5183 Commits (19eef2a6dc26e960059c46b7fc8adb08984845c1)

Author	SHA1	Message	Date
George Joseph	19eef2a6dc	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
Kevin Harwell	17c86dcfaa	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	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
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	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
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	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	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
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
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
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
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
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
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
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
Kevin Harwell	be0a61bc3d	res_rtp_asterisk: Add packet subtype during RTCP debug when relevant ... For some RTCP packet types the report count is actually the packet's subtype. This was not being reflected in the packet debug output. This patch makes it so for some RTCP packet types a "Packet Subtype" is now output in the debug replacing the "Reception reports" (i.e count). Change-Id: Id4f4b77bb37077a4c4f039abd6a069287bfefcb8	5 years ago
Boris P. Korzun	beb579bc99	res_config_pgsql: Limit realtime_pgsql() to return one (no more) record. ... Added a SELECT 'LIMIT' clause to realtime_pgsql() and refactored the function. ASTERISK-29293 #close Change-Id: If5a6d4b1072ea2e6e89059b21139d554a74b34f5	5 years ago
Ben Ford	83b0f5963f	res_pjsip_session.c: Check topology on re-invite. ... Removes an unnecessary check for the conditional that compares the stream topologies to see if they are equal to suppress re-invites. This was a problem when a Digium phone received an INVITE that offered codecs different than what it supported, causing Asterisk to send the re-invite. ASTERISK-29303 Change-Id: I04dc91befb2387904e28a9aaeaa3bcdbcaa7fa63	5 years ago
Jaco Kroon	7ab53fce7a	res_odbc_transaction: correctly initialise forcecommit value from DSN. ... Also improve the in-process documentation to clarify that the value is initialised from the DSN and not default false, but that the DSN's value is default false if unset. ASTERISK-29311 #close Change-Id: I46e2379f7b0656034442bce77cb37ccd4e61098d Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Joshua C. Colp	1af2a84c8b	res_pjsip_session: Always produce offer on re-INVITE without SDP. ... When PJSIP receives a re-INVITE without an SDP offer the INVITE session library will first call the on_create_offer callback and if unavailable then use the active negotiated SDP as the offer. In some cases this would result in a different SDP then was previously used without an incremented SDP version number. The two known cases are: 1. Sending an initial INVITE with a set of codecs and having the remote side answer with a subset. The active negotiated SDP would have the pruned list but would not have an incremented SDP version number. 2. Using re-INVITE for unhold. We would modify the active negotiated SDP but would not increment the SDP version. To solve these, and potential other unknown cases, the on_create_offer callback has now been implemented which produces a fresh offer with incremented SDP version number. This better fits within the model provided by the INVITE session library. ASTERISK-28452 Change-Id: I2d81048d54edcb80fe38fdbb954a86f0a58281a1	5 years ago
George Joseph	dbd8908f8d	res_pjsip_refer: Always serialize calls to refer_progress_notify ... refer_progress_notify wasn't always being called from the progress serializer. This could allow clearing notification->progress->sub in one thread while another was trying to use it. * Instances where refer_progress_notify was being called in-line, have been changed to use ast_sip_push_task(). Change-Id: Idcf1934c4e873f2c82e2d106f8d9f040caf9fa1e	5 years ago
Kevin Harwell	fad0cf12e6	AST-2021-002: Remote crash possible when negotiating T.38 ... When an endpoint requests to re-negotiate for fax and the incoming re-invite is received prior to Asterisk sending out the 200 OK for the initial invite the re-invite gets delayed. When Asterisk does finally send the re-inivite the SDP includes streams for both audio and T.38. This happens because when the pending topology and active topologies differ (pending stream is not in the active) in the delayed scenario the pending stream is appended to the active topology. However, in the fax case the pending stream should replace the active. This patch makes it so when a delay occurs during fax negotiation, to or from, the audio stream is replaced by the T.38 stream, or vice versa instead of being appended. Further when Asterisk sent the re-invite with both audio and T.38, and the endpoint responded with a declined T.38 stream then Asterisk would crash when attempting to change the T.38 state. This patch also puts in a check that ensures the media state has a valid fax session (associated udptl object) before changing the T.38 state internally. ASTERISK-29203 #close Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09	5 years ago
Alexander Traud	703158b903	rtp: Enable srtp replay protection ... Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	5 years ago
