asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	c4e0983742	func_odbc.conf.sample: Clarify sample documentation ASTERISK-20325 #close Change-Id: I06cb9b467b0fd06c8af2a5aee049f872c09cc4b6	6 years ago
Sean Bright	49cf84578e	chan_vpb: Fix 'catching polymorphic type ... by value' error Fixes the following compile error: chan_vpb.cc:2688:26: error: catching polymorphic type ‘class std::exception’ by value Change-Id: Ic87bc357d72427d77626735c83200fd278a7a649	6 years ago
Sean Bright	d68f940f6e	dns_txt: Add TXT record parsing support Change-Id: Ie0eca23b8e6f4c7d9846b6013d79099314d90ef5	6 years ago
Joshua C. Colp	98d10d0a16	audiohook: Don't allow audiohooks to attach to hung up channels. Given a scenario where MixMonitor was initiated over AMI it was possible for the channel and MixMonitor thread to remain alive past hang up of the channel. This scenario required the AMI initiated MixMonitor to retrieve the channel, a hangup to occur on the channel in another thread, and then for MixMonitor to actually start. If this occurred the MixMonitor thread would remain alive indefinitely and the channel reference would remain. This change ensures that audiohooks are never able to be attached to channels that have been hung up. An additional fix has also been done in app_mixmonitor to properly release the channel reference if this occurs. ASTERISK-28780 Change-Id: I8044c06daa06f0f16607788c596f55623be26f58	6 years ago
George Joseph	00a7e4b51d	CI: Create generic jenkinsfile This is a generic jenkinsfile to build Asterisk and optionally perform one or more of the following: * Publish the API docs to the wiki * Run the Unit tests * Run Testsuite Tests This job can be triggered manually from Jenkins or be triggered automatically on a schedule based on a cron string. Change-Id: Id9d22a778a1916b666e0e700af2b9f1bacda0852	6 years ago
Torrey Searle	a1dba820cf	res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use bridge_p2p_rtp_write will forward rtp to the bridged rtp instance without modifying the ssrc. However, it is not updating the SSRC in the bridged rtp. Thus, when SSRC packets are generated, they have the correct SSRC for the sender. ASTERISK-28773 #close Change-Id: I39f923bde28ebb4f0fddc926b92494aed294a478	6 years ago
George Joseph	0cde95ec89	Merge "res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated"	6 years ago
George Joseph	e828a596c5	Merge "chan_pjsip: Check audio frame when remote SSRC changes."	6 years ago
George Joseph	99efe1f868	Merge "codec negotiation: add incoming_call_offer_prefs option"	6 years ago
George Joseph	7b09d9c5aa	Merge "enum.c: Make ast_get_txt() actually do something."	6 years ago
George Joseph	2ff7c9d875	Merge "enum.c: Add support for regular expression flag in NAPTR record"	6 years ago
Joshua Colp	e8468eee13	Merge "res_rtp_asterisk: Add 'rtp show settings' cli command"	6 years ago
Torrey Searle	14ba1806f3	res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated If ICE support is enabled but not negotiated, the rtp->ice structure is not being destroyed. This leads to Asterisk waiting for ICE to complete instead of immediately starting the DTLS handshake, resulting in the call leg having no RTP. ASTERISK-28769 #close Change-Id: I17c137546dc9ecfb9583c24dcf4c2ced8bbd7a27	6 years ago
Paulo Vicentini	ed2a7e3eaf	chan_pjsip: Check audio frame when remote SSRC changes. If the SSRC of a received RTP packet differed from the previous SSRC an SSRC change control frame would be queued ahead of the media frame. In the case of audio this would result in the format of the audio frame not being checked, and if it differed or was not allowed then it could cause the call to drop due to failure to set up a translation path. The chan_pjsip module will now no longer assume the first frame will be the audio frame and instead goes through the complete list to find it. ASTERISK-28759 Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec	6 years ago
