4673 Commits (9fb166cf3b0191a1b16f883f1f8070bb40e95fa7)

Author	SHA1	Message	Date
George Joseph	597f612645	Merge "res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch" into 16	8 years ago
Joshua Colp	62afa54977	Merge "res_fax: Handle fax gateway being started more than once." into 16	8 years ago
Joshua Colp	ad37ab9a8f	Merge "res_pjsip_transport_websocket: Properly set src_name for IPv6" into 16	8 years ago
Richard Mudgett	4dd8b5bbb4	res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch ... ASTERISK-27988 Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843	8 years ago
Joshua Colp	6f27ad59f5	Merge "Create --disable-binary-modules option." into 16	8 years ago
Joshua Colp	390d0b42ca	res_fax: Handle fax gateway being started more than once. ... The T.38 fax gateway state machine can cause the fax gateway to be started more than once on a channel depending on the responses of the remote endpoint. This would previously leak the channel name, channel unique id, and underlying fax engine state. This change instead makes it so that if the fax gateway session is already present and not reserved the fax gateway is not started again. ASTERISK-27981 Change-Id: I552d95086860cb18f2522ee40ef47b13b6da2e0e	8 years ago
Sean Bright	245fb462d6	res_pjsip_transport_websocket: Properly set src_name for IPv6 ... SIP responses over WebSockets when the client is using IPv6 have invalid Via headers according to RFC 3261. The 'received' header parameter should not be wrapped in brackets if it is an IPv6 address. When src_name is populated by the built-in PJSIP transports, the code uses pj_sockaddr_print() with 'flags' set to 0, meaning that the brackets are not rendered around IPv6 addresses. This may be related to ASTERISK~27101. See also: https://github.com/onsip/SIP.js/pull/594 ASTERISK-28020 #close Change-Id: I8ea9d289901b837512bee2ca2535e3dc14f04d77	8 years ago
Corey Farrell	1b1f47bef6	Create --disable-binary-modules option. ... This new option can be passed for ./configure or ./tests/CI/buildAsterisk.sh to prevent download/install of binary modules. Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES will result in binary modules being enabled even if the build target is incompatible with those modules. This includes CI scripts which enable categories before disabling specific modules. If more binary modules are offered in the future this will help avoid accidentally downloading them if unwanted or incompatible. Adding a binary module will only require creating a new menuselect entry similar to the existing ones, it will not be necessary to modify the CI scripts. Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166	8 years ago
neutrino88	aa2755cbb3	res/res_rtp_asterisk: remove debug traces generated by an empty frame ... The realtime text timer pops regularly and sends text frames even if the buffer is empty. This causes a lot of unecessary debug logging. * Made red_write() test if we need to send a frame before calling ast_rtp_write() ASTERISK-28002 Reported by: Emmanuel BUU Tested by: Emmanuel BUU Change-Id: Icf81310c3b8080b615a42060afc02ab41f9523dd	8 years ago
Joshua Colp	378964f403	Merge "res_pjsip: Reduce processing when a Contact is updated." into 16	8 years ago
George Joseph	b523aaf699	Merge "res_sorcery_realtime.c: Fix unqualified fetch warning." into 16	8 years ago
George Joseph	0fe2eadbc3	Merge "res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response." into 16	8 years ago
Joshua Colp	b9cd4c6d92	res_pjsip: Reduce processing when a Contact is updated. ... When a Contact is updated the only material change that qualify support cares about is the underlying configuration for the AOR. In this case we will update things with the new AOR information but otherwise the callback to indicate the Contact has changed can be ignored. This is because it is only when a Contact is added or deleted that material changes occur within the qualify support. An update can't change the URI since it would result in a new Contact so it can be ignored. Change-Id: I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d	8 years ago
