8729 Commits (ba30b87e9172d1ab265178cdeda0010d8fa4fe69)

Author	SHA1	Message	Date
Sean Bright	351d853dfd	chan_sip.c: Fix __sip_reliable_xmit build error ... Fixes #954 (cherry picked from commit 909b93608a)	11 months ago
Naveen Albert	7057658110	chan_dahdi: Never send MWI while off-hook. ... In some circumstances, it is possible for the do_monitor thread to erroneously think that a line is on-hook and send an MWI FSK spill to it when the line is really off-hook and no MWI should be sent. Commit 0a8b3d3467 previously fixed this issue in a more readily encountered scenario, but it has still been possible for MWI to be sent when it shouldn't be. To robustly fix this issue, query DAHDI for the hook status to ensure we don't send MWI on a line that is actually still off hook. Resolves: #928 (cherry picked from commit fa614f755d)	11 months ago
Mike Bradeen	40d7ce1084	res_pjsip_notify: add dialplan application ... Add dialplan application PJSIPNOTIFY to send either pre-configured NOTIFY messages from pjsip_notify.conf or with headers defined in dialplan. Also adds the ability to send pre-configured NOTIFY commands to a channel via the CLI. Resolves: #799 UserNote: A new dialplan application PJSIPNotify is now available which can send SIP NOTIFY requests from the dialplan. The pjsip send notify CLI command has also been enhanced to allow sending NOTIFY messages to a specific channel. Syntax: pjsip send notify <option> channel <channel> (cherry picked from commit e94c5f0d3b)	1 year ago
Ben Ford	27e17dd4d5	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID. (cherry picked from commit 027127246e)	1 year ago
Sean Bright	d18159a008	pjsip: Add PJSIP_PARSE_URI_FROM dialplan function. ... Various SIP headers permit a URI to be prefaced with a `display-name` production that can include characters (like commas and parentheses) that are problematic for Asterisk's dialplan parser and, specifically in the case of this patch, the PJSIP_PARSE_URI function. This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that behaves identically to `PJSIP_PARSE_URI` except that the first argument is now a variable name and not a literal URI. Fixes #756 (cherry picked from commit 5068cc814f)	1 year ago
Naveen Albert	a2579ec402	callerid.c: Parse previously ignored Caller ID parameters. ... Commit f2f397c1a8 previously made it possible to send Caller ID parameters to FXS stations which, prior to that, could not be sent. This change is complementary in that we now handle receiving all these parameters on FXO lines and provide these up to the dialplan, via chan_dahdi. In particular: * If a redirecting reason is provided, the channel's redirecting reason is set. No redirecting number is set, since there is no parameter for this in the Caller ID protocol, but the reason can be checked to determine if and why a call was forwarded. * If the Call Qualifier parameter is received, the Call Qualifier variable is set. * Some comments have been added to explain why some of the code is the way it is, to assist other people looking at it. With this change, Asterisk's Caller ID implementation is now reasonably complete for both FXS and FXO operation. Resolves: #681 (cherry picked from commit 6bd0b67081)	1 year ago
Naveen Albert	6e7f26b795	chan_dahdi: Add DAHDIShowStatus AMI action. ... * Add an AMI action to correspond to the "dahdi show status" command, allowing span information to be retrieved via AMI. * Show span number and sig type in "dahdi show channels". Resolves: #673 (cherry picked from commit 37eab0cbd6)	1 year ago
George Joseph	ca6590426a	Fix incorrect application and function documentation references ... There were a few references in the embedded documentation XML where the case didn't match or where the referenced app or function simply didn't exist any more. These were causing 404 responses in docs.asterisk.org. (cherry picked from commit 9c2cc5bf24)	1 year ago
Naveen Albert	5ccedf22d2	chan_dahdi: Don't retry opening nonexistent channels on restart. ... Commit 729cb1d390 added logic to retry opening DAHDI channels on "dahdi restart" if they failed initially, up to 1,000 times in a loop, to address cases where the channel was still in use. However, this retry loop does not use the actual error, which means chan_dahdi will also retry opening nonexistent channels 1,000 times per channel, causing a flood of unnecessary warning logs for an operation that will never succeed, with tens or hundreds of thousands of open attempts being made. The original patch would have been more targeted if it only retried on the specific relevant error (likely EBUSY, although it's hard to say since the original issue is no longer available). To avoid the problem above while avoiding the possibility of breakage, this skips the retry logic if the error is ENXIO (No such device or address), since this will never succeed. Resolves: #669 (cherry picked from commit ec7e6fe2ed)	1 year ago
Naveen Albert	89a260d5ce	chan_dahdi: Allow specifying waitfordialtone per call. ... The existing "waitfordialtone" setting in chan_dahdi.conf applies permanently to a specific channel, regardless of how it is being used. This rather restrictively prevents a system from simultaneously being able to pick free lines for outgoing calls while also allowing barge-in to a trunk by some other arrangement. This allows specifying "waitfordialtone" using the CHANNEL function for only the next call that will be placed, allowing significantly more flexibility in the use of trunk interfaces. Resolves: #472 UserNote: "waitfordialtone" may now be specified for DAHDI trunk channels on a per-call basis using the CHANNEL function. (cherry picked from commit e10bca9f58)	1 year ago
