/12.4.0-rc1/ChangeLog   (with props)
    tags/12.4.0-rc1/contrib/realtime/mysql/mysql_cdr.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/mysql/mysql_config.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/oracle/oracle_cdr.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/oracle/oracle_config.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/postgresql/postgresql_cdr.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/postgresql/postgresql_config.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/sqlserver/mssql_cdr.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/sqlserver/mssql_config.sql   (with props)
    tags/12.4.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql   (with props)

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Added: tags/12.4.0-rc1/ChangeLog
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--- tags/12.4.0-rc1/ChangeLog (added)
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+2014-07-08  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 12.4.0-rc1 Released.
+
+2014-07-08 14:47 +0000 [r418172-418182]  Matthew Jordan <mjordan at digium.com>
+
+	* include/asterisk/manager.h, rest-api/api-docs/bridges.json,
+	  rest-api/api-docs/recordings.json,
+	  rest-api/api-docs/deviceStates.json,
+	  rest-api/api-docs/endpoints.json,
+	  rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
+	  rest-api/api-docs/asterisk.json,
+	  rest-api/api-docs/applications.json,
+	  rest-api/api-docs/playbacks.json, UPGRADE.txt,
+	  rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
+	  rest-api/resources.json: manager/ARI: Update version to
+	  2.4.0/1.4.0; Update UPGRADE.txt
+
+	* res/res_rtp_asterisk.c: res_rtp_asterisk: Fix undefined function
+	  when PJPROJECT is not installed The dtls_perform_handshake
+	  function was mistakenly placed under the guards for
+	  USE_PJPROJECT. If PJPROJECT was not installed, the function would
+	  not be defined, while other functions would attempt to still use
+	  it. This prevented res_rtp_asterisk from being loaded.
+	  ASTERISK-24001 #close Reported by: Don Fanning
+
+2014-07-07 16:05 +0000 [r418116]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c,
+	  include/asterisk/res_pjsip_presence_xml.h,
+	  include/asterisk/res_pjsip_body_generator_types.h,
+	  res/res_pjsip_dialog_info_body_generator.c (added):
+	  res_pjsip_dialog_info_body_generator: Add dialog-info+xml support
+	  for presence. This module implements dialog-info+xml for the
+	  purposes of presence. This means that phones such as Grandstreams
+	  can now subscribe to receive presence information for an
+	  extension. ASTERISK-21443 #close Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3705/
+
+2014-07-07 02:13 +0000 [r418089]  Matthew Jordan <mjordan at digium.com>
+
+	* res/ari/resource_channels.c, res/res_stasis.c, res/stasis/app.c,
+	  include/asterisk/stasis_app.h: ARI/res_stasis: Subscribe to both
+	  Local channel halves when originating to app This patch fixes two
+	  bugs: 1. When originating a channel into a Stasis application, we
+	  already create a subscription for the channel that is going into
+	  our Stasis app. Unfortunately, when you create a Local channel
+	  and pass it off to a Stasis app, you really aren't creating just
+	  one channel: you're creating two. This patch snags the second
+	  half of the Local channel pair (assuming it is a Local channel
+	  pair, but luckily core_local is kind about such assumptions) and
+	  subscribes to it as well. 2. Subscriptions are a bit sticky right
+	  now. If a subscription is made, the 'interest' count gets bumped
+	  on the Stasis subscription - but unless something explicitly
+	  unsubscribes the channel, said subscription sticks around. This
+	  is not much of a problem is a user is creating the subscription -
+	  if they made it, they must want it. However, when we are creating
+	  implicit subscriptions, we need to make sure something clears
+	  them out. This patch takes a pessimistic approach: it watches the
+	  cache updates coming from Stasis and, if we notice that the cache
+	  just cleared out an object, we delete our subscription object.
