6522 Commits (f252eae4389f0dd3a135eb608fb106743709ca72)

Author	SHA1	Message	Date
Joshua C. Colp	563f2f94d6	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	f59001d171	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	68b2a2f1ab	cli.c: Fix compiler error. ... Added default variable value to fix a compiler error. Change-Id: I7b592adbb1274dc5464dea1c5e5de0685c928553	6 years ago
Joshua C. Colp	51f1f78069	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
George Joseph	6a0c472374	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	023d73a0e7	tcptls: Fix notice when TLS is enabled but not supported. ... ASTERISK-28797 Change-Id: Iab364a2c2519fd9d11d1c28293fda43d61b64c28	6 years ago
Pirmin Walthert	6649b13b91	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	f56149d019	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	92169e6f8a	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	9e7fc21086	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	0ce4a77367	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
Guido Falsi	ca5c1e6336	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	6 years ago
Joshua C. Colp	2128eb1f47	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	6 years ago
Joshua C. Colp	62183bc777	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	6 years ago
Pirmin Walthert	9a928cae86	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	6 years ago
Jean Aunis	ef4255f6ed	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	6 years ago
Jaco Kroon	1e6b10c137	main/backtrace: binutils-2.34 fix. ... My tester missed this one previously, have confirmed a positive build this time round. Change-Id: Id06853375954a200f03f9a1b9c97fe0b10d31fbf	6 years ago
Kevin Harwell	267583f18d	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	6 years ago
sungtae kim	eb0493cfb8	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	6 years ago
Jaco Kroon	85fca26c24	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	6 years ago
Jaco Kroon	33b2c7f79b	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	7 years ago
Sean Bright	479723f3cc	dns_txt: Add TXT record parsing support ... Change-Id: Ie0eca23b8e6f4c7d9846b6013d79099314d90ef5	7 years ago
Joshua C. Colp	c40050d350	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	7 years ago
George Joseph	5e7055d082	Merge "enum.c: Make ast_get_txt() actually do something." into 16	7 years ago
Sean Bright	acaf24e23c	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	7 years ago
Sean Bright	9a111d188f	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	7 years ago
Kevin Harwell	fc1d93cf97	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. That being the case, and since this is technically an API breaking change (no one should really be affected since things never really worked) the ARI version was updated to reflect that. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	7 years ago
Kevin Harwell	ed68feeca4	Merge "say: Remove unused "plural" option from main/say" into 16	7 years ago
Kevin Harwell	a421759d63	Merge "format_cap: make function parameters 'const'" into 16	7 years ago
Kevin Harwell	333f64ce83	Merge "pjsip: Update ACLs on named ACL changes." into 16	7 years ago
Walter Doekes	ad5a81aac4	say: Remove unused "plural" option from main/say ... There are exceptions for plural objects, but they are detected using the supplied NUMBER, not using an extra option. Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113	7 years ago
Kevin Harwell	dd959bf8d0	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	7 years ago
George Joseph	dd313ceda5	Merge "tcptls.c: Log more informative OpenSSL errors" into 16	7 years ago
George Joseph	0cf988e72b	Merge "bridging: Add better support for adding/removing streams." into 16	7 years ago
George Joseph	97789b1749	Merge "app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used" into 16	7 years ago
Joshua C. Colp	b20a215ea7	pjsip: Update ACLs on named ACL changes. ... This change extends the Sorcery API to allow a wizard to be told to explicitly reload objects or a specific object type even if the wizard believes that nothing has changed. This has been leveraged by res_pjsip and res_pjsip_acl to reload endpoints and PJSIP ACLs when a named ACL changes. ASTERISK-28697 Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b	7 years ago
Sean Bright	c32b4c7dc0	tcptls.c: Log more informative OpenSSL errors ... Dump OpenSSL's error stack to the error log when things fail. ASTERISK-28750 #close Reported by: Martin Zeh Change-Id: Ib63cd0df20275586e68ac4c2ddad222ed7bd9c0a	7 years ago
Joshua C. Colp	423b0e68ce	bridging: Add better support for adding/removing streams. ... This change adds support to bridge_softmix to allow the addition and removal of additional video source streams. When such a change occurs each participant is renegotiated as needed to reflect the update. If another video source is added then each participant gets another source. If a video source is removed then it is removed from each participant. This functionality allows you to have both your webcam and screenshare providing video if you desire, or even more streams. Mapping has been changed to use the topology index on the source channel as a unique identifier for outgoing participant streams, this will never change and provides an easy way to establish the mapping. The bridge_simple and bridge_native_rtp modules have also been updated to renegotiate when the stream topology of a party changes allowing the same behavior to occur as added to bridge_softmix. If a screen share is added then the opposite party is renegotiated. If that screen share is removed then the opposite party is renegotiated again. Some additional fixes are also included in here. Stream state is now conveyed in SDP so sendonly/recvonly/inactive streams can be requested. Removed streams now also remove previous state from themselves so consumers don't get confused. ASTERISK-28733 Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5	7 years ago
Sean Bright	cd8b27dcc2	app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used ... When opening a file for writing, Asterisk silently converts filenames ending with 'wav49' to 'WAV.' We aren't taking that in to account when setting the MIXMONITOR_FILENAME variable in MixMonitor. * If the user wants to write to a wav49 file, make sure that it is reflected properly in MIXMONITOR_FILENAME. * Add a note to the documentation describing this behavior. * Add a note in main/file.c indicating that app_mixmonitor needs to be changed if the logic in build_filename was changed. ASTERISK-24798 #close Reported by: xrobau Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c	7 years ago
Joshua C. Colp	df52f713f5	stasis: Use format specifier for size_t. ... Change-Id: Ic9b4afcc5398e7f46314419fc3c90433d818e35c	7 years ago
George Joseph	d6574cb7c7	message.c: Add option to suppress the Message channel AMI and ARI events ... In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	7 years ago
Joshua Colp	60d4230eec	Merge "http: Add ability to disable /httpstatus URI" into 16	7 years ago
George Joseph	1b452ebb51	cdr.c: Set event time on party b when leaving a parking bridge ... When Alice calls Bob and Bob does a blind transfer to Charlie, Bob's bridge leave event generates a finalize on both the party_a and party_b CDRs but while the party_a CDR has the correct end time set from the event time, party_b's leg did not. This caused that CDR's end time to be equal to the answered time and resulted in a billsec of 0. * We now pass the bridge leave message event time to cdr_object_party_b_left_bridge_cb() and set it on that CDR before calling cdr_object_finalize() on it. NOTE: This issue affected transfers using chan_sip most of the time but also occasionally affected chan_pjsip probably due to message timing. ASTERISK-28677 Reported by: Maciej Michno Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca	7 years ago
Sean Bright	a2a4e1026c	http: Add ability to disable /httpstatus URI ... Add a new configuration option 'enable_status' which allows the /httpstatus URI handler to be administratively disabled. We also no longer unconditionally register the /static and /httpstatus URI handlers, but instead do it based upon configuration. Behavior change: If enable_static was turned off, the URI handler was still installed but returned a 403 when it was accessed. Because we now register/unregister the URI handlers as appropriate, if the /static URI is disabled we will return a 404 instead. Additionally: * Change 'enablestatic' to 'enable_static' but keep the former for backwards compatibility. * Improve some internal variable names ASTERISK-28710 #close Change-Id: I647510f796473793b1d3ce1beb32659813be69e1	7 years ago
Sean Bright	efecc9d139	translate.c: Fix silk 24kHz truncation in 'core show translation' ... SILK @ 24kHz is not shown in the 'core show translation' output because of an off-by-one-error. Discovered while looking into ASTERISK~19871. ASTERISK-28706 Reported by: Sean Bright Change-Id: Ie1a551a8a484e07b45c8699cc0c90f1061029510	7 years ago
Joshua Colp	5166088628	Merge "app_voicemail, say: Fix various leading whitespace problems" into 16	7 years ago
Sean Bright	13fa33588f	app_voicemail, say: Fix various leading whitespace problems ... In af90afd90c, Japanese language support was added to app_voicemail and main/say.c, but the leading whitespace is not consistent with Asterisk coding guidelines. This patch fixes that. Whitespace only, no functional change. ASTERISK~23324 Reported by: Kevin McCoy Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87	7 years ago
Sean Bright	f5a1e8b04d	pbx.c: Include filesystem cache in free memory calculation ... ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8	7 years ago
Joshua Colp	521f534712	Merge "res_pjsip_endpoint_identifier_ip.c: Add port matching support" into 16	7 years ago
Friendly Automation	932271939b	Merge "stasis.c: Use correct topic name in stasis_topic_pool_delete_topic" into 16	7 years ago