30374 Commits (731a23fba76d51cfef139895964d2dbe7af12afc)

 

All Branches   

Search

Author	SHA1	Message	Date
George Joseph	d178f497d2	res_pjsip: Filter out non SIP(S) requests ... Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460	8 years ago
Jenkins2	e9455d2264	Merge "chan_rtp: Use μ-law by default instead of signed linear"	8 years ago
Joshua Colp	ba27a6508a	Merge "tcptls: Change error message to debug."	8 years ago
Joshua Colp	01f2220bec	tcptls: Change error message to debug. ... The Websocket implementation will steal the underlying stream of TCP/TLS sessions. This results in an error message being output about a stream not being present when in reality this is actually fine. This change moves it to a debug message instead. Change-Id: I66cc639080b4b4599beadb4faa7d313f2721d094	8 years ago
Sean Bright	d8112cd98b	res_calendar: Various fixes ... * The way that we were looking at XML elements for CalDAV was extremely fragile, so use SAX2 for increased robustness. * Don't complain about a 'channel' not be specified if autoreminder is not set. Assume that if 'channel' is not set, we don't want to be notified. * Fix some truncated CLI output in 'calendar show calendar' and make the 'Autoreminder' description a bit more clear ASTERISK-24588 #close Reported by: Stefan Gofferje ASTERISK-25523 #close Reported by: Jesper Change-Id: I200d11afca6a47e7d97888f286977e2e69874b2c	8 years ago
Sean Bright	eec0396395	chan_rtp: Use μ-law by default instead of signed linear ... Multicast/Unicast RTP do not use SDP so we need to use a format that cleanly maps to one of the static RTP payload types. Without this change, an Originate to a Multicast or Unicast channel without a format specified would produce no audio on the receiving device. ASTERISK-21399 #close Reported by: Tzafrir Cohen Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3	8 years ago
George Joseph	446d48fd49	res_pjsip: Add handling for incoming unsolicited MWI NOTIFY ... A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c	8 years ago
Jenkins2	ec940f4fec	Merge "alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode"	8 years ago
Richard Mudgett	4889574ff5	res_rtp_asterisk.c: Add doxygen to RTCP payload types. ... Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998	8 years ago
Joshua Colp	619008f41c	Merge "cdr_pgsql: Refactor magic number by definition for version"	8 years ago
Jenkins2	97a7760b7a	Merge "alembic: Add support for MS-SQL"	8 years ago
George Joseph	f9bad3bd61	alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode ... The downgrade function was missing "_v2" at the end of the alter column type. Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e	8 years ago
Walter Doekes	680aba21ec	res/res_pjsip: Fix localnet checks in pjsip, part 2. ... In 45744fc53, I mistakenly broke SDP media address rewriting by misinterpreting which address was checked in the localnet comparison. Instead of checking the remote peer address to decide whether we need media address rewriting, we check our local media address: if it's local, then we rewrite. This feels awkward, but works and even made directmedia work properly if you set local_net. (For the record: for local peers, the SDP media rewrite code is not called, so the comparison does no harm there.) ASTERISK-27248 #close Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f	8 years ago
Rodrigo Ramírez Norambuena	c8d53a1638	cdr_pgsql: Refactor magic number by definition for version ... Change-Id: I43f25976aa3069793ddbe0086833965a6fb0a518	8 years ago
Florian Floimair	e9a81157ac	alembic: Add support for MS-SQL ... MS-SQL has no native Enum-type support and therefore needs to work with constraints. Since these constraints need unique names the suggested approach referenced in the following alembic documentation has been applied: http://bit.ly/2x9r8pb ASTERISK-27255 #close Change-Id: I8b579750dae0c549f1103ee50172644afb9b2f95	8 years ago
Jenkins2	68b506caaa	Merge "chan_sip: when getting sip pvt return failure if not found"	8 years ago
Jenkins2	e40be6c3ce	Merge "app_waitforsilence: Cleanup & don't treat missing frames as 'noise'"	8 years ago
Joshua Colp	973885df0d	Merge "res_srtp: Add support for libsrtp2.1."	8 years ago