+	  This keeps our ao2 container of Stasis forwards in an application
+	  from growing out of hand; it also is a bit more forgiving for end
+	  users who may not realize they were supposed to unsubscribe from
+	  that channel that just hung up. Review:
+	  https://reviewboard.asterisk.org/r/3710/ ASTERISK-23939 #close
+
+2014-07-07 01:18 +0000 [r418066-418071]  Kinsey Moore <kmoore at digium.com>
+
+	* tests/test_cel.c, main/cel.c, channels/chan_pjsip.c,
+	  res/res_pjsip_session.c: CEL: Fix incorrect/missing extra field
+	  information This corrects two issues with the extra field
+	  information in Asterisk 12+ in channel event logs. It is possible
+	  to inject custom values into the dialstatus provided by
+	  ast_channel_dial_type() Stasis messages that fall outside the
+	  enumeration allowed for the DIALSTATUS channel variable. CEL now
+	  filters for the allowed values and ignores other values. The
+	  "hangupsource" extra field key is always blank if the far end
+	  channel is a chan_pjsip channel. This is because the hangupsource
+	  is never set for the pjsip channel driver. This change sets the
+	  hangupsource whenever a hangup is queued for chan_pjsip channels.
+	  This corrects an issue with the pjsip channel driver where the
+	  hangupcause information was not being set properly. Review:
+	  https://reviewboard.asterisk.org/r/3690/
+
+	* main/http.c: HTTP: Fix build for gcc 4.10
+
+2014-07-03 22:06 +0000 [r417880-417958]  Richard Mudgett <rmudgett at digium.com>
+
+	* UPGRADE.txt, channels/sig_pri.c, channels/sig_pri.h,
+	  channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /:
+	  chan_dahdi: Add inband_on_setup_ack compatibility option. The new
+	  inband_on_setup_ack option causes Asterisk to assume inband audio
+	  may be present when a SETUP_ACKNOWLEDGE message is received.
+	  Q.931 Section 5.1.3 says that in scenarios with overlap dialing,
+	  when a dialtone is sent from the network side, progress indicator
+	  8 "Inband info now available" MAY be sent to the CPE if no digits
+	  were received with the SETUP. It is thus implied that the ie is
+	  mandatory if digits came with the SETUP and dialtone is needed.
+	  This option should be enabled, when the network sends dialtone
+	  and you want to hear it, but the network doesn't send the
+	  progress indicator when needed. NOTE: For Q.SIG setups this
+	  option should be enabled when outgoing overlap dialing is also
+	  enabled because Q.SIG does not send the progress indicator with
+	  the SETUP ACK. The commit -r413714 (AST-1338) which causes this
+	  issue was dealing with a SIP-to-ISDN interoperability issue. This
+	  commit is a merge of the two patches indicated below.
+	  ASTERISK-23897 #close Reported by: Pavel Troller Patches:
+	  pri-4.diff (license #6302) patch uploaded by Pavel Troller
+	  jira_asterisk_23897_v11.patch (license #5621) patch uploaded by
+	  rmudgett Review: https://reviewboard.asterisk.org/r/3633/
+	  ........ Merged revisions 417956 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 417957 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* res/res_ari.c, main/manager.c, res/ari/resource_channels.c:
+	  res_ari: Fix some off-nominal paths just dropping the HTTP
+	  connection. * Removed some incorrect newlines on ast_http_error()
+	  messages in manager.c. * Removed an incorrect newline in
+	  res_ari_channels.c. Addendum to ASTERISK-23552
+
+	* configs/http.conf.sample, include/asterisk/http.h, main/tcptls.c,
+	  res/res_ari.c, main/manager.c, res/res_phoneprov.c, main/http.c,
+	  UPGRADE.txt, include/asterisk/tcptls.h, res/res_http_post.c,
+	  res/res_http_websocket.c: HTTP: Add persistent connection
+	  support. Persistent HTTP connection support is needed due to the
+	  increased usage of the Asterisk core HTTP transport and the
+	  frequency at which REST API calls are going to be issued. * Add
+	  http.conf session_keep_alive option to enable persistent
+	  connections. * Parse and discard optional chunked body extension
+	  information and trailing request headers. * Increased the maximum
+	  application/json and application/x-www-form-urlencoded body size
+	  allowed to 4k. The previous 1k was kind of small. * Removed a
+	  couple inlined versions of ast_http_manid_from_vars() by calling
+	  the function. manager.c:generic_http_callback() and
+	  res_http_post.c:http_post_callback() * Add missing va_end() in
+	  ast_ari_response_error(). * Eliminated unnecessary RAII_VAR() use
+	  in http.c:auth_create(). ASTERISK-23552 #close Reported by: Scott
+	  Griepentrog Review: https://reviewboard.asterisk.org/r/3691/
+
+2014-07-03 16:07 +0000 [r417878]  sgalarneau <sgalarneau at localhost>:
