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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
2014-07-03pbx_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
2014-07-03main/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
2014-06-30Recorded 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
2014-06-30app_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
2014-06-27event.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
2014-06-26CEL: Add bridge tech to relevant CEL recordsKinsey Moore
Add the "bridge_technology" extra field key to BRIDGE_ENTER and BRIDGE_EXIT CEL events to convey the bridge technology in use at the time the record was generated. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-26Bridging: Allow channels to define bridging hooksKinsey Moore
This patch allows the current owner of a channel to define various feature hooks to be made available once the channel has entered a bridge. This includes any hooks that are setup on the ast_bridge_features struct such as DTMF hooks, bridge event hooks (join, leave, etc.), and interval hooks. Review: https://reviewboard.asterisk.org/r/3649/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-26udptl: Correct FEC to not consider negative sequence numbers as missingMatthew Jordan
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 ........ Merged revisions 417324 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-25ao2_container node object ignores REF_DEBUG in all places except oneCorey Farrell
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/ ........ Merged revisions 417212 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-24Move eid functions to utils.c, mark netsock.h deprecatedCorey Farrell
Move eid functions from netsock.c to utils.c. These functions were already published by utils.h. Flag netsock.h as deprecated and switch res_pjsip_session.h to use netsock2.h. The only code that still uses netsock.h is chan_iax2. ASTERISK-23920 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-23core_unreal: Fix off by one buffer overwrite error.Richard Mudgett
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(). ........ Merged revisions 417119 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-20Logger: Add manager command 'LoggerRotate' to rotate loggerJonathan Rose
Part of a series of AMI command equivalents to existing CLI commands Review: https://reviewboard.asterisk.org/r/3651/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-20voicemail API callbacks: Extract the sayname API call to its own registerd ↵Richard Mudgett
callback. * Extract the sayname API call to its own registerd callback. This allows the app_directory and app_chanspy applications to say a mailbox owner's name using an alternate provider when app_voicemail is not available because you are using res_mwi_external. app_directory still uses the voicemail.conf file. AFS-64 #close Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-20astobj2: Additional refactoring to push impl specific code down into the impls.George Joseph
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/ ........ Merged revisions 416806 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-19pjsip cli: Change Identify to show CIDR notation instead of netmasks.George Joseph
* 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/ ........ Merged revisions 416737 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-19Fix build warnings with TEST_FRAMEWORK enabledKinsey Moore
........ 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 ........ Merged revisions 416734 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-18stasis_channels: Update the stasis cache if manager variables are neededMatthew Jordan
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) ........ Merged revisions 416557 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-16We have faced situation when using CDR and CEL by sqlite3 modules. With ↵Igor Goncharovskiy
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 ........ Merged revisions 416338 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-15channel_internal_api: Publish a snapshot change when linkedids changeMatthew Jordan
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. ........ Merged revisions 416300 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13stasis: Reduce creation of channel snapshots to improve performanceMatthew Jordan
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/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13CEL: Expose parking retreiver in extra fieldKinsey Moore
This exposes the retreiver of a parked call under the "retreiver" key of the extra field when this information is available. Review: https://reviewboard.asterisk.org/r/3608/ ........ Merged revisions 416148 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages.Richard Mudgett
ASTERISK-23673 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/ ........ Merged revisions 416066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416067 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416070 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12main/pbx - documentation - enhance 'core show hints' and 'core show hint' ↵Rusty Newton
help text Adds descriptive help text to 'core show hints' and 'core show hint'. The text describes the various columns for the sake of clarity. It takes into account recent changes to the content displayed by the commands https://reviewboard.asterisk.org/r/3604/ and https://reviewboard.asterisk.org/r/3611/. ASTERISK-23764 Review: https://reviewboard.asterisk.org/r/3610/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections.Richard Mudgett
Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-11format.c: Fix misuse of hash container function.Richard Mudgett
The supplied hash function to a container must be idempotent given the object's key value to figure out which container bucket the object belongs in. Returning a random number or the current container count is not idempotent. The "computed hash" value doesn't help find the object later in those cases. * Fixed the format_list container to actually be a list since that is how the container is used. Conceptually, if more than 283 formats were added to the format_list then odd things may have happened before the fix. ........ Merged revisions 415728 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415729 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-11CLI: correct presence information on core show hintsScott Griepentrog