Richard Mudgett	236826a111	res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response. ... We were still getting crashes after the first fix. Somehow we receive a non-2xx final response before we get a 200 final response. With the failure response we had already cleaned up and destroyed some data structures. When the unexpected 200 response comes in we crash. * Add protection code to prevent processing another final T.38 reINVITE response. ASTERISK-27944 Change-Id: I8b5baba8d07fe4d63f0d7d05d3eb9a3d27d40a74	8 years ago
Richard Mudgett	19298141cf	res_sorcery_realtime.c: Fix unqualified fetch warning. ... The allow_unqualified_fetch option for the sorcery realtime backend blocked actually fetching all rows when the option is set to warn. * Made issue a warning and actually do the request when allow_unqualified_fetch=warn is set. Change-Id: I74456c80a03a62dce66fc3dc3cb0cf2351ac4312	8 years ago
Richard Mudgett	0874d5b316	res_rtp_asterisk.c: Fix unused variable warnings ... Compiling without SRTP support installed resulted in some unused variable warnings. These warnings also showed that the srtp variable was obtained and passed around some functions but not really used even when a system has SRTP installed. Change-Id: I6daad34be3e89b19adef6e2fbe738018975155fc	8 years ago
Joshua Colp	7d8e2389d6	Merge "res_resolver_unbound: Fix leak of config nameserver strings." into 16	8 years ago
Joshua Colp	14da6be84d	Merge "res_pjsip: Resolve transport management leak at shutdown." into 16	8 years ago
Kevin Harwell	80a331d96b	Merge "res_odbc: Allow unload at shutdown." into 16	8 years ago
George Joseph	4f95992d36	Merge "res/res_pjsip_sdp_rtp: put rtcp-mux in answer only if offered" into 16	8 years ago
Torrey Searle	0d4bde84d1	res/res_pjsip_sdp_rtp: put rtcp-mux in answer only if offered ... If in the initial sdp the caller doesn't include the line a=rtcp-mux Then asterisk shoud not include rtcp-mux in the response regardless of rtcp-mux being enabled on the endpoint ASTERISK-28007 #close Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7	8 years ago
Corey Farrell	167efe3a47	res_resolver_unbound: Fix leak of config nameserver strings. ... Change-Id: I3f396316bb40d1ae6e91f5f688042420f1a540ed	8 years ago
Corey Farrell	72dbc9fb70	res_pjsip: Resolve transport management leak at shutdown. ... Cleanup idle check scheduled events at shutdown. Change-Id: I61bfbb56bac69fe840c3242927d31ff3593be461	8 years ago
Corey Farrell	6e0f4a2127	res_pjsip: Fix leak in pjsip_options. ... sip_options_get_endpoint_state_compositor_state leaked a reference to the first available endpoint state compositor that was found. Change-Id: Idb6be19f7219b6eed1dfb19c1e740dd40cb3fdc7	8 years ago
Corey Farrell	b370482786	res_odbc: Allow unload at shutdown. ... This makes it possible for REF_DEBUG to report no leaks when loading res_odbc. Change-Id: I1a3dea786bd6e7f4820a6dd5cbaa197fa783ce93	8 years ago
George Joseph	100ffc6866	Merge "res_pjsip/rtp: No joint capabilities between streams." into 16	8 years ago
Joshua Colp	56c1285b8a	res_pjsip_caller_id: Add "party" parameter to RPID header. ... This change adds the "party" parameter to the Remote-Party-ID header which indicates which party the header information is applicable to. In Asterisk this is determined on whether we are the calling or called party. This is added to improve interoperability with some implementations. ASTERISK-28006 Change-Id: I1eec3e377ffff8633b5c1dd59a05e9533122cfca	8 years ago
