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2014-07-16cel_pgsql, cdr_pgsql, res_config_pgsql: Add PostgreSQL application_name supportMatthew Jordan
This patch adds support for the PostgreSQL application_name connection setting. When the appropriate PostgreSQL module's configuration is set with an application name, the name will be passed to PostgreSQL on connection and displayed in the database's pg_stat_activity view, as well as in CSV logs. This aids in managing which applications/servers are connected to a PostgreSQL database, as well as tracing the activity of those connections. Review: https://reviewboard.asterisk.org/r/3591 ASTERISK-23737 #close Reported by: Gergely Domodi patches: pgsql_application_name.patch uploaded by Gergely Domodi (License 6610) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-15codec_adpcm: Change description of codec "ADPCM" to "Dialogic ADPCM"Matthew Jordan
Technically, ADPCM is a method that can be applied to several codecs. Asterisk's ADPCM codec is the Dialogic ADPCM or VOX codec. See http://en.wikipedia.org/wiki/Dialogic_ADPCM for more information about said codec. Review: https://reviewboard.asterisk.org/r/3744 patches: rb3744.patch uploaded by dennis.guse (License 6513) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-15manager: Return ActionID on nominal responses to PresenceState actionMatthew Jordan
When the PresenceState action is executed, the nominal path fails to include the ActionID in the successful response. This patch adds a call to astman_start_ack, which guarantees that an ActionID (if provided) will be sent back to the AMI client. Unlike the Asterisk 11 and 12 patches, this patch also deprecates the duplicate Message key in the response to the action, replacing it with the key 'PresenceMessage'. Review: https://reviewboard.asterisk.org/r/3776/ ASTERISK-23985 #close ........ Merged revisions 418713 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 418714 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-15TEST_FRAMEWORK: Fix ref leak in feature activationKinsey Moore
This fixes two reference leaks that would occur when TEST_FRAMEWORK was enabled and features were successfully executed. ........ Merged revisions 418715 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-15func_uri: URIENCODE/URIDECODE - allow empty strings as argumentJonathan Rose
Previously these two dialplan functions would issue warnings and return failure when an empty string is used as the argument. Now they will not issue a warning and will successfully return an empty string. ASTERISK-23911 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3745/ ........ Merged revisions 418641 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 418649 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 418650 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-15Update Asterisk copyright year in main/asterisk.cSean Bright
It's been 2014 for like... 6 months. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-14logger.h: Extract DEBUG_ATLEAST() to complement VERBOSITY_ATLEAST().Richard Mudgett
........ Merged revisions 418586 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-14Actually delete the removed files.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-13astobj2: work around REF_DEBUG race which causes out of order log entriesCorey Farrell
* Update refcounter.py to use delta's to track the current reference count. * Use result from internal_ao2_ref to write old_refcount to refs_log. Review: https://reviewboard.asterisk.org/r/3756/ ........ Merged revisions 418504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 418505 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 418506 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-13astobj2: correct define for ao2_t_cleanupScott Griepentrog
This change maps the ao2_t_cleanup() function to the correct debug function so that it can be used. Review: https://reviewboard.asterisk.org/r/3764/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-13Fix minor reference leaks in app_skel and TEST_FRAMEWORKCorey Farrell
* Cleanup games object in app_skel. * Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+). Review: https://reviewboard.asterisk.org/r/3757/ ........ Merged revisions 418465 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 418466 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-13Remove files left behind on removal of h323, jingle and jabber.Corey Farrell
This change removes h323.conf.sample, jingle.h, jabber.h left behind by r3698. Review: https://reviewboard.asterisk.org/r/3755/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-11astobj2: Add tag variants for ao2_bump, ao2_cleanup, and ao2_replaceMatthew Jordan
Tags are useful in hunting down ref imbalances; this patch adds tag variants for these commonly used macros/functions. Review: https://reviewboard.asterisk.org/r/3750/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-11astobj2: tweak ao2_replace to do nothing when it would be a NoOpCorey Farrell
This change causes ao2_replace to do nothing when src == dst. This avoids REF_DEBUG logging when we're not actually doing anything. Review: https://reviewboard.asterisk.org/r/3743/ ........ Merged revisions 418396 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-11config: inform config hook of change when writing fileScott Griepentrog
When updated configuration is written back to the conf file - for example when a user changes their voicemail pin, make sure that any config hook that wants to know of changes is informed. Review: https://reviewboard.asterisk.org/r/3708/ ........ Merged revisions 418366 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 418369 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-10include/asterisk/xmpp.h: Convert indentation to tabsMatthew Jordan
This is a whitespace only change. ........ Merged revisions 418323 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 418324 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-10chan_dahdi/sig_pri: Fix type mismatch in the idledial feature's channel ↵Richard Mudgett
