4672 Commits (15)

Author	SHA1	Message	Date
Kevin Harwell	f361e65dc2	AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media ... After receiving a 200 OK with a declined stream in response to a T.38 initiated re-invite Asterisk would crash when attempting to dereference a NULL session media object. This patch checks to make sure the session media object is not NULL before attempting to use it. ASTERISK-28495 patches: ast-2019-004.patch submitted by Alexei Gradinari (license 5691) Change-Id: I168f45f4da29cfe739acf87e597baa2aae7aa572	7 years ago
George Joseph	ed649e7f5f	res_pjsip_messaging: Check for body in in-dialog message ... We now check that a body exists and it has a length > 0 before attempting to process it. ASTERISK-28447 Reported-by: Gil Richard Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f	7 years ago
George Joseph	94221a0c4b	res_pjsip_sdp_rtp: Fix return code from apply_negotiated_sdp_stream ... apply_negotiated_sdp_stream was returning a "1" when no joint capabilities were found on an outgoing call instead of a "-1". This indicated to res_pjsip_session that the handler DID handle the sdp when in fact it didn't. Without the appropriate setup, a subsequent media frame coming in would have an invalid stream_num and cause a seg fault when the stream was attempted to be retrieved. apply_negotiated_sdp_stream now returns the correct "-1" and any media is now discarded before it reaches the core stream processing. ASTERISK-28260 Reported by: Sotiris Ganouris Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f	8 years ago
Richard Mudgett	a2b6bbfc9e	res_smdi.c: Fix module ref counting and inverted test. ... I think this module is so screwed up that it doesn't work anymore. Even with these attempts to fix things it still won't gracefully shut down. The module refs will not go to zero to allow unloading the module. * Fix module ref counting dealing with the SMDI interface object. There were several off-nominal paths that unbalanced the module ref count. Also the destructor freed the ao2 object itself which is bad. Made the smdi_read thread not hold its own ref to the SMDI interface object so when all refs go away the destructor will stop the listener thread. * Fixed the smdi_load() return code of 1 concerning the number of listeners. The test was inverted. Change-Id: Ic288db51b58e395d6a2fc3847f77176c16988784	8 years ago
Joshua Colp	3be86d3f55	Merge "configure.ac: Check for unbound version >= 1.5" into 15	8 years ago
Joshua Colp	370abe7aac	Merge "res_pjsip: improve realtime performance on CLI 'pjsip show contacts'" into 15	8 years ago
Joshua Colp	d2805ca533	Merge "res_stasis: Fix stale data in ARI bridges" into 15	8 years ago
Alexei Gradinari	625b329f4e	res_pjsip: improve realtime performance on CLI 'pjsip show contacts' ... CLI command 'pjsip show contacts' inefficiently make a lot of DB requests. For example if there are 10k aors then asterisk requests these 10k records of aor and then does 10k requests of contact - one request per aor. Even if use 'like <pattern>' the asterisk requests all aor's and contact's records and then filters them by itself. This patch gathers contact's container by - retrieving all dynamic contacts by regex (filtered by reg_server) - retrieving all aors with permanent contacts - finally filters container by regex ASTERISK-28077 #close Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618	8 years ago
Kevin Harwell	d89812e356	Merge "res_odbc: fix missing SQL error diagnostic" into 15	8 years ago
George Joseph	b8a74b4f7d	Merge "res_rtp_asterisk.c: Add "seqno" strictrtp option" into 15	8 years ago
Moritz Fain	dfa926337c	res_stasis: Fix stale data in ARI bridges ... Fixed an issue that resulted in "Allocation failed" each time an ARI request was made to start playing MOH on a bridge. In bridge_moh_create() we were attaching the after bridge callbacks to chan which is the ;1 channel of the unreal channel pair. We should have attached them to the ;2 channel which is pushed into the bridge by ast_unreal_channel_push_to_bridge(). The callbacks are called when the specific channel leaves the bridging system. Since the ;1 channel is never put into a bridge the callbacks never get called. The callbacks then never remove the moh_wrapper from the app_bridges_moh container. As a result we cannot find the channel associated with the wrapper to start MOH because it has hungup. This is the reason causing the reported issue. * Rather than using after bridge callbacks to cleanup, we now have moh_channel_thread() doing the cleanup when the channel hangs up. * Fixed moh_channel_thread() accumulating control frames on the stasis bridge MOH channel until MOH is stopped. Control frames are no longer accumulated while MOH is playing. * Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may or may not return a channel ref. As a result ast_ari_bridges_start_moh() wouldn't know it may have a channel ref to release. stasis_app_bridge_moh_channel() will now return a ref with the channel it returns. * Eliminated RAII_VAR in bridge_moh_create(). ASTERISK-26094 #close Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c	8 years ago
