3828 Commits (31268e0a280110748f33314a2c09563c576243de)

Author	SHA1	Message	Date
Joshua Colp	2a4b24cc14	Merge "Fix IO conversion bug"	9 years ago
Joshua Colp	3fcc35c66c	Merge "res_pjsip: Fix 'A = B != C' kind."	9 years ago
Joshua Colp	cc9b30dd27	Merge "res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command"	9 years ago
Badalyan Vyacheslav	934aa2c768	res_pjsip: Fix 'A = B != C' kind. ... Consider reviewing the expression of the 'A = B != C' kind. The expression is calculated as following: 'A = (B != C)' Change-Id: Ibaa637dfda47d51a20e26069d3103e05ce80003d	9 years ago
Badalyan Vyacheslav	149d8db96c	Fix IO conversion bug ... Expression 'rlen < 0' is always false. Unsigned type value is never < 0. Change-Id: Id9f393ff25b009a6c4a6e40b95f561a9369e4585	9 years ago
Joshua Colp	5c89604a32	res_format_attr_opus: Fix crash when fmtp contains spaces. ... When an opus offer or answer was received that contained an fmtp line with spaces between the attributes the module would fail to properly parse it and crash due to recursion. This change makes the module handle the space properly and also removes the recursion requirement. ASTERISK-26579 Change-Id: I01f53e5d9fa9f1925a7365f8d25071b5b3ac2dc3	9 years ago
George Joseph	79b09b5f18	res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command ... The PJSIPShowRegistrationsInbound AMI command was just dumping out all AORs which was pretty useless and resource heavy since it had to get all endpoints, then all aors for each endpoint, then all contacts for each aor. PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail events which meets the intended purpose of the other command and has significantly less overhead. Also, some additional fields that were added to Contact since the original creation of the ContactStatusDetail event have been added to the end of the event. For compatibility purposes, PJSIPShowRegistrationsInbound is left intact. ASTERISK-26644 #close Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a	9 years ago
zuul	8bd9dc568a	Merge "res_pjsip_outbound_registration.c: Filter redundant statsd reporting."	9 years ago
Joshua Colp	2a415187c5	Merge "res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter"	9 years ago
Richard Mudgett	4b3d3fc741	res_pjsip_outbound_registration.c: Filter redundant statsd reporting. ... Increasing the testsuite shutdown timeout before forcibly killing Asterisk allowed more events to be sent out. Some tests failed as a result. The tests/channels/pjsip/statsd/registrations failed because we now get the statsd events that a comment in the test configuration stated couldn't be intercepted. Unfortunately, we get a variable number of events because of internal status state transition races generating redundant statsd events. We were reporting redundant statsd PJSIP.registrations.state changes for internal state changes that equated to the same thing publicly. * Made update_client_state_status() filter out redundant statsd updates. ASTERISK-26527 Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646	9 years ago
Joshua Colp	197e408395	Merge "PJPROJECT logging: Made easier to get available logging levels."	9 years ago
Joshua Colp	2679f80d3c	Merge "res_rtp: Fix regression when IPv6 is not available."	9 years ago
Joshua Colp	29596f1538	Merge "res_calendar_caldav: Add support reading gmail calendar"	9 years ago
Guido Falsi	75230f4c01	res_rtp: Fix regression when IPv6 is not available. ... The latest Release candidate fails to create RTP streams when IPv6 is not available. Due to the changes made in September the ast_sockaddr structure passed around to create these streams is always of AF_INET6 type, causing failure when used for IPv4. This patch adds a utility function to check for availability of IPv6 and applies such check at startup to determine how to create the ast_sockaddr structures. ASTERISK-26617 #close Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e	9 years ago
Richard Mudgett	1dfa11b65c	PJPROJECT logging: Made easier to get available logging levels. ... Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389	9 years ago
