asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	4e57652651	Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 \| mjordan \| 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) \| 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	94dc903e12	Add SLA to skinny. Adds sublines to skinny lines. Each subline can be attached to an SLA station/trunk combo. Includes the following functionality: Callid is persistent for both in/out calls on all skinny devices. Can join, hold, resume. All sublines appear under a single line button. See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ee2096fe55	Make hint watcher callback take const strings for context and exten parameters. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7caa2349af	Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 \| lmadsen \| 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) \| 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	7df5d0d416	Add autoanswer to skinny. Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER. Initial value must be the time to autoanswer in ms, then optionally :BEEP to play a tone when answered and :MUTE to mute the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and play a beep. just 3000 would answer afer 3 secs of ringing with no beep and full two way audio. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	9598e5bc2f	Remove skinny do_monitor and use ast_sched_start instead The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything. Review: https://reviewboard.asterisk.org/r/1256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	0bb90a372e	Remove extraneous line variables. The vars were either explicitly or implicitly not used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	069b70c522	Fix up skinny hints. Probably haven't been working for a couple of years. May still need some more love, but they are now working, both as a hint device and monitoring a hint. Changes centre around the long ago change to remove the requirement for a device name in a skinny line, and changes to the transmit_* functions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	969a317d81	Add activatesub and dialandactivate sub. When called, activatesub first cleans up the active sub and then handles the sub passed. dialandactivatesub first sets sub->exten and then calls activatesub. Revise handle_offhook to utilise the callid sent to chan_skinny. Some other minor fixes especially around d->hookstate (which still needs some more work). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	54bb8a0ca8	Move exten used for dialing from device to subchannel. There were some issues where if a simple switch was cancelled and a new switch started before the first had timed out where the d->exten would be used for both subchannels. This was bad leading to possible invalid extensions if some digits had been entered in the abandoned simple switch and the second one was completed before the first timed out, or the second would be cancelled because d->exten would be set to nothing on the time out of the first. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	c37c017781	Consolidate setsubstate_* into setsubstate and use a switch. Consolidate the functions and add some debugging info. Allows to be able to set a substate without explicitly knowing what the state is. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	bdbb3a506f	Add setsubstate_onhook. Add the setsubstate_onhook to complete the initial substate handling procedures. Added dumpsub(sub, forcehangup) which is the common way of calling setsubstate_onhook. Dumpsub attempts to activate another sub after setting the current one onhook. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	7002adcb3e	Add setsubstate_callwait. If a call is made to a line that already has a call and the device is offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	8c0b1115cd	Only allow voicemail if substate is OFFHOOK or no channel active (UNSET). (closes issue #17901) Reported by: salecha git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	a9beb8323e	Rename sub->parent to sub->line. Improve readability of code, eg, (sub->parent == d->activeline) becomes (sub->line == d->activeline). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	bc61836c1b	Move the hookstate from line to device. Long time coming, finally moving the hookstate from line to device. This may fix some issues where a device has multiple lines. Previously we had to run through all lines on a device to see if it was actually onhook or not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	0938974902	Merged revisions 317478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317478 \| russell \| 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) \| 12 lines Fix some consistency issues with jitterbuffer config. Store the defaults noted in the sample config files in the jitterbuffer config data structure. This makes the CLI commands that output these settings show the right thing. Also only show the settings that are relevant in the settings CLI commands, based on which jitterbuffer is selected and whether it's enabled. (closes issue #19083) Reported by: rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	e98ac1f0f4	Move hold stuff to the setsubstate arrangement. skinny_hold moved to setsubstate_hold and skinny_unhold integrated into setsubstate_connected. Removed sub->onhold and replaced with SUBSTATE_HOLD. Also fixed inbound call answering by queueing an AST_CONTROL_ANSWER on answering a SUBSTATE_RINGIN sub (was a typo). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	2ae06c3e6d	Add setsubstate_congestion and setsubstate_progress. Move handling of both state handling from skinny_indicate to it's own sub. Also, modified behaviour to not hangup the sub and let the dialplan have a chance in doing what it wants for congestion. Added various states to substate2str and added these states where applicable for other set_substate_ procs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317026 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	468b8229a7	Add setsubstate_busy. Move handling of setting busy state from skinny_indicate to it's own sub. Also, modified behaviour to not hangup the sub and let the dialplan have a chance in doing what it wants (eg busy(10); hangup() in the dialplan now gives a busy indication for 10 secs and then hangs up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	b9e763ecae	Add setsubstate_ringout (equivalent to AST_STATE ringing). Renamed previous setsubstate_ringout to setsubstate_dialing for a state when attempting to dial a number, substate ringout now for when core has indicated that the channel is actually ringing on the other end. Also added substate2str for debugging purposes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	99d0da2a2d	Add setsubstate_ringin. Added setsubstate_ringin. skinny_call now calls sss_ringin rather than inline. Fixed previous issue so that setsubstate_connected now use SUBSTATE_RINGIN to determine is an AST_CONTROL_ANSWER should be queued. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	bc814dc84a	Make skinny_answer use setsubsate_connected. Cosolidated the code so that skinny_answer now uses the setsubstate procedures rather than doing the handling inline. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	c6f189cd71	Cleanup skinny callinfo. Cosolidated the working out of the callinfo to be sent into transmit_callinfo. Replaced ambiguous sub->outgoing with calldirection which can be SKINNY_INCOMING or SKINNY_OUTGOING (same value as the skinny protocol). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	95561bd37a	Merged revisions 316336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316336 \| russell \| 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011) \| 8 lines Use htons() instead of ntohs() in some places. (closes issue #19200) Reported by: wdoekes Patches: issue19200-trunk.patch uploaded by wdoekes (license 717) issue19200-1.8.x.patch uploaded by wdoekes (license 717) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 \| russell \| 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) \| 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	079e794b1c	Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 \| mnicholson \| 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) \| 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 \| mnicholson \| 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) \| 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 \| mnicholson \| 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) \| 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	6b01eb3324	Consolidate all new call calls to run through new setsubstate_ringout. (closes issue #17907) Reported by: wedhorn Patches: cleanup.stateringout.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	d760e81f37	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	cc0b7e7df5	Some scheduler API cleanup and improvements. Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	27fbd5e156	Merged revisions 287643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287643 \| qwell \| 2010-09-20 16:29:46 -0500 (Mon, 20 Sep 2010) \| 15 lines Merged revisions 287642 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r287642 \| qwell \| 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) \| 8 lines Don't crash when parking a non-bridged call. (closes issue #17680) Reported by: jmhunter Patches: chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by: jmhunter, DEA ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	179ba271d0	Ignore redial hard button when no previous number. (closes issue #17887) Reported by: salecha Patches: skinny.redial.diff uploaded by wedhorn (license 30) Tested by: wedhorn, salecha git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	db994dbc6c	Hack to allow easy debugging of skinny in trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	530be85aad	Add additional AST_CONTROL_ states to control2str. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	101673347f	Fixes display issues on 7910 and older phones. Also correct the callinfo provided in skinny_answer. (closes issue #17876) Reported by: salecha Patches: skinny_cnd3.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	0e5b6069f4	Cleanup: consolidate offhook (new call). Consolidates all offhook (new call with dialtone) to setsubstate_offhook. This should be roughly equivalent to existing code, although a couple of calls now run through the full offhook sequence rather than an abbreviated one. (closes issue #17812) Reported by: wedhorn Patches: cleanup.stateoffhook.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	f8d64f8614	Fix up handling and indications during transfer. Cleaned up handling of onhook indications and added indications if more than one sub on device. Also fixes issue in 12324 so that the phone can call itself without locking up. (closes issue #17692) Reported by: jmhunter Patches: chan_skinny-transfer-v4.txt uploaded by DEA (license 3) skinnytransfver.v8.diff uploaded by wedhorn (license 30) Tested by: jmhunter, salecha, wedhorn Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	394fa75a0a	Move call answering stuff into new setsubstate_connected. Move call answering stuff into new setsubstate_connected. Also add sub->substate var and set it to SUBSTATE_CONNECTED in setsubstate_connected. (closes issue #17772) Reported by: wedhorn Patches: cleanup.stateconnected2.diff uploaded by wedhorn (license 30) Tested by: wedhorn, salecha Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	dcb865f68a	Start rtp on answer before the answer is queued (closes issue #17770) Reported by: salecha Patches: skinny.answercrash.diff uploaded by wedhorn (license 30) Tested by: salecha Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	a352ff7f60	Cleanup transmit_ for handle_register and keepalives Moved inline packet sending to transmit_ subs. Removed handle_keep_alive and handle_register_message to inline in handle_message. Also moved transmit_response(d) to transmit_response_bysessions(s) and created a wrapper transmit_response(d) that calls transmit_response_bysession(d->session). (closes issue #16980) Reported by: wedhorn Patches: skinny-clean06b.diff uploaded by wedhorn (license 30) Tested by: wedhorn, DEA Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	6fa79e8f77	Make ACLs IPv6-capable. ACLs can now be configured to match IPv6 networks. This is only relevant for ACLs in chan_sip for now since other channel drivers do not support IPv6 addressing. However, once those channel drivers are outfitted to support IPv6 addressing, the ACLs will already be ready for IPv6 support. https://reviewboard.asterisk.org/r/791 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cf7bbcc4c6	Expand the caller ANI field to an ast_party_id Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char tag; }; struct ast_party_dialed { struct { char str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	cd4ebd336f	Add IPv6 to Asterisk. This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Michiel van Baak	f4617ae2b5	Ignore Redial softkey when no previous dialed number is known (closes issue #17126) Reported by: wedhorn Patches: skinny79xx_redial1.diff uploaded by wedhorn (license 30) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Michiel van Baak	cc89bc0a27	Cleanup transmit_* functions Bulk lot of generally trivial changes for cleaning up the transmit stuff. Line state request has been modified for line only responses. (closes issue #16994) Reported by: wedhorn Patches: skinny-clean07.diff uploaded by wedhorn (license 30) Tested by: wedhorn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	68d1ded8dd	Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

1 2 3 4 5 ...

375 Commits (04224c20362f94f99886d6b51adb7b90db62855f)