asterisk

Commit Graph

Author	SHA1	Message	Date
Torrey Searle	be8cb75dd2	res_rtp_asterisk: Make P2P bridge Asymmetric codec aware Introduce a new property to rtp-engine to make it aware of the desire for assymetric codecs or not. If asymmetric codecs is not allowed, the bridge will compare read/write formats and shut down the p2p bridge if needed ASTERISK-26745 #close Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f	8 years ago
Joshua Colp	709b837c35	Merge "res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect" into 15	8 years ago
Jenkins2	11d1d51844	Merge "res_pjsip_session: Release media resources on session end quicker." into 15	8 years ago
Joshua Colp	9685a704a6	res_pjsip_session: Release media resources on session end quicker. A change was made long ago where the session was kept around until the underlying INVITE session had been destroyed. This had the side effect of also keeping the underlying media resources around for this time as well. This change ensures that when we are told to terminate the session we immediately release any media sessions associated with it. ASTERISK-27110 Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82	8 years ago
Jenkins2	b24854e16b	Merge "bridge: Fix stream topology/participant locking and video misrouting." into 15	8 years ago
Jenkins2	866dd2c816	Merge "channel: Fix leak on successful call to chan->tech->requester." into 15	8 years ago
Joshua Colp	027f3f665b	Merge "res_pjsip_nat.c: Remove unnecessary CMP_STOP." into 15	8 years ago
Jenkins2	f339069d09	Merge "Support GMIME 3.0" into 15	8 years ago
Joshua Colp	0e352ec510	bridge: Fix stream topology/participant locking and video misrouting. This change fixes a few locking issues and some video misrouting. 1. When accessing the stream topology of a channel the channel lock must be held to guarantee the topology remains valid. 2. When a channel was joined to a bridge the bridge specific implementation for stream mapping was not invoked, causing video to be misrouted for a brief period of time. ASTERISK-27182 Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03	8 years ago
Corey Farrell	3dbb1b9f48	channel: Fix leak on successful call to chan->tech->requester. joint_cap needs to be released unconditionally as chan->tech->requester does not steal the reference even on success. ASTERISK-27180 #close Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6	8 years ago
Kevin Harwell	09995fae53	res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect Currently, the handling of the msid attribute is not quite right. According to the spec the msid's between the offer/answer are not dependent upon one another. Meaning the same msid's given in an offer do not have to be returned in the answer for a given stream. And they probably shouldn't be (copied/reused) since this can potentially cause some browser side confusion. This patch generates new msids when both an offer and answer are sent from Asterisk. However, Asterisk does reuse the original msid it sent out for a reinvite. Also audio+video streams are paired together by sharing the same stream id, but a different track id. ASTERISK-27179 #close Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643	8 years ago
Corey Farrell	418e062ce5	app_privacy: remove unused header asterisk/image.h Change-Id: I56ed530633a642633b18383821069e806c92ae82	8 years ago
Jenkins2	2014a695f7	Merge "alembic/res_pjsip: Add "webrtc" configuration option" into 15	8 years ago
Joshua Colp	3517551b3a	Merge "Fix compile error for old versions of GCC." into 15	8 years ago
Jenkins2	3ac2da8974	Merge "Correct some leaks in unit tests." into 15	8 years ago
Jenkins2	4c7f81af6e	Merge "res_pjsip_transport_websocket.c: Fix serializer ref leak." into 15	8 years ago
Jenkins2	230cf2ee59	Merge "res_pjsip_outbound_registration.c: Misc fixes." into 15	8 years ago
Tzafrir Cohen	0cbfb95b7f	Support GMIME 3.0 Support building the Asterisk httpd with version 3.0 of gmime as well as earlier versions of that library. ASTERISK-27173 Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f	8 years ago
Corey Farrell	1a3e08326e	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	8 years ago
Richard Mudgett	563264c9b5	res_pjsip_transport_websocket.c: Fix serializer ref leak. Change-Id: Ib5a19bfd597f63d9021baeb645fc11153b3afa57	8 years ago
Richard Mudgett	347e148aa0	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	8 years ago
Richard Mudgett	db6b883087	res_pjsip_nat.c: Remove unnecessary CMP_STOP. Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da	8 years ago
Richard Mudgett	2cd7bec061	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	8 years ago
Kevin Harwell	b0c016cf6e	alembic/res_pjsip: Add "webrtc" configuration option 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	8 years ago
Corey Farrell	cd79a15b2f	Fix compile error for old versions of GCC. Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6	8 years ago
George Joseph	6edefc9480	Revert "res_pjsip_session: Release media resources on session end quicker." This reverts commit 6c374d863c420d2519a52fffba3de5f935a30cdb. Change-Id: Ia89836005df66ef540885afd6f171886de0db442	8 years ago
Joshua Colp	8f51443a08	res_pjsip_session: Release media resources on session end quicker. A change was made long ago where the session was kept around until the underlying INVITE session had been destroyed. This had the side effect of also keeping the underlying media resources around for this time as well. This change ensures that when we know the session is ending we release the media resources immediately. ASTERISK-27110 Change-Id: I3c6a82fe7d2c50b9dc9197cb12ef22f20d337501	8 years ago
Sean Bright	ef39998169	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	8 years ago
Joshua Colp	448317ddc7	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	8 years ago
Corey Farrell	7c3cdc81f7	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	8 years ago
Sean Bright	1078d4f7f6	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	8 years ago
George Joseph	63224d5d9c	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	8 years ago
Torrey Searle	c24cdbf4ae	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	8 years ago
Sean Bright	d056f6b2fe	res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours Change-Id: Ia578ede1a55b21014581793992a429441903278b	8 years ago
Joshua Colp	1c07c80eef	Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues." into 15	8 years ago
Jenkins2	fab609deeb	Merge "Core: Add support for systemd socket activation." into 15	8 years ago
Joshua Colp	28979a0f1c	Merge "res_stasis_device_state: Unsubscribe should remove old subscriptions" into 15	8 years ago
Jenkins2	a11f5fb51a	Merge "app_voicemail.c: Allow mailbox entry on authentication retry prompt." into 15	8 years ago
Joshua Colp	451d86d62e	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	8 years ago
Jenkins2	29a746c534	Merge "core: Add VP9 passthrough support." into 15	8 years ago
Sergej Kasumovic	2128dc7c87	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	8 years ago
Jenkins2	644af26f70	Merge "say.c: Fix file locations for second, seconds, minute, minutes files" into 15	8 years ago
Joshua Colp	927fc6bbd9	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	8 years ago
Matthew Fredrickson	9aa4942a49	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	8 years ago
Jenkins2	a4b55519d0	Merge "Restore the incorrectly deleted spandspflow2pcap.log" into 15	8 years ago
Richard Mudgett	85c631294a	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	8 years ago
Rusty Newton	e0ad75ec2a	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	8 years ago
Rusty Newton	715d79b60d	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	8 years ago
Corey Farrell	eea9da2f42	Core: Add support for systemd socket activation. 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	8 years ago
George Joseph	94de9d3eea	Restore the incorrectly deleted spandspflow2pcap.log Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5	8 years ago

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