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2017-08-03res_pjsip_outbound_registration.c: Misc fixes.Richard Mudgett
* Remove unnecessary CMP_STOP. * In handle_client_registration() use DEBUG_ATLEAST() to only do work needed for the debug log message when the debug log message is needed. * In sip_outbound_registration_state_destroy() check state->registration for NULL. Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80
2017-08-03res_pjsip_nat.c: Remove unnecessary CMP_STOP.Richard Mudgett
Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da
2017-08-03res_pjsip_registrar.c: Remove unnecessary CMP_STOP.Richard Mudgett
Most uses of CMP_STOP are superfluous and are only respected when OBJ_MULTIPLE is used to search the container. Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8
2017-08-03Support GMIME 3.0Tzafrir Cohen
Support building the Asterisk httpd with version 3.0 of gmime as well as earlier versions of that library. ASTERISK-27173 Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f
2017-08-03alembic/res_pjsip: Add "webrtc" configuration optionKevin Harwell
When the "webrtc" option was added in res_pjsip it was not added to the alembic scripts. This patch adds the option for alembic. Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of an OPT_BOOL_T so if this field is ever written to a database it will write out the correct value. ASTERISK-27119 #close Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b
2017-08-02chan_sip: Add dialplan function SIP_HEADERSkkm
Syntax: SIP_HEADERS([prefix]) If the argument is specified, only the headers matching the given prefix are returned. The function returns a comma-separated list of SIP header names from an incoming INVITE message. Multiple headers with the same name are included in the list only once. The returned list can be iterated over using the functions POP() and SIP_HEADER(). For example, '${SIP_HEADERS(Co)}' might return the string 'Contact,Content-Length,Content-Type'. Practical use is rather '${SIP_HEADERS(X-)}' to enumerate optional extended headers sent by a peer. ASTERISK-27163 Change-Id: I2076d3893d03a2f82429f393b5b46db6cf68a267
2017-08-02Fix compile error for old versions of GCC.Corey Farrell
Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6
2017-08-02app_privacy: remove unused header asterisk/image.hCorey Farrell
Change-Id: I56ed530633a642633b18383821069e806c92ae82
2017-08-01res_pjsip_pidf_eyebeam_body_supplement: Correct status presentationSean Bright
This change fixes PIDF content generation when the underlying device state is considered in use. Previously it was incorrectly marked as closed meaning they were offline/unavailable. The code now correctly marks them as open. Additionally: * Generate an XML element for our activity instead of a using a text node. * Consider every extension state other than "unavailable" to be 'open' status. * Update the XML namespaces and structure to reflect those documented in RFC 4480 * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the "in use" activity. This change results in eyeBeam using the appropriate icon for the watched user. This was tested on eyeBeam 1.5.20.2 build 59030 on Windows. ASTERISK-26659 #close Reported by: Abraham Liebsch patches: ASTERISK-26659.diff submitted by snuffy (license 5024) Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810
2017-08-01res_pjsip: Add support for dnsmgr to external_media_address.Joshua Colp
The "external_media_address" option on transports is now resolved using dnsmgr. This allows it to be automatically refreshed regularly if refreshes are enabled in dnsmgr. If the system is using a dynamic IP address a dynamic DNS hostname can be provided to keep the IP address up to date. Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2
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 "Core: Add support for systemd socket activation."Jenkins2
2017-07-26Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation ↵Joshua Colp
issues."
2017-07-26Merge "res_stasis_device_state: Unsubscribe should remove old subscriptions"Joshua Colp
2017-07-26Merge "SDP: Create declined m= SDP lines using remote SDP if applicable."Joshua Colp
2017-07-26Merge "SDP: Rework SDP offer/answer model and update capabilities merges."Joshua Colp
2017-07-26Merge "app_voicemail.c: Allow mailbox entry on authentication retry prompt."Jenkins2
2017-07-25Merge "core: Add VP9 passthrough support."Jenkins2
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"George Joseph
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
2017-07-24Merge "format.h: Fix a few minor errors in comments."Jenkins2
2017-07-24Merge "Update make_ari_stubs in master to make the version 16"Joshua Colp
2017-07-24Merge "Restore the incorrectly deleted spandspflow2pcap.log"Jenkins2
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
2017-07-21format.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
2017-07-21say.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
2017-07-21Sounds: 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
2017-07-21Update make_ari_stubs in master to make the version 16George Joseph
Ready for next major version Change-Id: If9dc99b3b78768529e69a297d8f87e23582ca6d0
2017-07-21Restore the incorrectly deleted spandspflow2pcap.logGeorge Joseph
Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5
2017-07-21Merge "corosync: Fix corosync library name in configure.ac"George Joseph
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
2017-07-20corosync: 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
2017-07-19Merge "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
2017-07-19bridge_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
2017-07-18pjsip: 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