33068 Commits (0adcdbd11894f9edef966410c2f87e65b8c89380)

 

All Branches   

Search

Author	SHA1	Message	Date
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	5 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	5 years ago
George Joseph	512d38868c	res_pjsip: Update documentation for the auth object ... Change-Id: I2f76867ce02ec611964925159be099de83346e38	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 years ago
Joshua C. Colp	623abc2b6a	res_pjsip: Give error when TLS transport configured but not supported. ... Change-Id: I058af496021ff870ccec2d8cbade637b348ab80b	5 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	5 years ago
Sean Bright	8db2a34065	app_queue: Add alembic migration to add ringinuse to queue_members. ... ASTERISK-28356 #close Change-Id: I53a1bfdd3113d620bea88349019173a2f3f0ae39	5 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	5 years ago
Ben Ford	25758670b8	logger.conf.sample: Add more debug documentation. ... Change-Id: Iff0e713f2120d8dce8e1e26924b99ed17f9d9dff	5 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	5 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	5 years ago
Sean Bright	cad843fe07	queues.conf.sample: Correct 'context' documentation. ... ASTERISK-24631 #close Change-Id: I8bf8776906a72ee02f24de6a85345940b9ff6b6f	5 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	5 years ago
Sean Bright	8d3d7bdb82	app_queue.c: Don't crash when realtime queue name is empty. ... ASTERISK-27542 #close Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a	5 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	5 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	5 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	5 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	5 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>	5 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	5 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	5 years ago
Joshua C. Colp	3330fb41f4	xml: Allow deprecated_in and removed_in for MODULEINFO. ... ASTERISK-29337 Change-Id: I2211b7da8d29369f8649aeabce07679da0787f2b	5 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	5 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	5 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	5 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>	5 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>	5 years ago
Sean Bright	8987de270f	app_dial.c: Only send DTMF on first progress event. ... ASTERISK-29329 #close Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601	5 years ago
Alexander Traud	1ae40e502d	res_format_attr_*: Parameter Names are Case-Insensitive. ... see RFC 4855: parameter names are case-insensitive both in media type strings and in the default mapping to the SDP a=fmtp attribute. This change is required for H.263+ because some implementations are known to use even mixed-case. This does not fix ASTERISK~29268 because H.264 was not fixed. This approach here lowers/uppers both parameter names and parameter values. H.264 needs a different approach because one of its parameter values is not case-insensitive: sprop-parameter-sets is Base64. Change-Id: Idf2a73457be231647aed3c87b1da197afba86892	5 years ago
Alexander Traud	8c461845c8	chan_iax2: System Header strings is included via asterisk.h/compat.h. ... The system header strings was included mistakenly with commit 3de0204. That header is included via asterisk.h and there via the compat.h. Change-Id: I3dc49060e275295f785670c87cc65fd3c3abd24a	5 years ago
Sean Bright	55bd104589	modules.conf: Fix differing usage of assignment operators. ... ASTERISK-24434 #close Change-Id: I0144e8d65d878128da59dcf3df12ca8cee47d6db	5 years ago
Sean Bright	30e509c2f9	strings.h: ast_str_to_upper() and _to_lower() are not pure. ... Because they modify their argument they are not pure functions and should not be marked as such, otherwise the compiler may optimize them away. ASTERISK-29306 #close Change-Id: Ibec03a08522dd39e8a137ece9bc6a3059dfaad5f	5 years ago
Sean Bright	df37b8181c	res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse. ... ao2_replace() bumps the reference count of the object that is doing the replacing, which is not what we want. We just want to drop the old ref on the old object and update the pointer to point to the new object. Pointed out by George Joseph in #asterisk-dev Change-Id: Ie8167ed3d4b52b9d1ea2d785f885e8c27206743d	5 years ago
Torrey Searle	8c247e2a94	res/res_rtp_asterisk: generate new SSRC on native bridge end ... For RTCP to work, we update the ssrc to be the one corresponding to the native bridge while active. However when the bridge ends we should generate a new SSRC as the sequence numbers will not continue from the native bridge left off. ASTERISK-29300 #close Change-Id: I23334b6934d2bf6490bda4bbf6414d96b8d17d10	5 years ago
Joshua C. Colp	304f8ddfb2	sorcery: Add support for more intelligent reloading. ... Some sorcery objects actually contain dynamic content that can change despite the underlying configuration itself not changing. A good example of this is the res_pjsip_endpoint_identifier_ip module which allows specifying hostnames. While the configuration may not change between reloads the DNS information of the hostnames can. This change adds the ability for a sorcery object to be marked as having dynamic contents which is then taken into account when reloading by the sorcery file based config module. If there is an object with dynamic content then a reload will be forced while if there are none then the existing behavior of not reloading occurs. ASTERISK-29321 Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1	5 years ago
George Joseph	607603cf89	res_pjsip_refer: Move the progress dlg release to a serializer ... Although the dlg session count was incremented in a pjsip servant thread, there's no guarantee that the last thread to unref this progress object was one. Before we decrement, we need to make sure that this is either a servant thread or that we push the decrement to a serializer that is one. Because pjsip_dlg_dec_session requires the dialog lock, we don't want to wait on the task to complete if we had to push it to a serializer. Change-Id: I8ff2d5d94be3ff04298394070434e22a7d3cbc41	5 years ago
Joshua C. Colp	6f67f24afd	res_pjsip_registrar: Include source IP and port in log messages. ... When registering it can be useful to see the source IP address and port in cases where multiple devices are using the same endpoint or when anonymous is in use. ASTERISK-29325 Change-Id: Ie178a6f55f53f8473035854c411bc3d056e0a2e0	5 years ago
Joshua C. Colp	f8d1758792	asterisk: Update copyright. ... ASTERISK-29326 Change-Id: Ia95dbfb66e2d11ac4d1228444283bb2e4d77396a	5 years ago
Ben Ford	fd560ad9fa	AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite. ... When Asterisk sends a reinvite negotiating T38 faxing, it's possible a crash can occur if the response contains a m=image and zero port. The reinvite callback code now checks session_media to see if it is null or not before trying to access the udptl variable on it. ASTERISK-29305 Change-Id: I1dfc51c5fa586e38579ede4bc228edee213ccaa9	5 years ago
Alexander Traud	a34e7de61c	res_format_attr_h263: Generate valid SDP fmtp for H.263+. ... Fixed: * RFC 4629 does not allow the value "0" for MPI, K, and N. * Allow value "0" for PAR. * BPP is printed only when specified because "0" has a meaning. New: * Added CPCF and MaxBR. * Some implementations provide CIF without MPI: a=fmtp:xx CIF;F=1 Although a violation of RFC 3555 section 3, we can support that. Changed: * Resorts the CIFs from large to small which partly fixes ASTERISK~29267. Change-Id: I95a650c715007b8dde11a77cb37d9c6c123a441e	5 years ago
Joshua C. Colp	2c1b6b7b15	res_pjsip_nat: Don't rewrite Contact on REGISTER responses. ... When sending a SIP response to an incoming REGISTER request we don't want to change the Contact header as it will contain the Contacts registered to the AOR and not our own Contact URI. ASTERISK-29235 Change-Id: I35a0723545281dd01fcd5cae497baab58720478c	5 years ago