136 Commits (master)

Author	SHA1	Message	Date
Luz Paz	d65af637ac	docs: Fix various typos in channels/ ... Found via `codespell -q 3 -S "./CREDITS,*.po" -L abd,asent,atleast,cachable,childrens,contentn,crypted,dne,durationm,enew,exten,inout,leapyear,mye,nd,oclock,offsetp,ot,parm,parms,preceeding,pris,ptd,requestor,re-use,re-used,re-uses,ser,siz,slanguage,slin,thirdparty,varn,varns,ues`	2 months ago
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	f1c72347e9	sig_analog: Fix regression with FGD and E911 signaling. ... Commit 466eb4a52b introduced a regression which completely broke Feature Group D and E911 signaling, by removing the call to analog_my_getsigstr, which affected multiple switch cases. Restore the original behavior for all protocols except Feature Group C CAMA (MF), which is all that patch was attempting to target. Resolves: #993	5 months ago
Naveen Albert	1ce9e1fa8f	sig_analog: Fix channel leak when mwimonitor is enabled. ... When mwimonitor=yes is enabled for an FXO port, the do_monitor thread will launch mwi_thread if it thinks there could be MWI on an FXO channel, due to the noise threshold being satisfied. This, in turns, calls analog_ss_thread_start in sig_analog. However, unlike all other instances where __analog_ss_thread is called in sig_analog, this call path does not properly set pvt->ss_astchan to the Asterisk channel, which means that the Asterisk channel is NULL when __analog_ss_thread starts executing. As a result, the thread exits and the channel is never properly cleaned up by calling ast_hangup. This caused issues with do_monitor on incoming calls, as it would think the channel was still owned even while receiving events, leading to an infinite barrage of warning messages; additionally, the channel would persist improperly. To fix this, the assignment is added to the call path where it is missing (which is only used for mwi_thread). A warning message is also added since previously there was no indication that __analog_ss_thread was exiting abnormally. This resolves both the channel leak and the condition that led to the warning messages. Resolves: #458	1 year 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
InterLinked1	200a3f1d68	chan_dahdi: Fix broken hidecallerid setting. (#101) ... The hidecallerid setting in chan_dahdi.conf currently is broken for a couple reasons. First, the actual code in sig_analog to "allow" or "block" Caller ID depending on this setting improperly used ast_set_callerid instead of updating the presentation. This issue was mostly fixed in ASTERISK_29991, and that fix is carried forward to this code as well. Secondly, the hidecallerid setting is set on the DAHDI pvt but not carried forward to the analog pvt properly. This is because the chan_dahdi config loading code improperly set permhidecallerid to permhidecallerid from the config file, even though hidecallerid is what is actually set from the config file. (This is done correctly for call waiting, a few lines above.) This is fixed to read the proper value. Thirdly, in sig_analog, hidecallerid is set to permhidecallerid only on hangup. This can lead to potential security vulnerabilities as an allowed Caller ID from an initial call can "leak" into subsequent calls if no hangup occurs between them. This is fixed by setting hidecallerid to permcallerid when calls begin, rather than when they end. This also means we don't need to also set hidecallerid in chan_dahdi.c when copying from the config, as we would have to otherwise. Fourthly, sig_analog currently only allows dialing *67 or *82 if that would actually toggle the presentation. A comment is added clarifying that this behavior is okay. Finally, a couple log messages are updated to be more accurate. Resolves: #100 ASTERISK-30349 #close	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	48b5a4def0	sig_analog: Fix no timeout duration. ... ASTERISK_28702 previously attempted to fix an issue with flash hook hold timing out after just under 17 minutes, when it should have never been timing out. It fixed this by changing 999999 to INT_MAX, but it did so in chan_dahdi, which is the wrong place since ss_thread is now in sig_analog and the one in chan_dahdi is mostly dead code. This fixes this by porting the fix to sig_analog. ASTERISK-30336 #close Change-Id: I05eb69cc0b5319d357842a70bd26ef64d145cb15	2 years ago
Naveen Albert	67186aad56	chan_dahdi: Allow FXO channels to start immediately. ... Currently, chan_dahdi will wait for at least one ring before an incoming call can enter the dialplan. This is generally necessary in order to receive the Caller ID spill and/or distinctive ringing detection. However, if neither of these is required, then there is nothing gained by waiting for one ring and this unnecessarily delays call setup. Users can now use immediate=yes to make FXO channels (FXS signaled) begin processing dialplan as soon as Asterisk receives the call. ASTERISK-30305 #close Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5	2 years ago
