3772 Commits (13.33)

Author	SHA1	Message	Date
Jean Aunis	947a6e8674	func_volume: Accept decimal number as argument ... Allow voice volume to be multiplied or divided by a floating point number. ASTERISK-28813 Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c	6 years ago
Jaco Kroon	5ddac396f0	acl: implement a centralized ACL output mechanism for HAs and ACLs. ... named_acl.c (which is really a named_ha) now uses ast_ha_output. I've also updated main/manager.c to output the actual ACL on "manager show user <username>" if one is set. If this works then we can add similar to other modules as required. Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f	6 years ago
Jaco Kroon	6b2aea9ce2	netsock2: compile fixes. ... This fixes ast_addressfamily_to_sockaddrsize to reference the provided argument, and ast_sockaddr_from_sockaddr to not use the name of a structure as argument. Change-Id: Ibf5db469c47c3b4214edf8456326086174e8edd7	6 years ago
Joshua Colp	994853658b	Merge "message & stasis/messaging: make text message variables work in ARI" into 13	6 years ago
Kevin Harwell	c5ababbd2b	Merge "res/res_pjsip_sdp_rtp: Fix MOH transitions" into 13	6 years ago
Kevin Harwell	0fb06121cb	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	ed670ca0c6	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	6af57d5750	Merge "say: Remove unused "plural" option from main/say" into 13	6 years ago
Kevin Harwell	2241f828ca	Merge "format_cap: make function parameters 'const'" into 13	6 years ago
Kevin Harwell	ff87627140	Merge "pjsip: Update ACLs on named ACL changes." into 13	6 years ago
Walter Doekes	c3e9d37ce7	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	d13cd25d4c	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
Ben Ford	f928600a98	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
Joshua C. Colp	a60071610c	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
Sean Bright	bfc93cc954	chan_pjsip: Ignore RTP that we haven't negotiated ... If chan_pjsip receives an RTP packet whose payload differs from the channel's native format, and asymmetric_rtp_codec is disabled (the default), Asterisk will switch the channel's native format to match that of the incoming packet without regard to the negotiated payloads. We now keep track of the formats that have been negotiated and check before switching payloads which results in these packets being dropped instead of causing the session to terminate. ASTERISK-28139 #close Reported by: Paul Brooks Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3	7 years ago
Joshua Colp	0d1fcf0626	Merge "netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr." into 13	7 years ago
Friendly Automation	3e446508b4	Merge "netsock2: backport ast_sockaddr_copy_sockaddr to asterisk 13." into 13	7 years ago
Jaco Kroon	e7d2ab666b	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>	7 years ago
Kevin Harwell	fa700acf39	Merge "res_pjsip_pubsub: Add ability to persist generator state information." into 13	7 years ago
Joshua Colp	4cefb2f3cb	Merge "res_pjsip_endpoint_identifier_ip.c: Add port matching support" into 13	7 years ago
Joshua C. Colp	9720b29d8e	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	7 years ago
Sean Bright	49bb7d85cd	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	7 years ago
George Joseph	e4e5e07e3d	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	7 years ago
Joshua Colp	5a3826e334	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 13	7 years ago
Jaco Kroon	e8ca2af820	netsock2: backport ast_sockaddr_copy_sockaddr to asterisk 13. ... This function is already present in 15, 16 and master. Change-Id: I491ddecdccc87620c264567c549217e48f3d4ff1 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	7 years ago
Friendly Automation	66970af0d3	Merge "confbridge: Add support for specifying maximum sample rate." into 13	7 years ago
Joshua C. Colp	7cea1c4306	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	7 years ago
Joshua C. Colp	ca35de4282	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	7 years ago
Kevin Harwell	754fca9297	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	7 years ago
Jaco Kroon	6e79028020	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>	7 years ago
Kevin Harwell	6ca76798e1	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	7 years ago
Sean Bright	f1e3a13579	utils.h: Set lower bound for thread stack size to PTHREAD_STACK_MIN ... ASTERISK-28590 #close Change-Id: I51abce00c04d0a06550bda5205580705185b9c1c	7 years ago
Kevin Harwell	2042020abf	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	7 years ago
Kevin Harwell	0b7efc1687	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	7 years ago
Torrey Searle	6fd195ca91	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	7 years ago
Joshua Colp	d775bc5add	chan_pjsip: Add support for passing hold and unhold requests through. ... This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ Change-Id: Ib6294e906e577e1a4245cb1f058d3976ff484c52	7 years ago
Sean Bright	622e487cd4	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	7 years ago
Sean Bright	ddc64ca059	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	1496aef51b	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	3449240130	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
George Joseph	f22dedc597	Merge "app_confbridge: Attended transfer event fixup" into 13	7 years ago
George Joseph	41f5d15763	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	1ea9bad34d	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
George Joseph	6edef49525	Merge "mwi core: Move core MWI functionality into its own files" into 13	7 years ago
Friendly Automation	10b46eea9e	Merge "app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings" into 13	7 years ago
Antoni Goldstein	001e7762e4	app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings ... Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled at the earliest received PROGRESS or RINGING. Added millisecond versions of DIALEDTIME and ANSWEREDTIME. Added millisecond versions of ast_channel_get_up_time and ast_channel_get_duration in channel.c. ASTERISK-28363 Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1	7 years ago
Kevin Harwell	4ea20c9c85	mwi core: Move core MWI functionality into its own files ... There is enough MWI functionality to warrant it having its own 'c' and header files. This patch moves all current core MWI data structures, and functions into the following files: main/mwi.h main/mwi.c Note, code was simply moved, and not modified. However, this patch is also in preparation for core MWI changes, and additions to come. Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0	7 years ago
Dan Cropp	00588226c6	res_pjsip: Added a norefersub configuration setting ... Added a new PJSIP global setting called norefersub. Default is true to keep support working as before. res_pjsip_refer: Configures PJSIP norefersub capability accordingly. Checks the PJSIP global setting value. If it is true (default) it adds the norefersub capability to PJSIP. If it is false (disabled) it does not add the norefersub capability to PJSIP. This is useful for Cisco switches that do not follow RFC4488. ASTERISK-28375 #close Reported-by: Dan Cropp Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9	7 years ago
Friendly Automation	e7de71ee24	Merge "main/json.c: Added app_name, app_data to channel type" into 13	7 years ago
sungtae kim	707897e5f7	main/json.c: Added app_name, app_data to channel type ... It was difficult to check the channel's current application and parameters using ARI for current channels. Added app_name, app_data items to show the current application information. ASTERISK-28343 Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c	7 years ago