Ben Ford	a46fcaca7b	res_pjsip/rtp: No joint capabilities between streams. ... When a conference contained a mixture of audio/video and audio-only users, a NOTICE message would pop up stating there are no joint capabilities between streams. This happens because streams can never be removed, but they can be in a REMOVED state. If we have the scenario where user A joins with audio/video, user B joins with audio-only, and user C joins with audio/video, then user A leaves, the message would be triggered. That removed stream is still in the SDP, but Asterisk would pass it through, causing it to be seen as a ulaw stream. A check has been added for removed streams, setting their status to REMOVED when handling negotiated SDPs. Also addressed an issue where user A joins, then user B joins but does not receive video until much later. Full frames were not being sent, causing some PLI from the browser. Because the video was flowing in one direction, the browser sets the SSRC to 1, but Asterisk was dropping the PLI because of that. Added a check to see if the SSRC is 1 or not, which sends full frames and allows video to flow between user A and user B. This should only happen when dealing with PSFB or FUR, and in the case of PSFB, only for PLI. ASTERISK-27398 Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e	8 years ago
Kevin Harwell	1d1473d408	Merge "res_pjsip_registrar: Improve performance on inbound handling." into 16	8 years ago
Joshua Colp	0df8ab0adc	Merge "res_pjsip: Make pjlib.h consistently included." into 16	8 years ago
Joshua Colp	ef029a3224	Merge "pjproject_bundled: Fix for Solaris builds. Do not undef s_addr." into 16	8 years ago
Alexander Traud	04974a0ca2	pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. ... The authors of PJProject undef s_addr because of some issue in Microsoft Windows. However in Oracle Solaris, s_addr is not a structure member, but defined to map to the real structure member. Updates the patch from ASTERISK_20366 ASTERISK-27997 Change-Id: I8223026d4d54e2a46521085fcc94bfa6ebe35b11	8 years ago
Richard Mudgett	99a0586ec1	res_pjsip: Make pjlib.h consistently included. ... * Don't include pjlib.h twice in res_pjsip.h * Consistently use #include <> form for pjproject includes. (pjsip.h and pjlib.h) Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7	8 years ago
Salah Ahmed	523b7b2ffc	dialplan_functions: wrong srtp use status report of a dialplan function ... If asterisk offer an endpoint with SRTP and that endpoint respond with non srtp, in that case channel(rtp,secure,audio) reply wrong status. Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key: Currently this flag has being set redundantly. In either case identical or different remote_key this flag has being set. So if we don't set it while we receive identical remote_key or non SRTP SDP response then we can take decision of srtp use by using that flag. ASTERISK-27999 Change-Id: I29dc2843cf4e5ae2604301cb4ff258f1822dc2d7	8 years ago
Kevin Harwell	07c23cea37	Merge "res_pjsip_endpoint_identifier_ip.c: Added regex support to match_header" into 16	8 years ago
Joshua Colp	1e837e13f5	res_pjsip_registrar: Improve performance on inbound handling. ... This change removes a sorcery lookup for retrieving all contacts at the end of the registration process by keeping track of the contacts that are added/updated/deleted. This ensures at the end of the process the container of contacts we have is the current state. Pool usage has also been reduced by allocating one for usage throughout the handling of a REGISTER and resetting it to a clean state. This ensures that in most cases we allocate once and just reuse it. ASTERISK-28001 Change-Id: I1a78b2d46f9a2045dbbff1a3fd6dba84b612b3cb	8 years ago
Joshua Colp	927f68bb9d	Merge "res_rtp_asterisk: In Developer Mode, do not require OpenSSL." into 16	8 years ago
Joshua Colp	ee9794d741	res_pjsip_pubsub: Use ast_true for "prune_on_boot". ... Change-Id: Iedec4e7390b3e821987681da24d0298632b9873d	8 years ago
Richard Mudgett	32ce8e5cf3	res_pjsip_endpoint_identifier_ip.c: Added regex support to match_header ... This patch adds regular expression support to make the identify section's match_header option more useful when attempting to match complex headers like the 'To' or 'From' headers. The 'From' header has variable components such as the tag parameter that you cannot predict. To specify a regular expression put slashes around the regular expression in place of the header value. [identify-alice] type=identify endpoint=alice match_header=From: /<sip:alice@127\\.0\\.0\\.1>/ * Added regex support to match_header so you could match a 