Ben Ford	166c130eac	res_rtp_asterisk.c: Add "seqno" strictrtp option ... When networks experience disruptions, there can be large gaps of time between receiving packets. When strictrtp is enabled, this created issues where a flood of packets could come in and be seen as an attack. Another option - seqno - has been added to the strictrtp option that ignores the time interval and goes strictly by sequence number for validity. Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71	8 years ago
Alexei Gradinari	d03661bcaf	res_odbc: fix missing SQL error diagnostic ... On SQL error there is not diagnostic information about this error. There is only WARNING res_odbc.c: SQL Execute error -1! The function ast_odbc_print_errors calls a SQLGetDiagField to get the number of available diagnostic records, but the SQLGetDiagField returns 0. However SQLGetDiagRec could return one diagnostic records in this case. Looking at many example of getting diagnostics error information I found out that the best way it's to use only SQLGetDiagRec while it returns SQL_SUCCESS. Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR to res_config_odbc. ASTERISK-28065 #close Change-Id: Iba5ae5470ac49ecd911dd084effbe9efac68ccc1	8 years ago
George Joseph	4c636f5377	configure.ac: Check for unbound version >= 1.5 ... In order to do this and provide good feedback, a new macro was created (AST_EXT_LIB_EXTRA_CHECK) which does the normal check and path setups for the library then compiles, links and runs a supplied code fragment to do the final determination. In this case, the final code fragment compares UNBOUND_VERSION_MAJOR and UNBOUND_VERSION_MINOR to determine if they're greater than or equal to 1.5. Since we require version 1.5, some code in res_resolver_unbound was also simplified. ASTERISK-28045 Reported by: Samuel Galarneau Change-Id: Iee94ad543cd6f8b118df8c4c7afd9c4e2ca1fa72	8 years ago
Joshua Colp	af090e4fad	res_rtp_asterisk: Raise event when RTP port is allocated ... This change raises a testsuite event to provide what port Asterisk has actually allocated for RTP. This ensures that testsuite tests can remove any assumption of ports and instead use the actual port in use. ASTERISK-28070 Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044	8 years ago
George Joseph	2784fe688a	Merge "res_remb_modifier: Add module for controlling REMB from CLI." into 15	8 years ago
George Joseph	71f1e23173	Merge "res_rtp_asterisk: Fix crash on ast_rtp_new failure." into 15	8 years ago
Corey Farrell	21f472355e	res_rtp_asterisk: Fix crash on ast_rtp_new failure. ... ast_rtp_new free'd rtp upon failure, but rtp_engine.c would also call the destroy callback. Remove call to ast_free from ast_rtp_new, leave it to rtp_engine.c to initiate the full cleanup. Add error detection for the ssrc_mapping vector initialization. In rtp_allocate_transport set rtp->s = -1 in the failure path where we close that FD to ensure we don't try closing it twice. ASTERISK-27854 #close Change-Id: Ie02aecbb46228ca804e24b19cec2bb6f7b94e451	8 years ago
Sean Bright	65f63bd678	res_rtp_asterisk: Reset all settings on module reload ... 'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults if they are not present in the updated configuration file. Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670	8 years ago
Sean Bright	fd87b8e49c	AST-2018-009: Fix crash processing websocket HTTP Upgrade requests ... The HTTP request processing in res_http_websocket allocates additional space on the stack for various headers received during an Upgrade request. An attacker could send a specially crafted request that causes this code to overflow the stack, resulting in a crash. * No longer allocate memory from the stack in a loop to parse the header values. NOTE: There is a slight API change when using the passed in strings as is. We now require the passed in strings to no longer have leading or trailing whitespace. This isn't a problem as the only callers have already done this before passing the strings to the affected function. ASTERISK-28013 #close Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a	8 years ago