George Joseph	f3c3c5720d	pjsip show channelstats: Prevent possible segfault when faxing ... Under rare circumstances, it's possible for the original audio session in the active_media_state default_session to be corrupted instead of removed when switching to the t38/image media session during fax negotiation. This can cause a segfault when a "pjsip show channelstats" attempts to print that audio media session's rtp statistics. In these cases, the active_media_state topology is correctly showing only a single t38/image stream so we now check that there's an audio stream in the topology before attempting to use the audio media session to get the rtp statistics. Resolves: #592 (cherry picked from commit 1aaf9e6a1d)	2 years ago
Naveen Albert	a7dd513b5d	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 49ec57fc7c)	2 years ago
PeterHolik	ea0cd51885	chan_rtp.c: MulticastRTP missing refcount without codec option ... Fixes: #529 (cherry picked from commit 683da188c9)	2 years ago
PeterHolik	ae747ba0ee	chan_rtp.c: Change MulticastRTP nameing to avoid memory leak ... Fixes: asterisk#536 (cherry picked from commit 9044dcc80a)	2 years ago
Naveen Albert	b9373a206c	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	212befbcf1	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 069d480138)	2 years ago
Naveen Albert	2be4ffa5e9	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 147f014072)	2 years ago
Sean Bright	152f5bdd2e	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 7b297cb90d)	2 years ago
George Joseph	d8f1be3b81	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 9a93ce0409)	2 years ago
Sean Bright	311ae80860	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 744bd4f9ac)	2 years ago
Naveen Albert	161ec4eab6	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 beb9689288)	2 years ago
Naveen Albert	f83156f826	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 4f99db350a)	2 years ago
Naveen Albert	c5cd205f60	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 cd90c5a82b)	2 years ago
Naveen Albert	59bc6ceb61	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 201d554b4e)	2 years ago
Maximilian Fridrich	98760e932e	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 0cd336a518)	2 years ago
Naveen Albert	c48bbbdbbc	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 333858fb70)	2 years ago
Naveen Albert	de8890e0e4	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. (cherry picked from commit c08c458fa2)	2 years ago
Naveen Albert	b145eb611e	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. (cherry picked from commit a670655b7a)	2 years ago
Sean Bright	9918cf1de3	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 (cherry picked from commit 39912b9138)	2 years ago
Naveen Albert	69f5aed6f7	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. (cherry picked from commit 0d1ddb15db)	2 years ago
Naveen Albert	88fd0ccd63	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 (cherry picked from commit 3260434b96)	2 years ago
Naveen Albert	0dbcc9370c	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. (cherry picked from commit f56477a604)	2 years ago
Maximilian Fridrich	8b81c5a16b	chan_pjsip: Allow topology/session refreshes in early media state ... 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 (cherry picked from commit a4cd452246)	2 years ago
Naveen Albert	6ea755eea1	chan_dahdi: Fix broken hidecallerid setting. ... 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 (cherry picked from commit d496544d7b)	2 years ago
InterLinked1	5dbd625bd6	chan_dahdi: Add dialmode option for FXS lines. (#36) ... 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. (cherry picked from commit 5d1dd11143)	2 years ago
Naveen Albert	6aa28346cf	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
Igor Goncharovsky	de745157ca	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	345ff2d8ee	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	8067229418	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" ... This reverts commit 62745013a4. Reason for revert: Issue when transcoding to/from g722 Change-Id: I1665a5442bfb6d7bfa06fdcea3374f4581395b4a	3 years ago
George Joseph	62745013a4	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
Naveen Albert	f5ead740f3	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
Naveen Albert	0bbcda3040	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	9de862242f	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	d41694350f	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	6c88400b3d	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	1f685d6969	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	bb8b2259bc	res_crypto: Use EVP API's instead of legacy API's ... ASTERISK-30046 #close Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2	3 years ago
Sean Bright	b3bf415455	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	d76f0506e5	chan_iax2: Add missing options documentation. ... Adds missing dial resource option documentation. ASTERISK-30164 #close Change-Id: I674e1fc9b1e5d67a20599bd4b418ce294d48fc83	3 years ago
Naveen Albert	08afdcbd30	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