4594 Commits (3f64ca0c04f3b0482125d99c6b9566a406af48ed)

Author	SHA1	Message	Date
Kinsey Moore	cd6c774456	TEST_FRAMEWORK: Fix threewaytransfer reporting ... Ensure that three-way transfers can be reported even if featuremap is non-NULL. ........ Merged revisions 418810 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fee789dddb	codec_adpcm: Change description of codec "ADPCM" to "Dialogic ADPCM" ... 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	11 years ago
Matthew Jordan	96cbaa187a	manager: Return ActionID on nominal responses to PresenceState action ... 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	11 years ago
Kinsey Moore	ca587079c0	TEST_FRAMEWORK: Fix ref leak in feature activation ... 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	11 years ago
Sean Bright	a1eec851c6	Update Asterisk copyright year in main/asterisk.c ... It's been 2014 for like... 6 months. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	fcdc4ad0bf	astobj2: work around REF_DEBUG race which causes out of order log entries ... * 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	11 years ago
Corey Farrell	f4a30ad32e	Fix minor reference leaks in app_skel and TEST_FRAMEWORK ... * 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	11 years ago
Matthew Jordan	0d1288e2d2	astobj2: Add tag variants for ao2_bump, ao2_cleanup, and ao2_replace ... 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	11 years ago
Scott Griepentrog	6e5d843a56	config: inform config hook of change when writing file ... 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	11 years ago
Richard Mudgett	f962448eee	ARI: Make mixing bridges propagate linkedids and accountcodes. ... * 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	11 years ago
Kinsey Moore	edcaa54019	CEL: Fix incorrect/missing extra field information ... This corrects two issues with the extra field information in Asterisk 12+ in channel event logs. It is possible to inject custom values into the dialstatus provided by ast_channel_dial_type() Stasis messages that fall outside the enumeration allowed for the DIALSTATUS channel variable. CEL now filters for the allowed values and ignores other values. The "hangupsource" extra field key is always blank if the far end channel is a chan_pjsip channel. This is because the hangupsource is never set for the pjsip channel driver. This change sets the hangupsource whenever a hangup is queued for chan_pjsip channels. This corrects an issue with the pjsip channel driver where the hangupcause information was not being set properly. Review: https://reviewboard.asterisk.org/r/3690/ ........ 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	11 years ago
Kinsey Moore	9c589571b7	HTTP: Fix build for gcc 4.10 ... ........ 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	11 years ago
Matthew Jordan	9711bb7b54	main/Makefile: fix compilation error of buildinfo occurring on 'make install' ... 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	11 years ago
Matthew Jordan	0e844b7598	configure: Remove last vestiges of h323; DO create menuselect-deps ... 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	11 years ago
Matthew Jordan	97834718c2	Remove many deprecated modules ... 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	11 years ago
Richard Mudgett	9b10813a2b	res_ari: Fix some off-nominal paths just dropping the HTTP connection. ... * Removed some incorrect newlines on ast_http_error() messages in manager.c. * Removed an incorrect newline in res_ari_channels.c. Addendum to ASTERISK-23552 ........ 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	11 years ago
Jonathan Rose	04a9123309	pbx_config: Add manager actions to add/remove extensions ... 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	11 years ago
Richard Mudgett	dbec5e0d8d	HTTP: Add persistent connection support. ... Persistent HTTP connection support is needed due to the increased usage of the Asterisk core HTTP transport and the frequency at which REST API calls are going to be issued. * Add http.conf session_keep_alive option to enable persistent connections. * Parse and discard optional chunked body extension information and trailing request headers. * Increased the maximum application/json and application/x-www-form-urlencoded body size allowed to 4k. The previous 1k was kind of small. * Removed a couple inlined versions of ast_http_manid_from_vars() by calling the function. manager.c:generic_http_callback() and res_http_post.c:http_post_callback() * Add missing va_end() in ast_ari_response_error(). * Eliminated unnecessary RAII_VAR() use in http.c:auth_create(). ASTERISK-23552 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/3691/ ........ 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	11 years ago
Matthew Jordan	758b13858b	main/tcptls: Add checks for OpenSSL Elliptic Curve support ... 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	11 years ago
Matthew Jordan	eaee92198d	main/tcptls: Add support for Perfect Forward Secrecy ... 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	11 years ago
