29192 Commits (6b7697ed486fc3a8e5e7a72344437e66bd4ae507)

 

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Author	SHA1	Message	Date
Richard Begg	6b7697ed48	res_pjsip_session: Enable RFC3578 overlap dialing support. ... Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	8 years ago
Joshua Colp	f5603cb1ec	Merge "res/res_pjsip_session: Only check localnet if it is defined"	8 years ago
zuul	24ada76408	Merge "thread safety: Don't use getprotobyname()"	8 years ago
Sean Bright	fc71c18a9b	thread safety: Don't use getprotobyname() ... POSIX does not require getprotobyname() to be thread safe and some implementations use static memory which causes issues when multiple threads are used. Further, our usage of it today is just to ultimately get IPPROTO_TCP for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48	8 years ago
Sean Bright	516e028b44	res_rtp_asterisk: Pass correct data length to ast_rtcp_interpret ... We are currently passing in the capacity of the read buffer instead of the number of bytes that we actually read off the wire. Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36	8 years ago
Joshua Colp	b05d2fda0c	Merge "app_queue: Fix locking behavior in stasis message handlers"	8 years ago
Joshua Colp	3996c79328	Merge "chan_sip: Add rtcp-mux support"	8 years ago
Joshua Colp	77582634d7	Merge "res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped."	8 years ago
Joshua Colp	0db211dc64	Merge "res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed."	8 years ago
Joshua Colp	317405ce69	Merge "app_confbridge: Fix ConfbridgeTalking AMI event description."	8 years ago
Joshua Colp	7f87cd7b4e	Merge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error"	8 years ago
Joshua Colp	15c72b3239	Merge "res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport"	8 years ago
Sean Bright	c13ea6080e	app_queue: Fix locking behavior in stasis message handlers ... The queue_stasis_data structure contains various mutable fields that require appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and 'caller_uniqueid' fields need to be locked when read from or written to. Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088	8 years ago
Sean Bright	15aa3c0a23	chan_sip: Add rtcp-mux support ... ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	8 years ago
Richard Mudgett	57656e2b5b	app_confbridge: Fix ConfbridgeTalking AMI event description. ... Thanks to Chris Howard for pointing this out on the wiki. Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705	8 years ago
Richard Mudgett	82982a191c	res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed. ... struct ast_rtcp does not define the dtls member if SRTP is not enabled. ASTERISK-26732 Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e	8 years ago
Joshua Colp	732367e806	Merge "res_pjsip: Symmetric transports"	8 years ago
Richard Mudgett	49b1f1ca16	res_pjsip_sdp_rtp.c: Fix cut-n-paste error ... We were inadvertenly referencing the cos_video option to determine if we should set the tos_audio and cos_audio value on the RTP instance. Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0	8 years ago
Matt Jordan	e6dc28b78f	res/res_pjsip_session: Only check localnet if it is defined ... If local_net is not defined on a transport, transport_state->localnet will be NULL. ast_apply_ha will, be default, return AST_SENSE_ALLOW in this case, causing the external_media_address, if set, to be skipped. This patch causes us to only check if we are sending within a network if local_net is defined. ASTERISK-26879 #close Change-Id: Ib661c31a954cabc9c99f1f25c9c9a5c5b82cbbfb	8 years ago
Joshua Colp	76e64f5589	Merge "RFC sdp: Initial SDP creation"	8 years ago
Richard Begg	44568fc712	res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport ... Currently a wildcard address is used for the local RTP socket, which will not always result in the same address as used by the SIP socket (e.g. if explicit transport addresses are configured). Use the transport's host address when binding new local RTP sockets if available. ASTERISK-26851 Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a	8 years ago
George Joseph	5013d8f5d3	res_pjsip: Symmetric transports ... A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	8 years ago
Joshua Colp	68749a9fa7	res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped. ... This change removes an assumption that when DTLS is stopped an RTCP session will be present on the RTP session. This is not always the case. ASTERISK-26732 Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611	8 years ago
Joshua Colp	84f0871cba	Merge "Add rtcp-mux support"	8 years ago
Joshua Colp	aec2a087ed	Merge "chan_iax2: Reload of iax peer results in loss of host address/port"	8 years ago
zuul	5fdd61b556	Merge "res/res_pjsip_refer: call xfer w/o extension"	8 years ago
zuul	300b257045	Merge "app_queue: Handle the caller being redirected out of a queue bridge"	8 years ago
zuul	bcc9a07db2	Merge "funcs/func_devstate: Remove new line in Device field of during module load"	8 years ago
zuul	3f30ce1272	Merge "pbx.c: Fix crash from malformed exten pattern."	8 years ago
zuul	941671b27a	Merge "res_pjsip_endpoint_identifier_ip: Don't output error if no header_match."	8 years ago
zuul	3fe1d8afba	Merge "core: Add stream topology changing primitives with tests."	8 years ago
Mark Michelson	10fa49e327	Add rtcp-mux support ... This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5	8 years ago
Joshua Colp	e536ef7afb	Merge "res_pjsip_endpoint_identifier_ip: Add an option to match requests by header"	8 years ago
Joshua Colp	2cb449a621	Merge "configure: Don't use the progress bar with curl when downloading to stdout"	8 years ago
Torrey Searle	dc4cdafd42	res/res_pjsip_refer: call xfer w/o extension ... When transfering to a URI without an extension, ensure that the s extension of the dialplan is entered ASTERISK-26869 #close Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525	8 years ago