creation. Square pegs in round holes don't work very well. ........ Merged revisions 418261 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 418262 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 418263 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-09ARI: Make mixing bridges propagate linkedids and accountcodes.Richard Mudgett
* Create a Stasis bridge sub-class to propagate linkedids and accountcodes. * Fixed the basic bridge sub-class to update peeraccount codes when the number of channels in the bridge drops back down to two parties. * Refactored ast_bridge_channel_update_accountcodes() to handle channels joining/leaving the bridge. * Fixed the basic bridge sub-class to not call the base bridge class pull method twice. AFS-105 #close ASTERISK-23852 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3720/ ........ Merged revisions 418225 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-08manager/ARI: Update version to 2.4.0/1.4.0; Update UPGRADE.txtMatthew Jordan
........ Merged revisions 418182 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-08res_rtp_asterisk: Fix undefined function when PJPROJECT is not installedMatthew Jordan
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 ........ Merged revisions 418172 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07res_pjsip_dialog_info_body_generator: Add dialog-info+xml support for presence.Joshua Colp
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/ ........ Merged revisions 418116 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07ARI/res_stasis: Subscribe to both Local channel halves when originating to appMatthew Jordan
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 ........ Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07CEL: Fix incorrect/missing extra field informationKinsey Moore
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/ ........ Merged revisions 418071 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07HTTP: Fix build for gcc 4.10Kinsey Moore
........ Merged revisions 418066 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-04main/Makefile: fix compilation error of buildinfo occurring on 'make install'Matthew Jordan
Egads. Another bad deletion of too much when attempting to remove h323 stuff. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-04configure: Remove last vestiges of h323; DO create menuselect-depsMatthew Jordan
The previous patch (r418034) fixed the 'glitch' that the channels/h323 Makefile no longer existed. Unfortunately, removing the entire line was a bit of a blunder, as it meant that build_tools/menuselect-deps was never generated. Hilarity ensued when actually trying to compile. But hey! At least configure worked. This patch fixes *that* glitch, and removes some more of the vestiges of h323. (It had tendrils in the main Makefile? Crazy.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-04configure: Update script to pass if channels/h323/Makefile.in does not existMatthew Jordan
This simply removes that check from the configure script, as r418019 removed chan_h323. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-04Remove many deprecated modulesMatthew Jordan
Billing records are fair, To get paid is quite bright, You should really use ODBC; Good-bye cdr_sqlite. Microsoft did once push H.323, Hell, we all remember NetMeeting. But try to compile chan_h323 now And you will take quite a beating. The XMPP and SIP war was fierce, And in the distant fray Was birthed res_jabber/chan_jingle; But neither to stay. For everyone did care and chase what Google professed. "Free Internet Calling" was what devotees cried, But Google did change the specs so often That the developers were happy the day chan_gtalk died. And then there was that odd application Dedicated to the Polish tongue. app_saycountpl was subsumed by Say; One could say its bell was rung. To read and parse a file from the dialplan You could (I guess) use an application. app_readfile did fill that purpose, but I think A function is perhaps better in its creation. Barging is rude, I'm not sure why we do it. Inwardly, the caller will probably sigh. But if you really must do it, Don't use app_dahdibarge, use ChanSpy. We all despise the sound of tinny robots It makes our queues so cold. To control such an abomination It's better to not use Wait/SetMusicOnHold. It's often nice to know properties of a channel It makes our calls right We have a nice function called CHANNEL And so SIPCHANINFO is sent off into the night. And now things get odd; Apparently one could delimit with a colon Properties from the SIPPEER function! Commas are in; all others are done. Finally, a word on pipes and commas. We're sorry. We can't say it enough. But those compatibility options in asterisk.conf; To maintain them forever was just too tough. This patch removes: * cdr_sqlite * chan_gtalk * chan_jingle * chan_h323 * res_jabber * app_saycountpl * app_readfile * app_dahdibarge It removes the following applications/functions: * WaitMusicOnHold * SetMusicOnHold * SIPCHANINFO It removes the colon delimiter from the SIPPEER function. Finally, it also removes all compatibility options that were configurable from asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems. Review: https://reviewboard.asterisk.org/r/3698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03chan_dahdi: Add inband_on_setup_ack compatibility option.Richard Mudgett
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 ........ Merged revisions 417958 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03res_ari: Fix some off-nominal paths just dropping the HTTP connection.Richard Mudgett
* 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 ........ Merged revisions 417932 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03chan_dahdi: Add AMI commands for controlling PRI debugging outputJonathan Rose
Adds the following AMI commands: PRIDebugSet - Set PRI debug levels for a specific span PRIDebugFileSet - Set the file used for PRI debug message output PRIDebugFileUnset - Disables file output for PRI debug messages Review: https://reviewboard.asterisk.org/r/3681/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03pbx_config: Add manager actions to add/remove extensionsJonathan Rose
