asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	7af4934943	general: Fix broken links. This fixes a number of broken links throughout the tree, mostly caused by wiki.asterisk.org being replaced with docs.asterisk.org, which should eliminate the need for sporadic fixes as in `f28047db36`. Resolves: #430 (cherry picked from commit `8cc5fc2e39`)	1 year ago
George Joseph	8c378f78d4	MergeApproved.yml: Remove unneeded concurrency The concurrency parameter on the MergeAndCherryPick job has been rmeoved. It was a hold-over from earlier days. (cherry picked from commit `74904a0b3a`)	1 year ago
Maximilian Fridrich	94de48c80c	app_dial: Add option "j" to preserve initial stream topology of caller Resolves: #462 UserNote: The option "j" is now available for the Dial application which uses the initial stream topology of the caller to create the outgoing channels. (cherry picked from commit `8d7665db7b`)	1 year ago
George Joseph	8dcac793e6	ast_coredumper: Increase reliability Instead of searching for the asterisk binary and the modules in the filesystem, we now get their locations, along with libdir, from the coredump itself... For the binary, we can use `gdb -c <coredump> ... "info proc exe"`. gdb can print this even without having the executable and symbols. Once we have the binary, we can get the location of the modules with `gdb ... "print ast_config_AST_MODULE_DIR` If there was no result then either it's not an asterisk coredump or there were no symbols loaded. Either way, it's not usable. For libdir, we now run "strings" on the note0 section of the coredump (which has the shared library -> memory address xref) and search for "libasteriskssl\|libasteriskpj", then take the dirname. Since we're now getting everything from the coredump, it has to be correct as long as we're not crossing namespace boundaries like running asterisk in a docker container but trying to run ast_coredumper from the host using a shared file system (which you shouldn't be doing). There is still a case for using --asterisk-bin and/or --libdir: If you've updated asterisk since the coredump was taken, the binary, libraries and modules won't match the coredump which will render it useless. If you can restore or rebuild the original files that match the coredump and place them in a temporary directory, you can use --asterisk-bin, --libdir, and a new --moddir option to point to them and they'll be correctly captured in a tarball created with --tarball-coredumps. If you also use --tarball-config, you can use a new --etcdir option to point to what normally would be the /etc/asterisk directory. Also addressed many "shellcheck" findings. Resolves: #445 (cherry picked from commit `26c98bc68b`)	1 year ago
Sean Bright	b777ae258a	logger.c: Move LOG_GROUP documentation to dedicated XML file. The `get_documentation` awk script will only extract the first DOCUMENTATION block that it finds in a given file. This is by design (`9bc2127`) to prevent AMI event documentation from being pulled in to the core.xml documentation file. Because of this, the `LOG_GROUP` documentation added in `89709e2` was not being properly extracted and was missing fom the resulting XML documentation file. This commit moves the `LOG_GROUP` documentation to a separate `logger.xml` file. (cherry picked from commit `2f405d232a`)	1 year ago
Matthew Fredrickson	4d54627fe8	res_odbc.c: Allow concurrent access to request odbc connections There are valid scenarios where res_odbc's connection pool might have some dead or stuck connections while others are healthy (imagine network elements/firewalls/routers silently timing out connections to a single DB and a single IP address, or a heterogeneous connection pool connected to potentially multiple IPs/instances of a replicated DB using a DNS front end for load balancing and one replica fails). In order to time out those unhealthy connections without blocking access to other parts of Asterisk that may attempt access to the connection pool, it would be beneficial to not lock/block access around the entire pool in _ast_odbc_request_obj2 while doing potentially blocking operations on connection pool objects such as the connection_dead() test, odbc_obj_connect(), or by dereferencing a struct odbc_obj for the last time and triggering a odbc_obj_disconnect(). This would facilitate much quicker and concurrent timeout of dead connections via the connection_dead() test, which could block potentially for a long period of time depending on odbc.ini or other odbc connector specific timeout settings. This also would make rapid failover (in the clustered DB scenario) much quicker. This patch changes the locking in _ast_odbc_request_obj2() to not lock around odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to lock around truly shared, non-immutable state like the connection_cnt member and the connections list on struct odbc_class. Fixes: #465 (cherry picked from commit `058ead0ea5`)	1 year ago
