[svn-commits] bebuild: tag 11.0.0-beta1 r371123 - /tags/11.0.0-beta1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Aug 10 15:16:14 CDT 2012
Author: bebuild
Date: Fri Aug 10 15:16:10 2012
New Revision: 371123
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=371123
Log:
Importing files for 11.0.0-beta1 release.
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tags/11.0.0-beta1/.version (with props)
tags/11.0.0-beta1/ChangeLog (with props)
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+2012-08-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.0.0-beta1 Released.
+
+2012-08-10 20:08 +0000 [r371121] Matthew Jordan <mjordan at digium.com>
+
+ * Created Asterisk 11 branch.
+
+ Because its one greater then 10.
+
+2012-08-10 19:54 +0000 [r371120] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, channels/chan_misdn.c, channels/chan_sip.c,
+ main/channel_internal_api.c, main/features.c,
+ include/asterisk/channel.h, channels/sig_pri.c,
+ funcs/func_callerid.c, main/cli.c: Add private representation of
+ caller, connected and redirecting party ids. This patch adds the
+ feature "Private representation of caller, connected and
+ redirecting party ids", as previously discussed with us (DATUS)
+ and Digium. 1. Feature motivation Until now it is quite difficult
+ to modify a party number or name which can only be seen by
+ exactly one particular instantiated technology channel
+ subscriber. One example where a modified party number or name on
+ one channel is spread over several channels are supplementary
+ services like call transfer or pickup. To implement these
+ features Asterisk internally copies caller and connected ids from
+ one channel to another. Another example are extension
+ subscriptions. The monitoring entities (watchers) are notified of
+ state changes and - if desired - of party numbers or names which
+ represent the involving call parties. One major feature where a
+ private representation of party names is essentially needed, i.e.
+ where a party name shall be exclusively signaled to only one
+ particular user, is a private user-specific name resolution for
+ party numbers. A lookup in a private destination-dependent
+ telephone book shall provide party names which cannot be seen by
+ any other user at any time. 2. Feature Description This feature
+ comes along with the implementation of additional private party
+ id elements for caller id, connected id and redirecting ids
+ inside Asterisk channels. The private party id elements can be
+ read or set by the user using Asterisk dialplan functions. When a
+ technology channel is initiating a call, receives an internal
+ connected-line update event, or receives an internal redirecting
+ update event, it merges the corresponding public id with the
+ private id to create an effective party id. The effective party
+ id is then used for protocol signaling. The channel technologies
+ which initially support the private id representation with this
+ patch are SIP (chan_sip), mISDN (chan_misdn) and PRI
+ (chan_dahdi). Once a private name or number on a channel is set
+ and (implicitly) made valid, it is generally used for any further
+ protocol signaling until it is rewritten or invalidated. To
+ simplify the invalidation of private ids all internally generated
+ connected/redirecting update events and also all
+ connected/redirecting update events which are generated by
+ technology channels -- receiving regarding protocol information -
+ automatically trigger the invalidation of private ids. If not
+ using the private party id representation feature at all, i.e. if
+ using only the 'regular' caller-id, connected and redirecting
+ related functions, the current characteristic of Asterisk is not
+ affected by the new extended functionality. 3. User interface
+ Description To grant access to the private name and number
+ representation from the Asterisk dialplan, the CALLERID,
+ CONNECTEDLINE and REDIRECTING dialplan functions are extended by
+ the following data types. The formats of these data types are
+ equal to the corresponding regular 'non-private' already existing
+ data types: CALLERID: priv-all priv-name priv-name-valid
+ priv-name-charset priv-name-pres priv-num priv-num-valid
+ priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid
+ priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE:
+ priv-name priv-name-valid priv-name-pres priv-name-charset
+ priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr
+ priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag
+ REDIRECTING: priv-orig-name priv-orig-name-valid
+ priv-orig-name-pres priv-orig-name-charset priv-orig-num
+ priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
+ priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type
+ priv-orig-subaddr-odd priv-orig-tag priv-from-name
+ priv-from-name-valid priv-from-name-pres priv-from-name-charset
+ priv-from-num priv-from-num-valid priv-from-num-pres
+ priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid
+ priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag
+ priv-to-name priv-to-name-valid priv-to-name-pres
+ priv-to-name-charset priv-to-num priv-to-num-valid
+ priv-to-num-pres priv-to-num-plan priv-to-subaddr
+ priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
+ priv-to-tag Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2030/
+
+2012-08-10 17:56 +0000 [r371113] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix a comparison that was causing presence
+ tests to fail. A recent change made it so that device state
+ changes that were not actual "changes" would not get reported to
+ subscribers. The problem was that this inadvertently blocked
+ presence updates as well.