+
+	* res/ari/resource_channels.h, rest-api/api-docs/events.json,
+	  res/ari/resource_events.h, rest-api/api-docs/channels.json: ARI:
+	  Improvements to body parameters documentation The variables body
+	  parameter under the originate and originate with id operations of
+	  the channel resource showed invalid JSON in its description. The
+	  variables body parameter under the userEvent operation of the
+	  event resource made no mention that the custom key/value pairs
+	  should be wrapped in a variables key in order to be added to the
+	  custom user event. ASTERISK-23975 #close Review:
+	  https://reviewboard.asterisk.org/r/3692/
+
+2014-07-03 11:26 +0000 [r417799]  Matthew Jordan <mjordan at digium.com>
+
+	* /, main/utils.c: main/untils: Prevent potential infinite loop in
+	  ast_careful_fwrite A loop in ast_careful_fwrite exists that will
+	  continually attempt to write to a file stream, even in the
+	  presence of EAGAIN/EINTR errors. However, if a connection that
+	  uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's
+	  call to fflush may return EAGAIN/EINTER along with EOF. A
+	  subsequent call to fflush will return EOF but not clear errno,
+	  resulting in an infinite loop. This patch clears errno after it
+	  is detected and handled the loop, such that any subsequent call
+	  to fflush will not get erroneously stuck. Review:
+	  https://reviewboard.asterisk.org/r/3704 #ASTERISK-23984 #close
+	  Reported by: Steve Davies patches: fflush_loop_fix uploaded by
+	  one47 (License 5012) ........ Merged revisions 417797 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 417798 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-01 14:40 +0000 [r417678-417705]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_rtp_asterisk.c: res_rtp_asterisk: Don't leak memory or
+	  reset state if DTLS configuration is set multiple times.
+
+	* main/sdp_srtp.c, res/res_pjsip_sdp_rtp.c,
+	  res/res_pjsip/pjsip_configuration.c, UPGRADE.txt,
+	  configs/sip.conf.sample, include/asterisk/rtp_engine.h,
+	  res/res_pjsip.c, channels/sip/include/sip.h,
+	  include/asterisk/res_pjsip.h, include/asterisk/sdp_srtp.h,
+	  res/res_rtp_asterisk.c,
+	  contrib/ast-db-manage/config/versions/51f8cb66540e_add_further_dtls_options.py
+	  (added), include/asterisk/res_pjsip_session.h, main/rtp_engine.c,
+	  channels/chan_sip.c: Recorded merge of revisions 417677 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........
+	  res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS
+	  negotiation on RTCP. This change fixes up DTLS support in
+	  res_rtp_asterisk so it can accept and provide a SHA-256
+	  fingerprint, so it occurs on RTCP, and so it occurs after ICE
+	  negotiation completes. Configuration options to chan_sip and
+	  chan_pjsip have also been added to allow behavior to be tweaked
+	  (such as forcing the AVP type media transports in SDP).
+	  ASTERISK-22961 #close Reported by: Jay Jideliov Review:
+	  https://reviewboard.asterisk.org/r/3679/ Review:
+	  https://reviewboard.asterisk.org/r/3686/
+
+2014-06-30 03:25 +0000 [r417589]  Matthew Jordan <mjordan at digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: be more tolerant of whitespace
+	  between attributes in SDP fmtp line This patch is essentially a
+	  backport of a small portion of r397526 from ASTERISK-21981. In
+	  that patch, pass through support and format attribute negotiation
+	  was added for Opus. Part of that included being more tolerant to
+	  whitespace in the fmtp line of an SDP; that part of the patch is
+	  being applied here. As the author of the backport pointed out, in
+	  SDP, the fmtp line is allowed to include whitespace between
+	  attributes. RFC 3267 chapter 8.3 (from 2001) includes an example
+	  for this. This was not removed in the updated RFC 4867 in 2007.
+	  Review: https://reviewboard.asterisk.org/r/3658 #ASTERISK-23916
+	  #close Reported by: Alexander Traud patches:
+	  sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud
+	  (License 6520) ........ Merged revisions 417587 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 417588 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-27 23:11 +0000 [r417565]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/event.c: event.c: Fix type mismatch errors in ie_maps[]. In
+	  v12+ the type values from the table are only used by the CEL unit
+	  tests. Since the unit tests were only comparing a generated
+	  expected event with a real event to see if the ie contents
+	  matched and using the same table IE_PLTYPE values to read the
+	  event contents, the type mismatches were not detected.