Sean Bright	17324c6cf9	res_pjsip_header_funcs.c: Check URI parameter length before copying. Fixes #477 (cherry picked from commit `0ed2e8200a`)	1 year ago
Sean Bright	ceeb4fba17	config.c: Log #exec include failures. If the script referenced by `#exec` does not exist, writes anything to stderr, or exits abnormally or with a non-zero exit status, we log that to Asterisk's error logging channel. Additionally, write out a warning if the script produces no output. Fixes #259 (cherry picked from commit `461c3c60e2`)	1 year ago
Sean Bright	4070a5f038	make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS. If LOCAL_MOD_SUBDIRS contains absolute paths, do not prefix them with the path to Asterisk's source tree. Fixes #142 (cherry picked from commit `e196ccb7c2`)	1 year ago
Sean Bright	5288c25dc1	app_voicemail.c: Completely resequence mailbox folders. Resequencing is a process that occurs when we open a voicemail folder and discover that there are gaps between messages (e.g. `msg0000.txt` is missing but `msg0001.txt` exists). Resequencing involves shifting the existing messages down so we end up with a sequential list of messages. Currently, this process stops after reaching a threshold based on the message limit (`maxmsg`) configured on the current folder. However, if `maxmsg` is lowered when a voicemail folder contains more than `maxmsg + 10` messages, resequencing will not run completely leaving the mailbox in an inconsistent state. We now resequence up to the maximum number of messages permitted by `app_voicemail` (currently hard-coded at 9999 messages). Fixes #86 (cherry picked from commit `242a0685b1`)	1 year ago
Naveen Albert	212befbcf1	sig_analog: Fix channel leak when mwimonitor is enabled. When mwimonitor=yes is enabled for an FXO port, the do_monitor thread will launch mwi_thread if it thinks there could be MWI on an FXO channel, due to the noise threshold being satisfied. This, in turns, calls analog_ss_thread_start in sig_analog. However, unlike all other instances where __analog_ss_thread is called in sig_analog, this call path does not properly set pvt->ss_astchan to the Asterisk channel, which means that the Asterisk channel is NULL when __analog_ss_thread starts executing. As a result, the thread exits and the channel is never properly cleaned up by calling ast_hangup. This caused issues with do_monitor on incoming calls, as it would think the channel was still owned even while receiving events, leading to an infinite barrage of warning messages; additionally, the channel would persist improperly. To fix this, the assignment is added to the call path where it is missing (which is only used for mwi_thread). A warning message is also added since previously there was no indication that __analog_ss_thread was exiting abnormally. This resolves both the channel leak and the condition that led to the warning messages. Resolves: #458 (cherry picked from commit `069d480138`)	1 year ago
Sean Bright	97f422e4f6	res_rtp_asterisk.c: Update for OpenSSL 3+. In `5ac5c2b0` we defined `OPENSSL_SUPPRESS_DEPRECATED` to silence deprecation warnings. This commit switches over to using non-deprecated API. (cherry picked from commit `410b734533`)	1 year ago
Sean Bright	a2d50679dd	alembic: Update list of TLS methods available on ps_transports. Related to #221 and #222. Also adds `*.ini` to the `.gitignore` file in ast-db-manage for convenience. (cherry picked from commit `b16d4a3420`)	1 year ago
Naveen Albert	2be4ffa5e9	func_channel: Expose previously unsettable options. Certain channel options are not set anywhere or exposed in any way to users, making them unusable. This exposes some of these options which make sense for users to manipulate at runtime. Resolves: #442 (cherry picked from commit `147f014072`)	1 year ago
Sean Bright	a8015816b1	app.c: Allow ampersands in playback lists to be escaped. Any function or application that accepts a `&`-separated list of filenames can now include a literal `&` in a filename by wrapping the entire filename in single quotes, e.g.: ``` exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world) ``` Fixes #172 UpgradeNote: Ampersands in URLs passed to the `Playback()`, `Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()` applications as filename arguments can now be escaped by single quoting the filename. Additionally, this is also possible when using the `CONFBRIDGE` dialplan function, or configuring various features in `confbridge.conf` and `queues.conf`. (cherry picked from commit `6b4d7f2125`)	1 year ago
Sean Bright	3344b19dac	uri.c: Simplify ast_uri_make_host_with_port() (cherry picked from commit `c25271f2ae`)	1 year ago
Sean Bright	56710cd1b1	func_curl.c: Remove CURLOPT() plaintext documentation. I assume this was missed when initially converting to XML documentation and we've been kicking the can down the road since. (cherry picked from commit `b91f97577d`)	1 year ago
Sean Bright	fbecf18b05	res_http_websocket.c: Set hostname on client for certificate validation. Additionally add a `assert()` to in the TLS client setup code to ensure that hostname is set when it is supposed to be. Fixes #433 (cherry picked from commit `992ea86981`)	1 year ago
Sean Bright	00498f4f67	live_ast: Add astcachedir to generated asterisk.conf. `astcachedir` (added in `b0842713`) was not added to `live_ast` so continued to point to the system `/var/cache` directory instead of the one in the live environment. (cherry picked from commit `04fd8704e1`)	1 year ago
George Joseph	2359af9ea3	SECURITY.md: Update with correct documentation URL (cherry picked from commit `08cadf999e`)	1 year ago
Naveen Albert	28028a372f	func_lock: Add missing see-also refs to documentation. Resolves: #423 (cherry picked from commit `44b0590e59`)	1 year ago
Matthew Fredrickson	d8434ca398	app_followme.c: Grab reference on nativeformats before using it Fixes a crash due to a lack of proper reference on the nativeformats object before passing it into ast_request(). Also found potentially similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c Fixes: #388 (cherry picked from commit `eeae61b2f8`)	1 year ago
Naveen Albert	1701775d07	configs: Improve documentation for bandwidth in iax.conf. This improves the documentation for the bandwidth setting in iax.conf by making it clearer what the ramifications of this setting are. It also changes the sample default from low to high, since only high is compatible with good codecs that people will want to use in the vast majority of cases, and this is a common gotcha that trips up new users. Resolves: #425 (cherry picked from commit `1f66a9fc28`)	1 year ago
Naveen Albert	aa43343687	logger: Add channel-based filtering. This adds the ability to filter console logging by channel or groups of channels. This can be useful on busy systems where an administrator would like to analyze certain calls in detail. A dialplan function is also included for the purpose of assigning a channel to a group (e.g. by tenant, or some other metric). ASTERISK-30483 #close Resolves: #242 UserNote: The console log can now be filtered by channels or groups of channels, using the logger filter CLI commands. (cherry picked from commit `318710cf6e`)	1 year ago
Sean Bright	152f5bdd2e	chan_iax2.c: Don't send unsanitized data to the logger. This resolves an issue where non-printable characters could be sent to the console/log files. (cherry picked from commit `7b297cb90d`)	1 year ago
George Joseph	acf0b399cb	codec_ilbc: Disable system ilbc if version >= 3.0.0 Fedora 37 started shipping ilbc 3.0.4 which we don't yet support. configure.ac now checks the system for "libilbc < 3" instead of just "libilbc". If true, the system version of ilbc will be used. If not, the version included at codecs/ilbc will be used. Resolves: #84 (cherry picked from commit `bee37b7404`)	1 year ago
Sean Bright	f4b2a34cf6	resource_channels.c: Explicit codec request when creating UnicastRTP. Fixes #394 (cherry picked from commit `59e296df01`)	1 year ago
Sean Bright	42cd03012a	doc: Update IP Quality of Service links. Fixes #328 (cherry picked from commit `092bd618fb`)	1 year ago
George Joseph	d8f1be3b81	chan_pjsip: Add PJSIPHangup dialplan app and manager action See UserNote below. Exposed the existing Hangup AMI action in manager.c so we can use all of it's channel search and AMI protocol handling without duplicating that code in dialplan_functions.c. Added a lookup function to res_pjsip.c that takes in the string represenation of the pjsip_status_code enum and returns the actual status code. I.E. ast_sip_str2rc("DECLINE") returns 603. This allows the caller to specify PJSIPHangup(decline) in the dialplan, just like Hangup(call_rejected). Also extracted the XML documentation to its own file since it was almost as large as the code itself. UserNote: A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range. (cherry picked from commit `9a93ce0409`)	1 year ago
Sean Bright	311ae80860	chan_iax2.c: Ensure all IEs are displayed when dumping frame contents. When IAX2 debugging was enabled (`iax2 set debug on`), if the last IE in a frame was one that may not have any data - such as the CALLTOKEN IE in an NEW request - it was not getting displayed. (cherry picked from commit `744bd4f9ac`)	1 year ago
Naveen Albert	161ec4eab6	chan_dahdi: Warn if nonexistent cadence is requested. If attempting to ring a channel using a nonexistent cadence, emit a warning, before falling back to the default cadence. Resolves: #409 (cherry picked from commit `beb9689288`)	1 year ago
Holger Hans Peter Freyther	ca92b2a2f7	stasis: Update the snapshot after setting the redirect The previous commit added the caller_rdnis attribute. Make it avialble during a possible ChanngelHangupRequest. (cherry picked from commit `ad46f59ca8`)	1 year ago
Holger Hans Peter Freyther	97042e0a3a	ari: Provide the caller ID RDNIS for the channels Provide the caller ID RDNIS when available. This will allow an application to follow the redirect. (cherry picked from commit `3c9e7ad4ba`)	1 year ago
Brad Smith	79bab53afd	main/utils: Implement ast_get_tid() for OpenBSD Implement the ast_get_tid() function for OpenBSD. OpenBSD supports getting the TID via getthrid(). (cherry picked from commit `d1f8978d91`)	1 year ago
Brad Smith	29934d21eb	res_rtp_asterisk.c: Fix runtime issue with LibreSSL The module will fail to load. Use proper function DTLS_method() with LibreSSL. (cherry picked from commit `9c70df1bb7`)	1 year ago
Naveen Albert	1f200b936d	app_directory: Add ADSI support to Directory. This adds optional ADSI support to the Directory application, which allows callers with ADSI CPE to navigate the Directory system significantly faster than is possible using the audio prompts. Callers can see the directory name (and optionally extension) on their screenphone and confirm or reject a match immediately rather than waiting for it to be spelled out, enhancing usability. Resolves: #356 (cherry picked from commit `72c6e95ed1`)	1 year ago
Naveen Albert	b5ed05ebf1	core_local: Fix local channel parsing with slashes. Currently, trying to call a Local channel with a slash in the extension will fail due to the parsing of characters after such a slash as being dial modifiers. Additionally, core_local is inconsistent and incomplete with its parsing of Local dial strings in that sometimes it uses the first slash and at other times it uses the last. For instance, something like DAHDI/5 or PJSIP/device is a perfectly usable extension in the dialplan, but Local channels in particular prevent these from being called. This creates inconsistent behavior for users, since using a slash in an extension is perfectly acceptable, and using a Goto to accomplish this works fine, but if specified through a Local channel, the parsing prevents this. This fixes this by explicitly parsing options from the last slash in the extension, rather than the first one, which doesn't cause an issue for extensions with slashes. ASTERISK-30013 #close Resolves: #248 (cherry picked from commit `63c5b119f4`)	1 year ago
Mark Murawski	c85e1d3243	Remove files that are no longer updated Fixes: #360 (cherry picked from commit `f32a1fa64e`)	1 year ago
Naveen Albert	af91eb9ea0	app_voicemail: Add AMI event for mailbox PIN changes. This adds an AMI event that is emitted whenever a mailbox password is successfully changed, allowing AMI consumers to process these. UserNote: The VoicemailPasswordChange event is now emitted whenever a mailbox password is updated, containing the mailbox information and the new password. Resolves: #398 (cherry picked from commit `70435a886c`)	1 year ago