+
+2012-08-10 16:49 +0000 [r371059-371091] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: remove ALREADYGONE flag on ooh323 call
+ data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone
+ there really. This indication arrive from asterisk core not h.323
+ stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov
+ Patches: ASTERISK-19308.patch ........ Merged revisions 371089
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371090 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * addons/ooh323c/src/ooGkClient.c, /: Send re-register packets by
+ GRQ (gatekeeper request) interval (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371061 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * addons/ooh323c/src/ooTimer.c: restore calling cb functions by
+ timer expire this was broken in rev 369602
+
+2012-08-10 02:07 +0000 [r371052] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c: Fix pickup extension channel reference error.
+ You cannot unref a pointer and then expect to ref it again later.
+ * Fix potential NULL pointer deref if the call pickup search
+ fails.
+
+2012-08-09 21:35 +0000 [r371036-371043] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: Introdue 'ooh323 show gk' cli command that
+ show status of connection to H.323 Gatekeeper (GkClient state)
+
+ * addons/ooh323c/src/ooGkClient.c, /: Fix to resend GRQ/RRQ if RRJ
+ (registration reject) is received (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094.patch ........ Merged revisions 371011
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371022 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-09 19:22 +0000 [r371030] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c, channels/sig_ss7.c: Use better libss7
+ detection test and move libpri compile test. ........ Merged
+ revisions 371012 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371013 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-09 18:28 +0000 [r371010] Alexandr Anikin <may at telecom-service.ru>
+
+ * /, addons/ooh323c/src/ooh323ep.c: change opening h323 logfile
+ with append mode instead of overwrite ........ Merged revisions
+ 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 370989 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-09 17:40 +0000 [r370987] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_meetme.c: Correct documentation for the MeetMe x flag
+ The documentation for the x flag for MeetMe incorrectly described
+ its function as closing down the conference when the last marked
+ user left. It actually causes the users with that flag to leave
+ the conference when the last marked user exits. The functionality
+ of this flag is not changing. ........ Merged revisions 370985
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 370986 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-09 14:52 +0000 [r370979] Mark Michelson <mmichelson at digium.com>
+
+ * main/pbx.c, channels/chan_sip.c, include/asterisk/pbx.h,
+ channels/sip/include/sip.h: Extend extension state callbacks to
+ have more information. Quote from review board: This patch
+ extends the extension state callbacks so that monitoring channels
+ (as chan_sip) get more information of the devices which are
+ responsible for an extension state change. The additional
+ information is needed by chan_sip to present names/numbers of the
+ caller and callee in an early-state SIP notification. Users of
+ extenstion state callback not interested in the additional
+ information are not affected by the changes. Motivation: to
+ present the involved party's name/number in an early-state
+ nofification (used by the notified device as a pickup offer) one
+ after another so that a user can see which call he will pick up
+ in an undirected pickup. Such a pickup offer to a user shall
+ indicate the same call (number/name-A calls number/name-B) as the
+ call which would be picked up when an undirected pickup is
+ executed. Users interested in additional state info must use the
+ new functions ast_extension_state_add_extended() resp.
