asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	78546868f9	sig_analog: Add Last Number Redial feature. This adds the Last Number Redial feature to simple switch. UserNote: Users can now redial the last number called if the lastnumredial setting is set to yes. Resolves: #437	3 months ago
Naveen Albert	301b0258bf	sig_analog: Add Called Subscriber Held capability. This adds support for Called Subscriber Held for FXS lines, which allows users to go on hook when receiving a call and resume the call later from another phone on the same line, without disconnecting the call. This is a convenience mechanism that most real PSTN telephone switches support. ASTERISK-30372 #close Resolves: #240 UserNote: Called Subscriber Held is now supported for analog FXS channels, using the calledsubscriberheld option. This allows a station user to go on hook when receiving an incoming call and resume from another phone on the same line by going on hook, without disconnecting the call.	2 years ago
Naveen Albert	e1a1ae933b	sig_analog: Allow three-way flash to time out to silence. sig_analog allows users to flash and use the three-way dial tone as a primitive hold function, simply by never timing it out. Some systems allow this dial tone to time out to silence, so the user is not annoyed by a persistent dial tone. This option allows the dial tone to time out normally to silence. ASTERISK-30004 #close Resolves: #205 UserNote: The threewaysilenthold option now allows the three-way dial tone to time out to silence, rather than continuing forever.	2 years ago
Naveen Albert	8cd7548e43	sig_analog: Allow immediate fake ring to be suppressed. When immediate=yes on an FXS channel, sig_analog will start fake audible ringback that continues until the channel is answered. Even if it answers immediately, the ringback is still audible for a brief moment. This can be disruptive and unwanted behavior. This adds an option to disable this behavior, though the default behavior remains unchanged. ASTERISK-30003 #close Resolves: #118 UserNote: The immediatering option can now be set to no to suppress the fake audible ringback provided when immediate=yes on FXS channels.	2 years ago
Naveen Albert	273ad73d99	sig_analog: Add fuller Caller ID support. A previous change, ASTERISK_29991, made it possible to send additional Caller ID parameters that were not previously supported. This change adds support for analog DAHDI channels to now be able to receive these parameters for on-hook Caller ID, in order to enhance the usability of CPE that support these parameters. Resolves: #94 ASTERISK-30331 UserNote: Additional Caller ID properties are now supported on incoming calls to FXS stations, namely the redirecting reason and call qualifier.	2 years ago
Naveen Albert	9a999242b2	chan_dahdi: Add dialmode option for FXS lines. Currently, both pulse and tone dialing are always enabled on all FXS lines, with no way of disabling one or the other. In some circumstances, it is desirable or necessary to disable one of these, and this behavior can be problematic. A new "dialmode" option is added which allows setting the methods to support on a per channel basis for FXS (FXO signalled lines). The four options are "both", "pulse", "dtmf"/"tone", and "none". Additionally, integration with the CHANNEL function is added so that this setting can be updated for a channel during a call. Resolves: #35 ASTERISK-29992 UserNote: A "dialmode" option has been added which allows specifying, on a per-channel basis, what methods of subscriber dialing (pulse and/or tone) are permitted. Additionally, this can be changed on a channel at any point during a call using the CHANNEL function.	2 years ago
Naveen Albert	bcc18ca9f5	general: Fix various typos. ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	3 years ago
Sarah Autumn	466eb4a52b	sig_analog: Changes to improve electromechanical signalling compatibility This changeset is intended to address compatibility issues encountered when interfacing Asterisk to electromechanical telephone switches that implement ANI-B, ANI-C, or ANI-D. In particular the behaviours that this impacts include: - FGC-CAMA did not work at all when using MF signaling. Modified the switch case block to send calls to the correct part of the signaling-handling state machine. - For FGC-CAMA operation, the delay between called number ST and second wink for ANI spill has been made configurable; previously all calls were made to wait for one full second. - After the ANI spill, previous behavior was to require a 'ST' tone to advance the call. This has been changed to allow 'STP' 'ST2P' or 'ST3P' as well, for compatibility with ANI-D. - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable. - For calls with an ANI failure, No. 1 Crossbar switches will send forward a single-digit failure code, with no calling number digits and no ST pulse to terminate the spill. I've made the ANI timeout configurable so to reduce dead air time on calls with ANI fail. - ANI info digits configurable. Modern digital switches will send 2 digits, but ANI-B sends only a single info digit. This caused the ANI reported by Asterisk to be misaligned. - Changed a confusing log message to be more informative. ASTERISK-29518 Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256	4 years ago
Tzafrir Cohen	6301531416	chan_dahdi: Configurable dialed digit timeouts Analog phones dial overlap dialing and it is chan_dahdi's job to read the numbers. It has three timeout constants that this commit converts to channel-level configuration options: * firstdigit_timeout: Default time (ms) to detect first digit * interdigit_timeout: Default time (ms) to detect following digits * matchdigit_timeout: Default time (ms) to wait in case of ambiguous match. This happens when the dialed digits match a number in the current context but are also the prefix of another number. Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213	7 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	7 years ago