+
+2014-06-27 19:27 +0000 [r417483-417509]  Corey Farrell <git at cfware.com>
+
+	* /, main/astobj2.c: Ensure REF_DEBUG records entrys for attempts
+	  to ao2_ref an invalid object This change ensures that
+	  __ao2_ref_debug writes to ref_log when given a non-NULL pointer
+	  to an invalid ao2 object. This is to ensure that we record any
+	  attempt manipulate references of already freed objects.
+	  ASTERISK-23948 #close Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3677/ ........ Merged
+	  revisions 417500 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 417505 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* contrib/scripts/refcounter.py, /: refcounter.py: prevent use of
+	  excessive RAM with large refs logs When processing a 212MB refs
+	  file, refcounter.py used over 3GB of RAM. This change greatly
+	  reduces memory usage in two ways: * Saving object history in
+	  whole lines instead of separated values. * Not saving
+	  normal/skewed/leaked object lists unless they are requested.
+	  ASTERISK-23921 #close Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3668/ ........ Merged
+	  revisions 417480 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 417481 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-27 13:48 +0000 [r417311-417460]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_pjsip_pubsub.c, res/res_pjsip_registrar.c,
+	  include/asterisk/res_pjsip.h,
+	  res/res_pjsip_outbound_registration.c,
+	  res/res_pjsip/pjsip_configuration.c: res_pjsip: Add ActionID to
+	  events created as a result of PJSIP AMI actions A number of
+	  various PJSIP AMI actions were failing to parse out and place the
+	  ActionID into their responses. This patch updates the various
+	  PJSIP actions such that the passed in ActionID is emitted on any
+	  event list complete events, as well as any intermediate events
+	  created as a result of the action. ASTERISK-23947 #close Reported
+	  by: Mark Michelson Review:
+	  https://reviewboard.asterisk.org/r/3675/
+
+	* res/res_http_websocket.exports.in, /: res_http_websocket: Export
+	  symbol for ast_websocket_set_timeout Thanks to Sean Bright for
+	  pointing out that this was missed in #asterisk-dev. ........
+	  Merged revisions 417419 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* channels/chan_pjsip.c: chan_pjsip: Add a test event for fast
+	  picture updates This will drive the test on review r3419. Note
+	  that the patch for this was done by Ben Ford, although it was
+	  slightly modified for this commit. ASTERISK-23562 Reported by:
+	  Matt Jordan
+
+	* main/udptl.c, /: udptl: Correct FEC to not consider negative
+	  sequence numbers as missing When using FEC, with span=3 and
+	  entries=4 Asterisk will attempt to repair the packet with
+	  sequence number 5, as it will see that packet -4 is missing. The
+	  result is Asterisk sending garbage packets that can kill a fax.
+	  This patch adds a check to see if the sequence number is valid
+	  before checking if the packet is missing. Review:
+	  https://reviewboard.asterisk.org/r/3657/ #ASTERISK-23908 #close
+	  Reported by: Torrey Searle patches: udptl_fec.patch uploaded by
+	  Torrey Searle (License 5334) ........ Merged revisions 417318
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  Merged revisions 417320 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* configs/pjsip.conf.sample, include/asterisk/http_websocket.h,
+	  configs/sip.conf.sample, res/res_pjsip/config_transport.c,
+	  res/ari/ari_websockets.c, res/res_pjsip_transport_websocket.c,
+	  res/ari/config.c, channels/sip/include/sip.h,
+	  include/asterisk/res_pjsip.h, res/res_ari.c, channels/chan_sip.c,
+	  /, UPGRADE.txt, res/ari/internal.h, configs/ari.conf.sample,
+	  res/res_pjsip.c, res/res_http_websocket.c: res_http_websocket:
+	  Close websocket correctly and use careful fwrite When a client
+	  takes a long time to process information received from Asterisk,
+	  a write operation using fwrite may fail to write all information.
+	  This causes the underlying file stream to be in an unknown state,
+	  such that the socket must be disconnected. Unfortunately, there
+	  are two problems with this in Asterisk's existing websocket code:
+	  1. Periodically, during the read loop, Asterisk must write to the
+	  connected websocket to respond to pings. As such, Asterisk
+	  maintains a reference to the session during the loop. When
+	  ast_http_websocket_write fails, it may cause the session to
+	  decrement its ref count, but this in and of itself does not break
+	  the read loop. The read loop's write, on the other hand, does not
+	  break the loop if it fails. This causes the socket to get in a
+	  'stuck' state, preventing the client from reconnecting to the
+	  server. 2. More importantly, however, is that the fwrite in
+	  ast_http_websocket_write fails with a large volume of data when
+	  the client takes awhile to process the information. When it does
+	  fail, it fails writing only a portion of the bytes. With some
+	  debugging, it was shown that this was failing in a similar
+	  fashion to ASTERISK-12767. Switching this over to
+	  ast_careful_fwrite with a long enough timeout solved the problem.
+	  Note that this version of the patch, unlike r417310 in Asterisk
+	  11, exposes configuration options beyond just chan_sip's
+	  sip.conf. Configuration options to configure the write timeout
+	  have also been added to pjsip.conf and ari.conf. #ASTERISK-23917
+	  #close Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3624/ ........ Merged
+	  revisions 417310 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-26 10:05 +0000 [r417212-417250]  Corey Farrell <git at cfware.com>
+
+	* /, channels/chan_sip.c: chan_sip: Fix handling of "From" headers
+	  longer than 256 characters From headers were processed using a
+	  256 character buffer on the stack. This change replaces that with
+	  a heap allocation by ast_strdup. ASTERISK-23790 #close Reported
+	  by: uniken1 Tested by: uniken1 Review:
+	  https://reviewboard.asterisk.org/r/3669/ Patches:
+	  chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes
+	  (license 5674) ........ Merged revisions 417248 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 417249 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* main/astobj2_container.c: ao2_container node object ignores
+	  REF_DEBUG in all places except one Almost every reference
+	  operation against container node's uses __ao2_alloc or __ao2_ref,
+	  thereby preventing ref logging for the nodes. One node reference
+	  is released with ao2_t_ref, causing refcounter.py to falsely
+	  report skews and leaks for many nodes. ASTERISK-23922 #close
+	  Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3670/
+
+2014-06-23 18:49 +0000 [r417142]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Return the length of
+	  data written when sending via ICE instead of 0. ASTERISK-23834
+	  #close Reported by: Richard Kenner ........ Merged revisions
+	  417141 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-23 15:53 +0000 [r417119]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/core_unreal.c: core_unreal: Fix off by one buffer overwrite
+	  error. Appending the ;2 to the user supplied ;1 uniqueid to
+	  create the ;2 version if the user did not also supply an extra
+	  uniqueid for the ;2 channel resulted in allocating a buffer that
+	  was one byte too small. * Fix off by one error in
+	  ast_unreal_new_channels() when generating the ;2 uniqueid from
+	  the user suppled ;1 version. * Pulled some long assignment lines
+	  from if tests to improve line break readability in
+	  ast_unreal_new_channels().
+
+2014-06-22 18:44 +0000 [r416995]  George Joseph <george.joseph at fairview5.com>
+
+	* include/asterisk/astobj2.h, Makefile.rules, Makefile: astobj2:
+	  Add an ao2_replace macro to astobj2.h This macro replaces one
+	  object reference with another cleaning up the original. param dst
+	  Pointer to the object that will be cleaned up. param src Pointer
+	  to the object replacing it. src's ref count is bumped if it's
+	  non-NULL. dst's ref count is decremented if it's non-NULL. src is
+	  assigned to dst, This patch was reviewed on IRC by coreyfarrell
+	  and mjordan. Tested by: George Joseph
+
+2014-06-20 23:16 +0000 [r416871-416931]  George Joseph <george.joseph at fairview5.com>
+
+	* /, configure, include/asterisk/autoconfig.h.in: build: Allow
+	  autoconf/ast_ext_tool_check to handle cross-compiling better.
+	  ast_ext_tool_check.m4 isn't handling cases where a path to a
+	  package is provided (E.G. --with-mysqlclient=/some/sysroot) and
+	  the package has a config tool (E.G. mysql_config) and the package
+	  has its own subdirectories in include or lib. For example,
+	  mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
+	  ast_ext_tool_check sets MYSQLCLIENT_LIB to
+	  ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
+	  includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
+	  directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
+	  fail and there are others in the same boat. The problem is caused
+	  by logic in ast_ext_tool_check that overrides the result of the
+	  config tool's --cflags and --libs options if package_DIR is set.
+	  This patch prepends package_DIR (if specified) to the -L and -I
+	  results from the package's config tool instead of overriding
+	  them. A regenerated ./configure and
+	  include/asterisk/autoconfig.h.in are included but can be
+	  regenerated by running ./bootstrap.sh at any time. Tested by:
+	  George Joseph Tested by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3550/ ........ Merged
+	  revisions 416929 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 416930 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* autoconf/ast_ext_tool_check.m4, /: build: Allow
+	  autoconf/ast_ext_tool_check to handle cross-compiling better.
+	  ast_ext_tool_check.m4 isn't handling cases where a path to a
+	  package is provided (E.G. --with-mysqlclient=/some/sysroot) and
+	  the package has a config tool (E.G. mysql_config) and the package
+	  has its own subdirectories in include or lib. For example,
+	  mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
+	  ast_ext_tool_check sets MYSQLCLIENT_LIB to
+	  ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
+	  includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
+	  directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
+	  fail and there are others in the same boat. The problem is caused
+	  by logic in ast_ext_tool_check that overrides the result of the
+	  config tool's --cflags and --libs options if package_DIR is set.
+	  This patch prepends package_DIR (if specified) to the -L and -I
+	  results from the package's config tool instead of overriding
+	  them. Tested by: George Joseph Tested by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3550/ ........ Merged
+	  revisions 416870 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-20 20:46 +0000 [r416849]  Jonathan Rose <jrose at digium.com>
+
+	* res/parking/parking_manager.c: res_parking: Make manager commands
+	  register with module information Previously module information
+	  was not included due to an oversight. Review:
+	  https://reviewboard.asterisk.org/r/3626/
+
+2014-06-20 15:22 +0000 [r416737-416806]  George Joseph <george.joseph at fairview5.com>
+
+	* main/astobj2_private.h, main/astobj2_container_private.h,
+	  main/astobj2_container.c, main/astobj2_hash.c,
+	  main/astobj2_rbtree.c, build_tools/cflags.xml,
+	  tests/test_astobj2.c: astobj2: Additional refactoring to push
+	  impl specific code down into the impls. Move some implementation
+	  specific code from astobj2_container.c into astobj2_hash.c and
+	  astobj2_rbtree.c. This completely removes the need for
+	  astobj2_container to switch on RTTI and it no longer has any
+	  knowledge of the implementation details. Also adds AO2_DEBUG as a
+	  new compile option in menuselect which controls astobj2 debugging
+	  independently of AST_DEVMODE and REF_DEBUG. Tested by: George
+	  Joseph Review: https://reviewboard.asterisk.org/r/3593/
+
+	* res/res_pjsip_endpoint_identifier_ip.c, main/acl.c,
+	  include/asterisk/netsock2.h, include/asterisk/acl.h,
+	  main/netsock2.c: pjsip cli: Change Identify to show CIDR notation
+	  instead of netmasks. * Added ast_sockaddr_cidr_bits() to count
+	  the 1 bits in an ast_sockaddr. * Added ast_ha_join_cidr() which
+	  uses ast_sockaddr_cidr_bits() for the netmask instead of
+	  ast_sockaddr_stringify_addr. * Changed
+	  res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr()
+	  instead of ast_ha_join() for the CLI output. This is a CLI change
+	  only. AMI was not affected. Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3652/
+
+2014-06-19 19:35 +0000 [r416734]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/sip/reqresp_parser.c, main/logger.c, main/test.c, /,
+	  main/bridge.c, res/parking/parking_tests.c: Fix build warnings
+	  with TEST_FRAMEWORK enabled ........ Merged revisions 416732 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 416733 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-19 16:03 +0000 [r416582-416669]  George Joseph <george.joseph at fairview5.com>
+
+	* /, pbx/pbx_lua.c: Remove the problematic and unneeded
+	  AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c
+	  AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be
+	  incorrectly loaded before pbx_config. pbx_config was therefore
+	  blowing away contexts that were created by pbx_lua. With
+	  AST_MODFLAG_DEFAULT the load order is now correct and contexs are
+	  being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed
+	  anyway since no other modules needed its global symbols that
+	  early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by:
+	  Dennis Guse Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3629/ ........ Merged
+	  revisions 416668 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* /, configs/extensions.lua.sample: Update extensions.lua.sample
