6564 Commits (28bae5e901b2595cac7b159734762128a2c591fe)

Author	SHA1	Message	Date
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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 years ago
Ben Ford	559fa0e89c	cli.c: Fix compiler error. ... Added default variable value to fix a compiler error. Change-Id: I7b592adbb1274dc5464dea1c5e5de0685c928553	5 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	5 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	5 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	5 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	5 years ago
traud	f9ea75d117	tcptls: Fix notice when TLS is enabled but not supported. ... ASTERISK-28797 Change-Id: Iab364a2c2519fd9d11d1c28293fda43d61b64c28	5 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	5 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	5 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	5 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	5 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	5 years ago
Guido Falsi	97494d8984	core/dns: Add system include required on FreeBSD ... While testing the latest RC on FreeBSD I noticed this new file fails to build. On FreeBSD inlcuding resolv.h requires sockaddr_in to be defined, and it's defined in netinet/in.h. So I added this include. ASTERISK-28853 #close Change-Id: I6997daf3956e6eb70ab6cb358628d162fad80079	5 years ago
Joshua C. Colp	1c5e68580a	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	5 years ago
Joshua C. Colp	6cfc6ff53c	confbridge: Add support for disabling text messaging. ... When in a conference bridge it may be necessary to have text messages disabled for specific participants or for all. This change adds a configuration option, "text_messaging", which can be used to enable or disable this on the user profile. By default existing behavior is preserved as it defaults to "yes". ASTERISK-28841 Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13	5 years ago
Pirmin Walthert	ca032d1e2e	res_rtp_asterisk: Free payload when error on insertion to data buffer ... When the ast_data_buffer_put rejects to add a packet, for example because the buffer already contains a packet with the same sequence number, the payload will never be freed, resulting in a memory leak. The data buffer will now return an error if this situation occurs allowing the caller to free the payload. The res_rtp_asterisk module has also been updated to do this. ASTERISK-28826 Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1	5 years ago
Jean Aunis	de66713fd5	func_volume: Accept decimal number as argument ... Allow voice volume to be multiplied or divided by a floating point number. ASTERISK-28813 Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c	5 years ago
Jaco Kroon	c5f3836bcc	main/backtrace: binutils-2.34 fix. ... My tester missed this one previously, have confirmed a positive build this time round. Change-Id: Id06853375954a200f03f9a1b9c97fe0b10d31fbf	5 years ago
George Joseph	2ee455958e	codec_negotiation: Implement outgoing_call_offer_pref ... Based on this new endpoint setting, a joint list of preferred codecs between those received from the Asterisk core (remote), and those specified in the endpoint's "allow" parameter (local) is created and is used to create the outgoing SDP offer. * Add outgoing_call_offer_pref to pjsip_configuration (endpoint) * Add "call_direction" to res_pjsip_session. * Update pjsip_session_caps.c to make the functions more generic so they could be used for both incoming and outgoing. * Update ast_sip_session_create_outgoing to create the pending_media_state->topology with the results of ast_sip_session_create_joint_call_stream(). * The endpoint "preferred_codec_only" option now automatically sets AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref. * A helper function ast_stream_get_format_count() was added to streams to return the current count of formats. ASTERISK-28777 Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437	5 years ago
Kevin Harwell	3c345ec56d	channel: write to a stream on multi-frame writes ... If a frame handling routine returns a list of frames (vs. a single frame) those frames are never passed to a tech's write_stream handler even if one is available. For instance, if a codec translation occurred and that codec returned multiple frames then those particular frames were always only sent to the tech's "write" handler. If that tech (pjsip for example) was stream capable then those frames were essentially ignored. Thus resulting in bad audio. This patch makes it so the "write_stream" handler is appropriately called for all cases, and for all frames if available. ASTERISK-28795 #close Change-Id: I868faea0b73a07ed5a32c2b05bb9cf4b586f739d	5 years ago
sungtae kim	dbddb6725d	dial.c: Removed dial string 80 character limitation ... The dial application had 80 characters of destination length limitation. But this limitation causes unexpected dial string cut if the dial string is long. Removed unnecessary limited buffer to support longer dial destination. ASTERISK-27946 Change-Id: I72c8f0319a4b47e8180817a66a7e9bde063cb330	5 years ago
Jaco Kroon	d32e559e8a	acl: implement a centralized ACL output mechanism for HAs and ACLs. ... named_acl.c (which is really a named_ha) now uses ast_ha_output. I've also updated main/manager.c to output the actual ACL on "manager show user <username>" if one is set. If this works then we can add similar to other modules as required. Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f	5 years ago
Jaco Kroon	2ad64e97c0	Update main/backtrace.c to deal with changes in binutils 2.34. ... binutils 2.34 merged this commit: https://sourceware.org/git/gitweb.cgi?p=binutils-gdb.git;a=commitdiff;\ h=fd3619828e94a24a92cddec42cbc0ab33352eeb4 Which effectively does things like: -#define bfd_section_size(bfd, ptr) ((ptr)->size) -#define bfd_get_section_size(ptr) ((ptr)->size) +#define bfd_section_size(sec) ((sec)->size) So in order to remain backwards compatible we need to detect this API change, and adjust accordingly. The simplest is to notice that the bfd_get_section_size and bfd_get_section_vma MACROs are no longer defined, and define then onto the new API. The alternative is to litter the code with a number of #ifdef #else #endif splatters right through the code. Change-Id: I3efe0f8e8f3e338d16fcbc2b26a505367b6e172f	5 years ago
Sean Bright	d68f940f6e	dns_txt: Add TXT record parsing support ... Change-Id: Ie0eca23b8e6f4c7d9846b6013d79099314d90ef5	5 years ago
Joshua C. Colp	98d10d0a16	audiohook: Don't allow audiohooks to attach to hung up channels. ... Given a scenario where MixMonitor was initiated over AMI it was possible for the channel and MixMonitor thread to remain alive past hang up of the channel. This scenario required the AMI initiated MixMonitor to retrieve the channel, a hangup to occur on the channel in another thread, and then for MixMonitor to actually start. If this occurred the MixMonitor thread would remain alive indefinitely and the channel reference would remain. This change ensures that audiohooks are never able to be attached to channels that have been hung up. An additional fix has also been done in app_mixmonitor to properly release the channel reference if this occurs. ASTERISK-28780 Change-Id: I8044c06daa06f0f16607788c596f55623be26f58	5 years ago
George Joseph	7b09d9c5aa	Merge "enum.c: Make ast_get_txt() actually do something."	5 years ago
Sean Bright	517224ce85	enum.c: Add support for regular expression flag in NAPTR record ... A regular expression in a NAPTR response record can have a trailing 'i' flag to indicate that the expression should be evaluated in a case-insensitive way. We were not checking for that flag which caused the record parsing to fail on otherwise valid input. Although this change will initially go into Asterisk 13, 16, and 17, it is my intention to replace the majority of this code in 16 and up - including this fix - by changing enum.c to consume the new DNS API which duplicates most of this logic already. Asterisk 13 doesn't have the DNS API, so this fix will be as good as it gets. ASTERISK-26711 #close Reported by: Vitold Change-Id: I33943a5b3e7539c6dca3a5079982ee15a08186f0	5 years ago
Sean Bright	ab63f0cd0f	enum.c: Make ast_get_txt() actually do something. ... The ast_get_txt() API function (and by extension, the TXTCIDNAME dialplan function) were broken in 65b8381550 such that we would never actually make a DNS TXT query as described. This patch restores the documented behavior. ASTERISK-19460 #close Reported by: George Joseph Change-Id: I1b19aea711488cb1ecd63843cddce05010e39376	5 years ago
Kevin Harwell	a715cf5aaa	message & stasis/messaging: make text message variables work in ARI ... When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	5 years ago
Kevin Harwell	d18af40431	Merge "say: Remove unused "plural" option from main/say"	5 years ago
Kevin Harwell	566f9a541f	Merge "format_cap: make function parameters 'const'"	5 years ago
Kevin Harwell	a3b3a9d2dc	Merge "pjsip: Update ACLs on named ACL changes."	5 years ago
Kevin Harwell	1e1651b4f4	format_cap: make function parameters 'const' ... There were a couple places where the format cap function parameter was not 'const' when it should have been. This patch makes them 'const'. Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e	5 years ago