Naveen Albert	f2f397c1a8	chan_dahdi: Fix buggy and missing Caller ID parameters ... There are several things wrong with analog Caller ID handling that are fixed by this commit: callerid.c's Caller ID generation function contains the logic to use the presentation to properly send the proper Caller ID. However, currently, DAHDI does not pass any presentation information to the Caller ID module, which means that presentation is completely ignored on all calls. This means that lines could be getting Caller ID information they aren't supposed to. Part of the reason this has been obscured is because the simple switch logic for handling the built in *67 and *82 is completely wrong. Rather than modifying the presentation for the call accordingly (which is what it's supposed to do), it simply blanks out the Caller ID or fills it in. This is wrong, so wrong that it makes a mockery of the specification. Additionally, it would leave to the "UNAVAILABLE" disposition being used for Caller ID generation as opposed to the "PRIVATE" disposition that it should have been using. This is now fixed to only update the presentation and not modify the number and name, so that the simple switch *67/*82 work correctly. Next, sig_analog currently only copies over the name and number, nothing else, when it is filling in a duplicated caller id structure. Thus, we also now copy over the presentation information so that is available for the Caller ID spill. Additionally, this meant that "valid" was implicitly 0, and as such presentation would always fail to "Unavailable". The validity is therefore also copied over so it can be used by ast_party_id_presentation. As part of this fix, new API is added so that all the relevant Caller ID information can be passed in to the Caller ID generation functions. Parameters that are also completely missing from the Caller ID spill have also been added, to enhance the compatibility, correctness, and completeness of the Asterisk Caller ID implementation. ASTERISK-29991 #close Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15	3 years ago
Naveen Albert	bcc18ca9f5	general: Fix various typos. ... ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	3 years ago
Naveen Albert	97f278a94a	sig_analog: Fix broken three-way conferencing. ... Three-way calling for analog lines is currently broken. If party A is on a call with party B and initiates a three-way call to party C, the behavior differs depending on whether the call is conferenced prior to party C answering. The post-answer case is correct. However, if A flashes before C answers, then the next flash disconnects B rather than C, which is incorrect. This error occurs because the subs are not swapped in the misbehaving case. This is because the flash handler only swaps the subs if C has answered already, which is wrong. To fix this, we swap the subs regardless of whether C has answered or not when the call is conferenced. This ensures that C is disconnected on the next hook flash, rather than B as can happen currently. ASTERISK-30043 #close Change-Id: I96c5bf6c9b7eb2636136b716c677c82c079b6f06	3 years ago
George Joseph	4aa541683b	GCC12: Fixes for 16+ ... Most issues were in stringfields and had to do with comparing a pointer to an constant/interned string with NULL. Since the string was a constant, a pointer to it could never be NULL so the comparison was always "true". gcc now complains about that. There were also a few issues where determining if there was enough space for a memcpy or s(n)printf which were fixed by defining some of the involved variables as "volatile". There were also a few other miscellaneous fixes. ASTERISK-30044 Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570	3 years ago
Alexander Traud	726d6dd166	channels: Fix for Doxygen. ... ASTERISK-29762 Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7	3 years ago
Josh Soref	d46ba42910	channels: Spelling fixes ... Correct typos of the following word families: appease permanently overriding residue silliness extension channels globally reference japanese group coordinate registry information inconvenience attempts cadence payloads presence provisioning mimics behavior width natively syslabel not owning unquelch mostly constants interesting active unequipped brodmann commanding backlogged without bitstream firmware maintain exclusive practically structs appearance range retransmission indication provisional associating always whether cyrillic distinctive components reinitialized initialized capability switches occurring happened outbound ASTERISK-29714 Change-Id: Ife52ee89cd2170b684fa651ca72b1cb911a57339	3 years ago
Naveen Albert	f9ba1ee7c9	sig_analog: Fix truncated buffer copy ... Fixes compiler warning caused by a truncated copy of the ANI2 into a buffer of size 10. This could prevent the null terminator from being copied if the copy value exceeds the size of the buffer. This increases the buffer size to 101 to ensure there is no way for truncation to occur. ASTERISK-29702 #close Change-Id: Ief9052212952840fa44de6463b8699fdb3e163d0	4 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
George Joseph	92d4ec2906	chan_dahdi: Address gcc9 issues ... Fixed format-truncation issues in chan_dahdi.c and sig_analog.c. Since they're related to fields provided by dahdi-tools we can't change the buffer sizes so we're just checking the return from snprintf and printing an errior if we overflow. Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5	6 years ago
Richard Mudgett	a470bb9e27	channel: Fix some more unprotected channel flag setting. ... Change-Id: I34c3b1201b1de539945bcfdcb264fff30332d48c	7 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