Jenkins2	b294a48a08	Merge "chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE"	8 years ago
Jenkins2	bee342fdd2	Merge "res_pjsip_session: Preserve stream name during renegotiation."	8 years ago
Jenkins2	fba2f65027	Merge "func_cdr: honour 'u' flag on dummy channel"	8 years ago
Jenkins2	2f03e11b0e	Merge "stasis/control.c: Fix set_interval_hook() ref leak."	8 years ago
Jacek Konieczny	525f84bb35	func_cdr: honour 'u' flag on dummy channel ... Fixes ${CDR(...,u)} when used in cdr_custom.conf ASTERISK-27165 #close Change-Id: Ia4e0b6ba93e03d27886354c279737790e2cd6a83	8 years ago
Sean Bright	2b3f903e6f	app_waitforsilence: Cleanup & don't treat missing frames as 'noise' ... * WaitForSilence completes successfully if it receives no media in the specified timeout, but when acting as WaitForNoise that logic needs to be reversed. * Use standard argument parsing macros and add some error checking for invalid values. * The documentation indicated that the first argument to both WaitForSilence and WaitForNoise was required when it was not. Update the documentation to reflect that. * Wrap up some behavior in structs to avoid boolean checks all over the place. ASTERISK-24066 #close Reported by: M vd S Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9	8 years ago
Scott Griepentrog	5553644284	chan_sip: when getting sip pvt return failure if not found ... In handle_request_invite, when processing a pickup, a call is made to get_sip_pvt_from_replaces to locate the pvt for the subscription. The pvt is assumed to be valid when zero is returned indicating no error, and is dereferenced which can cause a crash if it was not found. This change checks the not found case and returns -1 which allows the calling code to fail appropriately. ASTERISK-27217 #close Reported-by: Bryan Walters Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612	8 years ago
Richard Mudgett	23571f31ac	stasis/control.c: Fix set_interval_hook() ref leak. ... Change-Id: Ia0edb7dc0dbbb879c079ff7000f1b722d86ce7dc	8 years ago
George Joseph	94091c7b96	stasis/control: Fix possible deadlock with swap channel ... If an error occurs during a bridge impart it's possible that the "bridge_after" callback might try to run before control_swap_channel_in_bridge has been signalled to continue. Since control_swap_channel_in_bridge is holding the control lock and the callback needs it, a deadlock will occur. * control_swap_channel_in_bridge now only holds the control lock while it's actually modifying the control structure and releases it while the bridge impart is running. * bridge_after_cb is now tolerant of impart failures. Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3	8 years ago
George Joseph	2857a3334a	Merge "alembic: Fix enum creation for dtls_fingerprint"	8 years ago
Jenkins2	6c89ffad2a	Merge "alembic: fix erroneous commit for add_prune_on_boot"	8 years ago
Jenkins2	23f22a3647	Merge "res/res_pjsip: Standardize/fix localnet checks across pjsip."	8 years ago
Vitezslav Novy	67a2ca31f5	chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE ... If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE to both parties to set up media path directly between the endpoints. In this reINVITE msg SDP origin line (o=) contains IP address of endpoint instead of IP of asterisk. This behavior violates RFC3264, sec 8: "When issuing an offer that modifies the session, the "o=" line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP." This patch assures IP address of Asterisk is always sent in SDP origin line. ASTERISK-17540 Reported by: saghul Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e	8 years ago
Joshua Colp	3025b47e8f	Merge "formats: Restore previous fread() behavior"	8 years ago
George Joseph	0cbb17ce8f	alembic: Fix enum creation for dtls_fingerprint ... Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db	8 years ago
Jenkins2	4b606c25e3	Merge "res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel"	8 years ago
Jenkins2	9ec3a38c84	Merge "res_calendar*, res_smdi: Move to "extended" support"	8 years ago
