3910 Commits (d4ccd3a6c088f4d2fefda1a44e4555dd54335cda)

Author	SHA1	Message	Date
Jenkins2	d4ccd3a6c0	Merge "asterisk: Audit locking of channel when manipulating flags." into 13	9 years ago
Jenkins2	5715360ba5	Merge "res_agi: Fix malformed AGI usage response" into 13	9 years ago
George Joseph	a8f8c5d857	Merge "res_agi: Allow configuration of audio format of EAGI pipe" into 13	9 years ago
Jenkins2	558199e5dd	Merge "res_agi: Prevent crash when SET VARIABLE called without arguments" into 13	9 years ago
Jenkins2	a3684b74e6	Merge "res_agi: Clarify 'RECORD FILE' documentation" into 13	9 years ago
Jenkins2	a69905af69	Merge "res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm" into 13	9 years ago
Sean Bright	90237dca11	res_agi: Allow configuration of audio format of EAGI pipe ... This change allows the format of the EAGI audio pipe to be changed by setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of the loaded formats. ASTERISK-26124 #close Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd	9 years ago
Sean Bright	3eb7fbba72	res_agi: Clarify 'RECORD FILE' documentation ... Documented the 'beep' option in both the parameters list and the command description. ASTERISK-23839 #close Change-Id: I4970395c922dbdce3f7cf0f56d5b065ec9aa53ea	9 years ago
Sean Bright	f306e451f6	res_agi: Prevent crash when SET VARIABLE called without arguments ... Explicitly check that the appropriate number of arguments were passed to SET VARIABLE before attempting to reference them. Also initialize the arguments array to zeroes before populating it. ASTERISK-22432 #close Change-Id: I5143607d80a2724f749c1674f3126b04ed32ea97	9 years ago
Sean Bright	a007e438c3	res_agi: Fix malformed AGI usage response ... If the generated XML documentation for a command does not end with a \n, the postamble of the usage message does not appear on its own line. ASTERISK-25662 #close Change-Id: If190f1e9e37fe215fed95897d78d4a6e142b0020	9 years ago
Sean Bright	700ef6861a	res_format_attr_h26x: Trim blanks in fmtp attributes ... Some devices separate format attributes with a semicolon followed by a space, so trim blanks before trying to match them. ASTERISK-27008 #close Change-Id: Ia44cb2e4fef5c73dc541a29da79cb0e19c22d9cc	9 years ago
Kevin Harwell	e91efef2bb	res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm ... When using rtcp mux if an rtcp payload came in it would still use the srtp unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp data was being passed to the rtp unprotect method this would result in an error. This patch ensures that the correct unprotect method is chosen by making sure the passed in rtcp flag is appropriately set when rtcp mux is enabled and an rtcp payload is received. ASTERISK-26979 #close Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241	9 years ago
Yasin CANER	36628cc9c4	res_pjsip_session : fixed wrong From Header number On Re-invite ... ASTERISK-26964 #close Change-Id: I55a9caa7dc90e6c4c219cb09b5c2ec08af84a302	9 years ago
Sean Bright	c107ab4c04	res_hep_rtcp: Add support level to module info ... Change-Id: I5661478f9cf12d431f730e42be79323b62831e92	9 years ago
Joshua Colp	1618203964	asterisk: Audit locking of channel when manipulating flags. ... When manipulating flags on a channel the channel has to be locked to guarantee that nothing else is also manipulating the flags. This change introduces locking where necessary to guarantee this. It also adds helper functions that manipulate channel flags and lock to reduce repeated code. ASTERISK-26789 Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10	9 years ago
Richard Mudgett	b67363006f	res_pjsip_session.c: Process initial INVITE sooner. (key exists) ... Retransmissions of an initial INVITE could be queued in the serializer before we have processed the first INVITE message. If the first INVITE message doesn't get completely processed before the retransmissions are seen then we could try to setup the same call from the retransmissions. A symptom of this is seeing a (key exists) message associated with an INVITE. An earlier change attempted to address this kind of problem by calculating a distributor serializer to use for unassociated messages. Part of that change also made incoming calls keep using that distributor serializer. (ASTERISK-26088) However, some leftover code was still deferring the INVITE processing to the session's serializer even though we were already in that serializer. This not only is unnecessary but would cause the same call resetup problem. * Removed the code to defer processing the initial INVITE to the session's serializer because we are already running in that serializer. ASTERISK-26998 #close Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6	9 years ago
Joshua Colp	10a49ab362	res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages. ... This change adds the required logic to allow the SIP Call-ID to be placed into the HEP RTCP traffic if the chan_sip module is used. In cases where the option is enabled but the channel is not either SIP or PJSIP then the code will fallback to the channel name as done previously. Based on the change on Nir's branch at: team/nirs/hep-chan-sip-support ASTERISK-26427 Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d	9 years ago