Sean Bright	982d6173c5	app_queue: Handle the caller being redirected out of a queue bridge ... A caller can leave the Queue() application after being bridged with a member in a few ways: * Caller or member hangup * Caller is transferred somewhere else (blind or atx) * Caller is externally redirected elsewhere The first 2 scenarios are currently handled by subscribing to stasis messages, but the 3rd is not explicitly covered. If a caller is redirected away from the Queue() application, the member who was last bridged with that caller will remain in an "In use" state until the caller hangs up. This patch adds handling of the caller leaving the queue via redirection. We monitor the caller-member bridge, and if the caller is the one that leaves, we treat it the same as we would a caller hangup. ASTERISK-26400 #close Reported by: Etienne Lessard Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334	8 years ago
Joshua Colp	0b8a57af6d	res_pjsip_endpoint_identifier_ip: Don't output error if no header_match. ... This change ensures that if no header_match option is set on an identify an error message is not output stating the option is set to an invalid value. ASTERISK-26863 Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a	8 years ago
Matt Jordan	1475604eff	res_pjsip_endpoint_identifier_ip: Add an option to match requests by header ... This patch adds a new features to the endpoint identifier module, 'match_header'. When set, inbound requests are matched by a provided SIP header: value pair. This option works in conjunction with the existing 'match' configuration option, such that if any 'match*' attribute matches an inbound request, the request is associated with the specified endpoint. Since this module now identifies by more than just IP address, appropriate renaming of the module and/or variables can be done in a non-release branch. ASTERISK-26863 #close Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453 (cherry picked from commit 30f52d79d7)	8 years ago
George Joseph	71cc3fd969	Merge "res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue"	8 years ago
Joshua Colp	c5e1e85355	Merge "configs/samples/hep.conf.sample: Clarify how the HEP stack works"	8 years ago
Joshua Colp	2ffce60844	Merge "main/stasis_cache: Demote the ERROR message when removing a nonexistent item"	8 years ago
zuul	c152329932	Merge "res_pjsip_transport_websocket: Add support for IPv6."	8 years ago
Richard Mudgett	f997090877	pbx.c: Fix crash from malformed exten pattern. ... Forgetting to indicate an exten is a pattern can cause a crash if the "pattern" has a character set range. e.g., "9999[3-5]" The crash is due to a buffer overwrite because the '-' exten eye-candy wasn't removed as expected and overran the allocated space. The buffer overwrite is fixed two ways in this patch. 1) Fix ext_strncpy() to distinguish between pattern and non-pattern extens. Now '-' characters are removed when they are eye-candy and not when they are part of a pattern character set. Since the function is private to pbx.c, the return value now returns the number of bytes written to the destination buffer instead of the strlen() of the final buffer so the callers that care don't need to add one. 2) Fix callers to ext_strncpy() to supply the correct available buffer size of the destination buffer. ASTERISK-26668 Change-Id: I555d97411140e47e0522684062d174fbe32aa84a	8 years ago
Richard Begg	0dc007e94d	chan_iax2: Reload of iax peer results in loss of host address/port ... When using a non-dynamic peer address, build_peer() invalidates the peer address structure by setting the address family to unspecified. However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup() will not amend the peer address if the cache is still valid, resulting in peer connectivity failures. To fix this, we call ast_dnsmgr_refresh() instead. ASTERISK-26865 Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082	8 years ago
Matt Jordan	59130260e7	configure: Don't use the progress bar with curl when downloading to stdout ... In some scenarios, such as when there may not be a terminal (such as inside a Docker container), curl will apparently direct the progress bar to stdout. This can cause extra data to be appended to a file curl'd down to stdout, resulting in md5 verification failures. This patch removes the progress bar, and tells curl to download the file silently. ASTERISK-26872 #close Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c	8 years ago
zuul	2b611a8d93	Merge "chan_pjsip: Don't assume a session will have a channel."	8 years ago
George Joseph	8470c2bdea	RFC sdp: Initial SDP creation ... * Added additional fields to ast_sdp_options. * Re-organized ast_sdp. * Updated field names to correspond to RFC4566 terminology. * Created allocs/frees for SDP children. * Created getters/setters for SDP children where appropriate. * Added ast_sdp_create_from_state. * Refactored res_sdp_translator_pjmedia for changes. Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48	8 years ago
Joshua Colp	578bc33f6f	Merge "chan_sip: Call not cancelled after receiving a 422 response"	8 years ago
Matt Jordan	05713c36ea	configs/samples/hep.conf.sample: Clarify how the HEP stack works ... This patch updates the documenation in hep.conf.sample to better specify how the various HEP modules interact. ASTERISK-26717 #close Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124	8 years ago
Matt Jordan	0ded269bfa	funcs/func_devstate: Remove new line in Device field of during module load ... During module loading of func_devstate, Asterisk emits the current device state of all Custom device states currently stored in the AstDB. This was erroneously including a new line character ('\n') to the end of the device state, causing two new lines to be emitted in DeviceStateChange AMI events. Note that this only happened for those device state changes that occurred during startup. Regular device state changes for Custom device states are handled elsewhere, and did not have the newline. ASTERISK-26643 #close Reported by: Roman Bedros Tested by: Matt Jordan patches: ami_devstate.diff uploaded by Roman Bedros (License 6842) Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93	8 years ago