asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	567ea5abf8	app_voicemail: Configurable voicemail beep Hitherto, VoiceMail() played a non-customizable beep tone to indicate the caller could leave a message. In some cases, the beep may not be desired, or a different tone may be desired. To increase flexibility, a new option allows customization of the tone. If the t option is specified, the default beep will be overridden. Supplying an argument will cause it to use the specified file for the tone, and omitting it will cause it to skip the beep altogether. If the option is not used, the default behavior persists. ASTERISK-29349 Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280	4 years ago
Naveen Albert	0026aeada3	main/file.c: Don't throw error on flash event. AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c where it should be ignored. Adding this to the switch ensures a warning isn't thrown on RFC2833 flash events, since nothing's amiss. ASTERISK-29372 Change-Id: I4fa549bfb7ba1894a4044de999ea124877422fbc	4 years ago
Naveen Albert	fd40752954	chan_sip: Expand hook flash recognition. Some ATAs send hook flash events as application/hook-flash, rather than a DTMF event. Now, we also recognize hook-flash as a flash event. ASTERISK-29370 Change-Id: I1c3b82a040dff3affcd94bad8ce33edc90c04725	4 years ago
Joshua C. Colp	49c2e7e307	pjsip: Add patch for resolving STUN packet lifetime issues. In some cases it was possible for a STUN packet to be destroyed prematurely or even destroyed partially multiple times. This patch provided by Teluu fixes the lifetime of these packets and ensures they aren't partially destroyed multiple times. https://github.com/pjsip/pjproject/pull/2709 ASTERISK-29377 Change-Id: Ie842ad24ddf345e01c69a4d333023f05f787abca	4 years ago
Sean Bright	1b41629447	chan_pjsip: Correct misleading trace message ASTERISK-29358 #close Change-Id: I050daff67066873df4e8fc7f4bd977c1ca06e647	4 years ago
Ben Ford	0564d12280	STIR/SHAKEN: Switch to base64 URL encoding. STIR/SHAKEN encodes using base64 URL format. Currently, we just use base64. New functions have been added that convert to and from base64 encoding. The origid field should also be an UUID. This means there's no reason to have it as an option in stir_shaken.conf, as we can simply generate one when creating the Identity header. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c	4 years ago
Ben Ford	05f7bc9c66	STIR/SHAKEN: OPENSSL_free serial hex from openssl. We're getting the serial number of the certificate from openssl and freeing it with ast_free(), but it needs to be freed with OPENSSL_free() instead. Now we duplicate the string and free the one from openssl with OPENSSL_free(), which means we can still use ast_free() on the returned string. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Ia6e1a4028c1933a0e1d204b769ebb9f5a11f00ab	4 years ago
Ben Ford	259ecfa289	STIR/SHAKEN: Fix certificate type and storage. During OpenSIPit, we found out that the public certificates must be of type X.509. When reading in public keys, we use the corresponding X.509 functions now. We also discovered that we needed a better naming scheme for the certificates since certificates with the same name would cause issues (overwriting certs, etc.). Now when we download a public certificate, we get the serial number from it and use that as the name of the cached certificate. The configuration option public_key_url in stir_shaken.conf has also been renamed to public_cert_url, which better describes what the option is for. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d	4 years ago
George Joseph	09303e8e22	Updates for the MessageSend Dialplan App Enhancements: * The MessageSend dialplan application now takes an optional third argument that can set the message's "To" field on outgoing messages. It's an alternative to using the MESSAGE(to) dialplan function. NOTE: No channel driver currently implements this field. A follow-on commit for res_pjsip_messaging will implement it for the chan_pjsip channel driver. * To prevent confusion with the first argument, currently named "to", it's been renamed to "destination". Its function, creating the request URI, hasn't changed. * The documentation for MessageSend was updated to be more clear about the parameters and how they interact the MESSAGE() dialplan function. * With the rename of MessageSend's first parameter, and the fact that message.c references <info> elements in chan_sip.c, res_pjsip_messaging.c and res_xmpp, they each needed documentation updates to use MessageDestinationInfo instead of MessageToInfo. * appdocsxml.dtd was updated to include a missing element declaration for "dataType". This was showing up as an error in Eclipse's dtd editor. * Despite the changes in this commit, there should be no impact to current users of MessageSend. Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a	4 years ago
Sean Bright	e39efabd97	translate.c: Avoid refleak when checking for a translation path Change-Id: Idbd61ff77545f4a78b06a5064b55112e774b70e6	4 years ago
Sean Bright	b1807d440e	res_rtp_asterisk: More robust timestamp checking We assume that a timestamp value of 0 represents an 'uninitialized' timestamp, but 0 is a valid value. Add a simple wrapper to be able to differentiate between whether the value is set or not. This also removes the fix for ASTERISK~28812 which should not be needed if we are checking the last timestamp appropriately. ASTERISK-29030 #close Change-Id: Ie70d657d580d9a1f2877e25a6ef161c5ad761cf7	4 years ago
Joshua C. Colp	f142ca254e	chan_local: Skip filtering audio formats on removed streams. When a stream topology is provided to chan_local when dialing it filters the audio formats down. This operation did not skip streams which were removed (that have no formats) resulting in calling being aborted. This change causes such streams to be skipped. ASTERISK-29407 Change-Id: I1de8b98727cb2d10f4bc287da0b5fdcb381addd6	4 years ago
Sean Bright	4a843e00ef	res_pjsip.c: OPTIONS processing can now optionally skip authentication ASTERISK-27477 #close Change-Id: I68f6715bba92a525149e35d142a49377a34a1193	4 years ago
Jean Aunis	55279bfd9c	translate.c: Take sampling rate into account when checking codec's buffer size Up/down sampling changes the number of samples produced by a translation. This must be taken into account when checking the codec's buffer size. ASTERISK-29328 Change-Id: I9aebe2f8788e00321a7f5c47aa97c617f39e9055	4 years ago
Joshua C. Colp	531eb65cf3	svn: Switch to https scheme. Some versions of SVN seemingly don't follow the redirect to https. Change-Id: Ia7c76c18cb620bcf56f08e1211a7d80d321fe253	4 years ago
George Joseph	512d38868c	res_pjsip: Update documentation for the auth object Change-Id: I2f76867ce02ec611964925159be099de83346e38	4 years ago
Ben Ford	45a1977de4	res_aeap: Add basic config skeleton and CLI commands. Added support for a basic AEAP configuration read from aeap.conf. Also added 2 CLI commands for showing individual configurations as well as all of them: aeap show server <id> and aeap show servers. Only one configuration option is required at the moment, and that one is server_url. It must be a websocket URL. The other option, codecs, is optional and will be used over the codecs specified on the endpoint if provided. https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=45482453 Change-Id: I567ac5148c92b98d29d2ad83421b416b75ffdaa3	4 years ago
George Joseph	44aef0449a	bridge_channel_write_frame: Check for NULL channel There is a possibility, when bridge_channel_write_frame() is called, that the bridge_channel->chan will be NULL. The first thing bridge_channel_write_frame() does though is call ast_channel_is_multistream() which had no check for a NULL channel and therefore caused a segfault. Since it's still possible for bridge_channel_write_frame() to write the frame to the other channels in the bridge, we don't want to bail before we call ast_channel_is_multistream() but we can just skip the multi-channel stuff. So... bridge_channel_write_frame() only calls ast_channel_is_multistream() if bridge_channel->chan is not NULL. As a safety measure, ast_channel_is_multistream() now returns false if the supplied channel is NULL. ASTERISK-29379 Reported-by: Vyrva Igor Reported-by: Ross Beer Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce	4 years ago
Sean Bright	5a13e95c56	loader.c: Speed up deprecation metadata lookup Only use an XPath query once per module, then just navigate the DOM for everything else. Change-Id: Ia0336a7185f9180ccba4b6f631a00f9a22a36e92	4 years ago
George Joseph	53c702e1cc	res_prometheus: Clone containers before iterating The channels, bridges and endpoints scrape functions were grabbing their respective global containers, getting the count of entries, allocating metric arrays based on that count, then iterating over the container. If the global container had new objects added after the count was taken and the metric arrays were allocated, we'd run out of metric entries and attempt to write past the end of the arrays. Now each of the scape functions clone their respective global containers and all operations are done on the clone. Since the clone is stable between getting the count and iterating over it, we can't run past the end of the metrics array. ASTERISK-29130 Reported-By: Francisco Correia Reported-By: BJ Weschke Reported-By: Sébastien Duthil Change-Id: If0c8e40853bc0e9429f2ba9c7f5f358d90c311af	4 years ago
Joshua C. Colp	46ed6af9c2	loader: Output warnings for deprecated modules. Using the information from the MODULEINFO XML we can now output useful information at the end of module loading for deprecated modules. This includes the version it was deprecated in, the version it will be removed in, and the replacement if available. ASTERISK-29339 Change-Id: I2080dab97d2186be94c421b41dabf6d79a11611a	4 years ago
Kevin Harwell	0fc906a5e1	res_rtp_asterisk: Fix standard deviation calculation For some input to the standard deviation algorithm extremely large, and wrong numbers were being calculated. This patch uses a new formula for correctly calculating both the running mean and standard deviation for the given inputs. ASTERISK-29364 #close Change-Id: Ibc6e18be41c28bed3fde06d612607acc3fbd621f	4 years ago
Kevin Harwell	c4a376aac2	res_rtp_asterisk: Don't count 0 as a minimum lost packets The calculated minimum lost packets represents the lowest number of lost packets missed during an RTCP report interval. Zero of course is the lowest, but the idea is that this value contain the lowest number of lost packets once some have been missed. This patch checks to make sure the number of lost packets over an interval is not zero before checking and setting the minimum value. Also, this patch updates the rtp lost packet test to check for packet loss over several reports vs one. Change-Id: I07d6e21cec61e289c2326138d6bcbcb3c3d5e008	4 years ago
Kevin Harwell	65b68fd060	res_rtp_asterisk: Statically declare rtp_drop_packets_data object This patch makes the drop_packets_data object static. Change-Id: If4f9b21fa0c47d41a35b6b05941d978efb4da87b	4 years ago
Joshua C. Colp	8bd13a995a	res_rtp_asterisk: Only raise flash control frame on end. Flash in RTP is conveyed the same as DTMF, just with a specific digit. In Asterisk however we do flash as a single control frame. This change makes it so that only on end do we provide the flash control frame to the core. Previously we would provide a flash control frame on both begin and end, causing flash to work improperly. ASTERISK-29373 Change-Id: I1accd9c6e859811336e670e698bd8bd124f33226	4 years ago
Kevin Harwell	b86f1ef54c	res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command This patch makes it so when Asterisk is compiled in DEVMODE a CLI command is available that allows someone to drop incoming RTP packets. The command allows for dropping of packets once, or on a timed interval (e.g. drop 10 packets every 5 seconds). A user can also specify to drop packets by IP address. Change-Id: I25fa7ae9bad6ed68e273bbcccf0ee51cae6e7024	4 years ago
Joshua C. Colp	623abc2b6a	res_pjsip: Give error when TLS transport configured but not supported. Change-Id: I058af496021ff870ccec2d8cbade637b348ab80b	4 years ago
Kevin Harwell	eb92fb7298	time: Add timeval create and unit conversion functions Added a TIME_UNIT enumeration, and a function that converts a string to one of the enumerated values. Also, added functions that create and initialize a timeval object using a specified value, and unit type. Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392	4 years ago
Sean Bright	8db2a34065	app_queue: Add alembic migration to add ringinuse to queue_members. ASTERISK-28356 #close Change-Id: I53a1bfdd3113d620bea88349019173a2f3f0ae39	4 years ago
Sean Bright	c2dbfb9a8e	modules.conf: Fix more differing usages of assignment operators. I missed the changes in 18 and master in the previous review. ASTERISK-24434 #close Change-Id: Ieb132b2a998ce96daa9c9acf26535a974b895876	4 years ago
Ben Ford	25758670b8	logger.conf.sample: Add more debug documentation. Change-Id: Iff0e713f2120d8dce8e1e26924b99ed17f9d9dff	4 years ago
Ben Ford	55c53de022	logging: Add .log to samples and update asterisk.logrotate. Added .log extension to the sample logs in logger.conf.sample so that they will be able to be opened in the browser when attached to JIRA tickets. Because of this, asterisk.logrotate has also been updated to look for .log extensions instead of no extension for log files such as full and messages. Change-Id: I5de743c03f08047d6c6cc80cac5019ae0c4c200f	4 years ago
Sean Bright	aac442eecd	app_queue.c: Remove dead 'updatecdr' code. Also removed the sample documentation, and some oddly-placed documentation about the timeout argument to the Queue() application itself. There is a large section on the timeout behavior below. ASTERISK-26614 #close Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217	4 years ago
Sean Bright	cad843fe07	queues.conf.sample: Correct 'context' documentation. ASTERISK-24631 #close Change-Id: I8bf8776906a72ee02f24de6a85345940b9ff6b6f	4 years ago
Mark Murawski	b4347c4861	logger: Console sessions will now respect logger.conf dateformat= option The 'core' console (ie: asterisk -c) does read logger.conf and does use the dateformat= option. Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf and uses a hard coded dateformat option for printing received verbose messages: main/logger.c: static char dateformat[256] = "%b %e %T" This change will load logger.conf for each remote console session and use the dateformat= option to set the per-line timestamp for verbose messages Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1 ASTERISK-25358: #close Reported-by: Igor Liferenko	4 years ago