George Joseph	40da6434c1	build_system: Fix some warnings highlighted by clang ... Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd	9 years ago
Corey Farrell	c08fd275bf	Logger: Convert 'struct ast_callid' to unsigned int. ... Switch logger callid's from AO2 objects to simple integers. This helps in two ways. Copying integers is faster than referencing AO2 objects, so this will result in a small reduction in logger overhead. This also erases the possibility of an infinate loop caused by an invalid callid in threadstorage. ASTERISK-24833 #comment Committed callid conversion to trunk. Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4466/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 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	bb06603d5f	chan_dahdi/sig_analog: Put log message strings on one line. ... With the log messages on one line, you can search for the log message seen in the log and expect to find it. ........ Merged revisions 432032 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432034 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	05cc6d6d55	chan_dahdi: Remove some dead code. ... ........ Merged revisions 431992 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431993 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	eacbb4ceb5	chan_dahdi: Populate CALLERID(ani2) for incoming calls in featdmf signaling mode. ... For the featdmf signaling mode the incoming MF Caller-ID information is formatted as follows: *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}# Rather than discarding the ani2 digits, populate the CALLERID(ani2) value with what is received instead. AST-1368 #close Reported by: Denis Martinez Patches: extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett ........ Merged revisions 429783 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429784 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e04607f8a3	res_smdi: convert to astobj2 ... Remove functions: ast_smdi_interface_unref ast_smdi_md_message_putback ast_smdi_mwi_message_putback ast_smdi_md_message destructor ast_smdi_mwi_message destructor Includes for astobj.h are removed everywhere it's possible. ASTERISK-24066 #close Review: https://reviewboard.asterisk.org/r/3758/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	d00882108f	res_pjsip_refer: Fix bugs involving Parking/PJSIP/transfers ... PJSIP would never send the final 200 Notify for a blind transfer when transferring to parking. This patch fixes that. In addition, it fixes a reference leak when performing blind transfers to non-bridging extensions. Review: https://reviewboard.asterisk.org/r/3485/ ........ Merged revisions 414400 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abd3e4040b	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	28c0cb28d0	channel locking: Add locking for channel snapshot creation ... Original commit message by mmichelson (asterisk 12 r403311): "This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such." The above was initially committed and then reverted at r403398. The problem was found to be in core_local.c in the publish_local_bridge_message function. The ast_unreal_lock_all function locks and adds a reference to the returned channels and while they were being unlocked they were not being unreffed when no longer needed. Fixed by unreffing the channels. Also in bridge.c a lock was obtained on "other->chan", but then an attempt was made to unlock "other" and not the previously locked channel. Fixed by unlocking "other->chan" (closes issue ASTERISK-22709) Reported by: John Bigelow ........ Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	1212906351	Reverting r403311. It's causing ARI tests to hang. ... ........ Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8e8b329e14	Add channel locking for channel snapshot creation. ... This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such. ........ Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	7d17d5fb04	chan_dahdi: Fix analog parking using flash-hook. ... Transferring an analog call using a flash-hook to parking would fail to park the call and result in an invalid ao2 object unref. * Park the correct bridged channel. ........ Merged revisions 400236 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	868be02a2f	Fix uninitialized value in struct ast_control_pvt_cause_code usage. ... ........ Merged revisions 397744 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 397745 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	56aea1c030	Allow res_parking to be unloadable ... This change protects accesses of res_parking such that it can unload safely once transient uses of its registered functions are complete. The parking API has been restructured such that its consumers do not have access to the vtable exposed by the parking provider, but instead route through stubs to prevent consumers from holding on to function pointers. This adds calls to all the parking unload functions and moves application loading and unloading into functions in parking_applications.c similar to the rest of the parts of res_parking. Review: https://reviewboard.asterisk.org/r/2763/ (closes issue ASTERISK-22142) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4f07502000	Fixed chan_dahdi compilation failure ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5c4b482471	