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2017-08-01Fix compiler warnings on Fedora 26 / GCC 7.Corey Farrell
GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
2017-08-01app_queue: Add announce-position-only-up optionSean Bright
Setting this option will cause the Queue application to only announce the caller's position if it has improved since the last time that we announced it. Change-Id: I173a124121422209485b043e2bf784f54242fce6
2017-08-01bundled_pjproject: Improve SSL/TLS error handlingGeorge Joseph
OpenSSL has 2 levels or error processing. It's possible for the top layer to return SSL_ERROR_SYSCALL but the lower layer return no error, in which case processing should continue. Only the top layer was being examined though so connections were being torn down when they didn't need to be. This patch adds the examination of the lower level codes, and if they return no errors, allows processing to continue. ASTERISK-27001 Reported-by: Ian Gilmour patches: pjproject-2.6.patch submitted by Ian Gilmour (license 6889) Updated-by: George Joseph and Sauw Ming (Teluu) Merged to upstream pjproject on 7/27/2017 (commit 5631) Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2
2017-08-01chan_pjsip: add a new function PJSIP_DTMF_MODETorrey Searle
This function is a replica of SIPDtmfMode, allowing the DTMF mode of a PJSIP call to be modified on a per-call basis ASTERISK-27085 #close Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
2017-07-26res_rtp_asterisk: Fix mapping of pjsip's ICE roles to oursSean Bright
Change-Id: Ia578ede1a55b21014581793992a429441903278b
2017-07-26Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation ↵Joshua Colp
issues." into 15
2017-07-26Merge "Core: Add support for systemd socket activation." into 15Jenkins2
2017-07-26Merge "res_stasis_device_state: Unsubscribe should remove old subscriptions" ↵Joshua Colp
into 15
2017-07-26Merge "app_voicemail.c: Allow mailbox entry on authentication retry prompt." ↵Jenkins2
into 15
2017-07-26bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.Joshua Colp
This change does a few things to improve packet loss and renegotiation: 1. On outgoing RTP streams we will now properly reflect out of order packets and packet loss in the sequence number. This allows the remote jitterbuffer to better reorder things. 2. Video updates can now be discarded for a period of time after one has been sent to prevent flooding of clients. 3. For declined and removed streams we will now release any media session resources associated with them. This was not previously done and caused an issue where old state was being used for a new stream. 4. RTP bundling was not actually removing bundled RTP instances from the parent. This has been resolved by removing based on the RTP instance itself and not the SSRC. 5. The code did not properly handle explicitly unbundling an RTP instance from its parent. This now works as expected. ASTERISK-27143 Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45
2017-07-25Merge "core: Add VP9 passthrough support." into 15Jenkins2
2017-07-25res_stasis_device_state: Unsubscribe should remove old subscriptionsSergej Kasumovic
Case scenario with Applications ARI: * Once you subscribe to deviceState with Applications REST API, it will be added into subscription pool. * When you unsubscribe it will remove from the device_state_subscription hash table but not from the subscription pool. * When you subscribe again, it will add it to pool again. * Now you will have two subscriptions and you will receive same event twice. This fix should now remove deviceState subscription from pool and it should fix unsubscribe on deviceState. ASTERISK-27130 #close Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4
2017-07-25Merge "say.c: Fix file locations for second, seconds, minute, minutes files" ↵Jenkins2
into 15
2017-07-24core: Add VP9 passthrough support.Joshua Colp
This change adds VP9 as a known codec and creates a cached "vp9" media format for use. Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc
2017-07-24format.h: Fix a few minor errors in comments.Matthew Fredrickson
A few minor problems were found in comments in format.h. This patch fixes them. Change-Id: I07f0bdb47b93359b361c4c3d8ecc87cd3199dd94
2017-07-24Merge "Restore the incorrectly deleted spandspflow2pcap.log" into 15Jenkins2
2017-07-21app_voicemail.c: Allow mailbox entry on authentication retry prompt.Richard Mudgett
The following testsuite voicemail tests were failing to re-enter the mailbox after the first login attempt. tests/apps/voicemail/authenticate_invalid_mailbox tests/apps/voicemail/authenticate_invalid_password The tests were noting the start of the vm-incorrect-mailbox prompt and immediately sending the mailbox for the next login attempt. Since the invalid message playback had to complete before the digits were recognized, the test passed for the wrong reason and added approximately 20 seconds to the test times. * Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox digits like the initial vm-login prompt so the tests are able to enter the intended mailbox. Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8
2017-07-21Sounds: Update Makefile for Extra sounds 1.5.1 releaseRusty Newton
Incrementing version for the Extra sounds release. 1.5.1 Extra sounds removes two prompts that were moved into the Core packages in the 1.6 Core sounds release. ASTERISK-27142 #close Change-Id: I82f017812b0ea9599e19dd4635afd55611f13ee7
2017-07-21say.c: Fix file locations for second, seconds, minute, minutes filesRusty Newton
The seconds and minutes files have always existed in the base language directory of the Core package. So say.c has always been calling the wrong location (under digits/) for those two files and in the case of second and minute they didn't exist in the Core packages at all. The 1.6 sounds release moves the second and minute files into Core from Extra for the languages that already had them. A future release will include the second and minute files for languages that didn't already have them. This patch just changes all the target locations for second, seconds, minute, and minutes that were under the digits subdir to be under the root of sounds instead. Which is where the sounds will be for some languages after 1.6 sounds and for all languages after a future release. ASTERISK-25810 #close Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702 Reported-by: Nicolas Riendeau
