asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	47ebab959e	res_rtp_asterisk: Ensure DTLS timeout timer is -1 if DTLS is not used. This change fixes a bug where the DTLS timeout timer would be initialized to 0 if DTLS was not used for an RTP session. ASTERISK-25103 Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac	10 years ago
Joshua Colp	1ad827327a	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
Joshua Colp	895dbe532c	Merge "chan_sip: Fix early call pickup caused deadlock." into 11	10 years ago
Joshua Colp	6d9115767f	Merge "chan_vpb.cc: Fix compiler warning Jenkins found." into 11	10 years ago
Joshua Colp	89b0aaa652	Merge "res_timing: Don't close FD 0 when out of open files." into 11	10 years ago
Joshua Colp	077bfe7455	Merge "rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format." into 11	10 years ago
Joshua Colp	da5725dbc7	Merge "astfd: Fix buffer overflow in DEBUG_FD_LEAKS." into 11	10 years ago
Joshua Colp	aa04d4387a	Merge "chan_mgcp: Don't call close on fd -1." into 11	10 years ago
Walter Doekes	0e6d3f5ee5	chan_sip: Fix early call pickup caused deadlock. If non-magic pickup (no "pickup-" in callid) is used, chan_sip locks the channel it wants to pick up, and a bit further down, it locks the channel list when allocating a new channel. That causes a deadlock when another part of the code traverses over the channel list, locking all the channels one by one. This changeset fixes it by releasing the locks before calling sip_new and reacquiring them afterwards. Unfortunately this involves doing the checks we already did again (because the channel may have changed). While trying to avoid duplicate code, I did some refactoring for readability: - if refer_locked == 1, we guarantee there is a locked channel - magic_callid holds a cached version of !ast_strlen_zero(pickup.exten) This is for branch 11 only. It appears that the changed code in 13 does not lock the components like it does in 11 and below. Reproducing the deadlock on 13 has thusfar failed. ASTERISK-25213 #close Change-Id: Ie1d15bec7d634035f48892e1ed6227411d7de2c1	10 years ago
Walter Doekes	f6bbc4f16e	chan_mgcp: Don't call close on fd -1. ASTERISK-25220 #close Change-Id: Ic48f3a82f51ada87f2fb0e016c9efe0ad56f1ee3	10 years ago
Walter Doekes	c139956e95	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	028554faab	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	b090a8d40b	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 Change-Id: I147d7e33726c6e5a2751928d56561494f5800350	10 years ago
Richard Mudgett	1e186dc857	chan_vpb.cc: Fix compiler warning Jenkins found. Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0	10 years ago
Scott Griepentrog	8874d5fc94	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	9b74dcb687	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
Matt Jordan	96bbcf495a	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	10 years ago
Matt Jordan	353dd68cd0	Merge "chan_sip: Prevent deadlock when performing BYE with Also transfer." into 11	10 years ago
Damian Ivereigh	8b60998d29	chan_sip.c: Update dialog fromtag after request with auth If a client sends and INVITE which is 401 rejected, then subsequently sends a new INVITE with the auth info and uses a different fromtag from the first INVITE, Asterisk will accept the new INVITE as part of the original dialog - match_req_to_dialog() specifically ignores the fromtag. However it does not update the stored dialog with the new fromtag. This results in Asterisk being unable to match future packets that are part of this dialog (such as the ACK to the OK or the OK to the BYE), and the call is dropped. This problem was originally found when using an NEC-i SV8100-GE (NEC SIP Card). * After a successful match of a packet to the dialog, if the packet is not a SIP_RESPONSE, authentication is present and the fromtags are different, the stored fromtag is updated with the one from the recent INVITE. ASTERISK-25154 #close Reported by: Damian Ivereigh Tested by: Damian Ivereigh Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e	10 years ago
Mark Michelson	d821f56b02	chan_sip: Prevent deadlock when performing BYE with Also transfer. When a BYE with an Also header is successfully processed, and the sender of the BYE is bridged with another channel, chan_sip will unlock the owner of the dialog on which the BYE was received, call ast_async_goto() on the bridged channel, and then re-lock the owner. The reason for this locking behavior is that ast_async_goto() can result in a masquerade, which requires that the involved channels are unlocked. The problem here is that this causes a locking inversion since the dialog's lock is held when re-locking the owner channel after the async goto. The lock order is supposed to be channel and then sip_pvt. The fix proposed is simple. In addition to unlocking the owner channel before the ast_async_goto() call, also unlock the sip_pvt. Then relock both after ast_async_goto() returns, being sure to lock the channel and then the sip_pvt. ASTERISK-25139 #close Reported by Gregory Massel Change-Id: I72c4fc295ec8573bee599e8e9213c5350a3cd224	10 years ago
Joshua Colp	c164bd2613	Merge "weakref attribute detection broken with gcc 4.6 and higher" into 11	10 years ago