Adds presence to core show hint and changes presence string conversion to use the correct function. ASTERISK-23858 #close Review: https://reviewboard.asterisk.org/r/3611/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-10CLI: add presence information to core show hintsScott Griepentrog
Adds presence state value to output of core show hints. Also reformats the output slightly so it doesn't use as much space as it would otherwise. Was: 1000@demo : SIP/1000 State:Unavailable Watchers 0 Now: 1000@demo : SIP/1000 State:Unavailable Presence:Idle Watchers 0 AFS-53 #close Review: https://reviewboard.asterisk.org/r/3604/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-10Fix build in dev mode due to signed/unsigned mismatchKinsey Moore
........ Merged revisions 415678 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-10PJSIP: PJSIPNotify - Strip content-length headers and add documentationJonathan Rose
Documentation for how to add custom headers/content to notifies created with the PJSIPNotify manager action was a little sparse and it also wasn't vetting application of Content-length headers like its chan_sip equivalent was (so two Content-length headers could be applied... and PJSIP determines the content length anyway, so it just opens people up for error). This patch also flips the variable order so that the variables are interpreted in the same order as they are put in the AMI action. Review: https://reviewboard.asterisk.org/r/3587/ ........ Merged revisions 415658 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-09autoservice: stop thread on graceful shutdownCorey Farrell
This change adds thread shutdown to autoservice for graceful shutdowns only. ast_register_cleanup is backported to 1.8 to allow this. The logger callid is also released on shutdown in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3594/ ........ Merged revisions 415463 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415464 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415465 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-08bridges/bridge_native_rtp: Reconfigure bridge on removal of framehookMatthew Jordan
This patch is a re-do of r414122. When r414122 was merged, a major problem with it was uncovered. UNBRIDGE soft hangup flags have a catastrophic effect on the pbx core if they leak out from the bridge layer: the channel gets hung up. With the number of threads involved in a blind transfer, and with the initial patch, it was likely that this would occur. This caused a large number of test failures This patch is nearly identical with the one proposed in r414122, save for the following changes: - We explicitly clear the UNBRIDGE flag when setting an after goto on a channel in a bridge - Defensively, if we encounter an UNBRIDGE flag in the pbx core, we handle it https://reviewboard.asterisk.org/r/3585/ ........ Merged revisions 415443 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415444 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-06chan_sip: Fix order of variables specified in SIPNotify actionJonathan Rose
Prior to this patch, sequential variables would be ordered in reverse from the order specified in the manager action. Review: https://reviewboard.asterisk.org/r/3588/ ........ Merged revisions 415359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415390 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415410 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-06core uri: Custom uri parsing error when no query parametersKevin Harwell
If using the custom URI parsing code (not external uriparser lib) and there was no query parameters the resulting pointer would be NULL and then an attempt was made to subtract from it. The pointer is now set to a valid value if there is no query parameter(s). Also, in the 'ast_uri_make_host_with_port' function when setting the terminator on the resulting string it was writing it one past the end of allocated memory. It now writes the string terminator appropriately. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-06Split astobj2.c into more maintainable components.George Joseph
Split astobj2.c into the following files to improve maintainability. astobj2.c - object primitives, object primitive misc and initialization code. astobj2_private.h - internal object declarations needed by the containers. astobj2_container.c - generic conainer and container misc code. astobj2_container_hash.c - hash container specific code. astobj2_container_rbtree.c - rbtree container specific code. astobj2_container_private.h - generic container definitions and rtti prototypes. https://reviewboard.asterisk.org/r/3576/ ........ Merged revisions 415317 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-05config: Fix indentation and missing curlies in config_text_file_load().Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-05config: Fix config files not reloading when only an included file changes.Richard Mudgett
The twisted logic determining if a config file should be reloaded was mostly broken and disabled. The incorrect test that ASTERISK-23383 fixed actually reenabled the broken logic. The incorrect test was causing the timestamp to always be cleared which caused config files with includes to always be reloaded. * Made wildcard includes always cause a reload. Determining if a file was deleted cannot be determined without restructuring the cache to determine if any files are missing from the last files actually loaded. Also without refactoring config_text_file_load(), the glob loop couldn't check more than one file for changes anyway. * Made remove the cache entry if the file no longer exists when trying to get its timestamp or it is no longer a regular file. This fixes the corner case where the file was loaded, then deleted, then the config reloaded, then the file restored with the same timestamp, and then the config reloaded again. * Made remove the cache entry include list when actually loading the file. This gets rid of any stale includes the file had from the last time the file was loaded. ASTERISK-23683 #close Reported by: tootai Review: https://reviewboard.asterisk.org/r/3575/ ........ Merged revisions 415225 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415229 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415230 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-05res_http_websocket: Create a websocket clientKevin Harwell