2017-07-21Core: Add support for systemd socket activation.Corey Farrell
This change adds support for socket activation of certain SOCK_STREAM listeners in Asterisk: * AMI / AMI over TLS * CLI * HTTP / HTTPS Example systemd units are provided. This support extends to any socket which is initialized using ast_tcptls_server_start, so any unknown modules using this function will support socket activation. Asterisk continues to function as normal if socket activation is not enabled or if systemd development headers are not available during build. ASTERISK-27063 #close Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d
2017-07-21Restore the incorrectly deleted spandspflow2pcap.logGeorge Joseph
Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5
2017-07-21Merge "Update make_ari_stubs to correct version" into 15George Joseph
2017-07-21Update make_ari_stubs to correct versionGeorge Joseph
Change-Id: I18575b46db48d62edc72f37dc23b4ab22b43a8b1
2017-07-20corosync: Fix corosync library name in configure.acSean Bright
Also add new corosync packages to install_prereq. Reported by Travis Ryan in #asterisk-dev Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db
2017-07-20Update MAINLINE_BRANCH to 15George Joseph
Change-Id: I425d542b600ceabeef2342e9adfeb68c484a043d
2017-07-20Merge "Update AMI and ARI versions for master/15 and update UPDATE.txt"Jenkins2
2017-07-20Merge "pjsip: Increase maximum packet size."George Joseph
2017-07-20Update AMI and ARI versions for master/15 and update UPDATE.txtGeorge Joseph
AMI goes from 3.2.0 to 4.0.0 ARI goes from 2.0.0 to 3.0.0 Copied UPGRADE.txt -> UPGRADE-15.txt Created new UPGRADE.txt Removed a log file that was accidentally checked in a while ago Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7
2017-07-19Merge "core: Add digit filtering to ast_waitfordigit_full"Joshua Colp
2017-07-19Merge "app_playback.c: Use the timezonename parameter"George Joseph
2017-07-19Merge "bridge_softmix: Use removed stream spots when renegotiating."Jenkins2
2017-07-19Merge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO."Jenkins2
2017-07-18pjsip: Increase maximum packet size.Benjamin Keith Ford
The maximum packet size for PJSIP has been increased to handle the multiple streams being added for WebRTC. Change-Id: I9ea1e8d02668c544acadcb1c6200e1cc1bd588b3
2017-07-18Merge "app_queue: Add change priority of call"George Joseph
2017-07-18Merge "bridge_softmix: Don't reorder streams on participant leaving."Jenkins2
2017-07-17Merge "bridge/core_unreal: Fix SFU bugs with forwarding frames."Jenkins2
2017-07-17Merge "res_pjsip: Add "webrtc" configuration option"Jenkins2
2017-07-17Merge "res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use."Jenkins2
2017-07-17bridge_softmix: Don't reorder streams on participant leaving.Joshua Colp
When a participant leaves a bridge while operating in SFU mode their respective stream on every other participant needs to be removed. Leaving the stream out of the new topology results in every stream after it being moved and reordered. This causes problems with clients. Instead simply mark the stream as removed which leaves it in place in the SDP and doesn't reorder or touch any other streams. ASTERISK-27136 Change-Id: I4b3f840adcdf69b83842b0d8a737665ba0ef9cb1
2017-07-17Merge "app_confbridge: Make sure name recordings are always removed from the ↵Jenkins2
filesystem"
2017-07-17Merge "chan_iax2: On reload make sure to check for existing MWI subscription"George Joseph
2017-07-17Merge "res/res_stasis_snoop: generate silence when audiohook returns null"Jenkins2
2017-07-16bridge_softmix: Use removed stream spots when renegotiating.Joshua Colp
Streams are never truly removed in SDP, they still occupy a location within the SDP. This location can be reused by another stream if it so chooses. This change takes advantage of this such that if a new stream is needing to be added for a new participant any removed streams are instead replaced first. This reduces the size of the SDP and the number of streams. ASTERISK-27134 Change-Id: I95cdcfd55cf47e02ea52abb5d94008db3fb68b1d
2017-07-16res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use.Joshua Colp
This change makes it so that if an RTCP packet is being sent the RTP ICE component is used for sending if RTCP-MUX is in use. ASTERISK-27133 Change-Id: I6200f611ede709602ee9b89501720c29545ed68b
2017-07-14Merge "Sounds: Update for core sounds 1.6 release"Jenkins2
2017-07-14app_confbridge: Make sure name recordings are always removed from the filesystemSergej Kasumovic
This commit fixes two possible scenarios: * When recording name and if during recording you hangup, file is never removed. This is due to the fact file location is nulled. * When recording name and if you hangup during thank-you prompt, file is never removed. ASTERISK-27123 #close Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625
2017-07-14Merge "res/res_pjsip_t38 ensure t38 requests get rejected quickly"George Joseph
2017-07-14chan_iax2: On reload make sure to check for existing MWI subscriptionSergej Kasumovic
On every reload of chan_iax2 module, MWI subscription was added, which results in additional taskprocessors being accumulated over time. This commit fixes it by making sure we check for existing subscription first. This was verified with 'core show taskprocessors' CLI command. ASTERISK-27122 #close Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9
2017-07-13res_pjsip: Add "webrtc" configuration optionKevin Harwell
This patch creates a new configuration option called "webrtc". When enabled it defaults and enables the following options that are needed in order for webrtc to work in Asterisk: rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled media_encryption=dtls dtls_verify=fingerprint dtls_setup=actpass When "webrtc" is enabled, this patch also parses the "msid" media level attribute from an SDP. It will also appropriately add it onto the outgoing session when applicable. Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent. ASTERISK-27119 #close Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd
2017-07-13Sounds: Update for core sounds 1.6 releaseRusty Newton
Added necessary lines to make the en_NZ language set selectable and to get core sounds 1.6 pulled down. ASTERISK-26807 #close ASTERISK-25816 #close ASTERISK-26274 #close Change-Id: I84e4dd4696568cc1ba318d12ac4b075461d6eed4
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