Adds two new manager commands to pbx_config - DialplanExtensionAdd and DialplanExtensionRemove which allow manager users to create and delete extensions respectively. Review: https://reviewboard.asterisk.org/r/3650/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03HTTP: Add persistent connection support.Richard Mudgett
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/ ........ Merged revisions 417880 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03main/tcptls: Add checks for OpenSSL Elliptic Curve supportMatthew Jordan
The patch for ASTERISK-23905 that added PFS support in Asterisk depends on the elliptic curve library support being present in OpenSSL. As it turns out, some versions of OpenSSL don't have this library - notably the version running on our build agents. This patch fixes the build by providing a configure check for the specific library calls that the PFS patch relies on. Review: https://reviewboard.asterisk.org/r/3709/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03ARI: Improvements to body parameters documentationSam Galarneau
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/ ........ Merged revisions 417878 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03api.wiki.mustache: Update wiki template to support body parametersSam Galarneau
This patch updates the api.wiki.mustache template and the swagger_model python script to understand if an operation has a body parameter. If an operation does have a body parameter, it will now be displayed in the corresponding wiki entry. ........ Merged revisions 407389 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03dahdi_span_config_hook: automatically register new dahdi channelsTzafrir Cohen
Install a hook script for DAHDI to register new spans with Asterisk automatically by running: asterisk -rx 'dahdi create channel FIRST LAST' Review: https://reviewboard.asterisk.org/r/3157/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03main/tcptls: Add support for Perfect Forward SecrecyMatthew Jordan
This patch enables Perfect Forward Secrecy (PFS) in Asterisk's core TLS API. Modules that wish to enable PFS should consider the following: - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not specify a ECDHE cipher suite in a module's configuration, for example: tlscipher=AES128-SHA:DES-CBC3-SHA - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters into the private key file, i.e., tlsprivatekey. For an example, see the default dh2048.pem at http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt - Because clients expect the server to prefer PFS, and because OpenSSL sorts its cipher suites by bit strength, (see "openssl ciphers -v DEFAULT") consider re-ordering your cipher suites in the conf file. For example: tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH will use PFS when offered by the client. Clients which do not offer PFS fall-back to AES-128 (or even 3DES as recommend by RFC 3261). Review: https://reviewboard.asterisk.org/r/3647/ ASTERISK-23905 #close Reported by: Alexander Traud patches: tlsPFS_for_HEAD.patch uploaded by Alexander Traud (License 6520) tlsPFS.patch uploaded by Alexander Traud (License 6520) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03main/untils: Prevent potential infinite loop in ast_careful_fwriteMatthew Jordan
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 ........ Merged revisions 417799 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-02ARI: Remove unnecessary \briefs from automatically generated documentationJonathan Rose
Review: https://reviewboard.asterisk.org/r/3440/ ........ Merged revisions 412653 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-01res_rtp_asterisk: Don't leak memory or reset state if DTLS configuration is ↵Joshua Colp
set multiple times. ........ Merged revisions 417705 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-30Recorded merge of revisions 417677 from ↵Joshua Colp
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/ ........ Merged revisions 417678 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-30Reverse logic during subscription persistence recreation.Mark Michelson
In the abstraction effort, this bit of logic got messed up. We want to recreate the persistence if things go well, not if things fail. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-30apps/app_voicemail: Fix compilation error introduced in r417591Matthew Jordan
Not sure why that change to ast_channel_alloc was made but ... okay. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-30app_voicemail, say: Add support for Japanese LanguageMatthew Jordan
This patch adds support for the Japanese language to both the say family of applications, as well as for VoiceMail and VoiceMailMain. A new pack of language sounds will be released at the same time as the next major version of Asterisk to support the new language features. The language features can be enabled using a language code of 'ja'. Review: https://reviewboard.asterisk.org/r/3477 ASTERISK-23324 #close Reported by: Kevin McCoy patches: app_voicemail.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586) say.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-30chan_sip: be more tolerant of whitespace between attributes in SDP fmtp lineMatthew Jordan
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 ........ Merged revisions 417589 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-27event.c: Fix type mismatch errors in ie_maps[].Richard Mudgett
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. ........ Merged revisions 417565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-27Ensure REF_DEBUG records entrys for attempts to ao2_ref an invalid objectCorey Farrell
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 ........ Merged revisions 417509 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-27refcounter.py: prevent use of excessive RAM with large refs logsCorey Farrell
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 ........ Merged revisions 417483 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-27res_pjsip: Add ActionID to events created as a result of PJSIP AMI actionsMatthew Jordan
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/ ........ Merged revisions 417460 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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