Joshua Colp	ce1985b099	Merge "res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures" into 13	9 years ago
Kevin Harwell	bbe90d6aed	res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures ... When a call gets put on hold RTP is temporarily stopped and Asterisk was setting the remote RTCP address to NULL. Then when RTCP data was received from the remote endpoint, Asterisk would be missing this information when publishing the rtcp_message stasis event. Consequently, message subscribers (in this case res_hep_rtcp) trying to parse the "from" field output the following error: "ast_sockaddr_split_hostport: Port missing in (null)" This patch makes it so the remote RTCP address is no longer set to NULL when stopping RTP. There was only one place that appeared to check if the remote RTCP address was NULL as a way to tell if RTCP was running. This patch added an additional check on the RTCP schedid for that case to make sure RTCP was truly not running. ASTERISK-26860 #close Change-Id: I6be200fb20db647e48b5138ea4b81dfa7962974b	9 years ago
Jenkins2	20cc154dd3	Merge "res_pjsip_t38.c: Fix deadlock in T.38 framehook." into 13	9 years ago
Richard Mudgett	9d5df48968	res_pjsip_t38.c: Fix deadlock in T.38 framehook. ... A deadlock can happen between a channel lock and a pjsip session media container lock. One thread is processing a reINVITE's SDP and walking through the session's media container when it waits for the channel lock to put the determined format capabilities onto the channel. The other thread is writing a frame to the channel and processing the T.38 frame hook. The T.38 frame hook then waits for the pjsip session's media container lock. The two threads are now deadlocked. * Made the T.38 frame hook release the channel lock before searching the session's media container. This fix has been done to several other frame hooks to fix deadlocks. ASTERISK-26974 #close Change-Id: Ie984a76ce00bef6ec9aa239010e51e8dd74c8186	9 years ago
George Joseph	623832b94e	res_pjsip_outbound_authenticator_digest: Add context to log messages ... There was no context info in this module's log messages so it was impossible to toubleshoot. Added endpoint or host to all messages and added the realms in the challenge for the "No auth credentials for any realm" message. Change-Id: Ifeed2786f35fbea7d141237ae15625e472acff9b	9 years ago
Jenkins2	9bb683242c	Merge "res_pjsip_session: Add cleanup to ast_sip_session_terminate" into 13	9 years ago
Jenkins2	dc7166e59f	Merge "res_pjsip/res_pjsip_callerid: NULL check on caller id name string" into 13	9 years ago
Joshua Colp	5e3e309c82	Merge "res_rtp_asterisk.c: Fix crash in RTCP DTLS operation." into 13	9 years ago
George Joseph	c5b9ed20fd	res_pjsip_session: Add cleanup to ast_sip_session_terminate ... If you use ast_request to create a PJSIP channel but then hang it up without causing a transaction to be sent, the session will never be destroyed. This is due ot the fact that it's pjproject that triggers the session cleanup when the transaction ends. app_chanisavail was doing this to get more granular channel state and it's also possible for this to happen via ARI. * ast_sip_session_terminate was modified to explicitly call the cleanup tasks and unreference session if the invite state is NULL AND invite_tsx is NULL (meaning we never sent a transaction). * chan_pjsip/hangup was modified to bump session before it calls ast_sip_session_terminate to insure that session stays valid while it does its own cleanup. * Added test events to session_destructor for a future testsuite test. ASTERISK-26908 #close Reported-by: Richard Mudgett Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9	9 years ago
Kevin Harwell	c853cfdc7c	res_pjsip/res_pjsip_callerid: NULL check on caller id name string ... It's possible for a name in a party id structure to be marked as valid, but the name string itself be NULL (for instance this is possible to do by using the dialplan CALLERID function). There were a couple of places where the name was validated, but the string itself was not checked before passing it to functions like 'strlen'. This of course caused a crashed. This patch adds in a NULL check before attempting to pass it into a function that is not NULL tolerant. ASTERISK-25823 #close Change-Id: Iaa6ffe9d92f598fe9e3c8ae373fadbe3dfbf1d4a	9 years ago
George Joseph	95313c5f9e	Merge "res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions." into 13	9 years ago
George Joseph	f9369143db	Merge "res_hep: Add additional config initialization and validation" into 13	9 years ago
Sean Bright	1b88a3a4cf	res_hep: Add additional config initialization and validation ... * Initialize hepv3_runtime_data.sockfd to -1 so that our ao2 destructor does not close fd 0 * Add logging output when the required option - capture_address - is not specified. * Remove a no longer relevant #define and correct related documentation * Pass appropriate flags to aco_option_register so that capture_address cannot be the empty string. ASTERISK-26953 #close Change-Id: Ief08441bc6596d6f1718fa810e54a5048124f076	9 years ago