Sean Bright	517224ce85	enum.c: Add support for regular expression flag in NAPTR record A regular expression in a NAPTR response record can have a trailing 'i' flag to indicate that the expression should be evaluated in a case-insensitive way. We were not checking for that flag which caused the record parsing to fail on otherwise valid input. Although this change will initially go into Asterisk 13, 16, and 17, it is my intention to replace the majority of this code in 16 and up - including this fix - by changing enum.c to consume the new DNS API which duplicates most of this logic already. Asterisk 13 doesn't have the DNS API, so this fix will be as good as it gets. ASTERISK-26711 #close Reported by: Vitold Change-Id: I33943a5b3e7539c6dca3a5079982ee15a08186f0	6 years ago
Jared Smith	0a7fe3097f	indications.conf.sample: Add indication tones for Indonesia These tones come from http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf ASTERISK-23407 Change-Id: I48e2285f1e5bb29b3335f762006f66c423d0fcb8	6 years ago
Kevin Harwell	1ce0dfd144	Merge "res_pjsip_refer: ensure refer progress is still sent after Proceeding()"	6 years ago
Rodrigo Ramírez Norambuena	e089779908	res_rtp_asterisk: Add 'rtp show settings' cli command This change introduce a CLI command for the RTP to display the general configuration. In the first step add the follow fields of the configurations: - rtpstart - rtpend - dtmftimeout - rtpchecksum - strictrtp - learning_min_sequential - icesupport Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f	6 years ago
Sean Bright	ab63f0cd0f	enum.c: Make ast_get_txt() actually do something. The ast_get_txt() API function (and by extension, the TXTCIDNAME dialplan function) were broken in `65b8381550` such that we would never actually make a DNS TXT query as described. This patch restores the documented behavior. ASTERISK-19460 #close Reported by: George Joseph Change-Id: I1b19aea711488cb1ecd63843cddce05010e39376	6 years ago
Joshua Colp	5a8018875d	Merge "check_expr2: fix cross-compile/hardening issues"	6 years ago
Joshua Colp	bdf7b4eeb3	Merge "message & stasis/messaging: make text message variables work in ARI"	6 years ago
lvl	d1a2ff0aaf	res_pjsip_refer: ensure refer progress is still sent after Proceeding() ASTERISK-28766 #close Change-Id: I5ce2210062f9325db762edbf6e46075079bb2cd1	6 years ago
Kevin Harwell	06dada3f01	codec negotiation: add incoming_call_offer_prefs option Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that specifies the preferred order of codecs after receiving an offer. This patch does the following: Adds a new enumeration, ast_sip_call_codec_pref, used by the the new configuration option that's added to the endpoint media structure. Adds a new ast_sip_session_caps structure that's set for each session media object. Creates a new file, res_pjsip_session_caps that "implements" the new structure and option, and is compiled into the res_pjsip_session library. ASTERISK-28756 #close Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f	6 years ago
Joshua C. Colp	87fda066ea	res_rtp_asterisk: Improve video performance in certain networks. The receive buffer will now grow if we end up flushing the receive queue after not receiving the expected packet in time. This is done in hopes that if this is encountered again the extra buffer size will allow more time to pass and any missing packets to be received. The send buffer will now grow if we are asked for packets and can't find them. This is done in hopes that the packets are from the past and have simply been expired. If so then in the future with the extra buffer space the packets should be available. Sequence number cycling has been handled so that the correct sequence number is calculated and used in various places, including for sorting packets and for determining if a packet is old or not. NACK sending is now more aggressive. If a substantial number of missing sequence numbers are added a NACK will be sent immediately. Afterwards once the receive buffer reaches 25% a single NACK is sent. If the buffer continues to grow and reaches 50% or greater a NACK will be sent for each received future packet to aggressively ask the remote endpoint to retransmit. ASTERISK-28764 Change-Id: I97633dfa8a09a7889cef815b2be369f3f0314b41	6 years ago