+ ast_extension_state_add_destroy_extended() to register an
+ extended state callback. When the callback is registered this
+ way, an extra member device_state_info of struct
+ ast_state_cb_info is passed to the callback in addition to the
+ aggregated extension state. This container holds an object for
+ every device of the monitored extension hint consisting of the
+ device name, the device state and a channel reference to the
+ channel which (presumably) caused the device state. The
+ information is used by chan_sip for early-state notifications.
+ When the state of a device changes and the new state contains
+ AST_EVENT_RINGING, an early-state notification is sent to the
+ subscribed devices with the caller/callee names/numbers of the
+ oldest ringing channel of the monitored extension. The notified
+ user may then invoke a direct pickup, which will pickup exactly
+ this channel. Users of the old non-extended callbacks will only
+ be called when the aggregated state did change (same behavior as
+ before). Users of the extended callback will also be called when
+ the state is unchanged but does contain AST_EVENT_RINGING. That
+ could be the case if two channels are ringing at one device and
+ one of them hangs up, so the aggregated state does not change.
+ This way the monitoring channel can create a new early-state
+ notification with the now ringing party-ids. Review:
+ https://reviewboard.asterisk.org/r/2048 This contribution comes
+ from Guenther Kelleter
+
+2012-08-09 14:36 +0000 [r370978] Jonathan Rose <jrose at digium.com>
+
+ * pbx/pbx_dundi.c, CHANGES: DUNDi: Add CLI commands DUNDi show
+ cache and DUNDi show hints (closes issue ASTERISK-18390) Reported
+ by: Peter Racz Patches: dundi_cli_cache.patch.v2 uploaded by
+ Peter Racz (license #6290)
+ ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by
+ Jonathan Rose (license #6182)
+
+2012-08-08 22:45 +0000 [r370955] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When
+ a channel hangs up while being spied upon and the option to exit
+ the ChanSpy application when the spied on channel hangs up is
+ set, ast_autochan_destroy is not being called and therefore a
+ reference to the spied upon channel is not removed. The symptom
+ being reported was that when using func_group in the dialplan and
+ calling "group show channels" at the cli, the spied upon channel
+ was still being shown while "core show channels" showed that the
+ channel was not up. This patch calls ast_autochan_destroy when a
+ spied upon channel hangs up and the option to exit the ChanSpy
+ application is set, removing the reference to the channel
+ allowing the count for the group that the spied channel was part
+ of to be decremented. (closes issue ASTERISK-17515) Reported by:
+ Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young
+ Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael
+ L. Young (license 5026) ........ Merged revisions 370952 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370954 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-08 22:41 +0000 [r370951-370953] Mark Michelson <mmichelson at digium.com>
+
+ * CHANGES: Move a SIP change up to the other SIP changes in the
+ CHANGES file.
+
+ * main/channel.c, main/pbx.c, main/manager.c, pbx/pbx_spool.c,
+ apps/app_originate.c, include/asterisk/channel.h,
+ include/asterisk/pbx.h, CHANGES, res/res_clioriginate.c: Allow
+ support for early media on AMI originates and call files. This is
+ based on the work done by Olle Johansson on review board. The
+ idea is that the channel specified in an AMI originate or call
+ file is typically not connected to the outgoing extension until
+ the channel has been answered. With this change, an EarlyMedia
+ header can be specified for AMI originates and an early_media
+ option can be specified in call files. With this option set, once
+ early media is received on a channel, it will be connected with
+ the outgoing extension. (closes issue ASTERISK-18644) Reported by
+ Olle Johansson Review: https://reviewboard.asterisk.org/r/1472
+
+2012-08-08 21:22 +0000 [r370943] Terry Wilson <twilson at digium.com>
+
+ * main/manager.c, CHANGES: Add AMI_CLIENT dialplan function
+ Implementation of a dialplan function for checking manager
+ accounts. Right now it only returns the number of logged in
+ sessions for a manager account, but other attributes can be added
+ later. Patch by: Olle Johansson Review:
+ https://reviewboard.asterisk.org/r/421/
+
+2012-08-08 20:47 +0000 [r370927] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp_engine.c: Create the payload type if it does not exist
+ when setting information based on the 'm' line. An rtpmap
+ attribute is not required for defined payload numbers.
