29218 Commits (d08c69a9e28d98d386971bd8527dec3926f0502b)

 

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Author	SHA1	Message	Date
Sean Bright	d08c69a9e2	res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts ... chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL (44) when a channel is hung up due to an RTP timeout. So do the same when it happens with PJSIP for parity. Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8	9 years ago
Joshua Colp	6666deb907	Merge "res_xmpp: Don't crash when trying to send a message without a connection"	9 years ago
zuul	1a626ffb89	Merge "res_xmpp: Correctly check return value of SSL_connect"	9 years ago
zuul	d3ca0e0bc4	Merge "res_xmpp: Try to provide useful errors messages from OpenSSL"	9 years ago
Joshua Colp	59df60efb9	Merge "audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor."	9 years ago
Kevin Harwell	d2f2cdf476	AMI: Updated version ... Updated the AMI version for the following reason (see CHANGES for more details): The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now contains a new optional parameter, 'MatchHeader'. Change-Id: Ie206913ef1dcfa6a2ebe3282da2387e52d6f05b9	9 years ago
Sean Bright	be94105d6d	res_xmpp: Try to provide useful errors messages from OpenSSL ... If any errors occur during the TLS connection setup, we currently dump a fairly generic error message. So instead we try to pull in something useful from OpenSSL to report instead. ASTERISK-24712 Reported by: Matthias Urlichs Change-Id: I288500991a9681f447d92913b11fedaf426087f4	9 years ago
Sean Bright	9493981419	res_xmpp: Correctly check return value of SSL_connect ... SSL_connect returns non-zero for both success and some error conditions so simply negating is inadequate. Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1	9 years ago
Sean Bright	7657c279b5	res_xmpp: Don't crash when trying to send a message without a connection ... If we never establish a connection to our Jabber server, iksemel never sets up its internal transport pointer, so attempting to send a message dereferences a NULL pointer and causes a crash. ASTERISK-21855 #close Reported by: Jeremy Kister Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c	9 years ago
Joshua Colp	c1ab8ca74c	Merge "res_pjsip_session: Enable RFC3578 overlap dialing support."	9 years ago
Joshua Colp	3a50311c17	Merge "CHANNEL(callid): Give dialplan access to the callid."	9 years ago
zuul	06c9966608	Merge "res_pjsip_messaging: Check URI type before dereferencing"	9 years ago
zuul	58b42297fd	Merge "Revert "app_queue: Handle the caller being redirected out of a queue bridge""	9 years ago
zuul	30f011a460	Merge "app_queue: Member stuck as pending after forwarding previous call from queue"	9 years ago
zuul	f83a334434	Merge "pjsip: prevent memory corruption on creation of xml bodies"	9 years ago
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	9 years ago
zuul	9f64980e60	Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references."	9 years ago
zuul	bd4a16da04	Merge "res_hep: Capture actual transport type in use"	9 years ago
Sean Bright	d4fcf196a2	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	9 years ago
Sean Bright	1bf839d44b	Revert "app_queue: Handle the caller being redirected out of a queue bridge" ... This reverts commit 163e9e53dc. Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b	9 years ago
Sean Bright	6b4b87787c	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	9 years ago
Joshua Elson	65ad554c98	pjsip: prevent memory corruption on creation of xml bodies ... ASTERISK-26776 #close Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2	9 years ago
Sean Bright	fc794de756	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	9 years ago
Joshua Colp	f5603cb1ec	Merge "res/res_pjsip_session: Only check localnet if it is defined"	9 years ago
Aaron An	25016a74f8	audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor. ... Fixed a bug in function "ast_audiohook_write_frame" that checked the variable other_factory_samples and only flushed the factories, so they would be in sync, when other_factory_samples > 0. When there is not any rtp incoming the variable other_factory_samples will be 0, and although the result of "our_factory_ms - other_factory_ms" may be very large, this led to the record file not syncing. ASTERISK-26875 #close Reported-by: Aaron An Tested-by: Aaron An Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22	9 years ago
zuul	24ada76408	Merge "thread safety: Don't use getprotobyname()"	9 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	9 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	9 years ago
Joshua Colp	b05d2fda0c	Merge "app_queue: Fix locking behavior in stasis message handlers"	9 years ago
Joshua Colp	3996c79328	Merge "chan_sip: Add rtcp-mux support"	9 years ago
Joshua Colp	77582634d7	Merge "res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped."	9 years ago
Joshua Colp	0db211dc64	Merge "res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed."	9 years ago
Joshua Colp	317405ce69	Merge "app_confbridge: Fix ConfbridgeTalking AMI event description."	9 years ago
Joshua Colp	7f87cd7b4e	Merge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error"	9 years ago
Joshua Colp	15c72b3239	Merge "res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport"	9 years ago
Robert Mordec	79069f8ccb	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	9 years ago
Richard Mudgett	8cb4f9cea1	CHANNEL(callid): Give dialplan access to the callid. ... * Added CHANNEL(callid) to retrieve the call identifier log tag associated with the channel. Dialplan now has access to the call log search key associated with the channel so it can be saved in case there is a problem with the call. ASTERISK-26878 Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f	9 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	9 years ago
Sean Bright	15aa3c0a23	chan_sip: Add rtcp-mux support ... ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	9 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	9 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	9 years ago
Joshua Colp	732367e806	Merge "res_pjsip: Symmetric transports"	9 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	9 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	9 years ago
Joshua Colp	76e64f5589	Merge "RFC sdp: Initial SDP creation"	9 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	9 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	9 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	9 years ago
Joshua Colp	84f0871cba	Merge "Add rtcp-mux support"	9 years ago
Joshua Colp	aec2a087ed	Merge "chan_iax2: Reload of iax peer results in loss of host address/port"	9 years ago