asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	1b51b5efb6	rtp_engine.c: Tweak glue->update_peer() parameter nil value. Change glue->update_peer() parameter from 0 to NULL to better indicate it is a pointer. Change-Id: Iefbf9d4a708f2b64b7ad2b4e6c33bfaa12ccfa9d	11 years ago
Richard Mudgett	f5cd1fa0df	chan_sip.c: Tweak glue->update_peer() parameter nil value. Change glue->update_peer() parameter from 0 to NULL to better indicate it is a pointer. Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd	11 years ago
Mark Michelson	f2089dce3d	res_http_websocket: Properly encode 64 bit payload A test agent was continuously failing all ARI tests when run against Asterisk 13. As it turns out, the reason for this is that on those test runs, for some reason we decided to use the super extended 64 bit payload length for websocket text frames instead of the extended 16 bit payload length. For 64-bit payloads, the expected byte order over the network is 7, 6, 5, 4, 3, 2, 1, 0 However, we were sending the payload as 3, 2, 1, 0, 7, 6, 5, 4 This meant that we were saying to expect an absolutely MASSIVE payload to arrive. Since we did not follow through on this expected payload size, the client would sit patiently waiting for the rest of the payload to arrive until the test would time out. With this change, we use the htobe64() function instead of htonl() so that a 64-bit byte-swap is performed instead of a 32 bit byte-swap. Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a	11 years ago
Mark Michelson	7e8916908d	Local channels: Alternate solution to ringback problem. Commit `54b25c80c8` solved an issue where a specific scenario involving local channels and a native local RTP bridge could result in ringback still being heard on a calling channel even after the call is bridged. That commit caused many tests in the testsuite to fail with alarming consequences, such as not sending DialBegin and DialEnd events, and giving incorrect hangup causes during calls. This commit reverts the previous commit and implements and alternate solution. This new solution involves only passing AST_CONTROL_RINGING frames across local channels if the local channel is in AST_STATE_RING. Otherwise, the frame does not traverse the local channels. By doing this, we can ensure that a playtones generator does not get started on the calling channel but rather is started on the local channel on which the ringing frame was initially indicated. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39	11 years ago
Mark Michelson	534ed6744a	Merge "audiohook: Read the correct number of samples based on audiohook format." into 11	11 years ago
Mark Michelson	525bbf7d4e	Local channels: Do not block control -1 payloads. Control frames with a -1 payload are used as a special signal to stop playtones generators on channels. This indication is sent both by app_dial as well as by ast_answer() when a call is answered in case any tones were being generated on a calling channel. This control frame type was made to stop traversing local channel pairs as an optimization, because it was thought that it was unnecessary to send these indications, and allowing such unnecessary control frames to traverse the local channels would cause the local channels to optimize away less quickly. As it turns out, through some special magic dialplan code, it is possible to have a tones being played on a non-local channel, and it is important for the local channel to convey that the tones should be stopped. The result of having tones continue to be played on the non-local channel is that the tones play even once the channel has been bridged. By not blocking the -1 control frame type, we can ensure that this situation does not happen. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I40227e4249d6d61dc6a9b08bbe9ee3aa18be7e30	11 years ago
Joshua Colp	5606da754a	audiohook: Read the correct number of samples based on audiohook format. Due to changes in audiohooks to support different sample rates the underlying storage of samples is in the format of the audiohook itself and not of the format being requested. This means that if a channel is using G722 the samples stored will be at 16kHz. If something subsequently reads from the audiohook at a format which is not the same sample rate as the audiohook the number of samples needs to be adjusted. Given the following example: 1. Channel writing into audiohook at 16kHz (as it is using G722). 2. Chanspy reading from audiohook at 8kHz. The original code would read 160 samples from the audiohook for each 20ms of audio. This is incorrect. Since the audio in the audiohook is at 16kHz the actual number needing to be read is 320. Failure to read this much would cause the audiohook to reset itself constantly as the buffer became full. This change adjusts the requested number of samples by determining the duration of audio requested and then calculating how many samples that would be in the audiohook format. ASTERISK-25247 #close Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d	11 years ago
Rusty Newton	43a52df20d	Documentation: chan_sip doesn't support WS or WSS in outbound register. * In sip.conf.sample fix sentence where we said that WS or WSS are supported transports for use in an outbound register definition. They are not supported in that case. ASTERISK-24853 #close Reported by: PSDK Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca	11 years ago