Richard Mudgett	1213ac1ac5	res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions. ... If ICE is enabled and a STUN server does not respond then we will block until we give up on the STUN response. This will take nine seconds. In the mean time the peer that sent the INVITE will send retransmissions. * Restructure res_pjsip_session.c:new_invite() to send a 100 Trying out earlier to prevent these retransmissions. ASTERISK-26890 Change-Id: Ie3fc611e53a0eff6586ad55e4aacad81cf6319a8	9 years ago
Richard Mudgett	80fd7fd908	res_pjsip_session.c: Restructure ast_sip_session_alloc() ... * Restructure ast_sip_session_alloc() to need less cleanup on off nominal error paths. * Made ast_sip_session_alloc() and ast_sip_session_create_outgoing() avoid unnecessary ref manipulation to return a session. This is faster than calling a function. That function may do logging of the ref changes with REF_DEBUG enabled. Change-Id: I2a0affc4be51013d3f0485782c96b8fee3ddb00a	9 years ago
George Joseph	5860ad3ee8	Merge "rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes." into 13	9 years ago
George Joseph	98bb64f88a	Merge "res_stun_monitor: Don't fail to load if DNS resolution fails" into 13	9 years ago
Richard Mudgett	55f452884f	res_rtp_asterisk.c: Fix crash in RTCP DTLS operation. ... Occasionally a crash happens when processing the RTCP DTLS timeout handler. The RTCP DTLS timeout timer could be left running if we have not completed the DTLS handshake before we place the call on hold or we attempt direct media. * Made ast_rtp_prop_set() stop the RTCP DTLS timer when disabling RTCP. * Made some sanity tweaks to ast_rtp_prop_set() when switching from standard RTCP mode to RTCP multiplexed mode. ASTERISK-26692 #close Change-Id: If6c64c79129961acfa4b3d63a864e8f6b664acc0	9 years ago
Richard Mudgett	f856cfbb51	rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes. ... The struct ast_rtp_instance has historically been indirectly protected from reentrancy issues by the channel lock because early channel drivers held the lock for really long times. Holding the channel lock for such a long time has caused many deadlock problems in the past. Along comes chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock because sometimes there may not be an associated channel created yet or the channel pointer isn't available. In the case of ASTERISK-26835 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up changing the rtp->rtcp->local_addr_str string and interfering with each other. The classic reentrancy problem resulted in a crash. In the case of ASTERISK-26853 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up processing ICE candidates in PJPROJECT and interfering with each other. The classic reentrancy problem resulted in a crash. * rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP instance struct. * rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP instance struct for the API call. * res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy problem with rtp->rtcp->local_addr_str in the scheduler thread running ast_rtcp_write(). * res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in bridge_p2p_rtp_write() because there are two RTP instance structs involved. * res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler callbacks. We cannot hold the instance lock when trying to stop a scheduler callback. * res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads to prevent a race condition between starting and stopping a timeout timer. The race condition is no longer present between dtls_perform_handshake() and __rtp_recvfrom() because the instance lock prevents these functions from overlapping each other with regards to the timeout timer. * res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads using a condition signal to know when TURN negotiations complete. * res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN ioqueue_worker_thread(). We cannot hold the instance lock when trying to create or shut down the worker thread without a risk of deadlock. This patch exposed a race condition between a PJSIP serializer thread setting up an ICE session in ice_create() and another thread reading RTP packets. * res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we have re-locked the RTP instance to prevent the other thread from trying to process ICE packets on an incomplete ICE session setup. A similar race condition is between a PJSIP serializer thread resetting up an ICE session in ice_create() and the timer_worker_thread() processing the completion of the previous ICE session. * res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an uninitialized/null remote_address after calling update_address_with_ice_candidate(). * res_rtp_asterisk.c: Eliminate the chance of ice_reset_session() destroying and setting the rtp->ice pointer to NULL while other threads are using it by adding an ao2 wrapper around the PJPROJECT ice pointer. Now when we have to unlock the RTP instance object to call a PJPROJECT ICE function we will hold a ref to the wrapper. Also added some rtp->ice NULL checks after we relock the RTP instance and have to do something with the ICE structure. ASTERISK-26835 #close ASTERISK-26853 #close Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4	9 years ago