+	  with naming conflict guidance. The sample extensions.lua was
+	  causing pbx_lua to fail to load when parsing 'app.goto("default",
+	  "s", 1)' because in Lua 5.2, 'goto' is now a reserved word. This
+	  patch adds guidance to extensions.lua.sample and changed
+	  'app.goto("default", "s", 1)' to 'app.['goto']("default", "s",
+	  1)'. ASTERISK-23844 #close Reported by: rnewton Tested by:
+	  gtjoseph Review: https://reviewboard.asterisk.org/r/3627/
+	  ........ Merged revisions 416581 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-18 04:16 +0000 [r416557]  Matthew Jordan <mjordan at digium.com>
+
+	* main/stasis_channels.c: stasis_channels: Update the stasis cache
+	  if manager variables are needed In r416211, the publishing of
+	  variable changes was modified such that a cached channel snapshot
+	  was used if manager variables were not requested with each AMI
+	  event. This was done to reduce the amount of channel snapshots
+	  created. However, an assumption was made that generating a
+	  channel snapshot and publishing the snapshot to the channel topic
+	  was sufficient to ensure that the cache would be updated; this is
+	  not the case. The channel snapshot type must be used to force a
+	  snapshot update. This patch updates the publication of channel
+	  variables such that the cache is updated prior to publication of
+	  the channel variable message if manager variables are in use.
+	  This ensures that all AMI events receive the variable update when
+	  they are supposed to. Note that this issue was caught by the
+	  Asterisk Test Suite (go go testing)
+
+2014-06-17 18:43 +0000 [r416442-416502]  Mark Michelson <mmichelson at digium.com>
+
+	* /, funcs/func_strings.c: Allow the PUSH and UNSHIFT functions to
+	  set inheritable channel variables. ........ Merged revisions
+	  416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 416501 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* res/res_pjsip_xpidf_body_generator.c,
+	  res/res_pjsip_pidf_body_generator.c: Fix string growth algorithm
+	  for XML presence bodies. pjpidf_print() does not return < 0 if
+	  there is not enough room for the document to be printed. Rather,
+	  it returns 39, the length of the XML prolog. The algorithm also
+	  had a bug in that it would return if it attempted to grow the
+	  string larger.
+
+2014-06-17 16:26 +0000 [r416441]  Kinsey Moore <kmoore at digium.com>
+
+	* /, res/res_musiconhold.c: MoH: Don't restart stream on repeated
+	  start calls Currently, music on hold will stop and then start
+	  again from the beginning if ast_moh_start() is called multiple
+	  times. This can happen if a call is put on hold repeatedly (the
+	  channel receives multiple HOLD control frames) and can be
+	  triggered from ARI by starting MoH on a channel multiple times.
+	  This is fairly jarring/annoying to users. This change prevents
+	  MoH from being restarted if the requested music class is the same
+	  as the one currently playing. This includes an extra check to
+	  prevent the errors previously experienced in the testsuite and
+	  has 100+ test runs behind it. Review:
+	  https://reviewboard.asterisk.org/r/3615/ ........ Merged
+	  revisions 416439 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 416440 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-16 09:02 +0000 [r416338]  Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+	* main/db.c, res/res_config_sqlite3.c, cdr/cdr_sqlite3_custom.c, /,
+	  cel/cel_sqlite3_custom.c: We have faced situation when using CDR
+	  and CEL by sqlite3 modules. With system having high load (~100
+	  concurrent calls created by sipp) we found many cdr and cel
+	  records missed. There is special finction in sqlite3, that make
+	  able to fix this situation - sqlite3_wait_timeout, that also can
+	  replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this
+	  function can be used for aastdb and res_config_sqlite3 to avoid
+	  missed writes to sqlite db. #ASTERISK-23766 #close Reported by:
+	  Igor Goncharovsky Review:
+	  https://reviewboard.asterisk.org/r/3559/ ........ Merged
+	  revisions 416336 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 416337 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-16 02:39 +0000 [r416255-416318]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_sip.c: channels/chan_sip: Forbid remote bridging if
+	  T.38 is negotiated When a framehook is removed - such as the fax
+	  gateway framehook - the bridge framework will re-evaluate the
+	  bridge mixing technologies to see if it can improve the bridging.
+	  When this occurs, get_rtp_info will be called to determine if
+	  local or remote bridging can be used. Using remote bridging will
+	  cause a fax to fail, as direct media negotiation will cause some
+	  small number of packets to not arrive at the remote endpoint.
