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Author	SHA1	Message	Date
Joshua Colp	7386a761c1	Merge "PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error." into 13	10 years ago
Joshua Colp	4c99310468	Merge topic 'res_pjsip_mwi_cleanups' into 13 ... * changes: res_pjsip_mwi.c: Eliminate a simple RAII_VAR. res_pjsip_mwi.c: Fix mid-line log message line breaks.	10 years ago
Joshua Colp	f35d8f5670	Merge "PJSIP FAX: Fix T.38 automatic reject timer NULL channel pointer dereferences." into 13	10 years ago
Joshua Colp	618b97c143	Merge "res_pjsip_t38.c: Fix always false if test." into 13	10 years ago
Joshua Colp	2bc9245d9d	Merge "res_pjsip: Need to use the same serializer for a pjproject SIP transaction." into 13	10 years ago
Joshua Colp	6d4227bbba	Merge "res/res_http_websocket: Don't send HTTP response fragmented." into 13	10 years ago
Joshua Colp	05e8e14982	res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context. ... This change moves logic for setting up the DTLS SSL contexts to when the SDP is done being processed instead of when ICE negotiation completes. It also stops handshakes from being initiated when we are acting as a server. Manipulating the SSL context when ICE negotiation has completed is problematic as the SSL context is not protected and if acting as a client the remote side may have started DTLS negotiation already. The retransmission timeout timer code has also been split up and simplified some. Both RTP and RTCP now have their own timers and the points at which the timer is stopped and started is now more specific. When a packet is sent the timer is started. When a response is received but before it is processed the timer is stopped. This provides a guarantee that the timeout is not occurring while the response is processed. ASTERISK-22805 #close ASTERISK-24550 #close ASTERISK-24651 #close ASTERISK-24832 #close ASTERISK-25103 #close ASTERISK-25127 #close Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91	10 years ago
Richard Mudgett	38bace4fbb	res_pjsip_t38.c: Fix always false if test. ... Calling t38_change_state() sets the t38 state so it makes little sense to then check the state right after the call for something else. * Made the code in t38_interpret_parameters() reject or exit T.38 mode as intended but not implemented. Change-Id: Ib281263a6ed44da9448132c4e6df1e183b8a3df2	10 years ago
Richard Mudgett	2f7688c788	res_pjsip_mwi.c: Use safer loop coding in mwi_subscription_mailboxes_str(). ... Change-Id: I6f39d809a6d1b47b35bb32b298f5a12f35d6f907	10 years ago
Richard Mudgett	74be3a50d7	res_pjsip_mwi.c: Eliminate a simple RAII_VAR. ... Change-Id: Ib1843f81e826a6c760c424c88eb70c350d9d61da	10 years ago
Richard Mudgett	589e93617a	res_pjsip_mwi.c: Fix mid-line log message line breaks. ... * Add create_mwi_subscriptions_for_endpoint() doxygen comment. Change-Id: I3c3f921f4ec749fb65b62d2f6fa0d4d1888b94e2	10 years ago
Richard Mudgett	0d67e04359	res_pjsip_mwi.c: Fix MWI subscription memory corruption crash. ... MWI subscriptions can crash or corrupt memory when using the subscription datastore to access the MWI subscription object because the datastore is not holding a reference to the object. * Give the subscription datastore a ref to the MWI subscription object. It is unfortunate that the ref causes a circular ref chain that must be explicitly broken to allow the memory to get released. The loop is broken when the subscription is shutdown and if the subscription setup fails. ASTERISK-25168 #close Reported by: Carl Fortin Change-Id: Ice4fa823f138ff10a6c74d280699c41a82836d4f	10 years ago
Richard Mudgett	0422433f47	PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error. ... When res_pjsip body generator modules were generating XML or XPIDF response bodies, there was a chance that the generated body would be the exact size of the supplied buffer. Adding the nul string terminator would then write beyond the end of the buffer and potentially corrupt memory. * Fix MALLOC_DEBUG high fence violations caused by adding a nul string terminator on the end of a buffer for XML or XPIDF response bodies. * Made calls to pj_xml_print() safer if the XML prolog is requested. Due to a bug in pjproject, the return value could be -1 _or_ AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough. * Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the return value of pj_xml_print() when the supplied buffer is not large enough. ASTERISK-25168 Reported by: Carl Fortin Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de	10 years ago
