8776 Commits (6249324f40bff8d4ec3b529bb7ceb87974659679)

Author	SHA1	Message	Date
George Joseph	546cefb305	chan_websocket: Fix codec validation and add passthrough option. ... * Fixed an issue in webchan_write() where we weren't detecting equivalent codecs properly. * Added the "p" dialstring option that puts the channel driver in "passthrough" mode where it will not attempt to re-frame or re-time media coming in over the websocket from the remote app. This can be used for any codec but MUST be used for codecs that use packet headers or whose data stream can't be broken up on arbitrary byte boundaries. In this case, the remote app is fully responsible for correctly framing and timing media sent to Asterisk and the MEDIA text commands that could be sent over the websocket are disabled. Currently, passthrough mode is automatically set for the opus, speex and g729 codecs. * Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to ensure proper translation paths are set up when switching between native frames and slin silence frames. This fixes an issue with codec errors when transcode_via_sln=yes. Resolves: #1462	1 week ago
Joe Garlick	88d61c1d66	chan_websocket.c: Add DTMF messages ... Added DTMF messages to the chan_websocket feature. When a user presses DTMF during a call over chan_websocket it will send a message like: "DTMF_END digit:1" Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/70 (cherry picked from commit 219ec0141c)	3 weeks ago
Naveen Albert	7b0566f8f4	sig_analog: Skip Caller ID spill if usecallerid=no. ... If Caller ID is disabled for an FXS port, then we should not send any Caller ID spill on the line, as we have no Caller ID information that we can/should be sending. Resolves: #1394 (cherry picked from commit 54cab33a43)	3 weeks ago
Naveen Albert	0d45d465bc	chan_dahdi: Fix erroneously persistent dialmode. ... It is possible to modify the dialmode setting in the chan_dahdi/sig_analog private using the CHANNEL function, to modify it during calls. However, it was not being reset between calls, meaning that if, for example, tone dialing was disabled, it would never work again unless explicitly enabled. This fixes the setting by pairing it with a "perm" version of the setting, as a few other features have, so that it can be reset to the permanent setting between calls. The documentation is also clarified to explain the interaction of this setting and the digitdetect setting more clearly. Resolves: #1378 (cherry picked from commit 576a29b988)	3 weeks ago
George Joseph	58621854fc	chan_websocket: Allow additional URI parameters to be added to the outgoing URI. ... * Added a new option to the WebSocket dial string to capture the additional URI parameters. * Added a new API ast_uri_verify_encoded() that verifies that a string either doesn't need URI encoding or that it has already been encoded. * Added a new API ast_websocket_client_add_uri_params() to add the params to the client websocket session. * Added XML documentation that will show up with `core show application Dial` that shows how to use it. Resolves: #1352 UserNote: A new WebSocket channel driver option `v` has been added to the Dial application that allows you to specify additional URI parameters on outgoing connections. Run `core show application Dial` from the Asterisk CLI to see how to use it. (cherry picked from commit 5d822d64ef)	3 weeks ago
George Joseph	b3c4f4195f	chan_websocket: Fix buffer overrun when processing TEXT websocket frames. ... ast_websocket_read() receives data into a fixed 64K buffer then continually reallocates a final buffer that, after all continuation frames have been received, is the exact length of the data received and returns that to the caller. process_text_message() in chan_websocket was attempting to set a NULL terminator on the received payload assuming the payload buffer it received was the large 64K buffer. The assumption was incorrect so when it tried to set a NULL terminator on the payload, it could, depending on the state of the heap at the time, cause heap corruption. process_text_message() now allocates its own payload_len + 1 sized buffer, copies the payload received from ast_websocket_read() into it then NULL terminates it prevent the possibility of the overrun and corruption. Resolves: #1384 (cherry picked from commit 09802fbef2)	3 weeks ago
Naveen Albert	c7ea11c35a	sig_analog: Fix SEGV due to calling strcmp on NULL. ... Add an additional check to guard against the channel application being NULL. Resolves: #1380 (cherry picked from commit 6f2167989a)	3 weeks ago
Naveen Albert	8ef3e4fb3d	sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf. ... Previously, we were only using # (ST) as a terminator, and not handling A (STP), B (ST2P), or C (ST3P), which erroneously led to it being treated as part of the dialed number. Parse any of these as the start digit. Resolves: #1301	3 months ago
kodokaii	f9219284cb	chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA ... In the WebSocket channel driver, the FLUSH_MEDIA command clears all frames from the queue but does not reset the frame_queue_length counter. As a result, the driver incorrectly thinks the queue is full after flushing, which prevents new multimedia frames from being sent, especially after multiple flush commands. This fix sets frame_queue_length to 0 after flushing, ensuring the queue state is consistent with its actual content. Fixes: #1304	3 months ago
Martin Tomec	9e7788ad21	chan_pjsip.c: Change SSRC after media source change ... When the RTP media source changes, such as after a blind transfer, the new source introduces a discontinuous timestamp. According to RFC 3550, Section 5.1, an RTP stream's timestamp for a given SSRC must increment monotonically and linearly. To comply with the standard and avoid a large timestamp jump on the existing SSRC, a new SSRC is generated for the new media stream. This change resolves known interoperability issues with certain SBCs (like Sonus/Ribbon) that stop forwarding media when they detect such a timestamp violation. This code uses the existing implementation from chan_sip. Resolves: #927	3 months ago
Naveen Albert	415daae95f	users.conf: Remove deprecated users.conf integration. ... users.conf was deprecated in Asterisk 21 and is now being removed for Asterisk 23, in accordance with the Asterisk deprecation policy. This consists of: * Removing integration with app_directory, app_voicemail, chan_dahdi, chan_iax2, and AMI. * users.conf was also partially used for res_phoneprov, and this remaining functionality is consolidated to a separate phoneprov_users.conf, used only by res_phoneprov. Resolves: #1292 UpgradeNote: users.conf has been removed and all channel drivers must be configured using their specific configuration files. The functionality previously in users.conf for res_phoneprov is now in phoneprov_users.conf.	3 months ago
George Joseph	5963e624e2	Media over Websocket Channel Driver ... * Created chan_websocket which can exchange media over both inbound and outbound websockets which the driver will frame and time. See http://s.asterisk.net/mow for more information. * res_http_websocket: Made defines for max message size public and converted a few nuisance verbose messages to debugs. * main/channel.c: Changed an obsolete nuisance error to a debug. * ARI channels: Updated externalMedia to include chan_websocket as a supported transport. UserNote: A new channel driver "chan_websocket" is now available. It can exchange media over both inbound and outbound websockets and will both frame and re-time the media it receives. See http://s.asterisk.net/mow for more information. UserNote: The ARI channels/externalMedia API now includes support for the WebSocket transport provided by chan_websocket.	3 months ago
Mike Bradeen	579cd0b8bb	chan_pjsip: Serialize INVITE creation on DTMF attended transfer ... When a call is transfered via DTMF feature code, the Transfer Target and Transferer are bridged immediately. This opens the possibilty of a race condition between the creation of an INVITE and the bridge induced colp update that can result in the set caller ID being over-written with the transferer's default info. Fixes: #1234	5 months ago
Naveen Albert	2ced79259a	sig_analog: Add Call Waiting Deluxe support. ... Adds support for Call Waiting Deluxe options to enhance the current call waiting feature. As part of this change, a mechanism is also added that allows a channel driver to queue an audio file for Dial() to play, which is necessary for the announcement function. ASTERISK-30373 #close Resolves: #271 UserNote: Call Waiting Deluxe can now be enabled for FXS channels by enabling its corresponding option.	5 months ago
Naveen Albert	d60bcc56a7	chan_iax2: Minor improvements to documentation and warning messages. ... * Update Dial() documentation for IAX2 to include syntax for RSA public key names. * Add additional details to a couple warnings to provide more context when an undecodable frame is received. Resolves: #1206	6 months ago
Florent CHAUVEAU	a5bc39fa32	audiosocket: added support for DTMF frames ... Updated the AudioSocket protocol to allow sending DTMF frames. AST_FRAME_DTMF frames are now forwarded to the server, in addition to AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF with value 0x03 was added to the protocol. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...). UserNote: The AudioSocket protocol now forwards DTMF frames with payload type 0x03. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...).	6 months ago
Norm Harrison	4e546c35c7	audiosocket: fix timeout, fix dialplan app exit, server address in logs ... - Correct wait timeout logic in the dialplan application. - Include server address in log messages for better traceability. - Allow dialplan app to exit gracefully on hangup messages and socket closure. - Optimize I/O by reducing redundant read()/write() operations. Co-authored-by: Florent CHAUVEAU <florentch@pm.me>	6 months ago
Alexei Gradinari	1f42d20d9c	chan_pjsip: set correct Endpoint Device State on multiple channels ... 1. When one channel is placed on hold, the device state is set to ONHOLD without checking other channels states. In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN to calculate aggregate device state of all active channels. 2. The current implementation incorrectly classifies channels in use. The only channels that has the states: UP, RING and BUSY are considered as "in use". A channel should be considered "in use" if its state is anything other than DOWN or RESERVED. 3. Currently, if the number of channels "in use" is greater than device_state_busy_at, the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate device state. The endpoint device state should be BUSY if the number of channels "in use" is greater than or equal to device_state_busy_at. Fixes: #1181	6 months ago
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`	8 months ago
Holger Hans Peter Freyther	9fc631d028	ari/pjsip: Make it possible to control transfers through ARI ... Introduce a ChannelTransfer event and the ability to notify progress to ARI. Implement emitting this event from the PJSIP channel instead of handling the transfer in Asterisk when configured. Introduce a dialplan function to the PJSIP channel to switch between the "core" and "ari-only" behavior. UserNote: Call transfers on the PJSIP channel can now be controlled by ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.	8 months ago
Sean Bright	82bef2ed90	docs: Indent <since> tags. ... Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).	8 months ago
George Joseph	198b01c536	docs: Add version information to application and function XML elements ... * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml	8 months ago
George Joseph	77d5550d14	docs: Add version information to manager event instance XML elements ... * Do a git blame on the embedded XML managerEvent elements. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The events defined in res/res_pjsip/pjsip_manager.xml were done by hand because the XML was extracted from the C source file relatively recently. Two bugs were fixed along the way... * The get_documentation awk script was exiting after it processed the first DOCUMENTATION block it found in a file. We have at least 1 source file with multiple DOCUMENTATION blocks so only the first one in them was being processed. The awk script was changed to continue searching rather than exiting after the first block. * Fixing the awk script revealed an issue in logger.c where the third DOCUMENTATION block contained a XML fragment that consisted only of a managerEventInstance element that wasn't wrapped in a managerEvent element. Since logger_doc.xml already existed, the remaining fragments in logger.c were moved to it and properly organized.	8 months ago
George Joseph	316693ded9	docs: Add version information to configObject and configOption XML elements ... Most of the configObjects and configOptions that are implemented with ACO or Sorcery now have `<since>/<version>` elements added. There are probably some that the script I used didn't catch. The version tags were determined by the following... * Do a git blame on the API call that created the object or option. * From the commit hash, grab the summary line. * Do a `git log --grep <summary>` to find the cherry-pick commits in all branches that match. * Do a `git patch-id` to ensure the commits are all related and didn't get a false match on the summary. * Do a `git tag --contains <commit>` to find the tags that contain each commit. * Weed out all tags not <major>.<minor>.0. * Sort and discard any <major>.0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the API was last touched. configObjects and configOptions elements implemented with the base ast_config APIs were just not possible to find due to the non-deterministic way they are accessed. Also note that if the API call was on modified after it was added, the version will be the one it was last modified in. Final note: The configObject and configOption elements were introduced in 12.0.0 so options created before then may not have any XML documentation.	9 months ago
Naveen Albert	f6e1c163bb	chan_iax2: Avoid unnecessarily backlogging non-voice frames. ... Currently, when receiving an unauthenticated call, we keep track of the negotiated format in the chosenformat, which allows us to later create the channel using the right format. However, this was not done for authenticated calls. This meant that in certain circumstances, if we had not yet received a voice frame from the peer, only certain other types of frames (e.g. text), there were no variables containing the appropriate frame. This led to problems in the jitterbuffer callback where we unnecessarily bailed out of retrieving a frame from the jitterbuffer. This was logic intentionally added in commit 73103bdcd5 in response to an earlier regression, and while this prevents crashes, it also backlogs legitimate frames unnecessarily. The abort logic was initially added because at this point in the code, we did not have the negotiated format available to us. However, it should always be available to us as a last resort in chosenformat, so we now pull it from there if needed. This allows us to process frames the jitterbuffer even if voicefmt and peerfmt aren't set and still avoid the crash. The failsafe logic is retained, but now it shouldn't be triggered anymore. Resolves: #1054	9 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	9 months ago
George Joseph	114fa12f05	docs: Various XML fixes ... * channels/pjsip/dialplan_functions_doc.xml: Added xmlns:xi to docs element. * main/bucket.c: Removed XML completely since the "bucket" and "file" objects are internal only with no config file. * main/named_acl.c: Fixed the configFile element name. It was "named_acl.conf" and should have been "acl.conf" * res/res_geolocation/geoloc_doc.xml: Added xmlns:xi to docs element. * res/res_http_media_cache.c: Fixed the configFile element name. It was "http_media_cache.conf" and should have been "res_http_media_cache.conf".	9 months ago
Sean Bright	88abf2207b	dialplan_functions_doc.xml: Document PJSIP_MEDIA_OFFER's `media` argument. ... Resolves: #1023	9 months ago
Sean Bright	468601fd13	manager: Add `<since>` tags for all AMI actions.	9 months ago
Naveen Albert	5ed7f4d152	chan_dahdi: Fix wrong channel state when RINGING recieved. ... Previously, when AST_CONTROL_RINGING was received by a DAHDI device, it would set its channel state to AST_STATE_RINGING. However, an analysis of the codebase and other channel drivers reveals RINGING corresponds to physical power ringing, whereas AST_STATE_RING should be used for audible ringback on the channel. This also ensures the correct device state is returned by the channel state to device state conversion. Since there seems to be confusion in various places regarding AST_STATE_RING vs. AST_STATE_RINGING, some documentation has been added or corrected to clarify the actual purposes of these two channel states, and the associated device state mapping. An edge case that prompted this fix, but isn't explicitly addressed here, is that of an incoming call to an FXO port. The channel state will be "Ring", which maps to a device state of "In Use", not "Ringing" as would be more intuitive. However, this is semantic, since technically, Asterisk is treating this the same as any other incoming call, and so "Ring" is the semantic state (put another way, Asterisk isn't ringing anything, like in the cases where channels are in the "Ringing" state). Since FXO ports don't currently support Call Waiting, a suitable workaround for the above would be to ignore the device state and instead check the channel state (e.g. IMPORT(DAHDI/1-1,CHANNEL(state))) since it will be Ring if the FXO port is idle (but a call is ringing on it) and Up if the FXO port is actually in use. (In both cases, the device state would misleadingly be "In Use".) Resolves: #1029	9 months ago
Naveen Albert	d09359b9d9	chan_iax2: Add log message for rejected calls. ... Add a log message for a path that currently silently drops IAX2 frames without indicating that anything is wrong.	10 months ago
Maximilian Fridrich	0f1f0ef3ef	chan_pjsip: Send VIDUPDATE RTP frame for all H.264 streams ... Currently, when a chan_pjsip channel receives a VIDUPDATE indication, an RTP VIDUPDATE frame is only queued on a H.264 stream if WebRTC is enabled on that endpoint. This restriction does not really make sense. Now, a VIDUPDATE RTP frame is written even if WebRTC is not enabled (as is the case with VP8, VP9, and H.265 streams). Resolves: #1013	10 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	11 months ago
Naveen Albert	d4358d5f7a	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	1 year ago
Mike Bradeen	b2455b2732	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>	1 year ago
Ben Ford	fb0e46e464	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.	1 year ago
Sean Bright	25686a57da	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	1 year ago
Naveen Albert	c1b69d460d	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	1 year ago
Naveen Albert	0923a49cc2	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	2 years ago
George Joseph	f6b9d9e7d7	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.	2 years ago
Naveen Albert	2de1a68339	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	2 years ago
Naveen Albert	953dc3d127	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.	2 years ago
George Joseph	29a273618d	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	2 years ago
Naveen Albert	a3be6a455f	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	2 years ago
Naveen Albert	ac71e40042	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
PeterHolik	5216a1133d	chan_rtp.c: MulticastRTP missing refcount without codec option ... Fixes: #529	2 years ago
PeterHolik	25fec97428	chan_rtp.c: Change MulticastRTP nameing to avoid memory leak ... Fixes: asterisk#536	2 years 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	2 years ago
Naveen Albert	c4bf59b781	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	2 years ago
Sean Bright	fb74ef1c7a	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.	2 years ago