4032 Commits (066659a38327f988cba181c0ff23ec9073a7ddb8)

Author	SHA1	Message	Date
Jenkins2	066659a383	Merge "res_pjsip_session: Add cleanup to ast_sip_session_terminate"	9 years ago
Jenkins2	175297fe34	Merge "res_pjsip/res_pjsip_callerid: NULL check on caller id name string"	9 years ago
George Joseph	d6b2a58736	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
Jenkins2	54e27cad3c	Merge "res_rtp_asterisk.c: Fix crash in RTCP DTLS operation."	9 years ago
Kevin Harwell	c6b757fa05	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
Jenkins2	e478d2eb94	Merge "res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP."	9 years ago
Jenkins2	5a987fc5e9	Merge "res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions."	9 years ago
George Joseph	a3a77890dc	Merge "res_hep: Add additional config initialization and validation"	9 years ago
George Joseph	8df729517e	Merge "res_pjsip_session.c: Restructure ast_sip_session_alloc()"	9 years ago
Sean Bright	0611f2ca17	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
George Joseph	cebfe85aff	Merge "rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes."	9 years ago
Richard Mudgett	f1d20c84a1	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	835c209445	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	dd239e9f91	Merge "res_stun_monitor: Don't fail to load if DNS resolution fails"	9 years ago
Richard Mudgett	afad2ffd9f	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	d165079cbc	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	b55d21ad91	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: I94b97035311eccf52b0101b8590223265a7881d4	9 years ago
Sean Bright	f6600f2c2e	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
Alexander Traud	72c5f3b0ba	res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP. ... This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over UDP, if many codecs are allowed in Asterisk. This new feature is enabled together with the optional feature compact_headers=yes via the file pjsip.conf. ASTERISK-26932 #close Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689	9 years ago
George Joseph	6db0939b96	modules: change module LOAD_FAILUREs to LOAD_DECLINES (14) ... Change-Id: If99e3b4fc2d7e86fc3e61182aa6c835b407ed49e	9 years ago
George Joseph	747beb1ed1	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	7312cbe803	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	5b4e2ec267	res_pjsip: Fix pointer use after unref. ... Change-Id: I4b6e1b0070563eeaee223cb58326f1b962ed5bc1	9 years ago
Richard Mudgett	6f793ac149	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	270b485f04	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
zuul	d2e6eb2afe	Merge "Unused realtime MOH classes not purged on 'moh reload'"	9 years ago
zuul	a385db5333	Merge "res_pjsip_session: Allow BYE to be sent on disconnected session."	9 years ago
Richard Mudgett	f2ee8ac21e	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	6c3ae397cb	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	a889621b14	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	48be02c5d8	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
zuul	7898aad02d	Merge "res_pjsip_config_wizard: Add 2 new parameters to help with proxy config"	9 years ago
Joshua Colp	7581bb4f5f	Merge "srtp: Allow zero as tag value for a sRTP Crypto Suite."	9 years ago
zuul	410a5ac0fa	Merge "Add DTLS sanity check."	9 years ago
zuul	5a530171ca	Merge "res_musiconhold: Don't chdir() when scanning MoH files"	9 years ago
Alexander Traud	e76cc51d5e	srtp: Allow zero as tag value for a sRTP Crypto Suite. ... ASTERISK-25490 #close Change-Id: I1c5fc0942c33c96d62b24203aad0f1e1a1a0131f	9 years ago
Joshua Colp	f43cfb81d9	Merge "res_xmpp: Fix ref counting issue"	9 years ago
George Joseph	2fe52174de	res_pjsip_config_wizard: Add 2 new parameters to help with proxy config ... Two new parameters have been added to the pjsip config wizard. * Setting 'sends_line_with_registrations' to true will cause the wizard to skip the creation of an identify object to match incoming request to the endpoint and instead add the line and endpoint parameters to the outbound registration object. * Setting 'outbound_proxy' is a shortcut for adding individual endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy parameters. Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0 (cherry picked from commit a827892ff7) (cherry picked from commit 27344675be)	9 years ago
Joshua Colp	d0ada2246e	Merge "res_xmpp: Use incremental backoff when a read error occurs"	9 years ago
Richard Mudgett	3d8899bacf	Add DTLS sanity check. ... Change-Id: Ib32612cf6c7ce9213a11b9cba82f630f8cd3564b	9 years ago
Sean Bright	d22c678999	res_musiconhold: Don't chdir() when scanning MoH files ... There doesn't appear to be any reason that we are chdir'ing in moh_scan_files, and in the event of an Asterisk crash, the core files may not get written because we have changed into a read-only directory. ASTERISK-23996 #close Reported by: Walter Doekes Change-Id: Iac806dce01b3335963fbd62d4b4da9a65c614354	9 years ago
Joshua Colp	3bdf876b04	Merge "res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts"	9 years ago
Sean Bright	d5a8799c4b	res_xmpp: Use incremental backoff when a read error occurs ... If a read error occurs, we immediately attempt a reconnect without any delay. Instead, let's sleep and backoff up to 60 seconds before we try again. ASTERISK-24712 #close Reported by: Matthias Urlichs Change-Id: I6fe10ef4734837727437beab715e336777f13f48	9 years ago
zuul	f27c1c7dc2	Merge "res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus"	9 years ago
zuul	8982d0e007	Merge "res_xmpp: Include client name in connection related error messages"	9 years ago
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
Sean Bright	98a88e9ffa	res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus ... The documentation for JABBER_STATUS (and the deprecated JabberStatus app) indicate that a return value of 7 indicates that the specified buddy was not in the roster. It also indicates that you can specify a "bare" JID (one without a resource). Unfortunately the actual behavior does not match the documented behavior. Assuming that our roster includes the buddy online and available "valid@example.org/Valid" and does *not* include the buddy "invalid@example.org", the JABBER_STATUS() function returns the following before this patch: +------------------------------+------------+--------------------------+ | Buddy | Status | Result | +------------------------------+------------+--------------------------+ | valid@example.org | Online | 7 (Not in roster) | | valid@example.org/Valid | Online | 1 (Online) | | valid@example.org/Invalid | N/A | 7 (Not in roster) | | invalid@example.org | N/A | Error logged, no return | | invalid@example.org/Valid | N/A | Error logged, no return | +------------------------------+------------+--------------------------+ And after this patch: +------------------------------+------------+--------------------------+ | Buddy | Status | Result | +------------------------------+------------+--------------------------+ | valid@example.org | Online | 1 (Online) | | valid@example.org/Valid | Online | 1 (Online) | | valid@example.org/Invalid | N/A | 6 (Offline) | | invalid@example.org | N/A | 7 (Not in roster) | | invalid@example.org/Valid | N/A | 7 (Not in roster) | +------------------------------+------------+--------------------------+ This brings the behavior in line with the documentation. ASTERISK-23510 #close Reported by: Anthony Critelli Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf	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