Eduardo S. Libardi	0e214c4932	res_calendar_caldav: Add support reading gmail calendar ... The response from gmail calendar includes the string name "caldav:calendar-data". res_calendar_caldav implements the example included in RFC 4791: string "C:calendar-data". When reading the calendar, res_calendar_caldav compare the string and if does not match just discards the event. This commit compares the response to both strings, successfully loading gmail calendar events. Writing to gmail calendar is working prior to this fix. ASTERISK-26624 Reported by: Eduardo S. Libardi Change-Id: Ia1eef10552ae616efb645d390f5ffe81260d7d4a	9 years ago
Matt Jordan	a3f48be0da	res/res_pjsip: Fix documentation whitespace issues ... Tabs > Spaces. Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0	9 years ago
Matt Jordan	0e15760795	res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter ... Per RFC 7118 5.2, the SIP URI 'transport' parameter should advertise 'ws' when WebSockets are to be used as the transport. This applies to both secure and insecure WebSockets. There were two bugs in Asterisk with respect to this: (1) The most egregious occurs in res_pjsip. There, we advertise 'ws' for insecure websockets and 'wss' for secure websockets. While this would seem to make sense - since 'WS' and 'WSS' are used for the Via Transport parameter - this is not the case for the SIP URI. This patch corrects that by registering the secure websockets with pjproject using the shorthand 'WS', and by returning 'ws' when asked for the transport parameter. Note that in pjproject, it is perfectly valid to have multiple transports use the same shorthand. (2) In chan_sip, we return an upper-case version of the transport 'WS' instead of 'ws'. Since we should be strict in what we send and liberal in what we accept (within reason), this patch lower-cases the transport before appending it to the parameter. ASTERISK-24330 #close Reported by: cervajs, Inaki Baz Castillo Change-Id: Iff77b645f8cc3b7cd35168a6676c26b147f22f42	9 years ago
Joshua Colp	c9cc64b911	Merge "ast_format: Adds an identifier for interleaved audio formats to the ast_format"	9 years ago
gestoip2	d9b24cce0a	res_rtp_asterisk: RTT miscalculation in RTCP ... When retrieving RTCP stats for PJSIP channels, RTT values are unreliable. RTT calculation is correct, but the data representation isn't. RTT is represented by a 32-bit fixed-point number with the integer part in the first 16 bits and the fractional part in the last 16 bits. In order to get the RTT value, the fractional part is miscalculated, there is an unnecessary 16 bit shift that causes overflow. Besides this there is another mistake, when transforming the integer value to the fixed point fractional part via bitwise operation, that loses precision. * RTT fractional part is no longer shifted, avoiding overflow. * RTT fractional part is transformed to its fixed-point value more precisely. * Fixed timeval2ntp() and ntp2timeval() second fraction conversions. * Fixed NTP timestamp report logging. The usec was inexplicably multiplied by 4096. ASTERISK-26566 #close Reported by Hector Royo Concepcion Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f	9 years ago
snuffy	b546497fe0	Add support for older name resolving version libraries like openBSD ... Fix support of OS's like openBSD that use an older nameser.h, this change reverts the defines to the older style which on other systems is found in nameser_compat.h Tested on openBSD 6.0, Debian 8 ASTERISK-26608 #close Change-Id: Iffb36caab8c5aa9dece0ce2d009041f7b56cc86a	9 years ago
zuul	782985631e	Merge "build: Various OpenBSD issues"	9 years ago
Joshua Colp	d3dba74017	Merge "Implement internal abstraction for iostreams"	9 years ago
Joshua Colp	ed143a3b7c	Merge "res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak."	9 years ago
zuul	8fbdedb36e	Merge "res_format_attr_opus: Fix fmtp generation."	9 years ago
George Joseph	935f5d003b	build: Various OpenBSD issues ... OpenBSD's 'find' doesn't take the -delete argument so you have to pipe through 'xargs rm -rf'. 'echo -e' doesn't like \t starting a line. It just prints 't' which causes the libasteriskpj.exports file to be garbage. They were just cosmetic so they were removed. librt doesn't exist so the link of libasteriskpj.so fails. It's not actually needed for linux anyway so -lrt was removed from the link. res_rtp_asterisk was failing to load because of an undefined DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if so DTLSv1_method is used instead. ASTERISK-26608 Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c	9 years ago
