4123 Commits (dbc315dfc74a4c2ff2a3e3eaea63c53cc83cda9e)

Author	SHA1	Message	Date
Joshua Colp	1affc17d95	Merge "message & stasis/messaging: make text message variables work in ARI" into 16	6 years ago
Kevin Harwell	9617e530b6	Merge "res/res_pjsip_sdp_rtp: Fix MOH transitions" into 16	6 years ago
Kevin Harwell	fc1d93cf97	message & stasis/messaging: make text message variables work in ARI ... When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. That being the case, and since this is technically an API breaking change (no one should really be affected since things never really worked) the ARI version was updated to reflect that. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Torrey Searle	9ecb651564	res/res_pjsip_sdp_rtp: Fix MOH transitions ... Update the state of remote_hold immediately on receipt of remote SDP so that the information is available when building the SDP answer ASTERISK-28754 #close Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f	6 years ago
Kevin Harwell	ed68feeca4	Merge "say: Remove unused "plural" option from main/say" into 16	6 years ago
Kevin Harwell	a421759d63	Merge "format_cap: make function parameters 'const'" into 16	6 years ago
Kevin Harwell	333f64ce83	Merge "pjsip: Update ACLs on named ACL changes." into 16	6 years ago
Walter Doekes	ad5a81aac4	say: Remove unused "plural" option from main/say ... There are exceptions for plural objects, but they are detected using the supplied NUMBER, not using an extra option. Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113	6 years ago
Kevin Harwell	dd959bf8d0	format_cap: make function parameters 'const' ... There were a couple places where the format cap function parameter was not 'const' when it should have been. This patch makes them 'const'. Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e	6 years ago
George Joseph	0cf988e72b	Merge "bridging: Add better support for adding/removing streams." into 16	6 years ago
Joshua C. Colp	b20a215ea7	pjsip: Update ACLs on named ACL changes. ... This change extends the Sorcery API to allow a wizard to be told to explicitly reload objects or a specific object type even if the wizard believes that nothing has changed. This has been leveraged by res_pjsip and res_pjsip_acl to reload endpoints and PJSIP ACLs when a named ACL changes. ASTERISK-28697 Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b	6 years ago
Joshua C. Colp	423b0e68ce	bridging: Add better support for adding/removing streams. ... This change adds support to bridge_softmix to allow the addition and removal of additional video source streams. When such a change occurs each participant is renegotiated as needed to reflect the update. If another video source is added then each participant gets another source. If a video source is removed then it is removed from each participant. This functionality allows you to have both your webcam and screenshare providing video if you desire, or even more streams. Mapping has been changed to use the topology index on the source channel as a unique identifier for outgoing participant streams, this will never change and provides an easy way to establish the mapping. The bridge_simple and bridge_native_rtp modules have also been updated to renegotiate when the stream topology of a party changes allowing the same behavior to occur as added to bridge_softmix. If a screen share is added then the opposite party is renegotiated. If that screen share is removed then the opposite party is renegotiated again. Some additional fixes are also included in here. Stream state is now conveyed in SDP so sendonly/recvonly/inactive streams can be requested. Removed streams now also remove previous state from themselves so consumers don't get confused. ASTERISK-28733 Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5	6 years ago
Ben Ford	411d3a2f75	RTP/ICE: Send on first valid pair. ... When handling ICE negotiations, it's possible that there can be a delay between STUN binding requests which in turn will cause a delay in ICE completion, preventing media from flowing. It should be possible to send media when there is at least one valid pair, preventing this scenario from occurring. A change was added to PJPROJECT that adds an optional callback (on_valid_pair) that will be called when the first valid pair is found during ICE negotiation. Asterisk uses this to start the DTLS handshake, allowing media to flow. It will only be called once, either on the first valid pair, or when ICE negotiation is complete. ASTERISK-28716 Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867	6 years ago
George Joseph	d6574cb7c7	message.c: Add option to suppress the Message channel AMI and ARI events ... In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	6 years ago
Jaco Kroon	b92b0469ff	netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr. ... ast_addressfamily_to_sockaddrize will determine the size that's required, and ast_sockaddr_from_sockaddr then wraps this new function and ast_sockaddr_copy_sockaddr to copy arbitrary sockaddr's (without knowing the address family) into the ast_sockaddr structure. Change-Id: Iee604e96e9096c79b477d6e5ff310cf0b06dae86 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Friendly Automation	3746b1e09e	Merge "res_pjsip_pubsub: Add ability to persist generator state information." into 16	6 years ago