Richard Mudgett	8ea214aed7	PJSIP FAX: Fix T.38 automatic reject timer NULL channel pointer dereferences. ... When a caller calls a FAX number and then hangs up right after the call is answered then the T.38 re-INVITE automatic reject timer may still be running after the channel goes away. * Added session NULL channel checks on the code paths that get executed by t38_automatic_reject() to prevent a crash when the T.38 re-INVITE automatic reject timer expires. ASTERISK-25168 Reported by: Carl Fortin Change-Id: I07b6cd23815aedce5044f8f32543779e2f7a2403	10 years ago
Richard Mudgett	ada7346792	res_pjsip: Need to use the same serializer for a pjproject SIP transaction. ... All send/receive processing for a SIP transaction needs to be done under the same threadpool serializer to prevent reentrancy problems inside pjproject and res_pjsip. * Add threadpool API call to get the current serializer associated with the worker thread. * Pick a serializer from a pool of default serializers if the caller of res_pjsip.c:ast_sip_push_task() does not provide one. This is a simple way to ensure that all outgoing SIP request messages are processed under a serializer. Otherwise, any place where a pushed task is done that would result in an outgoing out-of-dialog request would need to be modified to supply a serializer. Serializers from the default serializer pool are picked in a round robin sequence for simplicity. A side effect is that the default serializer pool will limit the growth of the thread pool from random tasks. This is not necessarily a bad thing. * Made pjsip_distributor.c save the thread's serializer name on the outgoing request tdata struct so the response can be processed under the same serializer. This is a cherry-pick from master. **** ASTERISK-25115 Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a NOTE: session_inv_on_state_changed() is disassociating the dialog from the session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED. Unfortunately this is a tad too soon because our BYE request transaction has not completed yet. ASTERISK-25183 #close Reported by: Matt Jordan Change-Id: I8bad0ae1daf18d75b8c9e55874244b7962df2d0a	10 years ago
Joshua Colp	cba550df7a	Merge "chan_sip: Fix early call pickup channel leak." into 13	10 years ago
Joshua Colp	55137c3d12	res/res_http_websocket: Don't send HTTP response fragmented. ... This change makes it so that when accepting a WebSocket connection the HTTP response is sent as one packet instead of fragmented. Browsers don't like it when you send it fragmented. ASTERISK-25103 Change-Id: I9b82c4ec2949b0bce692ad0bf6f7cea9709e7f69	10 years ago
Matt Jordan	49f81ddb85	Makefile: Remove coverage files on 'make clean' ... This patch updates a variety of Makefiles in Asterisk's build system to remove .gcda and .gcno files when 'make clean' is executed. These files are generated when '--enable-coverage' is passed to the Asterisk configure script. Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602	10 years ago
Joshua Colp	a9f721303d	Merge "chan_vpb.cc: Fix compiler warning Jenkins found." into 13	10 years ago
Walter Doekes	e0f565663b	chan_sip: Fix early call pickup channel leak. ... When handle_invite_replaces() was called, and either ast_bridge_impart() failed or there was no bridge (because the channel we're picking up was still ringing), chan_sip would leak a channel. Thanks Matt and Corey for checking the bridge path. ASTERISK-25226 #close Change-Id: Ie736bb182170a73eef5bcef0ab0376f645c260c8	10 years ago
Matt Jordan	de3678c92e	Merge "sorcery/realtime: Add a bit of debug and warning messages for bad configs" into 13	10 years ago
Joshua Colp	ffde1073bb	Merge "res_timing: Don't close FD 0 when out of open files." into 13	10 years ago
Joshua Colp	43926568c3	Merge "rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format." into 13	10 years ago
Joshua Colp	c2f4bf142f	Merge "astfd: Fix buffer overflow in DEBUG_FD_LEAKS." into 13	10 years ago
Joshua Colp	c5c5c2a4ae	Merge "chan_mgcp: Don't call close on fd -1." into 13	10 years ago
Walter Doekes	a5a262be78	chan_mgcp: Don't call close on fd -1. ... ASTERISK-25220 #close Change-Id: Ic48f3a82f51ada87f2fb0e016c9efe0ad56f1ee3	10 years ago
Walter Doekes	b835312b4c	rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format. ... When running valgrind on Asterisk, it complained about: ==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304) ==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...) ==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292) ==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437) The code in question is a struct assignment, which may be performed by memcpy as a compiler optimization. It is changed to only copy the struct contents if source and destination are different. ASTERISK-25219 #close Change-Id: I6d3546c326b03378ca8e9b8cefd41c16e0088b9a	10 years ago