George Joseph	013e7dd4a6	Merge "Revert "Revert "AGI: Only defer frames when in an interception routine."""	9 years ago
zuul	d0474f6322	Merge "res/ari/resource_bridges: Add the ability to manipulate the video source"	9 years ago
Mark Michelson	dc8f99ee27	res_format_attr_opus: Fix fmtp generation. ... res_format_attr_opus assumed that the string being passed into it was empty. It tried to determine if the only thing it had written was a=fmtp:<num> And if it had, it would reset the string. Its calculation was off when working with chan_sip, though. chan_sip passes the entire built SDP rather than an empty string. This resulted in always putting an empty fmtp line in the SDP. ASTERISK-26520 #close Reported by scgm11 Change-Id: Ib2e8712d26a47067e5f36d5973577added01dbb5	9 years ago
Richard Mudgett	0cd0e70c16	res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak. ... Responding to authentication challenges leaks PJSIP memory pools. The leak was introduced with a pjproject 2.5.5 API change. https://trac.pjsip.org/repos/ticket/1929 changed the API usage of pjsip_auth_clt_init() to require the new API pjsip_auth_clt_deinit() to clean up cached authentication allocations that get allocated with pjsip_auth_clt_reinit_req(). ASTERISK-26516 #close Change-Id: I4473141b8c3961d0dc91c382beb3876b3efb45c8	9 years ago
Joshua Colp	ceefe483cf	Merge "res_ari: Add support for channel variables in ARI events."	9 years ago
Timo Teräs	070a51bf7c	Implement internal abstraction for iostreams ... fopencookie/funclose is a non-standard API and should not be used in portable software. Additionally, the way FILE's fd is used in non-blocking mode is undefined behaviour and cannot be relied on. This introduces internal abstraction for io streams, that allows implementing the desired virtualization of read/write operations with necessary timeout handling. ASTERISK-24515 #close ASTERISK-24517 #close Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85	9 years ago
Matt Jordan	a72ef38113	res/ari/resource_bridges: Add the ability to manipulate the video source ... In multi-party bridges, Asterisk currently supports two video modes: * Follow the talker, in which the speaker with the most energy is shown to all participants but the speaker, and the speaker sees the previous video source * Explicitly set video sources, in which all participants see a locked video source Prior to this patch, ARI had no ability to manipulate the video source. This isn't important for two-party bridges, in which Asterisk merely relays the video between the participants. However, in a multi-party bridge, it can be advantageous to allow an external application to manipulate the video source. This patch provides two new routes to accomplish this: (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} Sets a video source to an explicit channel (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource Removes any explicit video source, and sets the video mode to talk detection ASTERISK-26595 #close Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621	9 years ago
George Joseph	f62c9c42fa	Revert "Revert "AGI: Only defer frames when in an interception routine."" ... This reverts commit 6bce938c2f. Change-Id: Iadbf462bf2a52e8b2fa9ebc75b37b1f688ba51d9	9 years ago
Joshua Colp	d1739bcf07	Merge "res_pjsip.c: Rework endpt_send_request() req_wrapper code."	9 years ago
Sebastien Duthil	c6d755de11	res_ari: Add support for channel variables in ARI events. ... This works the same as for AMI manager variables. Set "channelvars=foo,bar" in your ari.conf general section, and then the channel variables "foo" and "bar" (along with their values), will appear in every Stasis websocket channel event. ASTERISK-26492 #close patches: ari_vars.diff submitted by Mark Michelson Change-Id: I5609ba239259577c0948645df776d7f3bc864229	9 years ago
Joshua Colp	7540036427	Merge "res_pjsip: Fix tdata leaks in off nominal paths."	9 years ago
Joshua Colp	1bd49040c4	res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp. ... When optimistic SRTP was on it was possible for us to still set up a call without an audio stream if an offer was received with required SRTP. This change makes it so this scenario will now fail with a 488 response. ASTERISK-26575 Change-Id: I7d14187037681f48879bd20319ac79d0877318f3	9 years ago