+	  This patch forces local native bridging if T.38 negotiation is in
+	  progress or has been established.
+
+	* main/channel_internal_api.c: channel_internal_api: Publish a
+	  snapshot change when linkedids change Snapshots are now not
+	  published *quite* as much as they used to. One instance where
+	  they are not published any longer is during bridge enter and exit
+	  - the state of the channel doesn't change, the bridge does.
+	  However, channels are changed when a linkedid is propagated;
+	  previously, the channel's state would be updated and published
+	  during the bridge enter event. Now this must be explicitly done.
+
+	* tests/test_stasis_endpoints.c: test_stasis_endpoints: Remove
+	  expected channel snapshot We no longer publish a channel snapshot
+	  when it is associated with an endpoint; after all, the channel
+	  itself hasn't changed - the endpoint state has changed. This
+	  updates the channel_messages unit test accordingly.
+
+	* res/res_musiconhold.c, /: MoH: Undo commit r416150 (1.8) This
+	  patch reverts r416150. When the comparison between mohclass->name
+	  and state->class->name is made, you are not guaranteed that (a)
+	  state->class is non-NULL or that state or state->class are in a
+	  safe state. Crashes caught by the bridges/transfer_capabilities
+	  test. ........ Merged revisions 416251 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 416252 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-13 18:16 +0000 [r416211]  Matthew Jordan <mjordan at digium.com>
+
+	* main/channel.c, main/dial.c, main/manager.c,
+	  include/asterisk/stasis_channels.h, res/res_agi.c,
+	  res/res_pjsip/pjsip_configuration.c, main/stasis_channels.c,
+	  res/ari/resource_channels.c, main/bridge_channel.c, main/pbx.c,
+	  main/stasis_cache.c, apps/app_meetme.c, main/pickup.c,
+	  main/channel_internal_api.c, include/asterisk/channel.h,
+	  main/core_local.c, main/aoc.c, main/endpoints.c, main/cel.c,
+	  apps/app_queue.c, main/stasis_bridges.c, apps/app_agent_pool.c,
+	  main/cli.c: stasis: Reduce creation of channel snapshots to
+	  improve performance During some performance testing of Asterisk
+	  with AGI, ARI, and lots of Local channels, we noticed that
+	  there's quite a hit in performance during channel creation and
+	  releasing to the dialplan (ARI continue). After investigating the
+	  performance spike that occurs during channel creation, we
+	  discovered that we create a lot of channel snapshots that are
+	  technically unnecessary. This includes creating snapshots during:
+	  * AGI execution * Returning objects for ARI commands * During
+	  some Local channel operations * During some dialling operations *
+	  During variable setting * During some bridging operations And
+	  more. This patch does the following: - It removes a number of
+	  fields from channel snapshots. These fields were rarely used,
+	  were expensive to have on the snapshot, and hurt performance.
+	  This included formats, translation paths, Log Call ID, callgroup,
+	  pickup group, and all channel variables. As a result, AMI Status,
+	  "core show channel", "core show channelvar", and "pjsip show
+	  channel" were modified to either hit the live channel or not show
+	  certain pieces of data. While this is unfortunate, the
+	  performance gain from this patch is worth the loss in behaviour.
+	  - It adds a mechanism to publish a cached snapshot + blob. A
+	  large number of publications were changed to use this, including:
+	  - During Dial begin - During Variable assignment (if no AMI
+	  variables are emitted - if AMI variables are set, we have to make
+	  snapshots when a variable is changed) - During channel pickup -
+	  When a channel is put on hold/unhold - When a DTMF digit is
+	  begun/ended - When creating a bridge snapshot - When an AOC event
+	  is raised - During Local channel optimization/Local bridging -
+	  When endpoint snapshots are generated - All AGI events - All ARI
+	  responses that return a channel - Events in the AgentPool,
+	  MeetMe, and some in Queue - Additionally, some extraneous channel
+	  snapshots were being made that were unnecessary. These were
+	  removed. - The result of ast_hashtab_hash_string is now cached in
+	  stasis_cache. This reduces a large number of calls to
+	  ast_hashtab_hash_string, which reduced the amount of time spent
+	  in this function in gprof by around 50%. ASTERISK-23811 #close
+	  Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3568/
+

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