asterisk

Commit Graph

Author	SHA1	Message	Date
Walter Doekes	2fcf45e9cf	chan_sip: Fix From header truncation for extremely long CALLERID(name). The CALLERID(num) and CALLERID(name) and other info are placed into the `char from[256]` in initreqprep. If the name was too long, the addr-spec and params wouldn't fit. Code is moved around so the addr-spec with params is placed there first, and then fitting in as much of the display-name as possible. ASTERISK-25396 #close Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260	10 years ago
Joshua Colp	59636e82b2	chan_sip: Allow call pickup to set the hangup cause. The call pickup implementation in chan_sip currently sets the channel hangup cause to "normal clearing" if call pickup is successfully performed. This action overwrites the "answered elsewhere" hangup cause set by the call pickup code and can result in the SIP device in question showing a missed call when it should not. This change sets the hangup cause to "normal clearing" as a default initially but allows the call pickup to change it as needed. ASTERISK-25346 #close Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff	10 years ago
Mark Michelson	6446061614	Merge "chan_sip.c: wrong peer searched in sip_report_security_event" into 11	10 years ago
Kevin Harwell	430db4333e	chan_sip.c: wrong peer searched in sip_report_security_event In chan_sip, after handling an incoming invite a security event is raised describing authorization (success, failure, etc...). However, it was doing a lookup of the peer by extension. This is fine for register messages, but in the case of an invite it may search and find the wrong peer, or a non existent one (for instance, in the case of call pickup). Also, if the peers are configured through realtime this may cause an unnecessary database lookup when caching is enabled. This patch makes it so that sip_report_security_event searches by IP address when looking for a peer instead of by extension after an invite is processed. ASTERISK-25320 #close Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4	10 years ago
Richard Mudgett	c777c9565d	chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF. Pressing DTMF digits on a phone to go out on a DAHDI channel can result in the digit not being recognized or even heard by the peer. Phone -> Asterisk -> DAHDI/channel Turns out the DAHDI behavior with DTMF generation (and any other generated tones) is exposed by the "buffers=" setting in chan_dahdi.conf. When Asterisk requests to start sending DTMF then DAHDI waits until its write buffer is empty before generating any samples for the DTMF tones. When Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI immediately stops generating the DTMF samples. As a result, the more samples there are in the DAHDI write buffer the shorter the time DTMF actually gets sent on the wire. If there are more samples in the write buffer than the time DTMF is supposed to be sent then no DTMF gets sent on the wire. With the "buffers=12,half" setting and each buffer representing 20 ms of samples then the DAHDI write buffer is going to contain around 120 ms of samples. For DTMF to be recognized by the peer the actual sent DTMF duration needs to be a minimum of 40 ms. Therefore, the intended duration needs to be a minimum of 160 ms for the peer to receive the minimum DTMF digit duration to recognize it. A simple and effective solution to work around the DAHDI behavior is for Asterisk to flush the DAHDI write buffer when sending DTMF so the full duration of DTMF is actually sent on the wire. When someone is going to send DTMF they are not likely to be talking before sending the tones so the flushed write samples are expected to just contain silence. * Made dahdi_digit_begin() flush the DAHDI write buffer after requesting to send a DTMF digit. ASTERISK-25315 #close Reported by John Hardin Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a	10 years ago
Richard Mudgett	f43ea74e9e	chan_dahdi.c: Lock private struct for ast_write(). There is a window of opportunity for DTMF to not go out if an audio frame is in the process of being written to DAHDI while another thread starts sending DTMF. The thread sending the audio frame could be past the currently dialing check before being preempted by another thread starting a DTMF generation request. When the thread sending the audio frame resumes it will then cause DAHDI to stop the DTMF tone generation. The result is no DTMF goes out. * Made dahdi_write() lock the private struct before writing to the DAHDI file descriptor. ASTERISK-25315 Reported by John Hardin Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb	10 years ago
Richard Mudgett	5311d18101	chan_sip.c: Move NULL check to where it will do some good. v11 only fix. Change-Id: I340512f86cfd3a6f7703971fa8acfffc7d47132b	10 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	10 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	10 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	10 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	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
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
Richard Mudgett	1e186dc857	chan_vpb.cc: Fix compiler warning Jenkins found. Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0	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	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
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
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
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
Richard Mudgett	f3bc0cc70f	v11: More files to ignore. Change-Id: If5eef47d03399ff93e3f2f490780144971f6b64a	10 years ago
Richard Mudgett	b54f5fda05	chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option. Some telco switches occasionally ignore ISDN RESTART requests. The fix for ASTERISK-19608 added an escape clause for B channels in the restarting state if the telco ignores a RESTART request. If the telco fails to acknowledge the RESTART then Asterisk will assume the telco acknowledged the RESTART on the second call attempt requesting the B channel by the telco. The escape clause is good for dealing with RESTART requests in general but it does cause the next call for the restarting B channel to be rejected if the telco insists the call must go on that B channel. chan_dahdi doesn't really need to issue a RESTART request in response to receiving a cause 44 (Requested channel not available) code. Sending the RESTART in such a situation is not required (nor prohibited) by the standards. I think chan_dahdi does this for historical reasons to deal with buggy peers to get channels unstuck in a similar fashion as the chan_dahdi.conf resetinterval option. * Add the chan_dahdi.conf force_restart_unavailable_chans compatability option that when disabled will prevent chan_dahdi from trying to RESTART the channel in response to a cause 44 code. ASTERISK-25034 #close Reported by: Richard Mudgett Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65	10 years ago
Richard Mudgett	e4b956b186	chan_dahdi/sig_pri: Make post AMI HangupRequest events on PRI channels. The chan_dahdi channel driver is a very old driver. The ability for it to support ISDN was added well after the initial analog support. Setting the softhangup flags is a carry over from the original analog code. The driver was not updated to call ast_queue_hangup() which will post the AMI HangupRequest event. * Changed sig_pri.c to call ast_queue_hangup() instead of setting the softhangup flag when the remote party initiates a hangup. ASTERISK-24895 #close Reported by: Andrew Zherdin Change-Id: I5fe2e48556507785fd8ab8e1c960683fd5d20325	10 years ago
George Joseph	b44aeb6998	.gitignore updates for 11 Added bootstrap products Added channels/h323/Makefile Added res/pjproject Change-Id: I6b3bc56bf7bdaee0554f36fc2ce3a77e9eaf8aa3	10 years ago
Matthew Jordan	efef678411	channels/chan_iax2: Improve POKE expiration time calculation for lossy networks POKE is used to check for peer availability; however, in networks with packet loss, the current calculations may result in POKE expiration times that are too short. This patch alters the expiration/retry time logic to take into account the last known qualify round trip time, as opposed to always using a static value for each peer. Review: https://reviewboard.asterisk.org/r/4536 ASTERISK-22352 #close Reported by: Frederic Van Espen ASTERISK-24894 #close Reported by: Y Ateya patches: poke_noanswer_duration.diff submitted by Y Ateya (License 6693) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@434564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	7c4efc490a	clang compiler warnings: Fix autological comparisons This fixes autological comparison warnings in the following: * chan_skinny: letohl may return a signed or unsigned value, depending on the macro chosen * func_curl: Provide a specific cast to CURLoption to prevent mismatch * cel: Fix enum comparisons where the enum can never be negative * enum: Fix comparison of return result of dn_expand, which returns a signed int value * event: Fix enum comparisons where the enum can never be negative * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be negative * presencestate: Use the actual enum value for INVALID state * security_events: Fix enum comparisons where the enum can never be negative * udptl: Don't bother to check if the return value from encode_length is less than 0, as it returns an unsigned int * translate: Since the parameters are unsigned int, don't bother checking to see if they are negative. The cast to unsigned int would already blow past the matrix bounds. Review: https://reviewboard.asterisk.org/r/4533 ASTERISK-24917 Reported by: dkdegroot patches: rb4533.patch submitted by dkdegroot (License 6600) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@434469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	ed191ca32c	chan_iax2: Fix mixup of code/declarations Interestingly enough, clang doesn't care about this. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@434334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	e61628db7b	chan_iax2: Fix crash caused by unprotected access to iaxs[peer->callno] This patch fixes an access to the peer callnumber that is unprotected by a corresponding mutex. The peer->callno value can be changed by multiple threads, and all data inside the iaxs array must be procted by a corresponding lock of iaxsl. The patch moves the unprotected access to a location where the mutex is safely obtained. Review: https://reviewboard.asterisk.org/r/4599/ ASTERISK-21211 #close Reported by: Jaco Kroon patches: asterisk-11.2.1-iax2_poke-segfault.diff submitted by Jaco Kroon (License 5671) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@434291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	7dd8c34e82	chan_sip: Handle IPv4 mapped IPv6 clients when NAT is enabled When udpbindaddr is set to the IPv6 bind all address of '::', Asterisk will attempt to handle both IPv4 and IPv6 addresses, although the information will be stored in a struct with an AF_INET6 address type. However, the current NAT handling code won't handle the IPv4 mapped IPv6 addresses correctly. This patch adds an additional check for the mapped address case, allowing the NAT code to handle clients even when the address is IPv6. Review: https://reviewboard.asterisk.org/r/4563/ ASTERISK-18032 #close Reported by: Christoph Timm patches: nat_with_ipv6.diff submitted by Valentin Vidić (License 6697) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@434288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	c224c44a16	clang compiler warnings: Fix non-literal-null-conversion warnings Clang will flag errors when a char pointer is set to '\0', as opposed to a value that the char pointer points to. This patch fixes this warning in a variety of locations. Review: https://reviewboard.asterisk.org/r/4551 ASTERISK-24917 Reported by: dkdegroot patches: rb4551.patch submitted by dkdegroot (License 6600) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@434187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
George Joseph	5e46127e17	build: Fixes for gcc 5 compilation These are fixes for compilation under gcc 5.0... chan_sip.c: In parse_request needed to make 'lim' unsigned. inline_api.h: Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99 inline semantics (same as clang). ccss.c: In ast_cc_set_parm, needed to fix weird comparison. dsp.c: Needed to work around a possible compiler bug. It was throwing an array-bounds error but neither sgriepentrog, rmudgett nor I could figure out why. manager.c: In action_atxfer, needed to correct an array allocation. This patch will go to 11, 13, trunk. Review: https://reviewboard.asterisk.org/r/4581/ Reported-by: Jeffrey Ollie Tested-by: George Joseph ASTERISK-24932 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@434113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	e159080c45	chan_sip: Fix expression in unit test /channels/chan_sip/test_sip_rtpqos. Fix misplaced parentheses in original fabs() expression. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@433816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	297b8df31b	clang compiler warnings: Fix -Wabsolute-value warnings This patch fixes several warnings caught by clang - in this case, usage of the abs function on non-integer values. This patch uses labs and fabs, as appropriate, in the various affected files. Review: https://reviewboard.asterisk.org/r/4525 ASTERISK-24917 Reported by: dkdegroot patches: rb4525.patch submitted by dkdegroot (License 6600) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@433749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	b8d7aa442d	clang compiler warnings: Fix invalid enum conversion This patch fixes some invalid enum conversion warnings caught by clang. In particular, several functions in chan_sip mixed usage of the st_refresher_param enum and st_refresher enum. This patch corrects that. ASTERISK-24917 Reported by: dkdegroot patches: rb4535.patch submitted by dkdegroot (License 6600) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@433746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	dc6249a271	clang compiler warnings: Fix -Wbitfield-constant-conversion warning In chan_iax2, we attempt to assign a -1 to a bitfield. This gets caught by clang, as it will truncate the -1 to a 1 implicitly. Instead, we just assign the value a '1'. Review: https://reviewboard.asterisk.org/r/4537/ ASTERISK-24917 Reported by: dkdegroot patches: rb4537.patch submitted by dkdegroot (License 6600) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@433683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	297f7b16ff	clang compiler warnings: Fix -Wunused-function This patch fixes clang compilers warnings for unused functions. Specifically: * channels/chan_iax2: removed user_ref function * main/dsp.c: removed goertzel_update function Review: https://reviewboard.asterisk.org/r/4527 ASTERISK-24917 Reported by: dkdegroot patches: rb4527.patch submitted by dkdegroot (License 6600) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@433678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	648fd9e2dd	chan_sip: Fix dialog reference leaked to scheduler for reinvite_timeout. Release the scheduler reference to the dialog for reinvite timeout during dialog_unlink_all. ASTERISK-24876 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4491/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@433112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	3e6e5a977e	chan_dahdi/sig_analog: Fix distinctive ring detection to suck less. The distinctive ring feature interferes with detecting Caller ID and appears to have been broken for years. What happens is if you have a ring-ring cadence as used in the UK you get too many DAHDI events for the distinctive ring pattern array and Caller ID detection is aborted. I think when Zapata/DAHDI added the ring begin event it broke distinctive ring. More events happen than before and the code does no filtering of which event times are recorded in the pattern array. * Made distinctive ring only record the ringt count when the ring ends instead of on just any DAHDI event. Distinctive ring can be ring, ring-ring, ring-ring-ring, or different ring durations for the up to three rings. * Fixed the distinctive ring detection enable (chan_dahdi.conf option usedistinctiveringdetection) to be per port instead of somewhat per port and somewhat global. This has been broken since v1.8. * Fixed using the default distinctive ring context when the detected pattern does not match any configured dringX patterns. The default context did not get set when the previous call was a matched distinctive ring pattern and the current call is not matched. This has been broken since v1.8. * Made distinctive ring have no effect on Caller ID detection when it is disabled. Caller ID detection just monitors for 10 seconds before giving up. * Fixed leak of struct callerid_state memory when a polarity reversal during Caller ID detection causes the incoming call to be aborted. DAHDI-1143 AST-1545 ASTERISK-24825 #close Reported by: Richard Mudgett ASTERISK-17588 Reported by: Daniel Flounders Review: https://reviewboard.asterisk.org/r/4444/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	2befe82721	chan_sip: Fix realtime locking inversion when poking a just built peer. When a realtime peer is built it can cause a locking inversion when the just built peer is poked. If the CLI command "sip show channels" is periodically executed then a deadlock can happen because of the locking inversion. * Push the