Sean Bright	8d3d7bdb82	app_queue.c: Don't crash when realtime queue name is empty. ASTERISK-27542 #close Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a	4 years ago
George Joseph	a03a05195a	res_pjsip_session: Make reschedule_reinvite check for NULL topologies When the check for equal topologies was added to reschedule_reinvite() it was assumed that both the pending and active media states would actually have non-NULL topologies. We since discovered this isn't the case. We now only test for equal topologies if both media states have non-NULL topologies. The logic had to be rearranged a bit to make sure that we cloned the media states if their topologies were non-NULL but weren't equal. ASTERISK-29215 Change-Id: I61313cca7fc571144338aac826091791b87b6e17	4 years ago
Joshua C. Colp	a8a08bcd1e	app_queue: Only send QueueMemberStatus if status changes. If a queue member was updated with the same status multiple times each time a QueueMemberStatus event would be sent which would be a duplicate of the previous. This change makes it so that the QueueMemberStatus event is only sent if the status actually changes. ASTERISK-29355 Change-Id: I580c60d992a0a8f2bea8b91c868771b3b490d116	4 years ago
Joshua C. Colp	970b84946e	core_unreal: Fix deadlock with T.38 control frames. When using the ast_unreal_lock_all function no channel locks can be held before calling it. This change unlocks the channel that indicate was called on before doing so and then relocks it afterwards. ASTERISK-29035 Change-Id: Id65016201b5f9c9519a216e250f9101c629e19e9	4 years ago
Joshua C. Colp	71dfbdc7b9	res_pjsip: Add support for partial transport reload. Some configuration items for a transport do not result in the underlying transport changing, but instead are just state we keep ourselves and use. It is perfectly reasonable to change these items. These include local_net and external_* information. ASTERISK-29354 Change-Id: I027857ccfe4419f460243e562b5f098434b3d43a	4 years ago
Jaco Kroon	fc03116d9b	menuselect: exit non-zero in case of failure on --enable\|disable options. ASTERISK-29348 Change-Id: I77e3466435f5a51a57538b29addb68d811af238d Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Joshua C. Colp	cce5ee5b7a	res_rtp_asterisk: Force resync on SSRC change. When an SSRC change occurs the timestamps are likely to change as well. As a result we need to reset the timestamp mapping done in the calc_rxstamp function so that they map properly from timestamp to real time. This previously occurred but due to packet retransmission support the explicit setting of the marker bit was not effective. ASTERISK-29352 Change-Id: I2d4c8f93ea24abc1030196706de2d70facf05a5a	4 years ago
Joshua C. Colp	efc61a96f0	menuselect: Add ability to set deprecated and removed versions. The "deprecated_in" and "removed_in" information can now be set in MODULEINFO for a module and is then displayed in menuselect so users can be aware of when a module is slated to be deprecated and then removed. ASTERISK-29337 Change-Id: I6952889cf08e0e9e99cf8b43f99b3cef4688087a	4 years ago
Joshua C. Colp	3330fb41f4	xml: Allow deprecated_in and removed_in for MODULEINFO. ASTERISK-29337 Change-Id: I2211b7da8d29369f8649aeabce07679da0787f2b	4 years ago
Joshua C. Colp	149e5e5b86	xml: Embed module information into core XML documentation. This change embeds the MODULEINFO block of modules into the core XML documentation. This provides a shared mechanism for use by both menuselect and Asterisk for information and a definitive source of truth. ASTERISK-29335 Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90	4 years ago
Joshua C. Colp	7438586d8e	documentation: Fix non-matching module support levels. Some modules have a different support level documented in their MODULEINFO XML and Asterisk module definition. This change brings the two in sync for the modules which were not matching. ASTERISK-29336 Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35	4 years ago
Joshua C. Colp	cc127a999c	channel: Fix crash in suppress API. There exists an inconsistency with framehook usage such that it is only on reads that the frame should be freed, not on writes as well. ASTERISK-29071 Change-Id: I5ef918ebe4debac8a469e8d43bf9d6b673e8e472	4 years ago
Jaco Kroon	41389bfdbd	func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds Change-Id: I75152cece8a00b7523d542e5ac22796f9595692b Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Jaco Kroon	8acb4fbd1e	app.h: Fix -Werror=zero-length-bounds compile errors in dev mode. Change-Id: I5c104dc1f8417ccd3d01faf86e84ccbf89bc3b31 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Sean Bright	8987de270f	app_dial.c: Only send DTMF on first progress event. ASTERISK-29329 #close Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601	4 years ago

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33131 Commits (241dbb1ec0b8c6b293e0bec4cc0f65adb26aa7a5) All Branches Search

33131 Commits (241dbb1ec0b8c6b293e0bec4cc0f65adb26aa7a5)

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