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Author	SHA1	Message	Date
Scott Griepentrog	0a99e4099b	astobj2: assert on invalid ref and backtrace cleanup ... If a reference count goes negative, instead of just logging that fact, be more helpful with a backtrace and an assert that will DO_CRASH. This patch also removes the duplicate ao2_bt() function and cleans up extraneous usage of the ast_log_backtrace() call. Review: https://reviewboard.asterisk.org/r/3765/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	f91989d44e	media formats: fix ref leak of peer for mwi subscription ... Holding a reference to the peer during mwi subscriptions resulted in a circular reference because the final event message would not be sent until destruction of the peer. Instead, pass the name of the peer to the event callback so that it can fail gracefully after the peer has gone. ASTERISK-23959 Review: https://reviewboard.asterisk.org/r/3754/ ........ Merged revisions 418636 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	3ad198c835	feature_config: insure featuregroups and applicationmaps are initialized ... If the features.conf is missing, the cfg->featurgroups and cfg->applicationmaps is not initialized, resulting in assert on ao2_find of a null container. This patch changes the initialization call and adds asserts for a safeguard. Review: https://reviewboard.asterisk.org/r/3809/ ........ Merged revisions 418886 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418961 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b71be2112e	func_audiohookinherit.c: Fixup some XML documentation wording. ... ........ Merged revisions 418937 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	af4cd65143	Channels: Masquerades to automatically move frame/audio hooks ... Whenever possible, audiohooks and framehooks will now be copied over to the channel that the masquerading channel gets cloned into. This should occur for all audiohooks and most framehooks. As a result, in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now deprecated and its behavior is essentially the new default for all audiohooks, plus some additional audiohooks/framehooks. Review: https://reviewboard.asterisk.org/r/3721/ ........ Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	5c988cc4e6	res_fax: Provide AMI equivalents for fax CLI commands ... Specifically the following equivalents were created: fax show session -> FAXSession fax show sessions -> FAXSessions fax show stats -> FAXStats Review: https://reviewboard.asterisk.org/r/3666/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	dd23637195	Update config.guess and config.sub ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	c1f52a5c8d	Add missing file from previous commit. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	84b9f5eff5	Import Asterisk's autoconf magic instead of using our own. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fc0fecb476	configs: Move sample config files into a subdirectory of configs ... This moves all samples configs from configs/ to configs/samples. This allows for additional sets of sample configuration files to be added in the future. Review: https://reviewboard.asterisk.org/r/3804/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1ce23d4534	chan_sip: Make progressinband=never really mean 'never' ... progressinband=never in sip.conf is easily defeated if an onward trunk sends a progress indication of its own. This is almost certain to happen if the onward trunk is ISDN or IAX as these technologies send a progress indication even if early media is not required. This progress message is passed to the caller, and causes the "never" option to be rather badly named. This patch changes the behaviour of this setting in the following ways: 1) In sip_write(), do not pass the media unless we have either progressed beyond INV_EARLY_MEDIA, or we are in INV_EARLY_MEDIA state, and early media is both set-up and wanted. This helps resolve double-ringing on some buggy handsets. 2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, but SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to avoid implicitly enabling early media. Avoid sending double ring indications. NOTE: the meaning of the SIP_PROGRESS_SENT flag changes slightly in this patch to also encapsulate the fact that a channel has *sent or received* a 183 Progress indication. This makes the updated code in sip_write() much more simple. Review: https://reviewboard.asterisk.org/r/3700 ASTERISK-23972 #close Reported by: Steve Davies patches: inband_never_present_early_media2 uploaded by Steve Davies (License 5012) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	96e5e491fa	Add svn:ignore property ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3f64ca0c04	configure: Fix libxml2 development library dependency checking ... The commit that added libxml2 support didn't fully check for the libxml2 development script in the Asterisk configure file. As a result, Asterisk could be configured, then fail on menuselect. This patch fixes it so that Asterisk should detect the libxml2 dependency failure first. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	26c7e684ea	menuselect: Add libxml2 support (Patch 3) ... This is the final patch in adding menuselect to Asterisk. - The first patch (r418832) added menuselect along with mxml - The second patch (r418833) removed mxml from menuselect This patch adds support for libxml2 to menuselect, and makes libxml2 a required library for Asterisk. Note that the libxml2 portion of this patch was written by Sean Bright, and was made available on a team branch: http://svn.digium.com/svn/menuselect/team/seanbright/libxml2/ Review: https://reviewboard.asterisk.org/r/3773/ ASTERISK-20703 #close patches: some_mysterious_team_branch uploaded by seanbright (License 5060) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	62f5e26d35	menuselect: Remove mxml from menuselect (Patch 2) ... This is the second patch that adds menuselect to Asterisk trunk. The previous commit (r418832) added menuselect along with mxml; this patch removes mxml completely from Menuselect. A subsequent patch will switch menuselect over to using libxml2, and make libxml2 a required dependency for Asterisk. ASTERISK-20703 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c7d3570248	menuselect: Add menuselect to Asterisk trunk (Patch 1) ... This is the first patch that adds menuselect to Asterisk trunk, and removes the svn:externals property. This is being done for two reasons: (1) The removal of external repositories eases a future migration to git (2) Asterisk is now the only thing that uses menuselect; as a result, there's little need to keep it in an external repository Subsequent patches will remove the mxml dependency from menuselect and tidy up the build system. ASTERISK-20703 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
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