Kevin Harwell	f8a852605d	Merge "res/res_pjsip_sdp_rtp: Fix MOH transitions"	6 years ago
Kevin Harwell	a715cf5aaa	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. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Kevin Harwell	8a8253e55b	Merge "addons/res_config_mysql: silense warnings about printf format errors."	6 years ago
Kevin Harwell	2d9ecd9cd1	Merge "app_queue: Refactor odd placement of if's around say_position"	6 years ago
Kevin Harwell	d18af40431	Merge "say: Remove unused "plural" option from main/say"	6 years ago
Kevin Harwell	999fdef335	Merge "app_mixmonitor: Turn on synchronization by default"	6 years ago
Kevin Harwell	566f9a541f	Merge "format_cap: make function parameters 'const'"	6 years ago
Kevin Harwell	a3b3a9d2dc	Merge "pjsip: Update ACLs on named ACL changes."	6 years ago
Sebastian Kemper	b7fbb9c41f	check_expr2: fix cross-compile/hardening issues When building check_expr2 with ASLR PIE hardening enabled the linker fails. This is resolved by adding the regular compiler flags when building the object files from ast_expr2f.c and ast_expr2.c. Note: The STANDALONE define is removed because it is already defined in _ASTCFLAGS. YY_NO_INPUT is defined so that the compile survives '--enable-dev-mode'. Also, a Makefile variable "CROSS_COMPILING" is added so that the build system doesn't try to run check_expr2 when cross-compiling, because that will fail the build as will. ASTERISK-28685 #close Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net> Change-Id: If435b7db9f9ad8266245bda51c81c220f9658915	6 years ago
Torrey Searle	77c9ba8e63	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
Walter Doekes	680e6b9774	app_queue: Refactor odd placement of if's around say_position Change-Id: Icba97905e331812f129e5966e91a59b104c7a748	6 years ago
Kevin Harwell	1e1651b4f4	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
Walter Doekes	0b5c6fddf1	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
Jaco Kroon	5cd7230f3c	addons/res_config_mysql: silense warnings about printf format errors. Warnings without this: res_config_mysql.c: In function 'update2_mysql': res_config_mysql.c:741:15: warning: format '%llu' expects argument of type 'long long unsigned int', but argument 6 has type 'my_ulonglong' {aka 'long unsigned int'} [-Wformat=] ast_debug(1, "MySQL RealTime: Updated %llu rows on table: %s\n", numrows, tablename); (reformatted for readability within line-wrap) Change-Id: I2af4d419a37c1a7eeee750cf9ae4a9a2b3a37fd3	6 years ago
George Joseph	838583783f	Merge "tcptls.c: Log more informative OpenSSL errors"	6 years ago
George Joseph	3854b561a5	Merge "bridging: Add better support for adding/removing streams."	6 years ago
George Joseph	9f25b4aa44	Merge "ast_tls_cert: Allow private key size to be set on command line"	6 years ago
George Joseph	4f1ab6404b	Merge "app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used"	6 years ago
George Joseph	12540a46a1	Merge "RTP/ICE: Send on first valid pair."	6 years ago
Joshua C. Colp	d6712790cd	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	7f2d56fc8c	tcptls.c: Log more informative OpenSSL errors Dump OpenSSL's error stack to the error log when things fail. ASTERISK-28750 #close Reported by: Martin Zeh Change-Id: Ib63cd0df20275586e68ac4c2ddad222ed7bd9c0a	6 years ago
Sean Bright	de6919f339	ast_tls_cert: Allow private key size to be set on command line The default size in release branches will be 1024 but we'll use 2048 in master. ASTERISK~28750 Change-Id: I435cea18bdd58824ed2b55259575c7ec7133842a	6 years ago
George Joseph	78b01f41ae	res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06	6 years ago
Joshua C. Colp	5a5be92b79	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
George Joseph	a6de4497e6	Merge "res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough"	6 years ago
Ben Ford	168637cc0c	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

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