47 Commits (f91cda9566ba4e7fd1e72a45a0b26faac5ab46d4)

Author	SHA1	Message	Date
Richard Mudgett	f91cda9566	Merged revisions 291656 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291656 | rmudgett | 2010-10-13 18:45:11 -0500 (Wed, 13 Oct 2010) | 34 lines Merged revisions 291655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500 (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) | 20 lines Deadlock between dahdi_exception() and dahdi_indicate(). There is a deadlock between dahdi_exception() and dahdi_indicate() for analog ports. The call-waiting and three-way-calling feature can experience deadlock if these features are trying to do something and an event from the bridged channel happens at the same time. Deadlock avoidance code added to obtain necessary channel locks before attemting an operation with call-waiting and three-way-calling. (closes issue #16847) Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/971/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	41b95ee887	Merged revisions 286931 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b3fa5ec3be	Merged revisions 286904-286905 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286904 | rmudgett | 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) | 12 lines Unable to originate calls using E&M over T1. When originating a call from Unit Under Test to Reference Unit using E&M RBS signaling mode, I get the following warning message: "Ring/Off-hook in strange state 3 on channel 1". Fixed the sig_analog outgoing flag. It was never set when sig_analog was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408 ........ r286905 | rmudgett | 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) | 1 line Simplify some code in sig_analog. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b8a71201dc	Merged revisions 281870 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281870 | rmudgett | 2010-08-11 15:30:29 -0500 (Wed, 11 Aug 2010) | 4 lines Fix a call to analog_set_pulsedial() not setting 0 or 1 only. * Also a couple minor tweaks. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281871 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
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	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	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
Jeff Peeler	0ef5550742	Change expected operation from error to debug message ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	00433d60e6	Fix no call waiting caller ID ... Clearing the callwaitcas flag in analog_call was causing the incoming D digit to be ignored which triggers sending the caller ID. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3194c061a7	Don't blow up if an ast_channel doesn't get allocated. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270726 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	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
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	b1ccb1a44e	Simplified dahdi_request() channel selection failed reason/cause code. ... Also avoid potential crash because cause could be NULL. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	e8a99a9962	Merged revisions 250480 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines Make sure to clear red alarm after polarity reversal. From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	568c057c4c	Extend max call limit duration from 24.8 days to 292+ million years. ... If the limit was set past MAX_INT upon answering, the call was immediately hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup). The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been changed to return an int64_t to prevent overflow. Also the reporter suggested adding a message indicating the reason for the call hanging up. Given that the new limit is so much higher, the message (which would only really be useful in the overflow scenario) has been made a debug message only. (closes issue #16006) Reported by: viraptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c5cfc2a867	Removed unused parameters from analog_available() and sig_pri_available(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ef9be94b35	Change all refererences to 1.6.3 to be 1.8, since that will be the next feature release ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	50b7338d02	Fix call forwarding for analog phones. ... (closes issue #16440) Reported by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. ... Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	10e8ee1746	Add a few missing events to analog_handle_event. ... The reported bug was actually only for pulsedigit, dtmfup, and dtmfdown handling. Also added recognition for fax events (just some verbose output) and fixed handling for the ec_disabled_event. In order to make comparing the analog version of events to the DAHDI events easier, the ordering has been changed to follow that of the DAHDI events. (closes issue #15924) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	843a724373	Merged revisions 218401 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor. After talking to rmudgett about some of his recent iflist locking changes, it was determined that the only place that would destroy a channel without being explicitly to do so was in handle_init_event. The loop to walk the interface list has been modified to wait to destroy the channel until the dahdi_pvt of the channel to be destroyed is no longer needed. (closes issue #15378) Reported by: samy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	3a718192c6	Allow do not disturb to be set on analog channels via the CLI and AMI. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5561ba19aa	Stop caller id transmission when offhook event detected. ... This fixes the problem that would occur if an analog phone was picked up while the caller id was being sent. The caller id before sent the whole spill even after pickup and is now corrected. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	fd561e871f	Fix memory leak of sig_xxx private structures. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Doug Bailey	8430c87faa	Added detection DTMF CID without polarity change alert. ... Added detection of DTMF tone energy levels on FXO channels in chan_dahdi monitoring loop so DTMF CID can be detected without the need of a polarity change precursor. (closes issue #9096) Reported by: fleed Patches: 9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819) Tested by: cyberplant, sum, maturs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Doug Bailey	eff8dd9a2f	Fix issue where DTMF CID detect was placing channels into signed linear mode ... made analog_set_linear_mode return back the mode that was being overwritten so it could be restored later. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	0d2ef8ac5c	Merged revisions 212430 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix uninitialized variable causing random MWI indications. (closes issue #15727) Reported by: doda Patches: dahdi_changes.patch uploaded by doda (license 853) ........ r212430 | rmudgett | 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix uninitialized variable. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	e60773a298	Add braces where missing and a few whitespace fixes in sig_analog ... (closes issue #15678) Reported by: alecdavis Patches: sig_analog_mainly_braces.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	132204459c	More code that somehow got left out of sig_analog ... * confirmanswer option now respected * check and set waiting for dialtone timer * unneeded needcallerid flag removed from analog_subchannel * ss_astchan does not need to be a void pointer * swap_channels callback updated to trunk * analog_hangup now resets channel to default law git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b65c0edd52	Fix chan_dahdi option ringtimeout ... dahdi_read relies on the dahdi_pvt copy of ringt which was not getting set in sig_analog. This patch adds a callback to do so. (closes issue #15288) Reported by: alecdavis Patches: chan_dahdi.ringtimeout.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	d669ba24d7	Miscellaneous minor fixes to sig_analog. ... * Sanity adjustments to __analog_ss_thread for sig_analog environment. * Deleted some duplicated code. * Fixed analog_ss_thread_start passing the wrong pointer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	95d037edad	Trim trailing whitespace. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6ac23c3eca	Merged revisions 207827 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines Wait for wink before dialing when using E&M wink signaling There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simply add SIG_EMWINK to the list. (closes issue #14434) Reported by: araasch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	8270339965	Fix segfault in sig_analog when using callwaiting, respect callwaiting options ... Sig_analog handles allocating the sub channel for callwaiting, so no longer try to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as allocated upon success of the alloc_sub callback, which was responsible for the segfault. Also, the callwaiting and callwaitingcallerid options were being unconditionally set to true. Now, the options are properly set from chan_dahdi.conf. (closes issue #15508) Reported by: elguero Tested by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	646cd02c09	The dialing flag was mistakingly removed from sig_pri. ... This readds the proper setting of the flag and is really a continuation of r205731. The flag was being set properly in sig_analog, but use of the newly added set_dialing callback allowed for some simplification in chan_dahdi. (closes issue #15486) Reported by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b9e898017e	Restore some missing functionality to sig_analog. ... The main purpose of this commit is to restore missing functionality present in the ss_thread before all the sig related work was done. Two of the biggest missing things were distinctive ring detection and cid handling for V23. fxsoffhookstate and associated mwi variables have been moved inside sig_analog as they were not being set properly as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	cf8395002d	Fix a deadlock in sig_analog ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	5e2a5d16b6	Add CEL transfer events to analog (chan_dahdi) transfers. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5ebf0f3c50	Check if polarityonanswerdelay has elapsed before setting a channel as answered ... after a polarity reversal. Previously on a polarity switch event chan_dahdi would set the channel immediately as answered. This would cause problems if a polarity reversal occurred when the line was picked up as the dial would not have yet occurred. Now if the polarity reversal occurs before delay has elapsed after coming off hook or an answer, it is ignored. Also, some refactoring was done in _handle_event. (closes issue #13917) Reported by: alecdavis Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. ... CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5c7da226e4	New signaling module to handle PRI/BRI operations in chan_dahdi ... This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into sig_pri.c. Functionality in theory should not change (mostly). A few trivial changes were made in sig_analog with verbose messages and commenting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	aaf5eb105e	New signaling module to handle analog operations in chan_dahdi ... This branch splits all the analog signaling logic out of chan_dahdi.c into sig_analog.c. Functionality in theory should not change at all. As noted in the code, there is still some unused code remaining that will be cleaned up in a later commit. Review: https://reviewboard.asterisk.org/r/253/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago