290 Commits (b75433a9e6139a40a477e7ea1b74c9f21a9e6ee6)

Author	SHA1	Message	Date
Richard Mudgett	b75433a9e6	Merged revisions 280235 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280235 | rmudgett | 2010-07-28 15:12:16 -0500 (Wed, 28 Jul 2010) | 9 lines Merged revisions 280229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28 Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	538d044aca	Merged revisions 279916 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010) | 12 lines Fix inband DTMF detection on outgoing ISDN calls. This is a regression from the sig_pri split from chan_dahdi. When a call is first initiated, the inband DTMF detector is not enabled if it's an outgoing ISDN call. However, it needs to be turned on once the media path starts up. This handling was put back in the open_media() callback of chan_dahdi. In sig_pri, open_media() calls were added to a few places where it was needed, including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING. Thanks to rmudgett for helping me with the patch! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	8bd241f238	Merged revisions 279636,279815 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279636 | russell | 2010-07-26 16:53:30 -0500 (Mon, 26 Jul 2010) | 2 lines Ignore a control subclass of -1 in ast_waitfordigit_full(). ........ r279815 | russell | 2010-07-27 11:06:58 -0500 (Tue, 27 Jul 2010) | 4 lines Support "channels" in addition to "channel" in chan_dahdi.conf. Review: https://reviewboard.asterisk.org/r/804 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	61c782df58	Merged revisions 279755 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279755 | pabelanger | 2010-07-26 22:57:33 -0400 (Mon, 26 Jul 2010) | 10 lines If dringXcontext is null, fallback to default context value. (closes issue #17693) Reported by: iasgoscouk Patches: issue17693.patch uploaded by pabelanger (license 224) Tested by: iasgoscouk Review: https://reviewboard.asterisk.org/r/803/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d7ca69ceea	Make "dahdi show channels" show an outgoing called number. ... The "dahdi show channels" extension column previously only showed the called number of an incoming call. It now shows the called number for an outgoing call as well. (closes issue #17653) Reported by: amazinzay Patches: issue17653_trunk.txt uploaded by rmudgett (license 664) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	301505c4c4	Rename sig_pri_pri to sig_pri_span. More descriptive of concept. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	8b3c00a824	missed FXS kewl start polarityswitch when finally on hook. ... (issue #17318) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	85bfe38f2f	Support FXS module Polarity Reversal on remote party Answer and Hangup ... FXS lines normally connect to a telephone. However, when FXS lines are routed to an external PBX or Key System to act as "external" or "CO" lines, it is extremely difficult, if not impossible for the external PBX to know when the call has been disconnected without receiving a polarity reversal on the line. Now using answeronpolarityswitch and hanguponpolarityswitch keywords that previously were used only for FXO ports, now applies like functionality for an FXS port, but from the connected equipment's point of view. (closes issue #17318) Reported by: armeniki Patches: fxs_linepolarity.diff5.txt uploaded by alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/797/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ab0b255455	DNID not cleared when channel hang up (Affects PRI and SS7) ... The "dahdi show channels" CLI command still reports the DNID of the previous call even if the call is already hang up. The "dahdi show channels" command of older releases clear the DNID once the channel is hang up. Regression from the sig_analog/sig_pri extraction from chan_dahdi. (closes issue #17623) Reported by: klaus3000 Patches: issue17623.patch uploaded by rmudgett (license 664) Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	16b4813599	Fix invalid test for rxisoffhook in FXO channels ... This fixes some cases of no outgoing calls on FXO before an incoming call. Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO (KS/GS) channels.In some cases the bit would not be initialized properly before the first inbound call and thus prevent an outgoing call. If those tests are actually required by anybody, they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c . (closes issue #14577) Reported by: jkroon Patches: asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by frawd (license 610) Tested by: frawd Review: https://reviewboard.asterisk.org/r/699/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	d1b0bf0f2d	include stat.h for everybody, needed for device2chan ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278361 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
Jeff Peeler	58061391a1	Fix regression with distinctive ring detection. ... The issue here is that passing an array to a function prohibits the ARRAY_LEN macro from returning the real size. To avoid this the size is now defined and use of ARRAY_LEN is avoided. (closes issue #15718) Reported by: alecdavis Patches: bug15718.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	34bc4b1dcb	Merged revisions 277419 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when reloading dahdi module During a reload, the priexclusive and outsignalling parameters are not read in from the config file as intended. Unfortunately, they get set to defaults as a result. This patch makes sure that they do not get set to defaults during a reload. (closes issue #17441) Reported by: mtryfoss Patches: issue17441_v1.4.patch uploaded by rmudgett (license 664) issue17441_v1.6.2.patch uploaded by rmudgett (license 664) issue17441_trunk.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277467 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
Eliel C. Sardanons	a1b89a6a50	Implement AstData API data providers as part of the GSOC 2010 project, ... midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	fd3297a272	Add missing conditional around chan_dahdi mfcr2_skip_category config parameter. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d20ca64e70	Merged revisions 274579 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	5754933045	Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the bitfield. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	c613897d1c	Fix various typos reported by Lintian ... (Also fix the typos in the comments) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	746d8e6013	Don't try to lock/unlock an uninitialized lock on a dahdi_pri. ... This small changes prevents destroy_all_channels() from accessing a lock on an unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when shutting Asterisk down gracefully. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	fb5fddd987	Extract sig_ss7_init_linkset() to sig_ss7. ... Also found a place where sig_pri_init_pri() was inlined and called it instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	93a5e74e37	Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn. ... Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn. Review: https://reviewboard.asterisk.org/r/696/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	8f92f95530	Attempt to fix a FreeBSD build error by including sys/stat.h. ... http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	717570899c	Add missing API function to sig_ss7: sig_ss7_fixup(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	6c6260d4c7	Eliminate deadlock potential in dahdi_fixup(). ... Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are in a potentially inconsistent state is a potentially bad thing in principle. However, calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	6d627b8c38	dial by name in chan_dahdi ... * chan_dahdi supports dialing configuring and dialing by device file name. DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise it may appear in chan_dahdi.conf as 'channel => span-name!local!1'. * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean. False by default. If set, chan_dahdi will ignore failed 'channel' entries. Handy for the above name-based syntax as it does not depend on initialization order. * have my_pri_make_cc_dialstring() only manupulate dial-strings of group (gGrR) dialing, which make it lsightly more complicated. https://reviewboard.asterisk.org/r/535/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	c672763af8	Fix some doxygen warnings. ... (closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	0122ccd29c	Extract sig_ss7 out of chan_dahdi. ... Extract the SS7 specific code out of chan_dahdi like what was done to ISDN/PRI and analog signaling. The new SS7 structures were modeled on sig_pri. The changes to sig_pri are an enhancement and a bug fix made possible because SS7 was extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable should have been set unconditionally in sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of SS7 extraction. 3) Module ref count error in dahdi_new() if startpbx failed to start the PBX for some reason. Review: https://reviewboard.asterisk.org/r/661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	1c67f208a7	Add ETSI Message Waiting Indication (MWI) support. ... Add the ability to report waiting messages to ISDN endpoints (phones). Relevant specification: EN 300 650 and EN 300 745 Review: https://reviewboard.asterisk.org/r/599/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267399 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	afcbc93dae	Add ETSI Call Waiting support. ... Add the ability to announce a call to an endpoint when there are no B channels available. A call waiting call is a SETUP message with no B channel selected. Relevant specification: EN 300 056, EN 300 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the "no_media_path" option. * Returns "0" if there is a B channel associated with the call. * Returns "1" if no B channel is associated with the call. The call is either on hold or is a call waiting call. If you are going to allow incoming call waiting calls then you need to use CHANNEL(no_media_path) do determine if you must drop a call to accept the new call. Review: https://reviewboard.asterisk.org/r/568/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	afd4454c44	Generic Advice of Charge. ... Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	48dd4d1249	Add ETSI Explicit Call Transfer (ECT) support. ... Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages to eliminate tromboned calls. Note: Asterisk already supported initiating the transfer of calls to eliminate tromboned calls to libpri so there was nothing to do for the asterisk portion. Review: https://reviewboard.asterisk.org/r/520/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	b5d5cc565f	Enhancements to connected line and redirecting work. ... From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	274eb8960c	Fix inverted logic in cli command: ss7 set debug on/off ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	093dbfdd52	Don't crash when destroying chan_dahdi pseudo channels. ... Must do a deep copy of the cc_params in duplicate_pseudo(). Otherwise, when the duplicate pseudo channel is destroyed, it frees the original pseudo channel cc_params. The original pseudo channel is then left with a dangling pointer for when the next duplicated pseudo channel is created. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	6b2e51b1ed	When failing to configure, don't destroy 'cfg' twice ... Fixes a crash when some config section had an incorrect channel config. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	f0a7adb309	Avoid a crash on SS7 channels. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	159f0d4b24	The inalarm flag is not passed up from the sig_analog and sig_pri submodules. ... The CLI "dahdi show channel" command was not correctly reporting the InAlarm status. The inalarm flag is now consistently passed between chan_dahdi and submodules. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	6dd80de93d	Merged revisions 260434 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines Ensure channel state is not incorrectly set in the case of a very early answer. The needringing bit was being read in dahdi_read after answering thereby setting the state to ringing from up. This clears needringing upon answering so that is no longer possible. (closes issue #17067) Reported by: tzafrir Patches: needringing.diff uploaded by tzafrir (license 46) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	00a1a96ff5	Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7, and MFCR2 users. ... Created SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS SIG_MFCR2_MAX_CHANNELS Also fixed the declaration of pollers[] in mfcr2_monitor(). It was dimensioned to the number of bytes in struct dahdi_mfcr2.pvts[] and not to the same dimension of the struct dahdi_mfcr2.pvts[]. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a3ce3441bb	Merged revisions 260195 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines DTMF CallerID detection problems. The code handling DTMF CallerID drops digits on long CallerID numbers and may timeout waiting for the first ring with shorter numbers. The DTMF emulation mode was not turned off when processing DTMF CallerID. When the emulation code gets behind in processing the DTMF digits it can skip a digit. For shorter numbers, the timeout may have been too short. I increased it from 2 seconds to 4 seconds. Four seconds is a typical time between rings for many countries. (closes issue #16460) Reported by: sum Patches: issue16460.patch uploaded by rmudgett (license 664) issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA AST-334 JIRA SWP-901 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4e3269c60d	Merged revisions 259531 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success" Changed the warning to "Failed to decode CallerID on channel 'name'". The message before it is likely more specific about why the CallerID decode failed. SWP-501 AST-283 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	3e04d6fe8e	Merged revisions 259270 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	3bda444f1e	Re-fix dahdi_request() iflist locking since CCSS merged. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Dwayne M. Hubbard	afedb856ae	Enable PRI SERVICE message support in chan_dahdi for the 'national' switchtype ... Revision 1072 of libpri added SERVICE message support for the 'national' switchtype. The attached patch enables the use of 'pri service' CLI commands on dahdi channels that are configured for the 'national' switchtype. (closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch uploaded by dhubbard (license 733) Tested by: elguero, dhubbard Review: https://reviewboard.asterisk.org/r/612/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. ... From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	1a2f96573a	Merged revisions 256225 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock. SWP-1231 ABE-2163 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a5a0a5f867	Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number. ... SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago