asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	0ad1ff8a72	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	1414b9cc57	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	b0d828f14a	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	b912b31853	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	65a4a3a4e6	res_pjsip: Give error when TLS transport configured but not supported. Change-Id: I058af496021ff870ccec2d8cbade637b348ab80b	5 years ago
Kevin Harwell	15de2f1727	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	35302efe73	app_queue: Add alembic migration to add ringinuse to queue_members. ASTERISK-28356 #close Change-Id: I53a1bfdd3113d620bea88349019173a2f3f0ae39	5 years ago
Sean Bright	be3153346b	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	bbfb8f2b9d	logger.conf.sample: Add more debug documentation. Change-Id: Iff0e713f2120d8dce8e1e26924b99ed17f9d9dff	5 years ago
Sean Bright	31364fa4c8	queues.conf.sample: Correct 'context' documentation. ASTERISK-24631 #close Change-Id: I8bf8776906a72ee02f24de6a85345940b9ff6b6f	5 years ago
Sean Bright	e27fa9eceb	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
Mark Murawski	a0009c807e	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	4393207751	app_queue.c: Don't crash when realtime queue name is empty. ASTERISK-27542 #close Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a	5 years ago
George Joseph	c78d0ce429	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	55c467eab1	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	ed2f637b47	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	f213833514	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	f47c5cbdf9	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	2e7fc84398	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	6aac148d59	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	be3e469f98	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	60fb559ccc	xml: Allow deprecated_in and removed_in for MODULEINFO. ASTERISK-29337 Change-Id: I2211b7da8d29369f8649aeabce07679da0787f2b	5 years ago
Joshua C. Colp	60800b038a	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
Kevin Harwell	263f906af4	manager: Increase the non breaking AMI version number ASTERISK~29244 added three new AMI events, so bump the version number. Change-Id: I0e77fa36d38fb27dec3481d4ef08131330da0632	5 years ago
Asterisk Development Team	0afd37e3b5	Update CHANGES and UPGRADE.txt for 18.3.0	5 years ago
Joshua C. Colp	f7bda066bb	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	23e41313a8	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	52707fba7f	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	94debe5085	app_dial.c: Only send DTMF on first progress event. ASTERISK-29329 #close Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601	5 years ago
Alexander Traud	262473c6d9	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	4fc0e16838	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	3084084648	modules.conf: Fix differing usage of assignment operators. ASTERISK-24434 #close Change-Id: I0144e8d65d878128da59dcf3df12ca8cee47d6db	5 years ago
Sean Bright	e4cd7a7d0b	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	16e4d1f36f	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	90ef6a14a7	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	a9acbd19f3	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	269bb08ea2	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	5f1c21e4ca	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	682f7d9437	asterisk: Update copyright. ASTERISK-29326 Change-Id: Ia95dbfb66e2d11ac4d1228444283bb2e4d77396a	5 years ago
Ben Ford	77328142b4	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	0323293142	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	976b1a1d7a	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
Joshua C. Colp	b43b81d953	channel: Fix memory leak in suppress API. A frame suppression API exists as part of channels which allows audio frames to or from a channel to be dropped. The MuteAudio AMI action uses this API to perform its job. This API uses a framehook to intercept flowing audio and drop it when appropriate. It is the responsibility of the framehook to free the frame it is given if it changes the frame. The suppression API failed to do this resulting in a leak of audio frames. This change adds the freeing of these frames. ASTERISK-29071 Change-Id: Ie50acd454d672d36af914050c327d2e120d8ba7b	5 years ago
Salah Ahmed	df8d335ad1	res_rtp_asterisk: Check remote ICE reset and reset local ice attrb This change will check is the remote ICE session got reset or not by checking the offered ufrag and password with session. If the remote ICE reset session then Asterisk reset its local ufrag and password to reject binding request with Old ufrag and Password. ASTERISK-29266 Change-Id: I9c55e79a7af98a8fbb497d336b828ba41bc34eeb	5 years ago