Michael Cargile	bd6fd55ea4	res/res_musiconhold: Add a warning when MOH does not exist Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b	11 years ago
Patric Marschall	0433f08903	sig_pri.h: force_restart_unavailable_chans in wrong scope In channels/sig_pri.h, struct sig_pri_span, the field force_restart_unavailable_chans is only defined if #if defined(HAVE_PRI_MCID) is true. All other occurences of force_restart_unavailable_chans are outside of the #if defined(HAVE_PRI_MCID) endif scope. ASTERISK-25257 #close Reported by: Patric Marschall Change-Id: I071de89cc2cd0d85927a013036e235851f672549	11 years ago
Matt Jordan	b36d670dc1	apps/app_dictate: Fix typo in attribution Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian (GameGamer43) for pointing that out. Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106	11 years ago
Matt Jordan	883f6f5d21	tests/test_devicestate: Add additional tests for the device state API This patch adds more tests that exercise the device state API. This includes: * Tests that cover adding a device state provider, as well as deleting a device state provider. This also verifies that you cannot add an already added device state provider, and cannot delete an already deleted device state provider. * A test that covers changing device state and receiving said updates from a device state subscriber. This also covers hitting both the device state cache as well as a custom device state provider. * A test that covers converting device state to channel state and device state values to a string representation and back. * A test that covers obtaining device state from an active channel and a channel driver that provides its own device state. Change-Id: I2adca67ffb405cd8625a5d6df1e3f9b3d945c08d	11 years ago
Matt Jordan	e545c05e35	main/devicestate: Prevent duplicate registration of device state providers Currently, the device state provider API will allow you to register a device state provider with the same case insensitive name more than once. This could cause strange issues, as the duplicate device state providers will not be queried when a device's state has to be polled. This patch updates the API such that a device state provider with the same name as one that has already registered will be rejected. Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2	11 years ago
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	11 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	11 years ago
Joshua Colp	895dbe532c	Merge "chan_sip: Fix early call pickup caused deadlock." into 11	11 years ago
Joshua Colp	6d9115767f	Merge "chan_vpb.cc: Fix compiler warning Jenkins found." into 11	11 years ago
Joshua Colp	89b0aaa652	Merge "res_timing: Don't close FD 0 when out of open files." into 11	11 years ago
Joshua Colp	077bfe7455	Merge "rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format." into 11	11 years ago
Joshua Colp	da5725dbc7	Merge "astfd: Fix buffer overflow in DEBUG_FD_LEAKS." into 11	11 years ago
Joshua Colp	aa04d4387a	Merge "chan_mgcp: Don't call close on fd -1." into 11	11 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	11 years ago
Walter Doekes	f6bbc4f16e	chan_mgcp: Don't call close on fd -1. ASTERISK-25220 #close Change-Id: Ic48f3a82f51ada87f2fb0e016c9efe0ad56f1ee3	11 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	11 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	11 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	11 years ago
Richard Mudgett	1e186dc857	chan_vpb.cc: Fix compiler warning Jenkins found. Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0	11 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	11 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	11 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	11 years ago
Matt Jordan	353dd68cd0	Merge "chan_sip: Prevent deadlock when performing BYE with Also transfer." into 11	11 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	11 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	11 years ago
Joshua Colp	c164bd2613	Merge "weakref attribute detection broken with gcc 4.6 and higher" into 11	11 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	11 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	11 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	11 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	11 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	11 years ago
Matt Jordan	8f431d679f	Merge "audiohook.c: Difference in read/write rates caused continuous buffer resets" into 11	11 years ago
Matt Jordan	5c047d4b0d	Merge "Logger: Reset defaults before processing config." into 11	11 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	11 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	11 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	11 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	11 years ago
Joshua Colp	81b0789906	Merge "main/manager.c: Bugfix sort action_manager by alphabetically" into 11	11 years ago
Joshua Colp	3bc8c96443	Merge "General: Fix recent menuselect-related cross compile regression" into 11	11 years ago
Joshua Colp	ffb60909a3	Merge "chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision." into 11	11 years ago
Joshua Colp	63d9ae01ee	Merge "res_config_mysql: Fix broken column type checking" into 11	11 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	11 years ago

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