Corey Farrell	53658a14cc	Fix unsafe uses of ast_context pointers. Although ast_context_find, ast_context_find_or_create and ast_context_destroy perform locking of the contexts table, any context pointer can become invalid at any time that the contexts table is unlocked. This change adds locking around all complete operations involving these functions. Places where ast_context_find was followed by ast_context_destroy have been replaced with calls ast_context_destroy_by_name. ASTERISK-25094 #close Reported by: Corey Farrell Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa	10 years ago
ibercom	eed9b1ce52	CLI: Cosmetic issue - core show uptime Show uptime information ends with an unnecessary space. Now NEEDCOMMA is better defined. Change-Id: I11b360504a0703309ff51772ff8f672287f3c5a1	10 years ago
ibercom	f98c458c92	weakref attribute detection broken with gcc 4.6 and higher GCC 4.7 Manual: http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html weakref ("target") A weak reference is an alias that does not by itself require a definition to be given for the target symbol. ASTERISK-22559 #close Reported by: Ibercom Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf	10 years ago
Ivan Poddubny	99c54fe42d	Fix buffer overflow in slin sample frames generation. The length of frames retured by sample functions was twice as large as real, what caused global buffer overflow caught by AddressSanitizer. ASTERISK-24717 #close Reported by: Badalian Vyacheslav Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6	10 years ago
Ivan Poddubny	f9877139df	Astobj2: Correctly treat hash_fn returning INT_MIN The code in astobj2_hash.c wrongly assumed that abs(int) is always > 0. However, abs(INT_MIN) = INT_MIN and is still negative, as well as abs(INT_MIN) % num_buckets, and as a result this led to a crash. One way to trigger the bug is using host=::80 or 0.0.0.128 in peer configuration section in chan_sip or chan_iax. This patch takes the remainder before applying abs, so that bucket number is always in range. ASTERISK-25100 #close Reported by: Mark Petersen Change-Id: Id6981400ad526f47e10bcf7b847b62bd2785e899	10 years ago
Matt Jordan	8f431d679f	Merge "audiohook.c: Difference in read/write rates caused continuous buffer resets" into 11	10 years ago
Matt Jordan	5c047d4b0d	Merge "Logger: Reset defaults before processing config." into 11	10 years ago
Corey Farrell	58de286467	Logger: Reset defaults before processing config. Reset options to default values before reloading config. This ensures that if a setting is removed or commented out of the configuration file it is unset on reload. ASTERISK-25112 #close Reported by: Corey Farrell Change-Id: Id24bb1fb0885c2c14cf8bd6f69a0c2ee7cd6c5bd	10 years ago
Corey Edwards	2d297c7b9a	chan_sip/sdp_crypto.c: allow SDP crypto tag to be up to 9 digits ASTERISK-24887 #close Reported by: Makoto Dei Tested by: tensai Change-Id: I6a96f572adb17f76b3acafe503a01c48eb5dd9bf	10 years ago
Kevin Harwell	58970f1475	audiohook.c: Difference in read/write rates caused continuous buffer resets Currently, everytime a sample rate change occurs (on read or write) the associated factory buffers are reset. If the requested sample rate on a read differed from that of a write then the buffers are continually reset on every read and write. This has the side effect of emptying the buffer, thus there being no data to read and then write to a file in the case of call recording. This patch fixes it so that an audiohook_list's rate always maintains the maximum sample rate among hooks and formats. Audiohook sample rates are only overwritten by this value when slin native compatibility is turned on. Also, the audiohook sample rate can only overwrite the list's sample rate when its rate is greater than that of the list or if compatibility is turned off. This keeps the rate from constantly switching/resetting. ASTERISK-24944 #close Reported by: Ronald Raikes Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f	10 years ago
Jonathan Rose	cf1190cc6b	Message.c: Clear message channel frames on cleanup The message channel is a special channel that doesn't actually process frames. However, certain actions can cause frames to be placed in the channel's read queue including the Hangup application which is called on the channel after each message is processed. Since the channel will continually be reused for many messages, it's necessary to flush these frames at some point. ASTERISK-25083 #close Reported by: Jonathan Rose Change-Id: Idf18df73ccd8c220be38743335b5c79c2a4c0d0f	10 years ago
Joshua Colp	81b0789906	Merge "main/manager.c: Bugfix sort action_manager by alphabetically" into 11	10 years ago
Joshua Colp	3bc8c96443	Merge "General: Fix recent menuselect-related cross compile regression" into 11	10 years ago
Joshua Colp	ffb60909a3	Merge "chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision." into 11	10 years ago
Joshua Colp	63d9ae01ee	Merge "res_config_mysql: Fix broken column type checking" into 11	10 years ago
Rodrigo Ramírez Norambuena	be7255b563	main/manager.c: Bugfix sort action_manager by alphabetically Fix the alphabetic order added on ast_manager_register_struct. The order for struct manager_action added is not working, this change fixes the problem. Change-Id: I149da0cd06c3c4445d7516cc303358e9f26f8b4b	10 years ago