+
+2012-08-08 20:32 +0000 [r370926] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Convert sig_analog to use a global
+ callback table.
+
+2012-08-08 20:30 +0000 [r370925] Kinsey Moore <kmoore at digium.com>
+
+ * main/channel.c, /: Do not define a cause that doesn't actually
+ exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there
+ is no cause information. As such, it should not be defined and
+ translatable as a cause. ........ Merged revisions 370923 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370924 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-08 20:17 +0000 [r370887-370902] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Fix the analog dial *0 flash-hook of
+ bridged peer feature. The flash-hook the bridged peer feature now
+ correctly determines if the bridged peer is another chan_dahdi
+ channel, that it is an analog channel, and that it has the
+ correct signaling for an FXO port. It now also flash-hooks the
+ correct channel. ........ Merged revisions 370900 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370901 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Convert sig_pri to use a global callback table.
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
+ Convert sig_ss7 to use a global callback table.
+
+2012-08-07 21:58 +0000 [r370881] Damien Wedhorn <voip at facts.com.au>
+
+ * build_tools/cflags-devmode.xml, channels/chan_skinny.c: Rewrite
+ of skinny debugging. Debugging messages and associated controls
+ only compiled in if configured with --enable-dev-mode. Debug
+ messages provide more detail (including thread id) and are
+ grouped so the user/dev can limit the type of messages displayed.
+ Functionally no real change to chan_skinny. Review:
+ https://reviewboard.asterisk.org/r/2040/
+
+2012-08-07 19:59 +0000 [r370860] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp_engine.c, include/asterisk/rtp_engine.h: Payload and RTP
+ code are must remain separate since in non-Asterisk format cases
+ they differ.
+
+2012-08-07 19:26 +0000 [r370851-370859] Kinsey Moore <kmoore at digium.com>
+
+ * /: Recorded merge of revisions 370858 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add
+ missing AST_CAUSE_* -> text translations ........ Merged
+ revisions 370856 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/channel.c: Add missing AST_CAUSE_* -> text translations A
+ few of these were missing from the list and are necessary for the
+ Who Hung Up? functionality.
+
+2012-08-07 17:47 +0000 [r370832-370845] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp_engine.c: Fix a bug uncovered by the test suite where
+ the RTP payload number was not getting set.
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
+ channels/chan_motif.c, include/asterisk/rtp_engine.h: Reduce
+ memory consumption significantly for users of the RTP engine API
+ by storing only the payloads present and in use instead of every
+ possible one. Review: https://reviewboard.asterisk.org/r/2052/
+
+2012-08-07 12:46 +0000 [r370820-370831] Matthew Jordan <mjordan at digium.com>
+
+ * main/channel.c, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, channels/chan_misdn.c,
+ channels/chan_sip.c, main/channel_internal_api.c,
+ channels/misdn/chan_misdn_config.h, main/features.c,
+ configs/misdn.conf.sample, include/asterisk/channel.h,
+ configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h,
+ channels/misdn_config.c: Add named callgroups/pickupgroups This
+ patch adds named calledgroups/pickupgroups to Asterisk. Named
+ groups are implemented in parallel to the existing numbered
+ callgroup/pickupgroup implementation. However, unlike the
+ existing implementation, which is limited to a maximum of 64
+ defined groups, the number of defined groups allowed for named
+ callgroups/pickupgroups is effectively unlimited. Named groups
+ are configured with the keywords "namedcallgroup" and
+ "namedpickupgroup". This corresponds to the numbered group
+ definitions of "callgroup" and "pickupgroup". Note that as the
+ implementation of named groups coexists with the existing
+ numbered implementation, a defined named group of "4" does not
+ equate to numbered group 4. Support for the named groups has been
+ added to the SIP, DAHDI, and mISDN channel drivers. Review:
+ https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther
+ Kelleter(license #6372)
+
+ * contrib/realtime/mysql/voicemail_data.sql: Revert r370820 That
+ change is wrong, wrong, wrong.
