29685 Commits (1f01106cfc273ea51fda4985cbd2bc3f33eee21e)

 

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Author	SHA1	Message	Date
Joshua Colp	1f01106cfc	Merge "chan_sip: Add dialplan function SIP_HEADERS"	9 years ago
Jenkins2	38c8080cdd	Merge "Fix compile error for old versions of GCC."	9 years ago
Jenkins2	9850034f6a	Merge "Correct some leaks in unit tests."	9 years ago
Jenkins2	7af10de1ba	Merge "res_pjsip_transport_websocket.c: Fix serializer ref leak."	9 years ago
Jenkins2	50d842b79a	Merge "res_pjsip_outbound_registration.c: Misc fixes."	9 years ago
Corey Farrell	7f8f3ca4dd	Correct some leaks in unit tests. ... * chan_sip: channel in test_sip_rtpqos_1. * test_config: config hook, config info and global config holder. * test_core_format: format in format_attribute_set_without_interface. * test_stream: unneeded frame duplication. * test_taskprocessor: task_data. Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31	9 years ago
Richard Mudgett	842e1414d0	res_pjsip_transport_websocket.c: Fix serializer ref leak. ... Change-Id: Ib5a19bfd597f63d9021baeb645fc11153b3afa57	9 years ago
Richard Mudgett	615b6a200a	res_pjsip_outbound_registration.c: Misc fixes. ... * 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	9 years ago
Richard Mudgett	5655cded78	res_pjsip_registrar.c: Remove unnecessary CMP_STOP. ... Most uses of CMP_STOP are superfluous and are only respected when OBJ_MULTIPLE is used to search the container. Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8	9 years ago
kkm	4c0798e91d	chan_sip: Add dialplan function SIP_HEADERS ... 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	9 years ago
Corey Farrell	4b03eb5c38	Fix compile error for old versions of GCC. ... Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6	9 years ago
Sean Bright	2be8d91c0f	res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation ... 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	9 years ago
Joshua Colp	2a4283f3e7	res_pjsip: Add support for dnsmgr to external_media_address. ... 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	9 years ago
Corey Farrell	58d032112b	Fix compiler warnings on Fedora 26 / GCC 7. ... 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	9 years ago
Sean Bright	3f98488279	app_queue: Add announce-position-only-up option ... 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	9 years ago
George Joseph	ac6d98b28d	bundled_pjproject: Improve SSL/TLS error handling ... 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	9 years ago
Torrey Searle	65c560894d	chan_pjsip: add a new function PJSIP_DTMF_MODE ... 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	9 years ago
Sean Bright	b3914df10b	res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours ... Change-Id: Ia578ede1a55b21014581793992a429441903278b	9 years ago
Jenkins2	1bec535df2	Merge "Core: Add support for systemd socket activation."	9 years ago
Joshua Colp	b610295b62	Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues."	9 years ago
Joshua Colp	7ea6c66968	Merge "res_stasis_device_state: Unsubscribe should remove old subscriptions"	9 years ago
Joshua Colp	3dff2711a3	Merge "SDP: Create declined m= SDP lines using remote SDP if applicable."	9 years ago
Joshua Colp	8412cc1e07	Merge "SDP: Rework SDP offer/answer model and update capabilities merges."	9 years ago
Jenkins2	8c6dcbcc6e	Merge "app_voicemail.c: Allow mailbox entry on authentication retry prompt."	9 years ago
Jenkins2	51ef6a8b6c	Merge "core: Add VP9 passthrough support."	9 years ago
Sergej Kasumovic	4f4936fd72	res_stasis_device_state: Unsubscribe should remove old subscriptions ... 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	9 years ago
George Joseph	63bcbb2a56	Merge "say.c: Fix file locations for second, seconds, minute, minutes files"	9 years ago
Joshua Colp	a6eb9ee7d2	core: Add VP9 passthrough support. ... This change adds VP9 as a known codec and creates a cached "vp9" media format for use. Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc	9 years ago
Jenkins2	8b8d5e4110	Merge "format.h: Fix a few minor errors in comments."	9 years ago
Joshua Colp	40e1b54bdb	Merge "Update make_ari_stubs in master to make the version 16"	9 years ago
Jenkins2	c54e2efc0f	Merge "Restore the incorrectly deleted spandspflow2pcap.log"	9 years ago
Richard Mudgett	922930753c	app_voicemail.c: Allow mailbox entry on authentication retry prompt. ... 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	9 years ago
Matthew Fredrickson	2697e45157	format.h: Fix a few minor errors in comments. ... A few minor problems were found in comments in format.h. This patch fixes them. Change-Id: I07f0bdb47b93359b361c4c3d8ecc87cd3199dd94	9 years ago
Rusty Newton	19b080b547	say.c: Fix file locations for second, seconds, minute, minutes files ... 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	9 years ago
Rusty Newton	a2f6028a51	Sounds: Update Makefile for Extra sounds 1.5.1 release ... 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	9 years ago
George Joseph	063c9a935f	Update make_ari_stubs in master to make the version 16 ... Ready for next major version Change-Id: If9dc99b3b78768529e69a297d8f87e23582ca6d0	9 years ago
George Joseph	ba52a36ff2	Restore the incorrectly deleted spandspflow2pcap.log ... Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5	9 years ago
George Joseph	25459ac083	Merge "corosync: Fix corosync library name in configure.ac"	9 years ago
Jenkins2	96b69f6a47	Merge "Update AMI and ARI versions for master/15 and update UPDATE.txt"	9 years ago
George Joseph	2720e2b2b2	Merge "pjsip: Increase maximum packet size."	9 years ago
George Joseph	3e8d628c0e	Update AMI and ARI versions for master/15 and update UPDATE.txt ... 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	9 years ago
Sean Bright	25c9464325	corosync: Fix corosync library name in configure.ac ... Also add new corosync packages to install_prereq. Reported by Travis Ryan in #asterisk-dev Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db	9 years ago
Joshua Colp	f43fc91911	Merge "core: Add digit filtering to ast_waitfordigit_full"	9 years ago
George Joseph	66b8c454d1	Merge "app_playback.c: Use the timezonename parameter"	9 years ago
Jenkins2	9b07d3ba18	Merge "bridge_softmix: Use removed stream spots when renegotiating."	9 years ago
Jenkins2	62c381afdb	Merge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO."	9 years ago
Joshua Colp	680c491a62	bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. ... 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	9 years ago
Benjamin Keith Ford	e7d9e42616	pjsip: Increase maximum packet size. ... The maximum packet size for PJSIP has been increased to handle the multiple streams being added for WebRTC. Change-Id: I9ea1e8d02668c544acadcb1c6200e1cc1bd588b3	9 years ago
George Joseph	47084ad09e	Merge "app_queue: Add change priority of call"	9 years ago
Jenkins2	fb3c7926b7	Merge "bridge_softmix: Don't reorder streams on participant leaving."	9 years ago