Florian Floimair	a133c5cc53	alembic: fix erroneous commit for add_prune_on_boot ... Added include for postgresql ENUM type and redefined values in the same way as in the other migration scripts. ASTERISK-27254 #close Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa	8 years ago
Alexander Traud	2d395793b7	res_srtp: Add support for libsrtp2.1. ... Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with previous versions of libSRTP, this change allows libSRTP 2.1.x as well. ASTERISK-27253 #close Change-Id: I2e6eb3c3bc844fee8a624060a2eb6f182dc70315	8 years ago
Ben Ford	bfc29de3ea	chan_pjsip: Suppress frame warnings. ... When rtp_keepalive is on for a PJSIP endpoint dialing to another Asterisk instance also using PJSIP, Asterisk will continue to print warning messages about not being able to send frames of a certain type. This suppresses that warning message. Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67	8 years ago
Sean Bright	c3a6c8fd2d	formats: Restore previous fread() behavior ... Some formats are able to handle short reads while others are not, so restore the previous behavior for the format modules so that we don't have spurious errors when playing back files. ASTERISK-27232 #close Reported by: Jens T. Change-Id: Iab7f52b25a394f277566c8a2a4b15a692280a300	8 years ago
Walter Doekes	f856d9b42b	res/res_pjsip: Standardize/fix localnet checks across pjsip. ... In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was confusion about whether the transport_state->localnet ACL has ALLOW or DENY semantics. For the record: the localnet has DENY semantics, meaning that "not in the list" means ALLOW, and the local nets are in the list. Therefore, checks like this look wrong, but are right: /* See if where we are sending this request is local or not, and if not that we can get a Contact URI to modify */ if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) { ast_debug(5, "Request is being sent to local address, " "skipping NAT manipulation\n"); (In the list == localnet == DENY == skip NAT manipulation.) And conversely, other checks that looked right, were wrong. This change adds two macro's to reduce the confusion and uses those instead: ast_sip_transport_is_nonlocal(transport_state, addr) ast_sip_transport_is_local(transport_state, addr) ASTERISK-27248 #close Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934	8 years ago
Joshua Colp	f556c31aea	Merge "app_directory: Handle a NULL mailbox without crashing"	8 years ago
Joshua Colp	68bcfccd52	res_pjsip_session: Preserve stream name during renegotiation. ... Stream names within Asterisk can have meaning so when an externally initiated renegotiation occurs we need to preserve the name of the stream if it already exists. Change-Id: I29f50d0cc7f3238287d6d647777e76e1bdf8c596	8 years ago
George Joseph	0ec95515f3	res_calendar*, res_smdi: Move to "extended" support ... Change-Id: I31eee8be30c6b0fc3dadb31111dd47742da8892d	8 years ago
Joshua Colp	f40b2901fc	Merge "chan_ooh323: Fix confusing indentation warning"	8 years ago
George Joseph	9b3f6d26bd	res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel ... t38_reinvite_response_cb can get called by res_pjsip_session's session_inv_on_tsx_state_changed in situations where session->channel is NULL. If it is, the ast_log warning segfaults because it tries to get the channel name from a NULL channel. * Check session->channel and print "unknown channel" when it's NULL. ASTERISK-27236 Reported by: Ross Beer Change-Id: I4326e288d36327f6c79ab52226d54905cdc87dc7	8 years ago
Sean Bright	60b44d1e38	rtp_engine: Prevent possible double free with DTLS config ... ASTERISK-27225 #close Reported by: Richard Kenner Change-Id: I097b81734ef730f8603c0b972909d212a3a5cf89	8 years ago
Sean Bright	ef8eb9d11b	chan_ooh323: Fix confusing indentation warning ... ASTERISK-27177 #close Reported by: Tzafrir Cohen Change-Id: I40311c404edb2302a7543ad5ca7a06b2a38f2d97	8 years ago
Sean Bright	1bdbefbe76	app_directory: Handle a NULL mailbox without crashing ... ASTERISK-27241 #close Reported by: David Moore Change-Id: Ibbbca85517b04c315406ebfe3b6f7e0763daedc6	8 years ago
Jenkins2	991750d3f1	Merge "chan_pjsip: Add tag info in CHANNEL function"	8 years ago
Joshua Colp	be8639f81b	Merge "res_rtp_asterisk: Allow remote SSRC to change on an RTP instance."	8 years ago