George Joseph	4ccaffe644	make ari-stubs so doc periodic jobs can run ... The periodic doc job does a make ari-stubs and checks that there are no changes before generating the docs. Since I changed the mustache template (and the generated code directly) recently and forgot to regenerate the stubs, the doc job thinks they're out of date. Change-Id: Ibd4bc649556615ff714d44534c45b6c2f6aa449d	9 years ago
Sean Bright	357d1fbdcc	res_stun_monitor: Don't fail to load if DNS resolution fails ... res_stun_monitor will fail to load if DNS resolution of the STUN server fails. Instead, we continue without the STUN server being resolved and we will re-attempt the resolution on the STUN refresh interval. ASTERISK-21856 #close Reported by: Jeremy Kister Change-Id: I6334c54a1cc798f8a836b4b47948e0bb4ef59254	9 years ago
George Joseph	f882ca2572	modules: change module LOAD_FAILUREs to LOAD_DECLINES ... In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting if a module can't be loaded. If the user wishes to retain the FAILURE behavior for a specific module, they can use the "require" or "preload-require" keyword in modules.conf. A new API was added to logger: ast_is_logger_initialized(). This allows asterisk.c/check_init() to print to the error log once the logger subsystem is ready instead of just to stdout. If something does fail before the logger is initialized, we now print to stderr instead of stdout. Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25	9 years ago
Richard Mudgett	cd80af508e	res_rtp_asterisk.c: Add stun_blacklist option ... Added the stun_blacklist option to rtp.conf. Some multihomed servers have IP interfaces that cannot reach the STUN server specified by stunaddr. Blacklist those interface subnets from trying to send a STUN packet to find the external IP address. Attempting to send the STUN packet needlessly delays processing incoming and outgoing SIP INVITEs because we will wait for a response that can never come until we give up on the response. Multiple subnets may be listed. ASTERISK-26890 #close Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342	9 years ago
Richard Mudgett	aecf19e7d2	res_pjsip: Fix pointer use after unref. ... Change-Id: I4b6e1b0070563eeaee223cb58326f1b962ed5bc1	9 years ago
Richard Mudgett	304f652cda	res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member. ... * create_rtp(): Eliminate use of deprecated transport struct member. That member and several others in the transport structure were deprecated because of an infinite loop created when using realtime configuration. See 2451d4e455 ASTERISK-26851 Change-Id: I0533aa13c9ce3c6cc394e0fd2b5bf1cd1b2ef3bc	9 years ago
Joshua Colp	b3f4a6365e	pjsip: Add Alembic for PUBLISH support. ... This change adds database tables for the PUBLISH support so it can be configured using realtime. A minor fix to the res_pjsip_publish_asterisk module was done so that it read the sorcery configuration from the correct section. Finally the sample configuration files have been updated. ASTERISK-26928 Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952	9 years ago
Joshua Colp	1a35801994	Merge "Unused realtime MOH classes not purged on 'moh reload'" into 13	9 years ago
Joshua Colp	f80b7f7014	Merge "res_pjsip_session: Allow BYE to be sent on disconnected session." into 13	9 years ago
Richard Mudgett	6906765381	res_pjsip_sdp_rtp.c: Don't alter global addr variable. ... * create_rtp(): Fix unexpected alteration of global address_rtp if a transport is bound to an address. * create_rtp(): Fix use of uninitialized memory if the endpoint RTP media address is invalid or the transport has an invalid address. ASTERISK-26851 Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7	9 years ago
Daniel Journo	70e5a2655d	Unused realtime MOH classes not purged on 'moh reload' ... Purge Realtime MOH classes on 'moh reload' even when musiconhold.conf hasn't changed. ASTERISK-25974 #close Change-Id: I42c78ea76528473a656f204595956c9eedcf3246	9 years ago
Richard Mudgett	27b556778d	res_pjsip: Fix transport ref leak. ... We were leaking a transport ref in multihomed_on_rx_message() which resulted in the FRACK about excessive ref counts. ASTERISK-26916 #close Change-Id: I7a96658a9614a060565bb9ad51cb1c9c11ee145f	9 years ago
Joshua Colp	bca9685d39	res_pjsip_session: Allow BYE to be sent on disconnected session. ... It is perfectly acceptable for a BYE to be sent on a disconnected session. This occurs when we respond to a challenge to the BYE for authentication credentials. ASTERISK-26363 Change-Id: I6ef0ddece812fea6665a1dd2549ef44fb9d90045	9 years ago
George Joseph	0667febf3d	Merge "res_pjsip_config_wizard: Add 2 new parameters to help with proxy config" into 13	9 years ago