Richard Mudgett	c7cc1b3059	chan_dahdi/sig_analog: Fix distinctive ring detection to suck less. The distinctive ring feature interferes with detecting Caller ID and appears to have been broken for years. What happens is if you have a ring-ring cadence as used in the UK you get too many DAHDI events for the distinctive ring pattern array and Caller ID detection is aborted. I think when Zapata/DAHDI added the ring begin event it broke distinctive ring. More events happen than before and the code does no filtering of which event times are recorded in the pattern array. * Made distinctive ring only record the ringt count when the ring ends instead of on just any DAHDI event. Distinctive ring can be ring, ring-ring, ring-ring-ring, or different ring durations for the up to three rings. * Fixed the distinctive ring detection enable (chan_dahdi.conf option usedistinctiveringdetection) to be per port instead of somewhat per port and somewhat global. This has been broken since v1.8. * Fixed using the default distinctive ring context when the detected pattern does not match any configured dringX patterns. The default context did not get set when the previous call was a matched distinctive ring pattern and the current call is not matched. This has been broken since v1.8. * Made distinctive ring have no effect on Caller ID detection when it is disabled. Caller ID detection just monitors for 10 seconds before giving up. * Fixed leak of struct callerid_state memory when a polarity reversal during Caller ID detection causes the incoming call to be aborted. DAHDI-1143 AST-1545 ASTERISK-24825 #close Reported by: Richard Mudgett ASTERISK-17588 Reported by: Daniel Flounders Review: https://reviewboard.asterisk.org/r/4444/ ........ Merged revisions 432530 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432534 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	062becab80	Convert sig_analog to use a global callback table. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f1dce57742	Fix the analog dial *0 flash-hook of bridged peer feature. The flash-hook the bridged peer feature now correctly determines if the bridged peer is another chan_dahdi channel, that it is an analog channel, and that it has the correct signaling for an FXO port. It now also flash-hooks the correct channel. ........ Merged revisions 370900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370901 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	a4f6d96b2e	push 'outgoing' flag from sig_XXX up to chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355851 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	23bc964e1c	Constify some more channel driver technology callback parameters. Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	938b642245	Restore call progress code for analog ports. Extracting sig_analog from chan_dahdi lost call progress detection functionality. * Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patches: chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.h.diff (license #5685) patch uploaded by Richard Miller ........ Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347007 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ccdc417ab5	Merged revisions 296167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296167 \| rmudgett \| 2010-11-24 16:49:48 -0600 (Wed, 24 Nov 2010) \| 57 lines Merged revisions 296166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296166 \| rmudgett \| 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) \| 50 lines Merged revisions 296165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296165 \| rmudgett \| 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) \| 43 lines Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip. The FXS connected phone has to have CW/CID support to fail, as it will send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal phone with no CID never fails. Also the SIP phone does not hear MOH when the CW call is answered. The DTMF end frame is suppressed when the phone acknowledges the CW signal for CID. The problem is the DTMF begin frame needs to be suppressed as well. The DTMF begin frame is causing SIP to start sending the DTMF RTP frames. Since the DTMF end frame is suppressed, SIP will not stop sending those DTMF RTP packets. * Suppress the DTMF begin and end frames when the channel driver is looking for DTMF digits. * Fixed a couple issues caused by not cleaning up the CID spill if you answer the CW call while it is sending the CID spill. * Fixed not sending CW/CID spill to the phone when the call is natively bridged. (Fixed by not using native bridge if CW/CID is possible.) * Suppress received audio when sending CW/CID spills. The other parties involved do not need to hear the CW/CID spills and may be confused if the CW call is for them. (closes issue #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett NOTE: * v1.4 does not have the main problem fixed by suppressing the DTMF start frames. The other three items fixed are relevant. * If you really must restore native bridging between analog ports, you need to disable CW/CID either by configuring chan_dahdi.conf callwaitingcallerid=no or dialing *70 before dialing the number to temporarily disable CW. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b1e7f85bce	Merged revisions 295747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295747 \| rmudgett \| 2010-11-19 21:11:15 -0600 (Fri, 19 Nov 2010) \| 13 lines One way audio before answering call waiting call on analog port. * Analog call waiting Caller ID spills could get stuck resulting in one way audio until the waiting call is answered. This only happens on the second (and later) call waiting call if the active call is not the first call. * The CLI/AMI "dahdi show channel" command could report the wrong channel information. Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer in sync. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	f6edd47dd6	Merged revisions 295516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295516 \| rmudgett \| 2010-11-19 10:47:11 -0600 (Fri, 19 Nov 2010) \| 13 lines Bring sig_analog extraction more into alignment with orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI support. * Fixed initial value of struct analog_pvt.use_callerid. It may get forced on depending upon other config options. * Call analog_dnd() instead of manual inlined code. * Removed unused struct analog_pvt.usedistinctiveringdetection. * Removed the struct analog_pvt.unknown_alarm flag. It was really the struct analog_pvt.inalarm flag. * Use ast_debug() instead of ast_log(LOG_DEBUG). * Rename several function's index variable to idx. * Some formatting tweaks. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	10cbc4a132	Merged revisions 293530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293530 \| rmudgett \| 2010-11-01 12:29:30 -0500 (Mon, 01 Nov 2010) \| 10 lines Analog 3-way call would not connect all parties if one was using sig_pri. Also the "dahdi show channel" would not show the correct 3-way call status. * Synchronized the inthreeway flag between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() sign error and made take an analog sub channel enum. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
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
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	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
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	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
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
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
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
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
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
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
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

43 Commits (master)