Joshua Colp	521f534712	Merge "res_pjsip_endpoint_identifier_ip.c: Add port matching support" into 16	6 years ago
Joshua C. Colp	8318b05f25	res_pjsip_pubsub: Add ability to persist generator state information. ... Some body generators, such as dialog-info+xml, require storing state information which is then conveyed in the NOTIFY request itself. Up until now there was no way for such body generators to persist this information. Two new API calls have been added to allow body generators to set and get persisted data. This data is persisted out alongside the normal persistence information and allows the body generator to restore state information or to simply use this for normal storage of state. State is stored in the form of JSON and it is up to the body generator to interpret this as needed. The dialog-info+xml body generator has been updated to take advantage of this to persist the version number. ASTERISK-27759 Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de	6 years ago
Friendly Automation	932271939b	Merge "stasis.c: Use correct topic name in stasis_topic_pool_delete_topic" into 16	6 years ago
Sean Bright	f8b0c2c933	res_pjsip_endpoint_identifier_ip.c: Add port matching support ... Adds source port matching support when IP matching is used: [example] type = identify match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 If the IP matches but the source port does not, we reject and search for alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), unless the configured FQDN includes a port number in which case just a host lookup is performed. ASTERISK-28639 #close Reported by: Mitch Claborn Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92	6 years ago
George Joseph	dd82ebecd3	stasis.c: Use correct topic name in stasis_topic_pool_delete_topic ... When a topic is created for an object, its name is only <object>:<uniqueid> For example: bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 When a topic is added to a pool, its name has the pool's topic name prepended. For example: bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 The topic_pool_entry's name however, is only what was passed in to stasis_topic_pool_get_topic which is bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 That's actually correct because the entry is qualified by the pool that's in. When you're ready to delete the entry from the pool, you retrieve the tropic name from the object but since it now has the pool's topic name prepended, it won't be found in the pool container. Fix: * Modified stasis_topic_pool_delete_topic() to skip past the pool topic's name, if it was prepended to the topic name, before searching the container for a pool entry. ASTERISK-28633 Reported by: Joeran Vinzens Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6	6 years ago
Sean Bright	47ba42f4a0	websocket: Consider pending SSL data when waiting for socket input ... When TLS is in use, checking the readiness of the underlying FD is insufficient for determining if there is data available to be read. So before polling the FD, check if there is any buffered data in the TLS layer and use that first. ASTERISK-28562 #close Reported by: Robert Sutton Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d	6 years ago
Friendly Automation	a992180091	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 16	6 years ago
Friendly Automation	a3edac10a6	Merge "confbridge: Add support for specifying maximum sample rate." into 16	6 years ago
Joshua C. Colp	82e2f0c662	Merge "configure: Add check for MySQL client bool and my_bool type usage." into 16	6 years ago
Joshua C. Colp	7167fd6d46	configure: Add check for MySQL client bool and my_bool type usage. ... Instead of trying to use the defined MySQL client version from the header use a configure check to determine whether the bool or my_bool type should be used for defining a boolean. ASTERISK-28604 Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7	6 years ago
Joshua C. Colp	5622df0a94	confbridge: Add support for specifying maximum sample rate. ... ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	6 years ago
Kevin Harwell	d17bbcb9f1	res_fax: wrap v21 detected Asterisk initiated negotiation with config option ... A previous patch: Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 made it so a T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38 versus the previous behavior of forwarding negotiation to the "other" channel once a preamble was detected. This had the unfortunate side effect of breaking some setups. Specifically ones that set the max datagram option on an endpoint configuration (configured max datagram was not propagated since Asterisk now initiates negotiations). This patch adds a configuration option, "negotiate_both", that when enabled makes it so Asterisk initiates the negotiation requests to both endpoints vs. the previous behavior of waiting, and forwarding the request. The default is disabled keeping with the old behavior. ASTERISK-28660 Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a	6 years ago