+
+ * contrib/realtime/mysql/voicemail_data.sql: Update the MySQL
+ voicemail_data contrib script to reflect Asterisk 11 changes All
+ voicemails now have a 'msg_id' included in their metadata. The
+ ODBC message storage backend now requires this column; as such,
+ the MySQL contrib script that creates the voicemail_data table
+ has been updated with the appropriate column information.
+
+2012-08-06 15:18 +0000 [r370801] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Improve debug message for temporary
+ outbound proxies. Thanks to Paul Belanger for pointing this out.
+ ........ Merged revisions 370797 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370798 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-03 21:52 +0000 [r370773] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/config_parser.c,
+ channels/sip/include/sip.h: Multiple revisions 370769-370771
+ ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri,
+ 03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a
+ SIP dialstring. This is based on the review request posted by
+ Walter Doekes (referenced lower in the commit message) The main
+ fix here is to treat the IPorHost portion of the dial string as a
+ temporary outbound proxy. This ensures requests get sent to the
+ proper location. Due to the age of the request, some parts were
+ no longer relevant. For instance, the request moved outbound
+ proxy parsing code into a single method. This is done in a
+ previous commit, so it was not necessary to do again. Also, the
+ review request fixed some errors with regards to request routing
+ for CANCEL and ACK requests. This has also been fixed in more
+ recent commits. (closes issue ASTERISK-19677) reported by Walter
+ Doekes Review https://reviewboard.asterisk.org/r/1859 ........
+ r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug
+ 2012) | 3 lines Remove unused variable. ........ r370771 |
+ mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5
+ lines Seriously? Another compilation error fixed. Somebody beat
+ me. ........ Merged revisions 370769-370771 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370772 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-02 15:51 +0000 [r370740] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Fix regression from r370636 When the
+ chan_sip cleanup went in, a typo was included that caused some
+ subscriptions of non-Polycom phones to be limited to the same
+ capabilities as Polycom phones. This resolves the failures in the
+ test suite resulting from this regression.
+
+2012-08-01 19:37 +0000 [r370726] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c: Fix a possible crash due to passing NULL to
+ ast_variables_dup()
+
+2012-08-01 18:52 +0000 [r370720] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: Make astobj2.h not
+ include linkedlists.h. Using astobj2 does not require
+ linkedlists.h be included even though astob2 uses linked lists
+ internally.
+
+2012-08-01 02:26 +0000 [r370699] Kinsey Moore <kmoore at digium.com>
+
+ * /, utils/extconf.c: Revert alloca changes for utils These changes
+ were a tad overzealous in the utils directory. Unfortunately,
+ these don't compile with a "make". ........ Merged revisions
+ 370697 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 370698 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-31 22:28 +0000 [r370681-370691] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: Add headers from SIPAddHeader to
+ outbound REFER requests. This is a patch from kkm from review
+ board. This is useful for adding headers to REFER requests that
+ emanate from a Transfer() dialplan application call. This also
+ fixes some uses of the Referred-by header, removing an extra set
+ of angle brackets. I've modified the reporter's original patch to
+ not require any additions to the sip_refer header and to just
+ remove the referred_by_name from sip_refer since it is no longer
+ needed or used. (closes Issue ASTERISK-17639) reported by Kirill
+ Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff
+ uploaded by Kirill Katsnelson (license #5845) Review:
+ https://reviewboard.asterisk.org/r/1159
+
+ * main/manager.c, configs/manager.conf.sample, CHANGES: Add
+ "setvar" option to manager.conf. With this option set, channel
+ variables can be set on every manager originate. The Variable
+ header can still be used to set additional channel variables for
+ individual calls if desired. This work was completed by Olle
+ Johansson on review board. I have applied the review feedback and
+ am committing it in order to get this into trunk before Asterisk
+ 11 is branched. Review: https://reviewboard.asterisk.org/r/1412
+
+2012-07-31 21:20 +0000 [r370677] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Schedule pokes of registered SIP peers
+ within a given timespan after SIP reload With a large number of
+ SIP peers registered, performing a SIP reload causes a flood of
+ SIP OPTIONS request packets. These are immediately sent out, and,
+ as responses come back, can cause peers to be flagged as 'lagged'
+ due to handling of the many response messages. This fix prevents
+ this "packet storm" and schedules the pokes for a random time.