peer poke off onto the scheduler thread to avoid the locking inversion of the just built realtime peer. AST-1540 ASTERISK-24838 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4454/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	8bb48e1f50	make: Remove 'res_features' from libraries to link against with cygwin/mingw32 Both the apps and channels Makefiles still listed 'res_features' as modules to link against when compiling for cygwin or mingw32. This module hasn't existed for quite some time. ASTERISK-18105 #close Reported by: feyfre git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	bbfc8cc778	channels/chan_sip: Don't send a BYE after final response when PBX thread fails When Asterisk fails to start a PBX thread for a new channel - for example, when the maxcalls setting in asterisk.conf is exceeded - we currently send a final response, and then attempt to send a BYE request to the UA. Since that's all sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt such that we don't get stuck sending BYE requests to something that does not want it. Note that this patch is a slight modification of the one on ASTERISK-15434. For clarity, it explicitly calls sipalreadygone with the calls to transmit a final response. ASTERISK-21845 ASTERISK-15434 #close Reported by: Makoto Dei Tested by: Matt Jordan patches: sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	6c4df2c704	channels/chan_sip: Clarify WARNING message in mismatched SRTP scenario When we receive an SDP as part of an offer/answer for a peer/friend has been configured to require encryption, and that SDP offer/answer failed to provide acceptable crypto attributes, we currently issue a WARNING that uses the phrase "we" and "requested". In this case, both of those terms are ambiguous - the user will probably think "we" is Asterisk (it most likely isn't) and it may not be a "request", so much as an SDP that was received in some fashion. This patch makes the WARNING messages slightly less bad and a bit more accurate as well. ASTERISK-23214 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	af498bf03e	channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI Prior to this patch, SDP offers negotiating SDES-SRTP crypto attributes would be rejected if those crypto attributes contained either a key lifetime or a MKI parameter. While from a theoretical point of view this was defensible - Asterisk does not support key lifetimes or multiple crypto keys - from a practical point of view, this is quite a problem. A large number of endpoints offer lifetimes/MKI, which Asterisk can tolerate so long as it doesn't actually have to support anything more than a single key or refresh the key. In reality, this is (so far as we've seen) always the case. This patch is a forward port of Olle's work in the lingon-srtp-key-lifetime-1.8 branch. To quote Olle from ASTERISK-17721, it handles lifetime/MKI parameters in the following fashion: > The Lingon branch now handle lifetime and MKI parameters. > > We only accept lifetimes up to max for the crypto and higher than 10 hours > for packetization of 20 ms (50 pps). > > We only handle MKI with index 1. > > We do not really bother with counting packets and reinviting at end of > lifetime, so the min of 10 hours kind of takes care of most calls. If there > are longer ones, we rely on the other side for re-invites. > > It's still not perfect, but I personally think this is an improvement. A > configuration option for minimum lifetime accepted could be added. When the patch was ported forward, I decided against adding a configuration option as Olle's handling was more than sufficient for every case I've seen come through the issue tracker or through interoperability testing. We can revisit that decision if it proves to be false. A few small other tweaks were made to the surrounding code to reduce indentation and provide better type safety for the 'tag' parameter. Review: https://reviewboard.asterisk.org/r/4419/ Review: https://reviewboard.asterisk.org/r/4418/ ASTERISK-17721 #close Reported by: Terry Wilson ASTERISK-17899 #close Reported by: Dwayne Hubbard patches: lingon-srtp-key-lifetime-1.8.diff uploaded by oej (License 5267) ASTERISK-20233 Reported by: tootai ASTERISK-22748 Reported by: Alejandro Mejia git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	789d4379b6	channels/chan_sip: Fix crash when transmitting packet after thread shutdown When the monitor thread is stopped, its pthread ID is set to a specific value (AST_PTHREADT_STOP) so that later portions of the code can determine whether or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit failed to check for that value, checking instead only for AST_PTHREAD_STOP. Passing the invalid yet very specific value to pthread_kill causes a crash. This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that it doesn't attempt to poke the thread if the thread has already been stopped. ASTERISK-24800 #close Reported by: JoshE git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	e16226c167	chan_dahdi/sig_analog: Put log message strings on one line. With the log messages on one line, you can search for the log message seen in the log and expect to find it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	13d0e9fe7d	chan_dahdi: Remove some dead code. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	f8254210de	Create work around for scheduler leaks during shutdown. * Added ast_sched_clean_by_callback for cleanup of scheduled events that have not yet fired. * Run all pending peercnt_remove_cb and replace_callno events in chan_iax2. Cleanup of replace_callno events is only run 11, since it no longer releases any references or allocations in 13+. ASTERISK-24451 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4425/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago

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