'To' header among other complex headers. Fixed reported crashes when trying to match special headers like 'Contact'. The identify section's match_header method used code that assumed you were matching a generic header. Any other type of header could cause a crash if the header structure variant did not match the generic header enough. * Made use code that will work for any header type instead of code specific to generic headers. Other fixes while in the area: * Made check all headers of the requested name. * Added some more sanity checks to the configured identify matching options when applying the configuration. ASTERISK-27548 Change-Id: I27dfd4ff5e2259b906640e3c330681b76b4ed1f1	8 years ago
Joshua Colp	59f53514ce	res_pjsip_pubsub: Treat "prune_on_boot" as a yes / no. ... The alembic for the PJSIP subscription persistence table has the "prune_on_boot" field as a boolean. While in Asterisk we are tolerant of many different definitions of true and false in the database we only accept "yes" and "no". This change makes the field treated as a yes/no instead of an integer, thus storing "yes" and "no" instead of "1" and "0". Change-Id: Ic8b9211b36babefe78f70def6828a135a6ae7ab6	8 years ago
Alexander Traud	8da81a208f	res_rtp_asterisk: In Developer Mode, do not require OpenSSL. ... OpenSSL is an optional external library and should stay optional even when Developer Mode is configured. ASTERISK-27990 Change-Id: Ia68a4cd5474b26d45e0f43b04032ad598022853b	8 years ago
neutrino88	d3809d61cb	res_rtp_asterisk: Avoid merging command and regular T.140 text packets ... When realtime text packets are to be sent, the text is accumulated in a buffer and sent regularly by a timer. It can happen that commands such as a backspace, CR, or LF get merged with regular text. This breaks some UAs. The proposed change: * We test if the current packet contains a command. If so we send the buffer immediately. * We test if the buffer contained a command. If so we send the buffer immediately. * We accumulate the text (or the command) in the buffer. ASTERISK-27970 Change-Id: Ifbe993311410fa855cb8aa4a12084db75f413462	8 years ago
Joshua Colp	84b5245476	Merge "res_pjsip: Change log message from error to warning for valid use cases" into 16	8 years ago
George Joseph	835489f76d	Merge "res_pjsip: Update default keepalive interval to 90 seconds." into 16	8 years ago
Florian Floimair	1f46e2e91c	res_pjsip: Change log message from error to warning for valid use cases ... If a SIP MESSAGE is triggered for an endpoint that is currently not registered - and therefore has no valid contact associated - an error message was logged. Since this is a valid request in a valid use cases this is now changed to a warning, as discussed with Matt Fredrickson on the asterisk-dev mailing list. Change-Id: I55eb62d2712818a58c7532119dec288bd98cf0c0	8 years ago
George Joseph	ebbc29f9f5	Merge "res_pjsip: Update endpoint transport option documentation." into 16	8 years ago
Joshua Colp	a2a3ad2438	res_pjsip: Update default keepalive interval to 90 seconds. ... A change recently went in which disabled the built-in PJSIP keepalive. This defaulted to 90 seconds and kept TCP/TLS connections alive. Disabling this functionality has resulted in a behavior change of not doing keepalives by default resulting in TCP/TLS connections dropping for some people. This change makes our default keepalive interval 90 seconds to match the previous behavior and preserve it. ASTERISK-27978 Change-Id: Ibd9a45f3cbe5d9bb6d2161268696645ff781b1d6	8 years ago
Richard Mudgett	dbffcdc561	res_pjsip: Update endpoint transport option documentation. ... Change-Id: I5394fdff6a296efc8e1695a156e616acd932ae52	8 years ago
Richard Mudgett	709b795cb0	pjsip_resolver.c: Use replacement function ... * Use the replacement function ast_sip_push_task_wait_servant() instead of the deprecated ast_sip_push_task_synchronous(). Change-Id: I145b550ba7054640c7faa3b644e63137f505c612	8 years ago
George Joseph	fa71763853	Merge "res_sorcery_config: Allow configuration section to be used based on name."	8 years ago