8733 Commits (2ec0b83e5bf439b74ba6c3b36caaa3c0c9d6d407)

Author	SHA1	Message	Date
Naveen Albert	76d33df366	chan_dahdi: Allow MWI to be manually toggled on channels. ... This adds a CLI command to manually toggle the MWI status of a channel, useful for troubleshooting or resetting MWI devices, similar to the capabilities offered with SIP messaging to manually control MWI status. UserNote: The 'dahdi set mwi' now allows MWI on channels to be manually toggled if needed for troubleshooting. Resolves: #440 (cherry picked from commit 4b908f364d)	2 years ago
PeterHolik	727a4cceec	chan_rtp.c: MulticastRTP missing refcount without codec option ... Fixes: #529 (cherry picked from commit 3b46872a8f)	2 years ago
PeterHolik	8a999b6706	chan_rtp.c: Change MulticastRTP nameing to avoid memory leak ... Fixes: asterisk#536 (cherry picked from commit 6fe045fd64)	2 years ago
Naveen Albert	e735ab8cfb	logger: Fix linking regression. ... Commit 008731b0a4 caused a regression by resulting in logger.xml being compiled and linked into the asterisk binary in lieu of logger.c on certain platforms if Asterisk was compiled in dev mode. To fix this, we ensure the file has a unique name without the extension. Most existing .xml files have been named differently from any .c files in the same directory or did not pose this issue. channels/pjsip/dialplan_functions.xml does not pose this issue but is also being renamed to adhere to this policy. Resolves: #539	2 years ago
Naveen Albert	4e774da45a	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 (cherry picked from commit c930230a73)	2 years ago
Naveen Albert	bef9a9422d	func_channel: Expose previously unsettable options. ... Certain channel options are not set anywhere or exposed in any way to users, making them unusable. This exposes some of these options which make sense for users to manipulate at runtime. Resolves: #442 (cherry picked from commit 9211fb5e97)	2 years ago
Sean Bright	a694931478	chan_iax2.c: Don't send unsanitized data to the logger. ... This resolves an issue where non-printable characters could be sent to the console/log files. (cherry picked from commit da4e6e7ddb)	2 years ago
George Joseph	c40496cb38	chan_pjsip: Add PJSIPHangup dialplan app and manager action ... See UserNote below. Exposed the existing Hangup AMI action in manager.c so we can use all of it's channel search and AMI protocol handling without duplicating that code in dialplan_functions.c. Added a lookup function to res_pjsip.c that takes in the string represenation of the pjsip_status_code enum and returns the actual status code. I.E. ast_sip_str2rc("DECLINE") returns 603. This allows the caller to specify PJSIPHangup(decline) in the dialplan, just like Hangup(call_rejected). Also extracted the XML documentation to its own file since it was almost as large as the code itself. UserNote: A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range. (cherry picked from commit af7e89ebf8)	2 years ago
Sean Bright	cd39daaed1	chan_iax2.c: Ensure all IEs are displayed when dumping frame contents. ... When IAX2 debugging was enabled (`iax2 set debug on`), if the last IE in a frame was one that may not have any data - such as the CALLTOKEN IE in an NEW request - it was not getting displayed. (cherry picked from commit 19507ae160)	2 years ago
Naveen Albert	27f3df7139	chan_dahdi: Warn if nonexistent cadence is requested. ... If attempting to ring a channel using a nonexistent cadence, emit a warning, before falling back to the default cadence. Resolves: #409 (cherry picked from commit cdcdca5199)	2 years ago
Naveen Albert	c58d5e9190	chan_dahdi: Clarify scope of callgroup/pickupgroup. ... Internally, chan_dahdi only applies callgroup and pickupgroup to FXO signalled channels, but this is not documented anywhere. This is now documented in the sample config, and a warning is emitted if a user tries configuring these settings for channel types that do not support these settings, since they will not have any effect. Resolves: #294 (cherry picked from commit 5b89e40541)	2 years ago
Naveen Albert	d01b047de3	chan_console: Fix deadlock caused by unclean thread exit. ... To terminate a console channel, stop_stream causes pthread_cancel to make stream_monitor exit. However, commit 5b8fea93d1 added locking to this function which results in deadlock due to the stream_monitor thread being killed while it's holding the pvt lock. To resolve this, a flag is now set and read to indicate abort, so the use of pthread_cancel and pthread_kill can be avoided altogether. Resolves: #308 (cherry picked from commit 3b027d1e47)	2 years ago
Naveen Albert	69338381ea	chan_iax2: Improve authentication debugging. ... Improves and adds some logging to make it easier for users to debug authentication issues. Resolves: #286 (cherry picked from commit f93138bcad)	2 years ago