Joshua Colp	87bac2057e	res_remb_modifier: Add module for controlling REMB from CLI. ... This adds a module which registers a CLI command that can set the REMB bitrate value for REMB as it enters or exits Asterisk. This allows you to ignore what Asterisk or a client produces and is useful for demonstrations. This does not generate REMB frames, however, but just modifies them as they flow to or from a channel. Change-Id: Ib089427c46a4a36d645cecfe02406adb38c17bec	8 years ago
Joshua Colp	e7bc861f88	Merge "res_pjsip_session: Don't add declined stream if one does not exist." into 15	8 years ago
Joshua Colp	b5959113f6	pjproject: Upgrade to 2.8. ... This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189	8 years ago
Joshua Colp	5626b8ecae	res_pjsip_session: Don't add declined stream if one does not exist. ... Given a scenario where a session refresh was done with a removed stream we would always add a removed stream to the outgoing SDP even if one did not already exist. This change makes it so that a removed stream is only placed into the SDP if one already exists. ASTERISK-28047 Change-Id: Ibb97d21cdeb87a8acae0c720861b0ff255708442	8 years ago
Sean Bright	5713fc0b49	autoconf: Check for srtp_get_version_string() before using it ... Change-Id: Id2a916ff9448706090e72ff2c7fb3f5ba24a05df	8 years ago
George Joseph	16ec2f2645	Merge "res_srtp.c: Show linked version of libsrtp on module init" into 15	8 years ago
Sean Bright	644f7ef413	res_srtp.c: Show linked version of libsrtp on module init ... Change-Id: Ib0a645d6985de5757cc4399ed2524b2d02c4f342	8 years ago
Sean Bright	5f6bef4e5b	res_pjsip: Log IPv6 addresses correctly ... Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name store IPv6 addresses without enclosing brackets. This causes some log output to be confusing because it is difficult to separate the IPv6 address from a port specification. * Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6 output. * When a pj_sockaddr is not available, explicitly wrap IPv6 addresses in brackets. * When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure to also set pjsip_rx_data.pkt_info.src_addr. Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8	8 years ago
George Joseph	fad0714c27	Merge "res_musiconhold.c: Restart MOH if previous hold just reached end-of-file" into 15	8 years ago
Jenkins2	609a895886	Merge "optional_api: Remove unused nonoptreq fields" into 15	8 years ago
Walter Doekes	4616c6508a	optional_api: Remove unused nonoptreq fields ... As they're not actively used, they only grow stale. The moduleinfo field itself is kept in Asterisk 13/15 for ABI compatibility. ASTERISK-28046 #close Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc	8 years ago
Sean Bright	b4cb51ca63	res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP ... The bundled version of pjproject has a patch for Solaris compatability that changes the definition of various socket structures which we need to account for when compiling against a non-bundled version. ASTERISK-28049 #close Change-Id: Ia1ea47c433fc2d915115193ee889a752373925f0	8 years ago
Frederic LE FOLL	d22c4d1999	res_musiconhold.c: Restart MOH if previous hold just reached end-of-file ... On MOH activation, moh_files_readframe() is called while the current stream attached to the channel is NULL and it calls ast_moh_files_next() immediately. However, it won't call ast_moh_files_next() again if sample reading fails. The failure may occur because res_musiconhold retains the last sample reading position in the channel data and MOH during the previous hold/retrieve just reached EOF. Obviously, a bit of bad luck is required here. * Restructured moh_files_readframe() to try a second time to start MOH if there was no stream setup and the saved position was at EOF. Also added comments describing what is going on for each step. ASTERISK-28029 Change-Id: I1508cf2c094f8feca22d6f76deaa9fdfa9944860	8 years ago
Sean Bright	9624cbff4f	res_pjproject: Add utility functions to convert between socket structures ... Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address needs to be rendered to a string and then parsed into the correct structure. This also involves a call to getaddrinfo(3). The same is true for the inverse operation. Instead, because we know the internal structure of both ast_sockaddr and pj_sockaddr, we can translate directly between the two without the need for an intermediate string. Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761	8 years ago
George Joseph	432f65fb19	Merge "res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch" into 15	8 years ago
Joshua Colp	304859d151	Merge "res_fax: Handle fax gateway being started more than once." into 15	8 years ago
Joshua Colp	59d92e96a9	Merge "res_pjsip_transport_websocket: Properly set src_name for IPv6" into 15	8 years ago
Richard Mudgett	4f30cd4b52	res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch ... ASTERISK-27988 Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843	8 years ago
Joshua Colp	b844477683	Merge "Create --disable-binary-modules option." into 15	8 years ago
Joshua Colp	58968366fc	res_fax: Handle fax gateway being started more than once. ... The T.38 fax gateway state machine can cause the fax gateway to be started more than once on a channel depending on the responses of the remote endpoint. This would previously leak the channel name, channel unique id, and underlying fax engine state. This change instead makes it so that if the fax gateway session is already present and not reserved the fax gateway is not started again. ASTERISK-27981 Change-Id: I552d95086860cb18f2522ee40ef47b13b6da2e0e	8 years ago
Sean Bright	7ec4f0cb6a	res_pjsip_transport_websocket: Properly set src_name for IPv6 ... SIP responses over WebSockets when the client is using IPv6 have invalid Via headers according to RFC 3261. The 'received' header parameter should not be wrapped in brackets if it is an IPv6 address. When src_name is populated by the built-in PJSIP transports, the code uses pj_sockaddr_print() with 'flags' set to 0, meaning that the brackets are not rendered around IPv6 addresses. This may be related to ASTERISK~27101. See also: https://github.com/onsip/SIP.js/pull/594 ASTERISK-28020 #close Change-Id: I8ea9d289901b837512bee2ca2535e3dc14f04d77	8 years ago
Corey Farrell	588e4ecf50	Create --disable-binary-modules option. ... This new option can be passed for ./configure or ./tests/CI/buildAsterisk.sh to prevent download/install of binary modules. Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES will result in binary modules being enabled even if the build target is incompatible with those modules. This includes CI scripts which enable categories before disabling specific modules. If more binary modules are offered in the future this will help avoid accidentally downloading them if unwanted or incompatible. Adding a binary module will only require creating a new menuselect entry similar to the existing ones, it will not be necessary to modify the CI scripts. Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166	8 years ago
neutrino88	b4e1a2c8fe	res/res_rtp_asterisk: remove debug traces generated by an empty frame ... The realtime text timer pops regularly and sends text frames even if the buffer is empty. This causes a lot of unecessary debug logging. * Made red_write() test if we need to send a frame before calling ast_rtp_write() ASTERISK-28002 Reported by: Emmanuel BUU Tested by: Emmanuel BUU Change-Id: Icf81310c3b8080b615a42060afc02ab41f9523dd	8 years ago
George Joseph	8c09e32b05	Merge "res_sorcery_realtime.c: Fix unqualified fetch warning." into 15	8 years ago
George Joseph	2dc3795cb3	Merge "res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response." into 15	8 years ago
Richard Mudgett	49c06adc9d	res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response. ... We were still getting crashes after the first fix. Somehow we receive a non-2xx final response before we get a 200 final response. With the failure response we had already cleaned up and destroyed some data structures. When the unexpected 200 response comes in we crash. * Add protection code to prevent processing another final T.38 reINVITE response. ASTERISK-27944 Change-Id: I8b5baba8d07fe4d63f0d7d05d3eb9a3d27d40a74	8 years ago
Richard Mudgett	3f98f49e43	res_sorcery_realtime.c: Fix unqualified fetch warning. ... The allow_unqualified_fetch option for the sorcery realtime backend blocked actually fetching all rows when the option is set to warn. * Made issue a warning and actually do the request when allow_unqualified_fetch=warn is set. Change-Id: I74456c80a03a62dce66fc3dc3cb0cf2351ac4312	8 years ago
Richard Mudgett	ef9ef0f856	res_rtp_asterisk.c: Fix unused variable warnings ... Compiling without SRTP support installed resulted in some unused variable warnings. These warnings also showed that the srtp variable was obtained and passed around some functions but not really used even when a system has SRTP installed. Change-Id: I6daad34be3e89b19adef6e2fbe738018975155fc	8 years ago
Joshua Colp	e814fb406c	Merge "res_resolver_unbound: Fix leak of config nameserver strings." into 15	8 years ago
Joshua Colp	d983b0c470	Merge "res_pjsip: Resolve transport management leak at shutdown." into 15	8 years ago