Jaco Kroon	77941efad9	ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging. ... Due to use in res_rtp_asterisk there is a need to be able to apply an ACL without logging any invalid/denies. It's probably sensible to at least validate the ACL once directly after load and report invalid ACLs. Change-Id: I256169229d945ca7c1bbf228fc492d91df345843 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Kevin Harwell	30c0af7257	various files - fix some alerts raised by lgtm code analysis ... This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25	6 years ago
Sean Bright	e627db7928	utils.h: Set lower bound for thread stack size to PTHREAD_STACK_MIN ... ASTERISK-28590 #close Change-Id: I51abce00c04d0a06550bda5205580705185b9c1c	6 years ago
Kevin Harwell	931ef77e21	res_pjsip/res_pjsip_mwi: use centralized serializer pools ... Both res_pjsip and res_pjsip_mwi made use of serializer pools. However, they both implemented their own serializer pool functionality that was pretty much identical in each of the source files. This patch removes the duplicated code, and uses the new 'ast_serializer_pool' object instead. Additionally res_pjsip_mwi enables a shutdown group on the pool since if the timing was right the module could be unloaded while taskprocessor threads still needed to execute, thus causing a crash. Change-Id: I959b0805ad024585bbb6276593118be34fbf6e1d	6 years ago
Kevin Harwell	afc10c25ac	serializer: move/add asterisk serializer pool functionality ... Serializer pools have previously existed in Asterisk. However, for the most part the code has been duplicated across modules. This patch abstracts the code into an 'ast_serializer_pool' object. As well the code is now centralized in serializer.c/h. In addition serializer pools can now optionally be monitored by a shutdown group. This will prevent the pool from being destroyed until all serializers have completed. Change-Id: Ib1e906144b90ffd4d5ed9826f0b719ca9c6d2971	6 years ago
Torrey Searle	9a933c3adc	channel/chan_pjsip: add dialplan function for music on hold ... Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows the on-hold behavior to be controlled on a per-call basis ASTERISK-28542 #close Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8	6 years ago
Corey Farrell	fec6e1bd87	core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option. ... Previous to this patch passing a NULL tag to ao2_alloc or ao2_ref based functions would result in the reference not being logged under REF_DEBUG. This could sometimes cause inaccurate logging if NULL was accidentally passed to a reference action. Now reference logging is only disabled by option passed to the allocation method. Change-Id: I3c17d867d901d53f9fcd512bef4d52e342637b54	6 years ago
Friendly Automation	a687c7919d	Merge "func_jitterbuffer: Add audio/video sync support." into 16	6 years ago
Joshua Colp	120f2fb5f9	Merge "core: Add H.265/HEVC passthrough support" into 16	6 years ago
Joshua Colp	6647be69ac	func_jitterbuffer: Add audio/video sync support. ... This change adds support to the JITTERBUFFER dialplan function for audio and video synchronization. When enabled the RTCP SR report is used to produce an NTP timestamp for both the audio and video streams. Using this information the video frames are queued until their NTP timestamp is equal to or behind the NTP timestamp of the audio. The audio jitterbuffer acts as the leader deciding when to shrink/grow the jitterbuffer when adaptive is in use. For both adaptive and fixed the video buffer follows the size of the audio jitterbuffer. ASTERISK-28533 Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492	6 years ago
George Joseph	913c8b48b7	Merge "channels: Allow updating variable value" into 16	6 years ago
Sean Bright	518b6bfb5c	channels: Allow updating variable value ... When modifying an already defined variable in some channel drivers they add a new variable with the same name to the list, but that value is never used, only the first one found. Introduce ast_variable_list_replace() and use it where appropriate. ASTERISK-23756 #close Patches: setvar-multiplie.patch submitted by Michael Goryainov Change-Id: Ie1897a96c82b8945e752733612ee963686f32839	6 years ago
Florian Floimair	f85631cf82	core: Add H.265/HEVC passthrough support ... This change adds H.265/HEVC as a known codec and creates a cached "h265" media format for use. Note that RFC 7798 section 7.2 also describes additional SDP parameters. Handling of these is not yet supported. ASTERISK-28512 Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2	6 years ago
Ben Ford	922d3e02df	res_rtp: Add unit tests for RTCP stats. ... Added unit tests for RTCP video stats. These tests include NACK, REMB, FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR tests are currently disabled due to a bug. We expect to receive a compound packet, but the code sends this out as a single packet, which the browser accepts, but makes Asterisk upset. While writing these tests, I noticed an issue with NACK as well. Where it is handling a received NACK request, it was reading in only the first 8 bits of following packets that were also lost. This has been changed to the correct value of 16 bits. Also made a minor fix to the data buffer unit test. Change-Id: I56107c7411003a247589bbb6086d25c54719901b	6 years ago