Sean Bright	9c1eaa01e0	app_queue.c: Emit unpause reason with PauseQueueMember event. Fixes #395 (cherry picked from commit `6c3ad4ceaf`)	1 year ago
George Joseph	5dcd79ceb0	bridge_simple: Suppress unchanged topology change requests In simple_bridge_join, we were sending topology change requests even when the new and old topologies were the same. In some circumstances, this can cause unnecessary re-invites and even a re-invite flood. We now suppress those. Resolves: #384 (cherry picked from commit `11a586b65e`)	1 year ago
Naveen Albert	d1c62f7fa0	res_pjsip: Include cipher limit in config error message. If too many ciphers are specified in the PJSIP config, include the maximum number of ciphers that may be specified in the user-facing error message. Resolves: #396 (cherry picked from commit `8397077e75`)	1 year ago
Mike Bradeen	8f0fa854c5	res_speech: allow speech to translate input channel * Allow res_speech to translate the input channel if the format is translatable to a format suppored by the speech provider. Resolves: #129 UserNote: res_speech now supports translation of an input channel to a format supported by the speech provider, provided a translation path is available between the source format and provider capabilites. (cherry picked from commit `6aa80d1bb8`)	1 year ago
Sean Bright	5a30ec962c	res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. Fixes #386 (cherry picked from commit `2aa6a63188`)	1 year ago
Sean Bright	3fbd753b01	res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. Fixes #376 (cherry picked from commit `47578d385a`)	1 year ago
George Joseph	481ffa1438	api.wiki.mustache: Fix indentation in generated markdown The '*' list indicator for default values and allowable values for path, query and POST parameters need to be indented 4 spaces instead of 2. Should resolve issue 38 in the documentation repo. (cherry picked from commit `848628b795`)	1 year ago
Sean Bright	ad64862b4b	pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. Per RFC8827: Implementations MUST NOT implement DTLS renegotiation and MUST reject it with a "no_renegotiation" alert if offered. So we disable it when webrtc=yes is set. Fixes #378 UpgradeNote: The dtls_rekey will be disabled if webrtc support is requested on an endpoint. A warning will also be emitted. (cherry picked from commit `0f06787469`)	1 year ago
Samuel Olaechea	8ade68ee1d	configs: Fix typo in pjsip.conf.sample. (cherry picked from commit `12017263c8`)	1 year ago
George Joseph	aadd293163	res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown Commit `f66f77f` last year prevents the res_pjsip_exten_state and res_pjsip_mwi modules from unloading due to possible pjproject asserts if the modules are reloaded. A side effect of the implementation is that the taskprocessors these modules use aren't being released. When asterisk is doing a graceful shutdown, it waits AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT seconds for all taskprocessors to stop but since those 2 modules don't release theirs, the shutdown hangs for that amount of time. This change allows the modules to be unloaded and their resources to be released when ast_shutdown_final is true. Resolves: #379 (cherry picked from commit `3e4024ee20`)	1 year ago
sungtae kim	eeea910890	res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters This commit introduces an extension to the endpoint and relevant resource sizes for PJSIP, transitioning from its current 40-character constraint to a more versatile 255-character capacity. This enhancement significantly overcomes limitations related to domain qualification and practical usage, ultimately delivering improved functionality. In addition, it includes adjustments to accommodate the expanded realm size within the ARI, specifically enhancing the maximum realm length. Resolves: #345 UserNote: With this update, the PJSIP realm lengths have been extended to support up to 255 characters. UpgradeNote: As part of this update, the maximum allowable length for PJSIP endpoints and relevant resources has been increased from 40 to 255 characters. To take advantage of this enhancement, it is recommended to run the necessary procedures (e.g., Alembic) to update your schemas. (cherry picked from commit `96420f3d48`)	1 year ago

1 2 3 4 5 ...

33775 Commits (7af4934943fb01bee7032f7db55e3cbe2fe9de4c) All Branches Search

33775 Commits (7af4934943fb01bee7032f7db55e3cbe2fe9de4c)

All Branches