Matthew Jordan	d1c6a9e69e	main/untils: Prevent potential infinite loop in ast_careful_fwrite ... A loop in ast_careful_fwrite exists that will continually attempt to write to a file stream, even in the presence of EAGAIN/EINTR errors. However, if a connection that uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's call to fflush may return EAGAIN/EINTER along with EOF. A subsequent call to fflush will return EOF but not clear errno, resulting in an infinite loop. This patch clears errno after it is detected and handled the loop, such that any subsequent call to fflush will not get erroneously stuck. Review: https://reviewboard.asterisk.org/r/3704 #ASTERISK-23984 #close Reported by: Steve Davies patches: fflush_loop_fix uploaded by one47 (License 5012) ........ Merged revisions 417797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 417798 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ 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	11 years ago
Joshua Colp	6e60f5d317	Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/branches/11 ... ........ res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP. This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation completes. Configuration options to chan_sip and chan_pjsip have also been added to allow behavior to be tweaked (such as forcing the AVP type media transports in SDP). ASTERISK-22961 #close Reported by: Jay Jideliov Review: https://reviewboard.asterisk.org/r/3679/ Review: https://reviewboard.asterisk.org/r/3686/ ........ 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	11 years ago
Matthew Jordan	af90afd90c	app_voicemail, say: Add support for Japanese Language ... 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	11 years ago
Richard Mudgett	3148dfbd11	event.c: Fix type mismatch errors in ie_maps[]. ... In v12+ the type values from the table are only used by the CEL unit tests. Since the unit tests were only comparing a generated expected event with a real event to see if the ie contents matched and using the same table IE_PLTYPE values to read the event contents, the type mismatches were not detected. ........ 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	12 years ago
Corey Farrell	024316fa3a	Ensure REF_DEBUG records entrys for attempts to ao2_ref an invalid object ... This change ensures that __ao2_ref_debug writes to ref_log when given a non-NULL pointer to an invalid ao2 object. This is to ensure that we record any attempt manipulate references of already freed objects. ASTERISK-23948 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3677/ ........ Merged revisions 417500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 417505 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ 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	12 years ago
Kinsey Moore	6145a57c83	CEL: Add bridge tech to relevant CEL records ... 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	12 years ago
Kinsey Moore	e977b7936b	Bridging: Allow channels to define bridging hooks ... 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	12 years ago
Matthew Jordan	f27074eeb7	udptl: Correct FEC to not consider negative sequence numbers as missing ... When using FEC, with span=3 and entries=4 Asterisk will attempt to repair the packet with sequence number 5, as it will see that packet -4 is missing. The result is Asterisk sending garbage packets that can kill a fax. This patch adds a check to see if the sequence number is valid before checking if the packet is missing. Review: https://reviewboard.asterisk.org/r/3657/ #ASTERISK-23908 #close Reported by: Torrey Searle patches: udptl_fec.patch uploaded by Torrey Searle (License 5334) ........ Merged revisions 417318 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 417320 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ 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	12 years ago
Corey Farrell	4a7a36a0a1	ao2_container node object ignores REF_DEBUG in all places except one ... Almost every reference operation against container node's uses __ao2_alloc or __ao2_ref, thereby preventing ref logging for the nodes. One node reference is released with ao2_t_ref, causing refcounter.py to falsely report skews and leaks for many nodes. ASTERISK-23922 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3670/ ........ 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	12 years ago
Corey Farrell	db6a8a6347	Move eid functions to utils.c, mark netsock.h deprecated ... 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	12 years ago
Richard Mudgett	eb2aded25c	core_unreal: Fix off by one buffer overwrite error. ... Appending the ;2 to the user supplied ;1 uniqueid to create the ;2 version if the user did not also supply an extra uniqueid for the ;2 channel resulted in allocating a buffer that was one byte too small. * Fix off by one error in ast_unreal_new_channels() when generating the ;2 uniqueid from the user suppled ;1 version. * Pulled some long assignment lines from if tests to improve line break readability in ast_unreal_new_channels(). ........ 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	12 years ago
Jonathan Rose	e087ae0c02	Logger: Add manager command 'LoggerRotate' to rotate logger ... 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	12 years ago
Richard Mudgett	86e8ab5ed4	voicemail API callbacks: Extract the sayname API call to its own registerd 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	12 years ago