+ That time varies between 1 ms and the peer's qualify time, or, if
+ the qualify time is unknown, the global qualifyfreq setting. The
+ committed patch has some very small modifications to the patch
+ schmidts wrote for the review. (closes issue ASTERISK-19154)
+ Reported by: Nicolo Mazzon patches: issue19154.patch license
+ #6034 uploaded by schmidts Review:
+ https://reviewboard.asterisk.org/r/1652 ........ Merged revisions
+ 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 370672 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-31 20:33 +0000 [r370664] Russell Bryant <russell at russellbryant.com>
+
+ * main/event.c: Move event cache updates into event processing
+ thread. Prior to this patch, updating the device state cache was
+ done by the thread that originated the event. It would update the
+ cache and then queue the event up for another thread to dispatch.
+ This thread moves the cache updating part to be in the same
+ thread as event dispatching. I was working with someone on a
+ heavily loaded Asterisk system and while reviewing backtraces of
+ the system while it was having problems, I noticed that there
+ were a lot of threads contending for the lock on the event cache.
+ By simply moving this into a single thread, this helped
+ performance *a lot* and alleviated some deadlock-like symptoms.
+ Review: https://reviewboard.asterisk.org/r/2066/
+
+2012-07-31 20:21 +0000 [r370655] Kinsey Moore <kmoore at digium.com>
+
+ * /, main/say.c, main/threadstorage.c, funcs/func_strings.c,
+ channels/chan_iax2.c, main/config.c, channels/chan_dahdi.c,
+ pbx/pbx_spool.c, channels/sig_analog.c, main/strcompat.c,
+ main/features.c, pbx/pbx_ael.c, main/http.c, pbx/pbx_realtime.c,
+ channels/chan_alsa.c, channels/sig_ss7.c, main/db.c,
+ include/asterisk/utils.h, main/pbx.c, funcs/func_cut.c,
+ tests/test_linkedlists.c, funcs/func_channel.c, apps/app_macro.c,
+ apps/app_mixmonitor.c, main/asterisk.c, apps/app_voicemail.c,
+ addons/app_mysql.c, apps/app_meetme.c, apps/app_dictate.c,
+ main/utils.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
+ channels/chan_gtalk.c, res/res_jabber.c,
+ res/res_http_websocket.c, res/ael/pval.c, main/channel.c,
+ main/manager.c, apps/app_osplookup.c, res/res_agi.c,
+ apps/app_minivm.c, main/logger.c, main/app.c,
+ addons/chan_mobile.c, apps/app_while.c, res/res_config_pgsql.c,
+ channels/chan_sip.c, apps/app_festival.c, pbx/pbx_lua.c,
+ channels/sig_pri.c, apps/app_getcpeid.c, funcs/func_global.c,
+ channels/chan_jingle.c, main/tcptls.c,
+ apps/app_directed_pickup.c, main/file.c, main/callerid.c,
+ apps/app_sms.c, main/astmm.c, main/event.c, pbx/pbx_dundi.c,
+ include/asterisk/strings.h, utils/extconf.c, main/dsp.c,
+ addons/res_config_mysql.c: Clean up and ensure proper usage of
+ alloca() This replaces all calls to alloca() with ast_alloca()
+ which calls gcc's __builtin_alloca() to avoid BSD semantics and
+ removes all NULL checks on memory allocated via ast_alloca() and
+ ast_strdupa(). (closes issue ASTERISK-20125) Review:
+ https://reviewboard.asterisk.org/r/2032/ ........ Merged
+ revisions 370642 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370643 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-31 19:57 +0000 [r370644] Mark Michelson <mmichelson at digium.com>
+
+ * CHANGES, pbx/pbx_config.c: Add "dialplan remove context" and
+ modify "dialplan add include" From corruptor's review board