Walter Doekes	6551e16e03	astfd: Fix buffer overflow in DEBUG_FD_LEAKS. ... If DEBUG_FD_LEAKS was used and more file descriptors than the default of 1024 were available, some DEBUG_FD_LEAKS-patched functions would overwrite memory past the fixed-size (1024) fdleaks buffer. This change: - adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe - consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024 - stores pointers to constants instead of copying the contents - reorders the fdleaks struct for possibly tighter packing - adds a tiny bit of documentation ASTERISK-25212 #close Change-Id: Iacb69e7701c0f0a113786bd946cea5b6335a85e5	10 years ago
Walter Doekes	f4dd9560cf	res_timing: Don't close FD 0 when out of open files. ... This fixes so a failure to get a timer file descriptor does not cascade to closing FD 0. On error, both res_timing_kqueue and res_timing_timerfd would call the destructor before setting the file handle. The file handle had been initialized to 0, causing FD 0 to be closed. This in turn, resulted in floods of "CLI>" messages and an unusable terminal. ASTERISK-19277 #close Reported by: Barry Chern For the 13 branch, this was already fixed. This patch only ensures that we do not attempt to close a negative file descriptor. Change-Id: I147d7e33726c6e5a2751928d56561494f5800350	10 years ago
Richard Mudgett	78a1f4aa46	chan_vpb.cc: Fix compiler warning Jenkins found. ... Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0	10 years ago
Scott Griepentrog	6b16fbfc22	Channel alert pipe: improve diagnostic error return ... When a frame is queued on a channel, any failure in ast_channel_alert_write is logged along with errno. This change improves the diagnostic message through aligning the errno value with actual failure cases. ASTERISK-25224 Reported by: Andrey Biglari Change-Id: I1bf7b3337ad392789a9f02c650589cd065d20b5b	10 years ago
Matt Jordan	8e07ab145d	sorcery/realtime: Add a bit of debug and warning messages for bad configs ... When a mapping does not exist between a sorcery.conf defined object and a realtime mapping in extconf, currently, the user will receive a slew of ERROR messages that don't really tell what is happening. Some ERROR messages may even be misleading, as they occur after the sorcery API has already given up on the attempt to load and create the sorcery object. This patch adds a bit of debug and a useful WARNING message for when a wizard's open callback fails for a particular object type. In the bad configurations that resulted in this patch, this provided a 'root cause' WARNING message that pointed in the right direction of the configuration problem. Change-Id: I1cc7344f2b015b8b9c85a7e6ebc8cb4753a8f80b	10 years ago
Joshua Colp	69bfa518a0	Merge "res_sorcery_realtime: Fix leak of sorcery object type." into 13	10 years ago
Matt Jordan	4c0b26b9e8	Merge "channel: Remove ignore of answer on non-outgoing channels." into 13	10 years ago
Mark Michelson	156395e743	res_sorcery_realtime: Fix leak of sorcery object type. ... This prevents a leak of a sorcery object type when realtime sorcery objects are retrieved by fields or when multiple objects are retrieved. The extent of this leak is that sorcery object types would be leaked. These are allocated whenever an object type is registered with sorcery, meaning that on module shutdown, these objects would be leaked. This could be problematic if many reloads were performed, but it is not as severe as if every sorcery object retrieved from realtime were being leaked. ASTERISK-25165 #close Reported by Corey Farrell Change-Id: I625c3b50eee4576670b7eeb013c81ad043b4b4f8	10 years ago
Matt Jordan	bfa5d38a1b	Merge "res_pjsip_nat: Adjust when contact should be rewritten." into 13	10 years ago
Matt Jordan	d84247a481	Merge "res/res_corosync: Always decline module load, instead of failing" into 13	10 years ago
Matt Jordan	a5e9c4e9b2	res/res_corosync: Always decline module load, instead of failing ... Returns a 'failure' from the module load routine indicates to Asterisk that it should abort loading completely. This is rarely - in fact, really, never - a good option. Aborting load of Asterisk from a dynamic module implies that the core, and the rest of the dynamic modules, don't matter: we should abandon all processing. res_corosync is really not that important. This patch updates the module such that, if it fails to load, it politely declines (emitting ERROR messages along the way), and allows Asterisk to continue to function. Note that this issue was keeping Asterisk unit tests from running on certain build agents. Change-Id: I252249e81fb9b1a68e0da873f54f47e21d648f0f	10 years ago
Matt Jordan	399cd8bcd9	main/pbx: Resolve case sensitivity regression in PBX hints ... When 8297136f was merged for ASTERISK-25040, a regression was introduced surrounding the case sensitivity of device names within hints. Previously, device names - such as 'sip/foo' - were compared in a case insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After that patch, only the case sensitive name would match, i.e., 'SIP/foo'. As a result, some dialplan hints stopped working. This patch re-introduces case insensitive matching for device names in hints. ASTERISK-25040 ASTERISK-25202 #close Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c (cherry picked from commit 96bbcf495a)	10 years ago
Mark Michelson	24eec5a10b	res_pjsip_nat: Adjust when contact should be rewritten. ... A previous change made the contact only get rewritten if the dialog's route set was not marked frozen. Unfortunately, while the intent of this is correct, the dialog's route set actually gets marked as frozen earlier than expected, especially for UAS dialogs. Instead, the idea is that the contact needs to not be rewritten if there is a pre-existing route set on the dialog. This is now accomplished by checking the dialog's route set list instead of checking if the route set is frozen. Doing this causes some broken tests to begin passing again. ASTERISK-25196 Reported by Mark Michelson Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e	10 years ago
Richard Mudgett	0ec461a637	res_pjsip_outbound_registration.c: Add a serializer shutdown group. ... The client_state objects contain a serializer used to send the outbound REGISTER messages. Once all those message transactions are complete then the module can shutdown. ASTERISK-24907 #close Reported by: Kevin Harwell Change-Id: Ibb2fe558f98190f2a06da830e0fadfa25516f547	10 years ago
Matt Jordan	0a1700d286	Merge "threadpool, res_pjsip: Add serializer group shutdown API calls." into 13	10 years ago
Matt Jordan	d8e61c04a4	Merge "res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs" into 13	10 years ago
Matt Jordan	9b70a078f6	Merge "res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API" into 13	10 years ago
Matt Jordan	c69c8d4e80	Merge "res_pjsip_refer: Prevent sending duplicate headers." into 13	10 years ago
Matt Jordan	e5499c3233	Merge "sorcery: Add ast_sorcery_object_unregister() API call." into 13	10 years ago
Matt Jordan	dc067e585f	Merge "res_pjsip_outbound_registration.c: Reorder load_module() and unload_module()." into 13	10 years ago
Mark Michelson	05a2cc1293	res_pjsip_refer: Prevent sending duplicate headers. ... res_pjsip_refer will attempt to add Referred-By or Replaces headers to outbound INVITEs at times. If the INVITE gets challenged for authentication, then we will resend the INVITE. Prior to this patch, the Referred-By or Replaces header would be re-added to the outbound INVITE, resulting in duplicated headers. ASTERISK-25204 #close Reported by Mark Michelson Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d	10 years ago
Matt Jordan	71c1c559cf	Merge "AMI: Add Linkedid to the standard channel snapshot AMI event headers." into 13	10 years ago
Mark Michelson	028fa54620	res_pjsip_nat: Rewrite route set when required. ... When performing some provider testing, the rewrite_contact option was interfering with proper construction of a route set when sending an ACK after receiving a 200 OK response to an INVITE. The initial INVITE was sent to address sip:foo. The 200 OK had a Contact header with URI sip:bar. In addition, the 200 OK had Record-Route headers for sip:baz and sip:foo, in that order. Since the Record-Route headers had the lr parameter, the result should have been: * Set R-URI of the ACK to sip:bar. * Add Route headers for sip:foo and sip:baz, in that order. However, the rewrite_contact option resulted in our rewriting the Contact header on the 200 OK to sip:foo. The result was: * R-URI remained sip:foo. * We added Route headers for sip:foo and sip:baz, in that order. The result was that sip:bar was not indicated in the ACK at all, so the far end never received our ACK. The call eventually dropped. The intention of rewrite_contact is to rewrite the most immediate destination of our SIP request to be the same address on which we received a request or response. In the case of processing a SIP response with Record-Route headers, this means that instead of rewriting the Contact header, we should instead rewrite the bottom-most Record-Route header. In the case of processing a SIP request with Record-Route headers, this means we rewrite the top-most Record-route header. Like when we rewrite the Contact header, we also ensure to update the dialog's route set if it exists. ASTERISK-25196 #close Reported by Mark Michelson Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f	10 years ago