George Joseph	6407ccd2d9	dns_core: Create new API ast_dns_resolve_ipv6_and_ipv4 ... The new function takes in a pointer to an ast_sockaddr structure, a hostname and an optional port and then dispatches parallel "AAAA" and "A" record queries. If an "AAAA" record is returned, it's parsed into the ast_sockaddr structure along with the port if it was supplied. If no "AAAA" record was returned, the first "A" record returned (if any) is parsed instead. This is a synchronous call. If you need asynchronous lookups, use ast_dns_query_set_resolve_async and roll your own. Change-Id: I194b0b0e73da94b35cc35263a868ffac3a8d0a95	7 years ago
Sean Bright	51fd43206b	audiohook.c: Substitute silence for unavailable audio frames ... There are 4 scenarios to consider when capturing audio from a channel with an audiohook: 1. There is no rx and no tx audio, so return nothing. 2. There is rx but no tx audio, so return rx. 3. There is tx but no rx audio, so return tx. 4. There is rx and tx audio, so mix them and return. The file passed as the primary argument to MixMonitor will be written to in scenarios 2, 3, and 4. However, if you pass the r() and t() options to MixMonitor, a frame will only be written to the r() file if there was rx audio and a frame will only be written to the t() file if there was tx audio. If you subsequently take the r() and t() files and try to mix them, the sides of the conversation will 'drift' and be non-representative of the user experience. This patch adds a new 'S' option to MixMonitor that injects a frame of silence on either the r() side or the t() side of the channel so that when later mixed, there is no such drift. Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e	7 years ago
Joshua Colp	6350f4e278	cdr / cel: Use event time at event creation instead of processing. ... When updating times on CDR or CEL records using the time at which it is done can result in times being incorrect if the system is heavily loaded and stasis message processing is delayed. This change instead makes it so CDR and CEL use the time at which the stasis messages that drive the systems are created. This allows them to be backed up while still producing correct records. ASTERISK-28498 Change-Id: I6829227e67aefa318efe5e183a94d4a1b4e8500a	7 years ago
Walter Doekes	64d25d36fb	sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread ... When fixing ASTERISK~24212, a change was done so a scheduled callback could not be removed while it was running. The caller of ast_sched_del would have to wait. However, when the caller of ast_sched_del is the callback itself (however wrong this might be), this new check would cause a deadlock: it would wait forever for itself. This changeset introduces an additional check: if ast_sched_del is called by the callback itself, it is immediately rejected (along with an ERROR log and a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the after-ast_sched_del-refcall function is only run if ast_sched_del returned success. This should fix the following spurious race condition found in chan_sip: - thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE) - thread 2: run sip_poke_peer_now - thread 2: blank out sched-ID (too soon!) - thread 1: set sched-ID (too late!) - thread 2: try to delete the currently running sched-ID After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and other madness) should occur. (Thanks Richard Mudgett for reviewing/improving this "scary" change.) Note that this change does not fix the observed race condition: unlocked access to peer->pokeexpire (and potentially other scheduled items in chan_sip), causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the deadlock go away. And in the observed case, it will not have adverse affects (like memory leaks) because the scheduled item is removed through a different path. ASTERISK-28282 Change-Id: Ic26777fa0732725e6ca7010df17af77a012aa856	7 years ago
Friendly Automation	a464847a1b	Merge "app_confbridge: Attended transfer event fixup" into 16	7 years ago
George Joseph	ccc92b6ecb	app_confbridge: Attended transfer event fixup ... When a channel already in a conference bridge is attended transfered to another extension, or when an existing call is attended transferred into a conference bridge, we now generate ConfbridgeJoin and ConfbridgeLeave events for the entering and departing channels. Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1	7 years ago
Joshua Colp	82789aafd6	res_rtp_asterisk: Add support for DTLS packet fragmentation. ... This change adds support for larger TLS certificates by allowing OpenSSL to fragment the DTLS packets according to the configured MTU. By default this is set to 1200. This is accomplished by implementing our own BIO method that supports MTU querying. The configured MTU is returned to OpenSSL which fragments the packet accordingly. When a packet is to be sent it is done directly out the RTP instance. ASTERISK-28018 Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06	7 years ago
Kevin Harwell	edc3e0df1a	conversions.c: Add conversions for largest max sized integer ... Added a conversion for umax (largest maximum sized integer allowed). Adjusted the other current conversion functions (uint and ulong) to be derivatives of the umax conversion since they are simply subsets of umax. Also made the negative check move the pointer on spaces since strtoumax does it anyways. Change-Id: I56c2ef2629d49b524c8df58af12951c181f81f08	7 years ago