+ posting: "I've noticed that we can remove particular extension
+ from context with dialplan remove extension command but in order
+ to remove all extensions in the context we should delete them on
+ by one. I've created dialplan remove context command which uses
+ ast_context_destroy to destroy the whole context with all
+ extensions. I've created to functions for in pbx_config.c:
+ handle_cli_dialplan_remove_context which actually removes context
+ and complete_dialplan_remove_context which completes input. They
+ are based on other similar functions and pretty trivial but I can
+ be mistaken somewhere. "I've also modified dialplan add include
+ <context2> into <context1>. I've made it similar dialplan add
+ extension ... command. It creates <context1> if it doesn't exist
+ and I've also modified complete_dialplan_add_include and removed
+ check for existance of <context2> because we can include
+ non-existent context into another one. (I usually include empty
+ (non-existent) contexts in advance). Should we raise warning in
+ this case as it's raised while reading extensions.conf? "I use
+ those functions with AMI. I think manager commands should be
+ created in addition to those CLI commands." I've addressed the
+ latest comments on review board and have made some other coding
+ guidelines-related cleanup. I also have modified the CHANGES file
+ to mention these new commands. (closes issue ASTERISK-19292)
+ reported by Andrey Solovyev Patches: dialplan_add_include.patch
+ uploaded by Andrey Solovyev (license #5214)
+ dialplan_remove_context.patch uploaded by Andrey Solovyev
+ (license #5214) Review: https://reviewboard.asterisk.org/r/2042
+
+2012-07-31 19:10 +0000 [r370636] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c, channels/sip/security_events.c,
+ channels/sip/include/sip.h: Clean up chan_sip This clean up was
+ broken out from https://reviewboard.asterisk.org/r/1976/ and
+ addresses the following: - struct sip_refer converted to use the
+ stringfields API. - sip_{refer|notify}_allocate ->
+ sip_{notify|refer}_alloc to match other *alloc functions. -
+ Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not
+ get_pidf_msg_text_body3 but get_content, to match add_content. -
+ get_body doesn't get the request body, renamed to
+ get_content_line. - get_body_by_line doesn't get the body line,
+ and is just a simple if test. Moved code inline and removed
+ function. - Remove camelCase in struct sip_peer peer state
+ variables, onHold -> onhold, inUse -> inuse, inRinging ->
+ ringing. - Remove camelCase in struct sip_request rlPart1 ->
+ rlpart1, rlPart2 -> rlpart2. - Rename instances of pvt->randdata
+ to pvt->nonce because that is what it is, no need to update
+ struct sip_pvt because _it already has a nonce field_. - Removed
+ struct sip_pvt randdata stringfield. - Remove useless (and
+ inconsistent) 'header' suffix on variables in
+ handle_request_subscribe. - Use ast_strdupa on Event header in
+ handle_request_subscribe to avoid overly complicated strncmp
+ calls to find the event package. - Move get_destination check in
+ handle_request_subscribe to avoid duplicate checking for packages
+ that don't need it. - Move extension state callback management in
+ handle_request_subscribe to avoid duplicate checking for packages
+ that don't need it. - Remove duplicate append_date prototype. -
+ Rename append_date -> add_date to match other add_xxx functions.