Holger Hans Peter Freyther	3286c04856	pjsip: Generate progress (once) when receiving a 180 with a SDP ASTERISK-29105 Change-Id: If1615fe7115fe544ef974b044d3cea5c48b94a38	5 years ago
Nico Kooijman	7b052ec965	main: With Dutch language year after 2020 is not spoken in say.c Implemented the english way of saying the year in ast_say_date_with_format_nl. Currently the numbers are spoken correctly until 2020 and stopped working this year. ASTERISK-29297 #close Reported-by: Jacek Konieczny Change-Id: If5918eed5ab05df31df4dd23f08a909a60f6aba4	5 years ago
Nick French	dedfb334bd	res_pjsip: dont return early from registration if init auth fails If set_outbound_initial_authentication_credentials() fails, handle_client_registration() bails early without creating or sending a register message. [set_outbound_initial_authentication_credentials() failures can occur during the process of retrieving an oauth access token.] The return from handle_client_registration is ignored, so returning an error doesn't do any good. This is a real problem when the registration request is a re-register, because then the registration will still be marked 'active' despite the re-register never being sent at all. So instead, log a warning but let the registration be created and sent (and probably fail) and follow the normal registration failed retry/abort logic. ASTERISK-29315 #close Change-Id: I2e03b1ea7fba1fa1a8279086aa4b17679e7fa7fa	5 years ago
Alexei Gradinari	d5e73d2121	res_fax: validate the remote/local Station ID for UTF-8 format If the remote Station ID contains invalid UTF-8 characters the asterisk fails to publish the Stasis and ReceiveFax status messages. json.c: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string. 0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28] 1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc] 2: /usr/sbin/asterisk(ast_channel_publish_varset+0x2b) [0x57aa0b] 3: /usr/sbin/asterisk(pbx_builtin_setvar_helper+0x121) [0x530641] 4: /usr/lib64/asterisk/modules/res_fax.so(+0x44fe) [0x7f27f4bff4fe] ... stasis_channels.c: Error creating message json.c: Error building JSON from '{s: s, s: s, s: s, s: s, s: s, s: s, s: o}': Invalid UTF-8 string. 0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28] 1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc] 2: /usr/lib64/asterisk/modules/res_fax.so(+0x5acd) [0x7f27f4c00acd] ... res_fax.c: Error publishing ReceiveFax status message This patch replaces the invalid UTF-8 Station IDs with an empty string. ASTERISK-29312 #close Change-Id: Ieb00b6ecf67db3bfca787649caa8517f29d987db	5 years ago
Sean Bright	6673c1b177	app_page.c: Don't fail to Page if beep sound file is missing ASTERISK-16799 #close Change-Id: I40367b0d6dbf66a39721bde060c8b2d734a61cf4	5 years ago
George Joseph	15afabdf8e	res_pjsip_refer: Refactor progress locking and serialization Although refer_progress_notify() always runs in the progress serializer, the pjproject evsub module itself can cause the subscription to be destroyed which then triggers refer_progress_on_evsub_state() to clean it up. In this case, it's possible that refer_progress_notify() could get the subscription pulled out from under it while it's trying to use it. At one point we tried to have refer_progress_on_evsub_state() push the cleanup to the serializer and wait for its return before returning to pjproject but since pjproject calls its state callbacks with the dialog locked, this required us to unlock the dialog while waiting for the serialized cleanup, then lock it again before returning to pjproject. There were also still some cases where other callers of refer_progress_notify() weren't using the serializer and crashes were resulting. Although all callers of refer_progress_notify() now use the progress serializer, we decided to simplify the locking so we didn't have to unlock and relock the dialog in refer_progress_on_evsub_state(). Now, refer_progress_notify() holds the dialog lock for its duration and since pjproject also holds the dialog lock while calling refer_progress_on_evsub_state() (which does the cleanup), there should be no more chances for the subscription to be cleaned up while still being used to send NOTIFYs. To be extra safe, we also now increment the session count on the dialog when we create a progress object and decrement the count when the progress is destroyed. ASTERISK-29313 Change-Id: I97a8bb01771a3c85345649b8124507f7622a8480	5 years ago

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