Support externally initiated parking requests; remove some dead code ... This patch does the following: * It adds support for externally initiated parking requests. In particular, chan_skinny has a protocol level message that initiates a call park. This patch now supports that option, as well as the protocol specific mechanisms in chan_dahdi/sig_analog and chan_mgcp. * A parking bridge features virtual table has been added that provides access to the parking functionality that the Bridging API needs. This includes requests to park an entire 'call' (with little or no additional information, thank you chan_skinny), perform a blind transfer to a parking extension, determine if an extension is a parking extension, as well as the actual "do the parking" request from the Bridging API. * Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new functions * The removal of some - but not all - dead parking code from features.c This also fixed blind transferring a multi-party bridge to a parking lot (which was implemented, but had at least one code path where using the parking features kK might not have worked) Review: https://reviewboard.asterisk.org/r/2710 (closes issue ASTERISK-22134) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cafc115896	A great big renaming patch ... This patch renames the bridging* files to bridge*. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: * channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	684c83b29b	Add transfer support to CEL ... This adds CEL support for blind and attended transfers and call pickup. During the course of adding this functionality I noticed that CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly useless without a bridge identifier, so I added that as well. This adds tests for blind transfers, several types of attended transfers, and call pickup. The extra field in CEL records now consists of a JSON blob whose fields are defined on a per-event basis. Review: https://reviewboard.asterisk.org/r/2658/ (closes issue ASTERISK-21565) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	909ee4bfb9	Refactor extraneous channel events ... This change removes JitterBufStats, ChannelReload, and ChannelUpdate and refactors the following events to travel over Stasis-Core: * LocalBridge * DAHDIChannel * AlarmClear * SpanAlarmClear * Alarm * SpanAlarm * DNDState * MCID * SIPQualifyPeerDone * SessionTimeout Review: https://reviewboard.asterisk.org/r/2627/ (closes issue ASTERISK-21476) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6114166237	Refactor chan_dahdi/sig_analog/sig_pri and chan_misdn to use the common transfer functions. ... (closes issue ASTERISK-21523) Reported by: Matt Jordan (closes issue ASTERISK-21524) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2600/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	2dc8a06006	Refactor the features configuration scheme. ... Features configuration is handled in its own API in features_config.h and features_config.c. This way, features configuration is accessible to anything that needs it. In addition, features configuration has been altered to be more channel-oriented. Most callers of features API code will be supplying a channel so that the individual channel's settings will be acquired rather than the global setting. Missing from this commit is XML documentation for the features configuration. That will be handled in a separate commit. Review: https://reviewboard.asterisk.org/r/2578/ (issue ASTERISK-21542) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	154fbf8cae	Split Hold event into Hold/Unhold, and move it into core. ... (closes issue ASTERISK-21487) Review: https://reviewboard.asterisk.org/r/2565/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5ab5715646	Remove some unnecessary calls to ast_bridged_channel() in chan_dahdi.c/sig_analog.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	f2bb9afe17	Multiple revisions 375993-375994 ... ........ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b5138fccf4	Add pause one second W dial modifier. ... * The following dialplan applications now recognize 'W' to pause sending DTMF for one second in addition to the previously existing 'w' that paused sending DTMF for half a second. Dial, ExternalIVR, and SendDTMF. * The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now distinguishes between 'w' and 'W'. The 'w' pauses dialing for half a second. The 'W' pauses dialing for one second. * Created dahdi_dial_str() in chan_dahdi that eliminated a lot of duplicated dialing code and diagnostic messages for the channel driver. (closes issue ASTERISK-20039) Reported by: Jeremiah Gowdy Patches: jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy Expanded patch to add support in chan_dahdi. Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	76d642ff69	Add HANGUPCAUSE information to callee channels ... This adds HANGUPCAUSE information to called channels so that hangup handlers can, in conjunction with predial dialplan execution, access the hangupcause information when the dialed channel hangs up on a one-to-one basis instead of a many-to-one basis as with HANGUPCAUSE usage on the caller channel. Review: https://reviewboard.asterisk.org/r/2069/ (closes issue ASTERISK-20198) ........ Merged revisions 371258 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago