6585 Commits (44d68bd56b1bfdbda718327e95174c52f7bc4e55)

Author	SHA1	Message	Date
Sean Bright	1b74555fcf	asterisk: Export additional manager functions ... Rename check_manager_enabled() and check_webmanager_enabled() to begin with ast_ so that the symbols are automatically exported by the linker. ASTERISK~29184 Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9	5 years ago
Alexander Traud	80c14f74bc	codecs: Remove test-law. ... This was dead code, test code introduced with Asterisk 13. This was found while analyzing ASTERISK_28416 and ASTERISK_29185. This change partly fixes, not closes those two issues. Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940	5 years ago
Sean Bright	357510cec3	app_chanspy: Spyee information missing in ChanSpyStop AMI Event ... The documentation in the wiki says there should be spyee-channel information elements in the ChanSpyStop AMI event. https://wiki.asterisk.org/wiki/x/Xc5uAg However, this is not the case in Asterisk <= 16.10.0 Version. We're using these Spyee* arguments since Asterisk 11.x, so these arguments vanished in Asterisk 12 or higher. For maximum compatibility, we still send the ChanSpyStop event even if we are not able to find any 'Spyee' information. ASTERISK-28883 #close Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f	5 years ago
George Joseph	7d4ae7dc18	logger.c: Automatically add a newline to formats that don't have one ... Scope tracing allows you to not specify a format string or variable, in which case it just prints the indent, file, function, and line number. The trace output automatically adds a newline to the end in this case. If you also have debugging turned on for the module, a debug message is also printed but the standard log functionality which prints it doesn't add the newline so you have messages that don't break correctly. * format_log_message_ap(), which is the common log message formatter for all channels, now adds a newline to the end of format strings that don't already have a newline. ASTERISK-29209 Reported by: Alexander Traud Change-Id: I994a7df27f88df343b7d19f3e81a4b562d9d41da	5 years ago
lvl	b08427134f	Introduce astcachedir, to be used for temporary bucket files ... As described in the issue, /tmp is not a suitable location for a large amount of cached media files, since most distributions make /tmp a RAM-based tmpfs mount with limited capacity. I opted for a location that can be configured separately, as opposed to using a subdirectory of spooldir, given the different storage profile (transient files vs files that might stay there indefinitely). This commit just makes the cache directory configurable, and changes the default location from /tmp to /var/cache/asterisk. ASTERISK-29143 Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d	5 years ago
Sean Bright	c8b6340023	media_cache: Fix reference leak with bucket file metadata ... Change-Id: Ia0e4124110df613ce5fdfa9ef8780016ebaa52c6	5 years ago
Boris P. Korzun	8cb439f7e4	bridge_basic: Fixed setup of recall channels ... Fixed a bug (like a typo) in retransfer_enter() at main/bridge_basic.c:2641. common_recall_channel_setup() setups common things on the recalled transfer target, but used same target as source instead trasfered. ASTERISK-29161 #close Change-Id: Ieb549654a621c38b1ad5e9d15b9f18823d9cc31f	5 years ago
Alexander Traud	28faafd1c4	Compiler fixes for GCC when printf %s is NULL ... ASTERISK-29146 Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41	5 years ago
Walter Doekes	1650d50e91	main/say: Work around gcc 9 format-truncation false positive ... Version: gcc (Ubuntu 9.3.0-10ubuntu2) 9.3.0 Warning: say.c:2371:24: error: ‘%d’ directive output may be truncated writing between 1 and 11 bytes into a region of size 10 [-Werror=format-truncation=] 2371 | snprintf(buf, 10, "%d", num); say.c:2371:23: note: directive argument in the range [-2147483648, 9] That's not possible though, as the if() starts out checking for (num < 0), making this Warning a false positive. (Also replaced some else<TAB>if with else<SP>if while in the vicinity.) Change-Id: Ic7a70120188c9aa525a6d70289385bfce878438a	5 years ago
Sean Bright	fa023cbfa0	tcptls.c: Don't close TCP client file descriptors more than once ... ASTERISK-28430 #close Change-Id: Ib556b0a0c95cca939e956886214ec8d828d89606	5 years ago
Kevin Harwell	56028426de	Logging: Add debug logging categories ... Added debug logging categories that allow a user to output debug information based on a specified category. This lets the user limit, and filter debug output to data relevant to a particular context, or topic. For instance the following categories are now available for debug logging purposes: dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet These debug categories can be enable/disable via an Asterisk CLI command. While this overrides, and outputs debug data, core system debugging is not affected by this patch. Statements still output at their appropriate debug level. As well backwards compatibility has been maintained with past debug groups that could be enabled using the CLI (e.g. rtpdebug, stundebug, etc.). ASTERISK-29054 #close Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849	6 years ago
Sean Bright	51cba591e3	pbx.c: On error, ast_add_extension2_lockopt should always free 'data' ... In the event that the desired extension already exists, ast_add_extension2_lockopt() will free the 'data' it is passed before returning an error, so we should not be freeing it ourselves. Additionally, there were two places where ast_add_extension2_lockopt() could return an error without also freeing the 'data' pointer, so we add that. ASTERISK-29097 #close Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae	6 years ago
George Joseph	773f424c7f	app_confbridge/bridge_softmix: Add ability to force estimated bitrate ... app_confbridge now has the ability to set the estimated bitrate on an SFU bridge. To use it, set a bridge profile's remb_behavior to "force" and set remb_estimated_bitrate to a rate in bits per second. The remb_estimated_bitrate parameter is ignored if remb_behavior is something other than "force". Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a	6 years ago
Jasper van der Neut	e831952eba	channels: Don't dereference NULL pointer ... Check result of ast_translator_build_path against NULL before dereferencing. ASTERISK-29091 Change-Id: Ia3538ea190bd371f70c9dd49984b021765691b29	6 years ago
Sean Bright	16dfe8f03f	dsp.c: Update calls to ast_format_cmp to check result properly ... ASTERISK-28311 #close Change-Id: Ib1ce8fc1a8752751f5bf3615c59245532dfd9aa2	6 years ago
Sean Bright	30e08ce1bb	format_cap: Perform codec lookups by pointer instead of name ... ASTERISK-28416 #close Change-Id: I069420875ebdbcaada52d92599a5f7de3cb2cdf4	6 years ago
George Joseph	44bb0858cb	debugging: Add enough to choke a mule ... Added to: * bridges/bridge_softmix.c * channels/chan_pjsip.c * include/asterisk/res_pjsip_session.h * main/channel.c * res/res_pjsip_session.c There NO functional changes in this commit. Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3	6 years ago
George Joseph	86f1bce186	res_pjsip_session: Handle multi-stream re-invites better ... When both Asterisk and a UA send re-invites at the same time, both send 491 "Transaction in progress" responses to each other and back off a specified amount of time before retrying. When Asterisk prepares to send its re-invite, it sets up the session's pending media state with the new topology it wants, then sends the re-invite. Unfortunately, when it received the re-invite from the UA, it partially processed the media in the re-invite and reset the pending media state before sending the 491 losing the state it set in its own re-invite. Asterisk also was not tracking re-invites received while an existing re-invite was queued resulting in sending stale SDP with missing or duplicated streams, or no re-invite at all because we erroneously determined that a re-invite wasn't needed. There was also an issue in bridge_softmix where we were using a stream from the wrong topology to determine if a stream was added. This also caused us to erroneously determine that a re-invite wasn't needed. Regardless of how the delayed re-invite was triggered, we need to reconcile the topology that was active at the time the delayed request was queued, the pending topology of the queued request, and the topology currently active on the session. To do this we need a topology resolver AND we need to make stream named unique so we can accurately tell what a stream has been added or removed and if we can re-use a slot in the topology. Summary of changes: * bridge_softmix: * We no longer reset the stream name to "removed" in remove_all_original_streams(). That was causing multiple streams to have the same name and wrecked the checks for duplicate streams. * softmix_bridge_stream_sources_update() was checking the old_stream to see if it had the softmix prefix and not considering the stream as "new" if it did. If the stream in that slot has something in it because another re-invite happened, then that slot in old might have a softmix stream but the same stream in new might actually be a new one. Now we check the new_stream's name instead of the old_stream's. * stream: * Instead of using plain media type name ("audio", "video", etc) as the default stream name, we now append the stream position to it to make it unique. We need to do this so we can distinguish multiple streams of the same type from each other. * When we set a stream's state to REMOVED, we no longer reset its name to "removed" or destroy its metadata. Again, we need to do this so we can distinguish multiple streams of the same type from each other. * res_pjsip_session: * Added resolve_refresh_media_states() that takes in 3 media states and creates an up-to-date pending media state that includes the changes that might have happened while a delayed session refresh was in the delayed queue. * Added is_media_state_valid() that checks the consistency of a media state and returns a true/false value. A valid state has: * The same number of stream entries as media session entries. Some media session entries can be NULL however. * No duplicate streams. * A valid stream for each non-NULL media session. * A stream that matches each media session's stream_num and media type. * Updated handle_incoming_sdp() to set the stream name to include the stream position number in the name to make it unique. * Updated the ast_sip_session_delayed_request structure to include both the pending and active media states and updated the associated delay functions to process them. * Updated sip_session_refresh() to accept both the pending and active media states that were in effect when the request was originally queued and to pass them on should the request need to be delayed again. * Updated sip_session_refresh() to call resolve_refresh_media_states() and substitute its results for the pending state passed in. * Updated sip_session_refresh() with additional debugging. * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE to pjproject if a transaction is in progress. This stops us from creating a partial pending media state that would be invalid later on. * Updated reschedule_reinvite() to clone both the current pending and active media states and pass them to delay_request() so the resolver can tell what the original intention of the re-invite was. * Added a large unit test for the resolver. ASTERISK-29014 Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb	6 years ago
Sungtae Kim	aae0904c7d	res_stasis.c: Added video_single option for bridge creation ... Currently, it was not possible to create bridge with video_mode single. This made hard to put the bridge in a vidoe_single mode. So, added video_single option for Bridge creation using the ARI. This allows create a bridge with video_mode single. ASTERISK-29055 Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae	6 years ago
Ben Ford	80a609fcce	Bridging: Use a ref to bridge_channel's channel to prevent crash. ... There's a race condition with bridging where a bridge can be torn down causing the bridge_channel's ast_channel to become NULL when it's still needed. This particular case happened with attended transfers, but the crash occurred when trying to publish a stasis message. Now, the bridge_channel is locked, a ref to the ast_channel is obtained, and that ref is passed down the chain. Change-Id: Ic48715c0c041615d17d286790ae3e8c61bb28814	6 years ago
Kevin Harwell	1a5597741f	conversions: Add string to signed integer conversion functions ... Change-Id: Id603b0b03b78eb84c7fca030a08b343c0d5973f9	6 years ago
Joshua C. Colp	28bae5e901	pbx: Fix hints deadlock between reload and ExtensionState. ... When the ExtensionState AMI action is executed on a pattern matched hint it can end up adding a new hint if one does not already exist. This results in a locking order of contexts -> hints -> contexts. If at the same time a reload is occurring and adding its own hint it will have a locking order of hints -> contexts. This results in a deadlock as one thread wants a lock on contexts that the other has, and the other thread wants a lock on hints that the other has. This change enforces a hints -> contexts locking order by explicitly locking hints in the places where a hint is added when queried for. This matches the order seen through normal adding of hints. ASTERISK-29046 Change-Id: I49f027f4aab5d2d50855ae937bcf5e2fd8bfc504	6 years ago
George Joseph	54ddf19141	logger.c: Added a new log formatter called "plain" ... Added a new log formatter called "plain" that always prints file, function and line number if available (even for verbose messages) and never prints color control characters. It also doesn't apply any special formatting for verbose messages. Most suitable for file output but can be used for other channels as well. You use it in logger.conf like so: debug => [plain]debug console => [plain]error,warning,debug,notice,pjsip_history messages => [plain]warning,error,verbose Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d	6 years ago
Sean Bright	3553192900	bridge_channel: Ensure text messages are zero terminated ... T.140 data in RTP is not zero terminated, so when we are queuing a text frame on a bridge we need to ensure that we are passing a zero terminated string. ASTERISK-28974 #close Change-Id: Ic10057387ce30b2094613ea67e3ae8c5c431dda3	6 years ago
George Joseph	64ca2d48da	scope_trace: Added debug messages and added additional macros ... The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages at the same level as the scope level. This allows the same messages to be printed to the debug log when AST_DEVMODE isn't enabled. Also added a few variants of the SCOPE_EXIT macros that will also call ast_log instead of ast_debug to make it easier to use scope tracing and still print error messages. Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21	6 years ago
George Joseph	118cb3f0dd	stream.c: Added 2 more debugging utils and added pos to stream string ... * Added ast_stream_to_stra and ast_stream_topology_to_stra() macros which are shortcuts for ast_str_tmp(256, ast_stream_to_str(stream, &STR_TMP)) * Added the stream position to the string representation of the stream. * Fixed some formatting in ast_stream_to_str(). Change-Id: Idaf4cb0affa46d4dce58a73a111f35435331cc4b	6 years ago
George Joseph	647c53c41f	ACN: Changes specific to the core ... Allow passing a topology from the called channel back to the calling channel. * Added a new function ast_queue_answer() that accepts a stream topology and queues an ANSWER CONTROL frame with it as the data. This allows the called channel to indicate its resolved topology. * Added a new virtual function to the channel tech structure answer_with_stream_topology() that allows the calling channel to receive the called channel's topology. Added ast_raw_answer_with_stream_topology() that invokes that virtual function. * Modified app_dial.c and features.c to grab the topology from the ANSWER frame queued by the answering channel and send it to the calling channel with ast_raw_answer_with_stream_topology(). * Modified frame.c to automatically cleanup the reference to the topology on ANSWER frames. Added a few debugging messages to stream.c. Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c	6 years ago
Ben Ford	9ed6387c14	utils.c: NULL terminate ast_base64decode_string. ... With the addition of STIR/SHAKEN, the function ast_base64decode_string was added for convenience since there is a lot of converting done during the STIR/SHAKEN process. This function returned the decoded string for you, but did not NULL terminate it, causing some issues (specifically with MALLOC_DEBUG). Now, the returned string is NULL terminated, and the documentation has been updated to reflect this. Change-Id: Icdd7d05b323b0c47ff6ed43492937a03641bdcf5	6 years ago
Sean Bright	7d96b3e437	utf8.c: Add UTF-8 validation and utility functions ... There are various places in Asterisk - specifically in regards to database integration - where having some kind of UTF-8 validation would be beneficial. This patch adds: * Functions to validate that a given string contains only valid UTF-8 sequences. * A function to copy a string (similar to ast_copy_string) stopping when an invalid UTF-8 sequence is encountered. * A UTF-8 validator that allows for progressive validation. All of this is based on the excellent UTF-8 decoder by Björn Höhrmann. More information is available here: https://bjoern.hoehrmann.de/utf-8/decoder/dfa/ The API was written in such a way that should allow us to replace the implementation later should we determine that we need something more comprehensive. Change-Id: I3555d787a79e7c780a7800cd26e0b5056368abf9	6 years ago
sungtae kim	c10ed8d4d6	stasis_bridge.c: Fixed wrong video_mode shown ... Currently, if the bridge has created by the ARI, the video_mode parameter was not shown in the BridgeCreated event correctly. Fixed it and added video_mode shown in the 'bridge show <bridge id>' cli. ASTERISK-28987 Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295	6 years ago
Sean Bright	c3588d9c0b	acl.c: Coerce a NULL pointer into the empty string ... If an ACL is misconfigured in the realtime database (for instance, the "rule" is blank) and Asterisk attempts to read the ACL, Asterisk will crash. ASTERISK-28978 #close Change-Id: Ic1536c4df856231bfd2da00128f7822224d77610	6 years ago
George Joseph	9bd1d686a1	ACN: Add tracing to existing code ... Prior to making any modifications to the pjsip infrastructure for ACN, I've added the tracing functions to the existing code. This should make the final commit easier to review, but we can also now run a "before and after" trace. No functional changes were made with this commit. Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c	6 years ago
George Joseph	d093e44b1e	frame.c: Make debugging easier ... * ast_frame_subclass2str() and ast_frame_type2str() now return a pointer to the buffer that was passed in instead of void. This makes it easier to use these functions inline in printf-style debugging statements. * Added many missing control frame entries in ast_frame_subclass2str. Change-Id: Ifd0d6578e758cd644c96d17a5383ff2128c572fc	6 years ago
George Joseph	955b7b4fdb	Scope Trace: Make it easier to trace through synchronous tasks ... Tracing through synchronous tasks was a little troublesome because the new thread's stack counter reset to 0. This change allows a synchronous task to set its trace level to be the same as the thread that pushed the task. For now, the task's level has to be passed in the task's data structure but a future enhancement to the taskprocessor subsystem could automatically set the trace level of the servant to be that of the caller. This doesn't really make sense for async tasks because you never know when they're going to run anyway. Change-Id: Ib8049c0b815063a45d8c7b0cb4e30b7b87b1d825	6 years ago
Kevin Harwell	cfed0ea033	manager - Add Content-Type parameter to the SendText action ... This patch allows a user of AMI to now specify the type of message content contained within by setting the 'Content-Type' parameter. Note, the AMI version has been bumped for this change. ASTERISK-28945 #close Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb	6 years ago
George Joseph	8d1064eaaf	Streams: Add features for Advanced Codec Negotiation ... The Streams API becomes the home for the core ACN capabilities. These include... * Parsing and formatting of codec negotation preferences. * Resolving pending streams and topologies with those configured using configured preferences. * Utility functions for creating string representations of streams, topologies, and negotiation preferences. For codec negotiation preferences: * Added ast_stream_codec_prefs_parse() which takes a string representation of codec negotiation preferences, which may come from a pjsip endpoint for example, and populates a ast_stream_codec_negotiation_prefs structure. * Added ast_stream_codec_prefs_to_str() which does the reverse. * Added many functions to parse individual parameter name and value strings to their respectrive enum values, and the reverse. For streams: * Added ast_stream_create_resolved() which takes a "live" stream and resolves it with a configured stream and the negotiation preferences to create a new stream. * Added ast_stream_to_str() which create a string representation of a stream suitable for debug or display purposes. For topology: * Added ast_stream_topology_create_resolved() which takes a "live" topology and resolves it, stream by stream, with a configured topology stream and the negotiation preferences to create a new topology. * Added ast_stream_topology_to_str() which create a string representation of a topology suitable for debug or display purposes. * Renamed ast_format_caps_from_topology() to ast_stream_topology_get_formats() to be more consistent with the existing ast_stream_get_formats(). Additional changes: * A new function ast_format_cap_append_names() appends the results to the ast_str buffer instead of replacing buffer contents. Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56	6 years ago
Ben Ford	1274117102	res_stir_shaken: Add outbound INVITE support. ... Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is sent, the caller ID will be checked to see if there is a certificate that corresponds to it. If so, that information will be retrieved and an Identity header will be added to the SIP message. The format is: header.payload.signature;info=<public_key_url>alg=ES256;ppt=shaken Header, payload, and signature are all BASE64 encoded. The public key URL is retrieved from the certificate. Currently the algorithm and ppt are ES256 and shaken, respectively. This message is signed and can be used for verification on the receiving end. Two new configuration options have been added to the certificate object: attestation and origid. The attestation is required and must be A, B, or C. origid is the origination identifier. A new utility function has been added as well that takes a string, allocates space, BASE64 encodes it, then returns it, eliminating the need to calculate the size yourself. Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4	6 years ago
Joshua C. Colp	de2813cf23	core_unreal / core_local: Add multistream and re-negotiation. ... When requesting a Local channel the requested stream topology or a converted stream topology will now be placed onto the resulting channels. Frames written in on streams will now also preserve the stream identifier as they are queued on the opposite channel. Finally when a stream topology change is requested it is immediately accepted and reflected on both channels. Each channel also receives a queued frame to indicate that the topology has changed. ASTERISK-28938 Change-Id: I4e9d94da5230d4bd046dc755651493fce1d87186	6 years ago
Kevin Harwell	3d1bf3c537	Compiler fixes for gcc 10 ... This patch fixes a few compile warnings/errors that now occur when using gcc 10+. Also, the Makefile.rules check to turn off partial inlining in gcc versions greater or equal to 8.2.1 had a bug where it only it only checked against versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures any version above the specified version is correctly compared. Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9	6 years ago
Ben Ford	559fa0e89c	cli.c: Fix compiler error. ... Added default variable value to fix a compiler error. Change-Id: I7b592adbb1274dc5464dea1c5e5de0685c928553	6 years ago
Ben Ford	3927f79cb5	res_stir_shaken: Add inbound INVITE support. ... Integrated STIR/SHAKEN support with incoming INVITES. Upon receiving an INVITE, the Identity header is retrieved, parsing the message to verify the signature. If any of the parsing fails, AST_STIR_SHAKEN_VERIFY_NOT_PRESENT will be added to the channel for this caller ID. If verification itself fails, AST_STIR_SHAKEN_VERIFY_SIGNATURE_FAILED will be added. If anything in the payload does not line up with the SIP signaling, AST_STIR_SHAKEN_VERIFY_MISMATCH will be added. If all of the above steps pass, then AST_STIR_SHAKEN_VERIFY_PASSED will be added, completing the verification process. A new config option has been added to the general section for stir_shaken.conf. "signature_timeout" is the amount of time a signature will be considered valid. If an INVITE is received and the amount of time between when it was received and when it was signed is greater than signature_timeout, verification will fail. Some changes were also made to signing and verification. There was an error where the whole JSON string was being signed rather than the header combined with the payload. This has been changed to sign the correct thing. Verification has been changed to do this as well, and the unit tests have been updated to reflect these changes. A couple of utility functions have also been added. One decodes a BASE64 string and returns the decoded string, doing all the length calculations for you. The other retrieves a string value from a header in a rdata object. Change-Id: I855f857be3d1c63b64812ac35d9ce0534085b913	6 years ago
Joshua C. Colp	1fcb6b1b21	bridge_channel: Don't queue unmapped frames. ... If a frame is written to a channel in a bridge we would normally queue this frame up and the channel thread would then act upon it. If this frame had no stream mapping on the channel it would then be discarded. This change adds a check before the queueing occurs to determine if a mapping exists. If it does not exist then the frame is not even queued at all. This stops a frame duplication from happening and from the channel thread having to wake up and deal with it. Change-Id: I17189b9b1dec45fc7e4490e8081d444a25a00bda	6 years ago
sungtae kim	25ae412f75	bridge.c: Fixed null pointer exception ... If the bridge show all command could not get the bridge snapshot, it causes null pointer exception. Fixed it to check the snapshot is null. ASTERISK-28920 Change-Id: I3521fc1b832bfc69644d0833f2c78177e1e51f58	6 years ago
George Joseph	ca3c22c5f1	Scope Tracing: A new facility for tracing scope enter/exit ... What's wrong with ast_debug? ast_debug is fine for general purpose debug output but it's not really geared for scope tracing since it doesn't present its output in a way that makes capturing and analyzing flow through Asterisk easy. How is scope tracing better? Scope tracing uses the same "cleanup" attribute that RAII_VAR uses to print messages to a separate "trace" log level. Even better, the messages are indented and unindented based on a thread-local call depth counter. When output to a separate log file, the output is uncluttered and easy to follow. Here's an example of the output. The leading timestamps and thread ids are removed and the output cut off at 68 columns for commit message restrictions but you get the idea. --> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 --> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 --> res_pjsip_session.c:3669 handle_incoming_response PJSIP/ --> chan_pjsip.c:3265 chan_pjsip_incoming_response_after --> chan_pjsip.c:3194 chan_pjsip_incoming_response P chan_pjsip.c:3245 chan_pjsip_incoming_respon <-- chan_pjsip.c:3194 chan_pjsip_incoming_response P <-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after <-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/ <-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 <-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 The messages with the "-->" or "<--" were produced by including the following at the top of each function: SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session)); Scope isn't limited to functions any more than RAII_VAR is. You can also see entry and exit from "if", "for", "while", etc blocks. There is also an ast_trace() macro that doesn't track entry or exit but simply outputs a message to the trace log using the current indent level. The deepest message in the sample (chan_pjsip.c:3245) was used to indicate which "case" in a "select" was executed. How do you use it? More documentation is available in logger.h but here's an overview: * Configure with --enable-dev-mode. Like debug, scope tracing is #ifdef'd out if devmode isn't enabled. * Add a SCOPE_TRACE() call to the top of your function. * Set a logger channel in logger.conf to output the "trace" level. * Use the CLI (or cli.conf) to set a trace level similar to setting debug level... CLI> core set trace 2 res_pjsip.so Summary Of Changes: * Added LOG_TRACE logger level. Actually it occupies the slot formerly occupied by the now defunct "event" level. * Added core asterisk option "trace" similar to debug. Includes ability to specify global trace level in asterisk.conf and CLI commands to turn on/off and set levels. Levels can be set globally (probably not a good idea), or by module/source file. * Updated sample asterisk.conf and logger.conf. Tracing is disabled by default in both. * Added __ast_trace() to logger.c which keeps track of the indent level using TLS. It's #ifdef'd out if devmode isn't enabled. * Added ast_trace() and SCOPE_TRACE() macros to logger.h. These are all #ifdef'd out if devmode isn't enabled. Why not use gcc's -finstrument-functions capability? gcc's facility doesn't allow access to local data and doesn't operate on non-function scopes. Known Issues: The only know issue is that we currently don't know the line number where the scope exited. It's reported as the same place the scope was entered. There's probably a way to get around it but it might involve looking at the stack and doing an 'addr2line' to get the line number. Kind of like ast_backtrace() does. Not sure if it's worth it. Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027	6 years ago
traud	f9ea75d117	tcptls: Fix notice when TLS is enabled but not supported. ... ASTERISK-28797 Change-Id: Iab364a2c2519fd9d11d1c28293fda43d61b64c28	6 years ago
Pirmin Walthert	6b2d945174	app.c: make sure that no non-async-signal-safe syscalls are used after ... fork before exec Posix does only allow async-signal-safe syscalls after fork before exec. As asterisk ignores this, functions like TrySystem or System sometimes end up in a deadlocked child process. The patch prevents the use of non-async-signal-safe syscalls. ASTERISK-28776 Change-Id: Idc76365c0592ee3f3b3bd72a4f48f7a098978e8e	6 years ago
George Joseph	7fbfbe7da0	streams: Fix one memory leak and one formats ref issue ... ast_stream_topology_create_from_format_cap() was setting the stream->formats directly but not freeing the default formats. This causes a memory leak. * ast_stream_topology_create_from_format_cap() now calls ast_stream_set_formats() which properly cleans up the existing stream formats. When cloning a stream, the source stream's format caps _pointer_ is copied to the new stream and it's reference count bumped. If either stream is set to "removed", this will cause _both_ streams to have their format caps cleared. * ast_stream_clone() now creates a new format caps object and copies the formats from the source stream instead of just copying the pointer. ASTERISK-28870 Change-Id: If697d81c3658eb7baeea6dab413b13423938fb53	6 years ago
Nathan Bruning	f217fcdc62	app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions ... Add a new "masquarade" channel event, and use it in app_queue to track unique id's. Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210 ASTERISK-28829 #close ASTERISK-25844 #close Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6	6 years ago
Jaco Kroon	44e5dd288b	Remove #include <sys/cdefs.h> ... These are not provided by standards, and as a result causes failure to compile on musl. https://wiki.musl-libc.org/faq.html#Q:-When-compiling-something-against-musl,-I-get-error-messages-about-%3Ccode%3Esys/cdefs.h%3C/code%3E Change-Id: I6a357cefd106c72cfecafd898638f6b5692c2e05	6 years ago
Alexander Traud	29070b61f7	core_local: Local calls are always secure. ... In a Dialplan, the channel drivers 'chan_sip' and 'chan_iax2' support the channel items 'secure_bridge_media' and 'secure_bridge_signaling'. That way, a channel can be forced to use encryption even if not specified in its configuration. However, when the Local Proxy kicks in, for example, in case of a forwarding (SIP status 302), Local Proxy stated it does not know those items. Consequently, such a call could not be proxied how clever your Dialplan was. Because local calls within Asterisk are always secure, Local Proxy accepts such a request now. ASTERISK-22920 Change-Id: I4c143bb70f686790953cc04c5a4b810bbb03636c	6 years ago