+ - Added add_expires helper function, removed code that manually
+ added expires header. - Remove _header suffix on
+ add_diversion_header (no other header adding functions have
+ this). - Don't pass req->debug to request handle_request_XXXXX
+ handlers if req is also being passed. - Don't pass req->ignore to
+ check_auth as req is already being passed. - Don't create a
+ subscription in handle_request_subscribe if p->expiry == 0. -
+ Don't walk of the back of referred_by_name when splitting string
+ in get_refer_info - Remove duplicate check for no dialog in
+ handle_incoming when sipmethod == SIP_REFER, handle_request_refer
+ checks for that. Review: https://reviewboard.asterisk.org/r/1993/
+ Patch-by: gareth
+
+2012-07-30 23:26 +0000 [r370565-370598] Richard Mudgett <rmudgett at digium.com>
+
+ * main/test.c: Tweak unit test warning message.
+
+ * funcs/func_presencestate.c, main/test.c: Fix some presence-state
+ unit test typos.
+
+ * apps/app_confbridge.c: DECLINE to load confbridge if the config
+ fails to load.
+
+ * channels/chan_misdn.c, /: Release B channel allocation on error
+ path in chan_misdn. ........ Merged revisions 370563 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370564 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-30 14:52 +0000 [r370548] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Change app_meetme support level
+ to extended from deprecated (closes issue ASTERISK-20134)
+ Reported by: Leif Madsen ........ Merged revisions 370547 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-30 13:45 +0000 [r370534-370541] Russell Bryant <russell at russellbryant.com>
+
+ * tests/test_event.c: Fix ast_event_new unit test. One of my recent
+ commits broke this test. The error was:
+ [test_event.c:event_new_test:214]: Events expected to be
+ identical have different size: 69 != 59 The difference in size
+ occurred because the first event had the EID IE added to the
+ event twice. ast_event_new() now always adds it automatically.
+ Previously it only added it if there were no IEs specified, which
+ was kind of weird.
+
+ * include/asterisk/event_defs.h, res/res_corosync.c, main/event.c:
+ Add a "corosync ping" CLI command. This patch adds a new CLI
+ command to the res_corosync module. It is primarily used as a
+ debugging tool. It lets you fire off an event which will cause
+ res_corosync on other nodes in the cluster to place messages into
+ the logger if everything is working ok. It verifies that the
+ corosync communication is working as expected. I didn't put
+ anything in the CHANGES file for this, because this module is new
+ in Asterisk 11. There is already a generic "res_corosync new
+ module" entry in there so I figure that covers it just fine.
+
+ * addons/app_mysql.c, CHANGES: Allow specifying a port number for
+ the MySQL server. This patch allows you to specify a port number
+ for the MySQL server. It's useful if a MySQL server is running on
+ a non-standard port. Even though this module is deprecated in
+ favor of func_odbc, someone asked for this feature and it seems
+ pretty harmless to add. It has been tested using a number of
+ combinations of with/without a port number specified in the
+ dialplan and changing the port number for mysqld.
+
+2012-07-26 15:31 +0000 [r370510-370518] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c, CHANGES: chan_sip: Add SIPpeerstatus command
+ to AMI This patch was submitted by mnicholson a while back. It
+ adds a new AMI action which allows users to request SIP peer
+ status on demand similar to existing PeerStatus events and to the
+ output you would see from CLI with sip show peer Review:
+ https://reviewboard.asterisk.org/r/1098/
+
+ * /, res/res_agi.c: res_agi: Add message indicating need for \n
+ character in verbose message The while loop responsible for
+ reading AGI messages from a fastAGI service can end up looping
+ indefinitely when an AGI script fails to indicate the end of a
+ message with a \n character. This patch adds an indication that
+ we are expecting a \n character to end the message to make it
+ more clear to users that this is necessary if they are receiving
+ this warning over and over. (issue ASTERISK-20061) Reported by:
+ Eike Kuiper ........ Merged revisions 370494 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370495 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-25 14:27 +0000 [r370481-370488] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/Makefile: Repair editline builds using in-tree editline
+ sources. The previous change to the build system for using a
+ system-provided editline library was missing a crucial include
[... 17812 lines stripped ...]
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