Added a websocket server client in Asterisk. Asterisk has a websocket server, but not a client. The ability to have Asterisk be able to connect to a websocket server can potentially be useful for future work (for instance this could allow ARI to connect back to some external system, although more work would be needed in order to incorporate that). Also a couple of things to note - proxy connection support has not been implemented and there is limited http response code handling (basically, it is connect or not). Also added an initial new URI handling mechanism to core. Internet type URI's are parsed into a data structure that contains pointers to the various parts of the URI. (closes issue ASTERISK-23742) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/3541/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-30TALK_DETECT: A channel function that raises events when talking is detectedMatthew Jordan
This patch adds a new channel function TALK_DETECT that, when set on a channel, causes events indicating the start/stop of talking on a channel to be emitted to both AMI and ARI clients. The function allows setting both the silence threshold (the length of silence after which we decide no one is talking) as well as the talking threshold (the amount of energy that counts as talking). Parameters can be updated on a channel after talk detection has been enabled, and talk detection can be removed at any time. The events raised by the function use a nomenclature similar to existing AMI/ARI events. For AMI: ChannelTalkingStart/ChannelTalkingStop For ARI: ChannelTalkingStarted/ChannelTalkingFinished Review: https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close Reported by: Matt Jordan ........ Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-30main/config.c: AMI action UpdateConfig EmptyCat clears all categoriesMatthew Jordan
When invoking UpdateConfig AMI action with Action set to EmptyCat, Asterisk will make all categories empty in the config but the one requested with a Cat variable. This is due to a bug in ast_category_empty (main/config.c) that makes an incorrect comparison for a category name. This patch corrects the comparison such that only the requested category is cleared. Review: https://reviewboard.asterisk.org/r/3573/ #ASTERISK-23803 #close Reported by: zvision patches: manager.c.diff uploaded by zvision (License 5755) ........ Merged revisions 414880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414881 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414882 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-29PBX: Prevent incorrect hint parsingKinsey Moore
Dynamic and pattern matching hints should not be checked for their last known state until they are instantiated by subscribers. (closes issue AFS-56) Reported by: John Hardin Patch AFS-56-pbx.diff submitted by Matt Jordan (license 6283) ........ Merged revisions 414813 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414859 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414860 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messagesMatthew Jordan
This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-25core_unreal: Prevent double free of core_unreal pvtMatthew Jordan
When a channel is destroyed (such as via ast_channel_release in off nominal paths in core_unreal), it will attempt to free (via ast_free) the channel tech pvt. This is problematic for a few reasons: 1. The channel tech pvt is an ao2 object in core_unreal. Free'ing the pvt directly is no good. 2. The channel tech pvt's reference count is dropped just prior to calling ast_channel_release, resulting in the pvt's destruction. Hence, the channel destructor is free'ing an invalid pointer. This patch keeps the dropping of the reference count, but sets the pvt to NULL on the channel prior to releasing it. This models what would occur if the channel was hung up directly. ........ Merged revisions 414542 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-23Fix signed/unsigned build warningsKinsey Moore
........ Merged revisions 414474 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22ARI: Add ability to raise arbitrary User EventsScott Griepentrog
User events can now be generated from ARI. Events can be signalled with arbitrary json variables, and include one or more of channel, bridge, or endpoint snapshots. An application must be specified which will receive the event message (other applications can subscribe to it). The message will also be delivered via AMI provided a channel is attached. Dialplan generated user event messages are still transmitted via the channel, and will only be received by a stasis application they are attached to or if the channel is subscribed to. This change also introduces the multi object blob mechanism used to send multiple snapshot types in a single message. The dialplan app UserEvent was also changed to use multi object blob, and a new stasis message type created to handle them. ASTERISK-22697 #close Review: https://reviewboard.asterisk.org/r/3494/ ........ Merged revisions 414405 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22res_pjsip_refer: Fix bugs involving Parking/PJSIP/transfersJonathan Rose
PJSIP would never send the final 200 Notify for a blind transfer when transferring to parking. This patch fixes that. In addition, it fixes a reference leak when performing blind transfers to non-bridging extensions. Review: https://reviewboard.asterisk.org/r/3485/ ........ Merged revisions 414400 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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