Ivan Poddubnyi	2770cc5872	res_pjsip_diversion: Fix adding more than one histinfo to Supported ... New responses sent within a PJSIP sessions are based on those that were sent before. Therefore, adding/modifying a header once causes it to be sent on all responses that follow. Sending 181 Call Is Being Forwarded many times first adds "histinfo" duplicated more and more, and eventually overflows past the array boundary. This commit adds a check preventing adding "histinfo" more than once, and skipping it if there is no more space in the header. Similar overflow situations can also occur in res_pjsip_path and res_pjsip_outbound_registration so those were also modified to check the bounds and suppress duplicate Supported values. ASTERISK-29227 Reported by: Ivan Poddubny Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322	5 years ago
Sean Bright	5a6f2f913b	res_rtp_asterisk.c: Fix signed mismatch that leads to overflow ... ASTERISK-29205 #close Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea	5 years ago
roadkill	176274caa4	res/res_pjsip.c: allow user=phone when number contain *# ... if From number contain * or # asterisk will not add user=phone Currently only number that uses AST_DIGIT_ANYNUM can have "user=phone" but the validation should use AST_DIGIT_ANY this is a problem when you want to send call to ISUP as they will disregard the From header and either replace From with anonymous or with p-asserted-identity ASTERISK-29261 Reported by: Mark Petersen Tested by: Mark Petersen Change-Id: I3307bdbf757582740bfee4110e85f7b6c9291cc4	5 years ago
Boris P. Korzun	f1c88a497b	res_musiconhold: Add support of various URL-schemes by MoH. ... Provided a support of variuos URL-schemes for res_musiconhold, registered by ast_bucket_scheme_register(). ASTERISK-29262 #close Change-Id: If0ea8697587353dce358a70035d82649fd4632b6	5 years ago
Robert Cripps	017e09b40a	res/res_pjsip_session.c: Check that media type matches in ... function ast_sip_session_media_state_add. Check ast_media_type matches when a ast_sip_session_media is found otherwise when transitioning from say image to audio, the wrong session is returned in the first if statement. ASTERISK-29220 #close Change-Id: I6f6efa9b821ebe8881bb4c8c957f8802ddcb4b5d	5 years ago
Sean Bright	fb42b60326	res_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet ... The last argument to ast_copy_string() is the buffer size, not the number of characters, so we add 1 to avoid stamping out the final \n in the persisted SUBSCRIBE message. Change-Id: I019b78942836f57965299af15d173911fcead5b2	5 years ago
Alexander Traud	a25bcf70ed	pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang. ... Otherwise, Clang 10 warned because of logical-not-parentheses. Change-Id: Ia8fb493f727b08070eb2dcf520c08df34ed11d79	5 years ago
Alexander Traud	3f119192bb	res_pjsip_session: Avoid sometimes-uninitialized warning with Clang. ... ASTERISK-29248 Change-Id: I2b17bd5ffb246bc64c463402c9831413da78a556	5 years ago
Jean Aunis	c10557c401	Stasis/messaging: tech subscriptions conflict with endpoint subscriptions. ... When both a tech subscription and an endpoint subscription exist for a given endpoint, TextMessageReceived events are dispatched to the tech subscription only. ASTERISK-29229 Change-Id: I9eac4cba5f9e27285a282509395347abc58fc2b8	5 years ago
Ivan Poddubnyi	c3fad2fd01	chan_pjsip: Assign SIPDOMAIN after creating a channel ... session->channel doesn't exist until chan_pjsip creates it, so intead of setting a channel variable every new incoming call sets one and the same global variable. This patch moves the code to chan_pjsip so that SIPDOMAIN is set on a newly created channel, it also removes a misleading reference to channel->session used to fetch call pickup configuraion. ASTERISK-29240 Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755	5 years ago
George Joseph	b3927ff8bc	Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff" ... This reverts commit 860e40dd80. Reason for revert: Too many issues reported. Need to research and correct. ASTERISK-29230 ASTERISK-29231 Reported by: Michael Maier Change-Id: I9011e2eecda4e91e1cfeeda6d1a7f1a0453eab41	5 years ago
Nick French	3c8598ffef	res_pjsip: Prevent segfault in UDP registration with flow transports ... Segfault occurs during outbound UDP registration when all transport states are being iterated over. The transport object in the transport is accessed, but flow transports have a NULL transport object. Modify to not iterate over any flow transport ASTERISK-29210 #close Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a	5 years ago
Richard Mudgett	027f4e3a21	res_pjsip_session.c: Fix compiler warnings. ... AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an unsigned long on all machines. Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338	5 years ago
Sungtae Kim	d8b7a6f599	res_pjsip_session: Fixed NULL active media topology handle ... Added NULL pointer check to prevent Asterisk crash. ASTERISK-29215 Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95	5 years ago
Torrey Searle	a7aea71e60	res/res_pjsip_diversion: prevent crash on tel: uri in History-Info ... Add a check to see if the URI is a Tel URI and prevent crashing on trying to retrieve the reason parameter. ASTERISK-29191 ASTERISK-29219 Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37	5 years ago
Sean Bright	13682210e2	app_chanspy: Spyee information missing in ChanSpyStop AMI Event ... The documentation in the wiki says there should be spyee-channel information elements in the ChanSpyStop AMI event. https://wiki.asterisk.org/wiki/x/Xc5uAg However, this is not the case in Asterisk <= 16.10.0 Version. We're using these Spyee* arguments since Asterisk 11.x, so these arguments vanished in Asterisk 12 or higher. For maximum compatibility, we still send the ChanSpyStop event even if we are not able to find any 'Spyee' information. ASTERISK-28883 #close Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f	5 years ago