Joshua Colp	e77aa78dc7	Merge "res_pjsip: Perform resolution when explicit IPv6 transport is used."	9 years ago
Richard Mudgett	338f35edcc	res_pjsip.c: Rework endpt_send_request() req_wrapper code. ... * Don't hold the req_wrapper lock too long in endpt_send_request(). We could block the PJSIP monitor thread if the timeout timer expires. sip_get_tpselector_from_endpoint() does a sorcery access that could take awhile accessing a database. pjsip_endpt_send_request() might take awhile if selecting a transport. * Shorten the time that the req_wrapper lock is held in the callback functions. * Simplify endpt_send_request() req_wrapper->timeout code. * Removed some redundant req_wrapper->timeout_timer->id assignments. Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9	9 years ago
Richard Mudgett	bb196323f9	res_pjsip: Fix tdata leaks in off nominal paths. ... Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b	9 years ago
Richard Mudgett	9df59d9ff4	res_pjsip_registrar_expire.c: Remove extra linefeed in debug message. ... Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94	9 years ago
Joshua Colp	bdb6d928c5	res_pjsip: Perform resolution when explicit IPv6 transport is used. ... This change fixes the SIP resolver such that if an IPv6 transport is explicitly used it will resolve NAPTR, SRV, and AAAA records. You can explicitly use one by specifying it on an endpoint. ASTERISK-26571 Change-Id: I2ed3ce81b43a6a8a937c0ebc1b8ed2da5ac2ef36	9 years ago
George Joseph	7efac7b3d0	Merge "Revert "AGI: Only defer frames when in an interception routine.""	9 years ago
George Joseph	6bce938c2f	Revert "AGI: Only defer frames when in an interception routine." ... This reverts commit 28926d1c81. Multiple testsuite failures were detected after the fact. Change-Id: I8d4f5ccbb421a351d616254844ae7e5a31053edb	9 years ago
Mark Michelson	e5860ce07d	res_pjsip_session: Do not call session supplements when it's too late. ... res_pjsip_sesssion was hooking into transaction and invite state changes. One of the reasons for doing so was due to the PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the message sending process, and so we should call session supplements to alter the outgoing message. In reality, this event was meant to indicate that the message either a) had already been sent, or b) required a DNS lookup and would be sent when the DNS query completed. In case (a), this meant we were altering an already-sent request/response for no reason. In case (b), this potentially meant we could be trying to alter a request/response at the same time that the DNS resolution completed. In this case, it meant we might be stomping on memory being used by the thread actually sending the message. This caused potential crashes and memory corruption. This patch removes the calls to session supplements from the case where the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to alter the message at this point is too late, and it can cause nothing but harm to try to do it. Because there were no longer any calls to the handle_outgoing() function, it has been removed. Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92	9 years ago
Mark Michelson	28926d1c81	AGI: Only defer frames when in an interception routine. ... AGI recently was modified to defer important frames. This was because when AGI was used in a connected line interception routine, the resulting connected line frame would end up getting discarded by the AGI. However, this caused bad behavior in other cases. Specifically, during a transfer, if someone attempted to manually set the Caller ID on a channel in an AGI, the deferred connected line frame would end up overwriting what had been manually set in the AGI. Since the initial issue was specific to interception routines, this change removes the manual frame deferral from AGI and instead uses the new frame deferral API in interception routines. ASTERISK-26343 #close Reported by Morton Tryfoss Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208	9 years ago
Joshua Colp	61af0e6704	Merge "res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems"	9 years ago
Joshua Colp	3019dfd49f	Merge "res_stasis: Don't unsubscribe from a NULL bridge."	9 years ago
Joshua Colp	1ab943a425	Merge "stasis_recording/stored: remove calls to deprecated readdir_r function."	9 years ago