29081 Commits (a6761d2d58bd703452e45e7805ba1202eaeb98fe)

 

All Branches   

Search

Author	SHA1	Message	Date
zuul	a6761d2d58	Merge "res_pjsip_session: Enable RFC3578 overlap dialing support." into 14	8 years ago
zuul	ab0a05716d	Merge "res_pjsip_messaging: Check URI type before dereferencing" into 14	8 years ago
zuul	e528c2fc18	Merge "Revert "app_queue: Handle the caller being redirected out of a queue bridge"" into 14	8 years ago
zuul	db246cd87e	Merge "app_queue: Member stuck as pending after forwarding previous call from queue" into 14	8 years ago
Joshua Colp	df0fdf31b8	Merge "pjsip: prevent memory corruption on creation of xml bodies" into 14	8 years ago
Richard Begg	d7f50afb7c	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	d36a260fcf	Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references." into 14	8 years ago
zuul	a0824a20f9	Merge "res_hep: Capture actual transport type in use" into 14	8 years ago
Sean Bright	fa45b11727	res_hep: Capture actual transport type in use ... Rather than hard-coding UDP, allow consumers of the HEP API to specify which protocol is in use. Update the PJSIP provider to pass in the current protocol type. ASTERISK-26850 #close Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978	8 years ago
Sean Bright	b7a2d271c1	Revert "app_queue: Handle the caller being redirected out of a queue bridge" ... This reverts commit 163e9e53dc. Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b	8 years ago
Sean Bright	0a51c52cc0	res_pjsip_messaging: Check URI type before dereferencing ... We aren't validating that the URI we just parsed is a SIP/SIPS one before trying to access the user, host, and port members of a possibly uninitialized structure. Also update the MessageSend documentation to indicate what 'from' formats are accepted. ASTERISK-26484 #close Reported by: Vinod Dharashive Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30	8 years ago
Joshua Elson	cd80729193	pjsip: prevent memory corruption on creation of xml bodies ... ASTERISK-26776 #close Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2	8 years ago
Sean Bright	aac3c94ca4	bridge_softmix: Ignore non-voice frames from translator ... Some codecs - codec_speex specifically - take voice frames and return other types of frames, like CNG. If we subsequently treat those as voice frames, we'll run into trouble when destroying the frame because of the requirement that each voice frame have an associated format. ASTERISK-26880 #close Reported by: Kirsty Tyerman Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c	8 years ago
zuul	8c529c4f92	Merge "res/res_pjsip_session: Only check localnet if it is defined" into 14	8 years ago
zuul	74f59af52a	Merge "thread safety: Don't use getprotobyname()" into 14	8 years ago
Sean Bright	75c4b1c97d	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	a38a44880a	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	117092efac	Merge "app_queue: Fix locking behavior in stasis message handlers" into 14	8 years ago
Joshua Colp	3deed1f80e	Merge "chan_sip: Add rtcp-mux support" into 14	8 years ago
Joshua Colp	2295756e82	Merge "res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped." into 14	8 years ago
Joshua Colp	41baa7b4e8	Merge "res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed." into 14	8 years ago
Joshua Colp	0b3f807589	Merge "app_confbridge: Fix ConfbridgeTalking AMI event description." into 14	8 years ago
Joshua Colp	002dffe212	Merge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error" into 14	8 years ago
Robert Mordec	6ae6e16468	app_queue: Member stuck as pending after forwarding previous call from queue ... Queue member will get stuck in pending_members if queue calls a device that is different from the one observed for state changes. This patch removes members from pending_members as a result of channel stasis events such as blind or attended transfers and hangup. ASTERISK-26862 #close Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727	8 years ago
Sean Bright	e8764a93cb	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	217ba2e5d0	chan_sip: Add rtcp-mux support ... ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	8 years ago
Richard Mudgett	e7ed625d23	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	241f069184	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
Richard Mudgett	ea6ced0eef	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	486abb4210	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
Richard Begg	67204a0a3d	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
Joshua Colp	11e806abbb	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
George Joseph	6541a07083	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	a07015cd4d	Merge "Add rtcp-mux support" into 14	8 years ago
Joshua Colp	face057f45	Merge "res/res_pjsip_refer: call xfer w/o extension" into 14	8 years ago
Joshua Colp	d5479a9ad7	Merge "app_queue: Handle the caller being redirected out of a queue bridge" into 14	8 years ago
zuul	04b13666a3	Merge "chan_iax2: Reload of iax peer results in loss of host address/port" into 14	8 years ago
zuul	6e38a72763	Merge "pbx.c: Fix crash from malformed exten pattern." into 14	8 years ago
Richard Mudgett	dcc2d8df0d	autochan/mixmonitor/chanspy: Fix unsafe channel locking and references. ... Dereferencing struct ast_autochan.chan without first calling ast_autochan_channel_lock() is unsafe because the pointer could change at any time due to a masquerade. Unfortunately, ast_autochan_channel_lock() itself uses struct ast_autochan.chan unsafely and can result in a deadlock if the original channel happens to get destroyed after a masquerade in addition to the pointer getting changed. The problem is more likely to happen with v11 and earlier because masquerades are used to optimize out local channels on those versions. However, it could still happen on newer versions if the channel is executing a dialplan application when the channel is transferred or redirected. In this situation a masquerade still must be used. * Added a lock to struct ast_autochan to safely be able to use ast_autochan.chan while trying to get the channel lock in ast_autochan_channel_lock(). The locking order is the channel lock then the autochan lock. Locking in the other direction requires deadlock avoidance. * Fix unsafe ast_autochan.chan usages in app_mixmonitor.c. * Fix unsafe ast_autochan.chan usages in app_chanspy.c. * app_chanspy.c: Removed unused autochan parameter from next_channel(). ASTERISK-26867 Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592	8 years ago
Joshua Colp	7e8cad893f	Merge "res_pjsip_endpoint_identifier_ip: Don't output error if no header_match." into 14	8 years ago
Joshua Colp	2fc0c228a2	Merge "configure: Don't use the progress bar with curl when downloading to stdout" into 14	8 years ago
Torrey Searle	90ffb91cc4	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
Mark Michelson	a7764b0c36	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
Sean Bright	e0767d7a6d	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	a6dddc4a32	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
Joshua Colp	77334fcd6c	Merge changes from topic 'ASTERISK-26863' into 14 ... * changes: res_pjsip_endpoint_identifier_ip: Add an option to match requests by header res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue	8 years ago
Joshua Colp	bafea29fbc	Merge "configs/samples/hep.conf.sample: Clarify how the HEP stack works" into 14	8 years ago
Joshua Colp	1c94b1de05	Merge "funcs/func_devstate: Remove new line in Device field of during module load" into 14	8 years ago
zuul	8cbd95b018	Merge "main/stasis_cache: Demote the ERROR message when removing a nonexistent item" into 14	8 years ago
Joshua Colp	2416cf62d5	Merge "res_pjsip_transport_websocket: Add support for IPv6." into 14	8 years ago