Alexandre Fournier	9370db107a	res_config_mysql: Fix broken column type checking MySQL configuration engine contains a bug in require_mysql(). This function is used for column type checking in tables. This bug only affects DATETIME, DATE and FLOAT types. It came from mixing the first condition (switch-case-like if/then/else), to check the expected column type, with the second condition, to check the actual column type against the expected column type. Both conditions must be checked separately in order to avoid the execution of the wrong block. ASTERISK-18252 #comment This patch might fix the issue Reported by: Gareth Blades ASTERISK-25041 #close Reported by: Alexandre Fournier Tested by: Alexandre Fournier Change-Id: I0b8bf7e68ab938be8e6525a249260cb648cb0bfa	10 years ago
Yousf Ateya	e3129b84b1	res_rtp_asterisk: Correction for the limit which detects that a packet is DTLS. First byte of DTLS packet shall be in range 20-63, not 20-64. Refer to RFC https://tools.ietf.org/html/rfc5764#section-5.1.2 for correct values. Change-Id: Iae6fa0d72b37c36a27fe40686e0ae6fba3afec31	10 years ago
Joshua Colp	735d337d88	Merge "cdr_pgsql: Use PQescapeStringConn for escaping names." into 11	10 years ago
Richard Mudgett	bedc7bf825	chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision. If an ISDN call is hungup by both sides at the same time a crash could happen. * Added missing NULL checks for the owner channel after calling pri_queue_pvt_cause_data() in two places. Code after those calls need to check the owner channel pointer for NULL before use because pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the owner and the owner may get hung up. ASTERISK-21893 #close Reported by: Alexandr Gordeev Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a	10 years ago
Sebastian Kemper	53326e7ab0	General: Fix recent menuselect-related cross compile regression MAKE_MENUSELECT currently sets CC to CC, which is the compiler for the target platform. But menuselect is to be run on the build system, so BUILD_CC needs to be used instead - like it was in the past, before the recent changes (https://reviewboard.asterisk.org/r/4370/). This is the patch for ASTERISK-25074. ASTERISK-25074 #close Reported by: Sebastian Kemper Tested by: Sebastian Kemper Change-Id: I8a2b1fc5deb6ad2b80f49baca35b1b13d468ebf8	10 years ago
Joshua Colp	cd3e851a35	Merge "Fix processing of asterisk.conf debug=yes." into 11	10 years ago
Corey Farrell	57144feed4	Fix processing of asterisk.conf debug=yes. The code which reads asterisk.conf supports processing the debug option with ast_true, but ast_true returns -1. This causes debug to still be off, convert to 1 so debug will be on as requested. ASTERISK-25042 Reported by: Corey Farrell Change-Id: I3c898b7d082d914b057e111b9357fde46bad9ed6	10 years ago
Rodrigo Ramírez Norambuena	4c0eb9d4fb	cdr_pgsql: Use PQescapeStringConn for escaping names. Use function PQescapeStringConn for escaping the name of the table and schema instead of doing it manually. Change-Id: I6709165e2d00463e9c813d24f17830ad4910b599	10 years ago
Ivan Poddubny	cc39cfa213	pbx/pbx_spool: Fix issue when call files were executed too early pbx_spool used to delete/move the call file upon successful outgoing call completion, but did not delete it from in-memory list of files (dirlist, used only when compiled with inotify/kqueue support). That resulted in an extra attempt to process that filename after retrytime seconds. Then, if a new file with the same name appears that is scheduled in future further than the completed one plus its retrytime, then it gets executed earlier than expected. This patch fixes remove_from_queue function to also remove the entry from the dirlist. ASTERISK-17069 #close Reported by: Jeremy Kister ASTERISK-24442 #close Reported by: tootai Change-Id: If9ec9b88073661ce485d6b008fd0b2612e49a28b	10 years ago
Matt Jordan	fb27395f75	Merge "tcptls: Avoiding ERR_remove_state in OpenSSL." into 11	10 years ago
Matt Jordan	767d96cf7e	Merge "res_rtp_asterisk: Issue ERROR if res_srtp is not found." into 11	10 years ago
Sean Bright	151c3316d5	res_rtp_asterisk: Issue ERROR if res_srtp is not found. While trying to get WebRTC working with chan_pjsip, I was running into the following error: Attempted to set an invalid DTLS-SRTP configuration on RTP instance... Josh helpfully pointed out that res_srtp.so might not be loaded, and sure enough, it wasn't. This patch adds a ERROR indiciating as much to hopefully help others having a similar problem. Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f	10 years ago
Rusty Newton	a2f96d3c1a	sounds: Add Swedish sounds to Makefile and XML Added the necessary lines to the Makefile and sounds.xml so we'll have the Swedish sounds in all available formats in menuselect. See also: Swedish sounds were added into the core sounds release 1.4.27. ASTERISK-24744 #close Reported by: Tove Hjelm Tested by: Rusty Newton Change-Id: Ib6f4fd177afd1667b2402735034001d4d055a908	10 years ago

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24134 Commits (47ebab959edc93a769be4e6cc959c43dad3c7dd0) All Branches Search

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