Maximilian Fridrich	3829c94d71	chan_rtp: Implement RTP glue for UnicastRTP channels ... Resolves: #298 UserNote: The dial string option 'g' was added to the UnicastRTP channel which enables RTP glue and therefore native RTP bridges with those channels. (cherry picked from commit 1af2ae177c)	2 years ago
Naveen Albert	e2f0538dea	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. (cherry picked from commit cd0bfe193f)	2 years ago
Naveen Albert	d16046e41f	chan_dahdi: Allow autoreoriginating after hangup. ... Currently, if an FXS channel is still off hook when all calls on the line have hung up, the user is provided reorder tone until going back on hook again. In addition to not reflecting what most commercial switches actually do, it's very common for switches to automatically reoriginate for the user so that dial tone is provided without the user having to depress and release the hookswitch manually. This can increase convenience for users. This behavior is now supported for kewlstart FXS channels. It's supported only for kewlstart (FXOKS) mainly because the behavior doesn't make any sense for ground start channels, and loop start signalling doesn't provide the necessary DAHDI event that makes this easy to implement. Likely almost everyone is using FXOKS over FXOLS anyways since FXOLS is pretty useless these days. ASTERISK-30357 #close Resolves: #224 UserNote: The autoreoriginate setting now allows for kewlstart FXS channels to automatically reoriginate and provide dial tone to the user again after all calls on the line have cleared. This saves users from having to manually hang up and pick up the receiver again before making another 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
Sean Bright	508657879e	chan_iax2.c: Avoid crash with IAX2 switch support. ... A change made in 82cebaa0 did not properly handle the case when a channel was not provided, triggering a crash. ast_check_hangup(...) does not protect against NULL pointers. Fixes #180	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	4176f57938	chan_dahdi: Fix Caller ID presentation for FXO ports. ... Currently, the presentation for incoming channels is always available, because it is never actually set, meaning the channel presentation can be nonsensical. If the presentation from the incoming Caller ID spill is private or unavailable, we now update the channel presentation to reflect this. Resolves: #120 ASTERISK-30333 ASTERISK-21741	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
Maximilian Fridrich	18f0b6661a	chan_pjsip: Allow topology/session refreshes in early media state (#74) ... With this change, session modifications in the early media state are possible if the SDP was sent reliably and confirmed by a PRACK. For details, see RFC 6337, escpecially section 3.2. Resolves: #73	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	73103bdcd5	chan_iax2: Fix jitterbuffer regression prior to receiving audio. ... ASTERISK_29392 (a security fix) introduced a regression by not processing frames when we don't have an audio format. Currently, chan_iax2 only calls jb_get to read frames from the jitterbuffer when the voiceformat has been set on the pvt. However, this only happens when we receive a voice frame, which means that prior to receiving voice frames, other types of frames get stalled completely in the jitterbuffer. To fix this, we now fallback to using the format negotiated during call setup until we've actually received a voice frame with a format. This ensures we're always able to read from the jitterbuffer. ASTERISK-30354 #close ASTERISK-30162 #close Change-Id: Ie4fd1e8e088a145ad89e0427c2100a530e964fe9	3 years ago
Mike Bradeen	6b03d60c7d	res_monitor: Remove deprecated module. ... ASTERISK-30303 Change-Id: I0462caefb4f9544e2e2baa23c498858310b52d50	3 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. ... For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	3 years ago
Igor Goncharovsky	410150235a	res_pjsip_rfc3326: Add SIP causes support for RFC3326 ... Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). ASTERISK-30319 #close Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645	3 years ago
George Joseph	7dc8773178	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	3 years ago
George Joseph	3a3d6c7dcb	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" ... This reverts commit e66c5da145. Reason for revert: Issue when transcoding to/from g722 Change-Id: I12853c5b1d3a77f5b9200f41908fd238a17159dc	3 years ago
Mike Bradeen	4095a382da	chan_sip: Remove deprecated module. ... ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996	3 years ago
George Joseph	e66c5da145	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	3 years ago
Mike Bradeen	de3ce178ab	chan_alsa: Remove deprecated module. ... ASTERISK-30298 Change-Id: I5c8afb781528afdf55d237e3bffa5e4a862ae8c7	3 years ago
Mike Bradeen	89a7d30a97	chan_mgcp: Remove deprecated module. ... Also removes res_pktcops to avoid merge conflicts with ASTERISK~30301. ASTERISK-30299 Change-Id: I41a316d327646a197b6f112f7f637aceb5111b41	3 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	3 years ago
Mike Bradeen	8d652ab4be	chan_skinny: Remove deprecated module. ... ASTERISK-30300 Change-Id: I8be11455010b8ec552e62b0719368342e8a1bae9	3 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	3 years ago
George Joseph	b6670e2440	chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer ... unicast_rtp_request() was setting the channel variables like this: pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS", ast_sockaddr_stringify_addr(&local_address)); ast_rtp_instance_get_local_address(instance, &local_address); pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT", ast_sockaddr_stringify_port(&local_address)); ...which made it appear that UNICASTRTP_LOCAL_ADDRESS was being set before local_address was set. In fact, the address part of local_address was set earlier in the function, just not the port. This was confusing however so ast_rtp_instance_get_local_address() is now being called before setting UNICASTRTP_LOCAL_ADDRESS. ASTERISK-30281 Change-Id: I872ac49477100f4eb33891d46efc6ca21ec81aa4	3 years ago
Naveen Albert	e4117d7290	chan_dahdi: Fix unavailable channels returning busy. ... This fixes dahdi_request to properly set the cause code to CONGESTION instead of BUSY if no channels were actually available. Currently, the cause is erroneously set to busy if the channel itself is found, regardless of its current state. However, if the channel is not available (e.g. T1 down, card not operable, etc.), then the channel itself may not be in a functional state, in which case CHANUNAVAIL is the correct cause to use. This adds a simple check to ensure that busy tone is only returned if a channel is encountered that has an owner, since that is the only possible way that a channel could actually be busy. ASTERISK-30274 #close Change-Id: Iad5870223c081240c925b19df8d6af136953b994	3 years ago
Naveen Albert	9bc733758d	chan_dahdi: Resolve format truncation warning. ... Fixes a format truncation warning in notify_message. ASTERISK-30256 #close Change-Id: I983a423c0214641ca4f8c9dfe0b19c47448fdee1	3 years ago
Ben Ford	31b3addce7	res_pjsip: Add TEL URI support for basic calls. ... This change allows TEL URI requests to come through for basic calls. The allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To headers will now allow TEL URIs, as well as the request URI. Support is only for TEL URIs present in traffic from a remote party. Asterisk does not generate any TEL URIs on its own. ASTERISK-26894 Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a	3 years ago
Philip Prindeville	97b3459bd2	res_crypto: Use EVP API's instead of legacy API's ... ASTERISK-30046 #close Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2	3 years ago
Sean Bright	e0fe864874	chan_dahdi.c: Resolve a format-truncation build warning. ... With gcc (Ubuntu 11.2.0-19ubuntu1) 11.2.0: > chan_dahdi.c:4129:18: error: ‘%s’ directive output may be truncated > writing up to 255 bytes into a region of size between 242 and 252 > [-Werror=format-truncation=] This removes the error-prone sizeof(...) calculations in favor of just doubling the size of the base buffer. Change-Id: I2d276785286730d3d5d0a921bcea2e065dbf27c5	3 years ago
Naveen Albert	79e1447dc2	chan_iax2: Add missing options documentation. ... Adds missing dial resource option documentation. ASTERISK-30164 #close Change-Id: I674e1fc9b1e5d67a20599bd4b418ce294d48fc83	3 years ago
Naveen Albert	f5378f63c4	general: Improve logging levels of some log messages. ... Adjusts some logging levels to be more or less important, that is more prominent when actual problems occur and less prominent for less noteworthy things. ASTERISK-30153 #close Change-Id: Ifc8f7df427aa018627db462125ae744986d3261b	3 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	8a21417095	chan_dahdi: Add POLARITY function. ... Adds a POLARITY function which can be used to retrieve the current polarity of an FXS channel as well as set the polarity of an FXS channel to idle or reverse at any point during a call. ASTERISK-30000 #close Change-Id: If6f50998f723e4484bf68e2473f5cedfeaf9b8f1	3 years ago
Naveen Albert	bcc18ca9f5	general: Fix various typos. ... ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	3 years ago
Naveen Albert	5f60caa402	chan_iax2: Allow compiling without OpenSSL. ... ASTERISK_30007 accidentally made OpenSSL a required depdendency. This adds an ifdef so the relevant code is compiled only if OpenSSL is available, since it only needs to be executed if OpenSSL is available anyways. ASTERISK-30083 #close Change-Id: Iad05c1a9a8bd2a48e7edf8d234eaa9f80779e34d	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