George Joseph	577632dec9	astobj2: Additional refactoring to push impl specific code down into the impls. ... Move some implementation specific code from astobj2_container.c into astobj2_hash.c and astobj2_rbtree.c. This completely removes the need for astobj2_container to switch on RTTI and it no longer has any knowledge of the implementation details. Also adds AO2_DEBUG as a new compile option in menuselect which controls astobj2 debugging independently of AST_DEVMODE and REF_DEBUG. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3593/ ........ 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	12 years ago
George Joseph	d87f8c429e	pjsip cli: Change Identify to show CIDR notation instead of netmasks. ... * Added ast_sockaddr_cidr_bits() to count the 1 bits in an ast_sockaddr. * Added ast_ha_join_cidr() which uses ast_sockaddr_cidr_bits() for the netmask instead of ast_sockaddr_stringify_addr. * Changed res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr() instead of ast_ha_join() for the CLI output. This is a CLI change only. AMI was not affected. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3652/ ........ 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	12 years ago
Kinsey Moore	bd36288efa	Fix build warnings with TEST_FRAMEWORK enabled ... ........ Merged revisions 416732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416733 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ 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	12 years ago
Matthew Jordan	a965613cf7	stasis_channels: Update the stasis cache if manager variables are needed ... In r416211, the publishing of variable changes was modified such that a cached channel snapshot was used if manager variables were not requested with each AMI event. This was done to reduce the amount of channel snapshots created. However, an assumption was made that generating a channel snapshot and publishing the snapshot to the channel topic was sufficient to ensure that the cache would be updated; this is not the case. The channel snapshot type must be used to force a snapshot update. This patch updates the publication of channel variables such that the cache is updated prior to publication of the channel variable message if manager variables are in use. This ensures that all AMI events receive the variable update when they are supposed to. Note that this issue was caught by the Asterisk Test Suite (go go testing) ........ 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	12 years ago
Igor Goncharovskiy	a1e0a5e4b0	We have faced situation when using CDR and CEL by sqlite3 modules. With system having high load (~100 concurrent calls created by sipp) we found many cdr and cel records missed. There is special finction in sqlite3, that make able to fix this situation - sqlite3_wait_timeout, that also can replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this function can be used for aastdb and res_config_sqlite3 to avoid missed writes to sqlite db. ...	12 years ago
Matthew Jordan	072b61bbed	channel_internal_api: Publish a snapshot change when linkedids change ... Snapshots are now not published *quite* as much as they used to. One instance where they are not published any longer is during bridge enter and exit - the state of the channel doesn't change, the bridge does. However, channels are changed when a linkedid is propagated; previously, the channel's state would be updated and published during the bridge enter event. Now this must be explicitly done. ........ 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	12 years ago
Matthew Jordan	9cc1a8e893	stasis: Reduce creation of channel snapshots to improve performance ... During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ 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	12 years ago
Kinsey Moore	b2012ccb0a	CEL: Expose parking retreiver in extra field ... 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	12 years ago
Richard Mudgett	13e697f8c0	AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages. ... 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	12 years ago
Rusty Newton	9ec5064383	main/pbx - documentation - enhance 'core show hints' and 'core show hint' 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	12 years ago
Richard Mudgett	4ca5745dbe	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. ... 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	12 years ago
Richard Mudgett	71b3c9a749	format.c: Fix misuse of hash container function. ... 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	12 years ago
Scott Griepentrog	d5298f2a1b	CLI: correct presence information on core show hints ... 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	12 years ago
Scott Griepentrog	d7ed0a1ece	CLI: add presence information to core show hints ... 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	12 years ago
Kinsey Moore	2bd6a010a6	Fix build in dev mode due to signed/unsigned mismatch ... ........ 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	12 years ago
Jonathan Rose	a0adb8a26b	PJSIP: PJSIPNotify - Strip content-length headers and add documentation ... 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	12 years ago
Corey Farrell	8da7f0248f	autoservice: stop thread on graceful shutdown ... 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	12 years ago