Corey Farrell	7b7132710b	Remove include of astobj.h from channels/dahdi/bridge_native_dahdi.c. ... The include was unneeded, this is split off from r3758 as it applies to 12. ........ Merged revisions 418787 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fd94fea599	res_pjsip: Support setting a default accountcode on endpoints ... Most channel drivers let you specify a default accountcode to be set on channels associated with a particular peer/endpoint/object. Prior to this patch, chan_pjsip/res_pjsip did not support such a setting. This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'. When a channel is created that is associated with an endpoint with this value set, the channel will automatically have its accountcode property set to the value configured for the endpoint. Review: https://reviewboard.asterisk.org/r/3724/ ASTERISK-24000 #close Reported by: Matt Jordan ........ Merged revisions 418756 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	03e9c598e5	cel_pgsql, cdr_pgsql, res_config_pgsql: Add PostgreSQL application_name support ... 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	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
Jonathan Rose	4420ce6e5c	func_uri: URIENCODE/URIDECODE - allow empty strings as argument ... 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	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
Richard Mudgett	874f075025	logger.h: Extract DEBUG_ATLEAST() to complement VERBOSITY_ATLEAST(). ... ........ 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	11 years ago
Richard Mudgett	4339183c3e	Actually delete the removed files. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418566 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
Scott Griepentrog	3e245920d8	astobj2: correct define for ao2_t_cleanup ... 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	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
Corey Farrell	6461d90d8a	Remove files left behind on removal of h323, jingle and jabber. ... 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	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
Corey Farrell	694b68e544	astobj2: tweak ao2_replace to do nothing when it would be a NoOp ... 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	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
Matthew Jordan	ded0d16174	include/asterisk/xmpp.h: Convert indentation to tabs ... 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	11 years ago
Richard Mudgett	d834be9faf	chan_dahdi/sig_pri: Fix type mismatch in the idledial feature's channel 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	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
Matthew Jordan	5a3023a114	manager/ARI: Update version to 2.4.0/1.4.0; Update UPGRADE.txt ... ........ 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	11 years ago
Matthew Jordan	3126d18c1b	res_rtp_asterisk: Fix undefined function when PJPROJECT is not installed ... The dtls_perform_handshake function was mistakenly placed under the guards for USE_PJPROJECT. If PJPROJECT was not installed, the function would not be defined, while other functions would attempt to still use it. This prevented res_rtp_asterisk from being loaded. ASTERISK-24001 #close Reported by: Don Fanning ........ 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	11 years ago
Joshua Colp	534ffd8481	res_pjsip_dialog_info_body_generator: Add dialog-info+xml support for presence. ... This module implements dialog-info+xml for the purposes of presence. This means that phones such as Grandstreams can now subscribe to receive presence information for an extension. ASTERISK-21443 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3705/ ........ 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	11 years ago
Matthew Jordan	d4b436d0ea	ARI/res_stasis: Subscribe to both Local channel halves when originating to app ... This patch fixes two bugs: 1. When originating a channel into a Stasis application, we already create a subscription for the channel that is going into our Stasis app. Unfortunately, when you create a Local channel and pass it off to a Stasis app, you really aren't creating just one channel: you're creating two. This patch snags the second half of the Local channel pair (assuming it is a Local channel pair, but luckily core_local is kind about such assumptions) and subscribes to it as well. 2. Subscriptions are a bit sticky right now. If a subscription is made, the 'interest' count gets bumped on the Stasis subscription - but unless something explicitly unsubscribes the channel, said subscription sticks around. This is not much of a problem is a user is creating the subscription - if they made it, they must want it. However, when we are creating implicit subscriptions, we need to make sure something clears them out. This patch takes a pessimistic approach: it watches the cache updates coming from Stasis and, if we notice that the cache just cleared out an object, we delete our subscription object. This keeps our ao2 container of Stasis forwards in an application from growing out of hand; it also is a bit more forgiving for end users who may not realize they were supposed to unsubscribe from that channel that just hung up. Review: https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close ........ 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	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	46b00b9c64	configure: Update script to pass if channels/h323/Makefile.in does not exist ... 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	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	3bd495a688	chan_dahdi: Add inband_on_setup_ack compatibility option. ... The new inband_on_setup_ack option causes Asterisk to assume inband audio may be present when a SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a dialtone is sent from the network side, progress indicator 8 "Inband info now available" MAY be sent to the CPE if no digits were received with the SETUP. It is thus implied that the ie is mandatory if digits came with the SETUP and dialtone is needed. This option should be enabled, when the network sends dialtone and you want to hear it, but the network doesn't send the progress indicator when needed. NOTE: For Q.SIG setups this option should be enabled when outgoing overlap dialing is also enabled because Q.SIG does not send the progress indicator with the SETUP ACK. The commit -r413714 (AST-1338) which causes this issue was dealing with a SIP-to-ISDN interoperability issue. This commit is a merge of the two patches indicated below. ASTERISK-23897 #close Reported by: Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded by Pavel Troller jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3633/ ........ Merged revisions 417956 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 417957 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ 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	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	bc4b236d71	chan_dahdi: Add AMI commands for controlling PRI debugging output ... 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	11 years ago