asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	6630c560fd	chan_iax2: Reduce indentation in __attempt_transmit(). * Reduce indentation in __attempt_transmit(). * Don't update the static last error time variable every time in __schedule_action() and socket_read(). ........ Merged revisions 398456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	aa8405923a	chan_iax2: Fix stray reference to worker thread idle_list. * Fix stray reference to idle_list in cleanup_thread_list(). This may be the reason for the note in iax2_process_thread() about threads not being removed from the task lists. * Move cleanup_thread_list(&idle_list) to after the other lists are cleaned up. ........ Merged revisions 398416 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	ed61d21419	chan_iax2: Fix bridgecallno deadlock avoidance. * Fix bridgecallno deadlock avoidance. When doing deadlock avoidance, you need to retest the status of values for each loop to see if you still need the lock for bridgecallno. * As a safety check, after acquiring the bridgecallno lock you should check if iaxs[bridgecallno] is NULL just like the current callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE to after processing any deferred frames to ensure that the iostate is IDLE when it is placed back into the idle list. defer_full_frame() tries to ensure iax2_process_thread() wakes up to process the frame. ........ Merged revisions 398379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a4db8b381a	chan_iax2: Add missing control frame names to debug frame decode output. (Part 2) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5f7e74ba56	chan_iax2: Add missing control frame names to debug frame decode output. ........ Merged revisions 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a3241cb426	chan_misdn: Fix misdn debug output printed with arbitrary verbose levels. Fix the misdn debug output to remote consoles. chan_misdn uses ast_console_puts() which doesn't know about verbose levels. Better to use ast_verbose() instead. Without this patch the misdn debug messages are appended to the verbose level which ever was set by the message sent to the console before, i.e. any undefined level. (closes issue AST-1218) Reported by: Guenther Kelleter Patches: misdnlog.patch (license #6372) patch uploaded by Guenther Kelleter ........ Merged revisions 398235 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	71857a4a5e	Fix various memory leaks main/config.c - cleanup cache fie includes res/res_security_log.c - unregister logger level channesl/chan_sip.c - cleanup io context and notify_types main/translator.c - cleanup at shutdown main/named_acl.c - cleanup cli commands main/indications.c - ast_get_indication_tone() unref default_tone_zone if used (closes issues ASTERISK-22378) Reported by: Corey Farrell Patches: config_shutdown.patch uploaded by coreyfarrell (license 5909) res_security_log.patch uploaded by coreyfarrell (license 5909) chan_sip-11.patch uploaded by coreyfarrell (license 5909) indications_refleak.patch uploaded by coreyfarrell (license 5909) named_acl-cli_unreg-11.patch uploaded by coreyfarrell (license 5909) translate_shutdown.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 398102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	15994e3bf7	Verbose logging discrepancies Refactored cases where a combination of ast_verbose/options_verbose were present. Also in general tried to eliminate, in as many places as possible, where the options_verbose global variable was being used. Refactored the way local and remote consoles handle verbose message logging in an attempt to solve the various discrepancies that sometimes would show between the two. (closes issue AST-1193) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2798/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c58bab8ce3	AST-2013-005: Fix crash caused by invalid SDP If the SIP channel driver processes an invalid SDP that defines media descriptions before connection information, it may attempt to reference the socket address information even though that information has not yet been set. This will cause a crash. This patch adds checks when handling the various media descriptions that ensures the media descriptions are handled only if we have connection information suitable for that media. Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing the solution to this problem. (closes issue ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches: issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674) ........ Merged revisions 397756 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397757 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	fdc86bb44c	Fix uninitialized value in struct ast_control_pvt_cause_code usage. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4fd979228d	AST-2013-004: Fix crash when handling ACK on dialog that has no channel A remote exploitable crash vulnerability exists in the SIP channel driver if an ACK with SDP is received after the channel has been terminated. The handling code incorrectly assumed that the channel would always be present. This patch adds a check such that the SDP will only be parsed and applied if Asterisk has a channel present that is associated with the dialog. Note that the patch being applied was modified only slightly from the patch provided by Walter Doekes of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches: issueA21064_fix.patch uploaded by wdoekes (License 5674) ........ Merged revisions 397710 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397711 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0cd0977454	Fix memory corruption when trying to get "core show locks". Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch in memory pools but had a math error determining the buffer size and didn't address other similar memory pool mismatches. * Effectively reverted the previous patch to go in the same direction as trunk for the returned memory pool of ast_bt_get_symbols(). * Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is defined. * Fixed some formatting in ast_bt_get_symbols(). * Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is enabled. * Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when MALLOC_DEBUG is enabled. * Moved __dump_backtrace() because of compile issues with the utils directory. (closes issue ASTERISK-22221) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2778/ ........ Merged revisions 397525 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	142c5d4816	Prevent a crash on outbound SIP MESSAGE requests. If a From header on an outbound out-of-call SIP MESSAGE were malformed, the result could crash Asterisk. In addition, if a From header on an incoming out-of-call SIP MESSAGE request were malformed, the message was happily accepted rather than being rejected up front. The incoming message path would not result in a crash, but the behavior was bad nonetheless. (closes issue ASTERISK-22185) reported by Zhang Lei git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	88a5f18dec	Fix Not Storing Current Incoming Recv Address In 1.8, r384779 introduced a regression by retrieving an old dialog and keeping the old recv address since recv was already set. This has caused a problem when a proxy is involved since responses to incoming requests from the proxy server, after an outbound call is established, are never sent to the correct recv address. In 11, r382322 introduced this regression. The fix is to revert that change and always store the recv address on incoming requests. Thank you Walter Doekes for helping to point out this error and Mark Michelson for your input/review of the fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin Tested by: Alex Zarubin, Karsten Wemheuer Patches: asterisk-22071-store-recvd-address.diff by Michael L. Young (license 5026) ........ Merged revisions 397204 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	3b91cde004	Remove REF_DEBUG definition. ........ Merged revisions 397156 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	e510fa1514	Fix refcounting of sip_pvt in test_sip_rtpqos test and unlink it from the list of pvts. (closes issue ASTERISK-22248) reported by Corey Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell (license #5909) ........ Merged revisions 397112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	f83b144899	chan_sip: Convert 'just did sched_add waitid...' from warning to debug message. Patches: reviewboard-2377.patch uploaded by Paul Belanger Review: https://reviewboard.asterisk.org/r/2377/ ........ Merged revisions 396582 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@396583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	16160ea357	chan_sip: Fix IP-addr in warning when rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/ ........ Merged revisions 396579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@396580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	8d9eff176e	- Fix different issues with call transfer cancel. In case 3rd party busy or congestion call was not returned.	12 years ago
Michael L. Young	1e03a50878	Fix Registration Failure When A Peer And TLS Are Used If a peer is used in a register line and TLS is defined as the transport, the registration fails since the transport on the dialog is never set properly resulting in UDP being used instead of TLS. This patch sets the dialog's transport based on the transport that was defined in the register line. If the register line does not specify a transport, the parsing function for the register line always defaults back to UDP. (closes issue ASTERISK-21964) Reported by: Doug Bailey Tested by: Doug Bailey Patches: asterisk-21964-set-reg-dialog-transport.diff by Michael L. Young (license 5026) ........ Merged revisions 396240 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@396248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Moises Silva	bc78bfee41	Fix a longstanding issue with MFC-R2 configuration that prevented users from mixing different variants or general MFC-R2 settings within the same E1 line. Most users do not have a problem with this since MFC-R2 lines are usually fractional E1s, or the whole E1 has the same country variant and R2 settings. In Venezuela however is common to have inbound MFC-R2 and outbound DTMF-R2 within the same E1. This fix now properly parses the chan_dahdi.conf file to generate a new openr2 context every time a new channel => section is found and the configuration was changed. (closes issue ASTERISK-21117) Reported by: Rafael Angulo Related Elastix issue: http://bugs.elastix.org/view.php?id=1612 ........ Merged revisions 394106 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c03b11466d	chan_dahdi: Fix segfault reloading chan_dahdi when round robin is used. * Clear round_robin[] in dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 393627 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@393628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	9ce8896d15	Fix issue with inability to cancell call transfer made by on-sceen menus.	12 years ago
Matthew Jordan	2ffb648a20	Fix memory/ref counting leaks in a variety of locations This patch fixes the following memory leaks: * http.c: The structure containing the addresses to bind to was not being deallocated when no longer used * named_acl.c: The global configuration information was not disposed of * config_options.c: An invalid read was occurring for certain option types. * res_calendar.c: The loaded calendars on module unload were not being properly disposed of. * chan_motif.c: The format capabilities needed to be disposed of on module unload. In addition, this now specifies the default options for the maxpayloads and maxicecandidates in such a way that it doesn't cause the invalid read in config_options.c to occur. (issue ASTERISK-21906) Reported by: John Hardin patches: http.patch uploaded by jhardin (license 6512) named_acl.patch uploaded by jhardin (license 6512) config_options.patch uploaded by jhardin (license 6512) res_calendar.patch uploaded by jhardin (license 6512) chan_motif.patch uploaded by jhardin (license 6512) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@392810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	13b2c25687	Fix issue with no sound in both way in case of previous call to chan_unistim phone was canceled.	12 years ago
Alec L Davis	a90ad16e55	IAX2: Transfer Reject: Lock bridgecallno before touching it, refactor 1). When touching the bridgecallno, we need to lock it. 2). Remove magic number '0' and replace with TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce indentation. Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2613/ ........ Merged revisions 391333 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	f09521a0d5	chan_iax2: nativebridge refactor, missed unlock bridgecallno ........ Merged revisions 391143 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	30cfce07f7	fix bad edit after conflict resolution ........ Merged revisions 391107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	20b9dac9fc	IAX2: refactor nativebridge transfer remove triple checking of iaxs[fr->callno]->transferring reduce indentation. Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2602/ ........ Merged revisions 391065 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	9fca44e6d4	IAX2: fix race condition with nativebridge transfers. 1). When touching the bridgecallno, we need to lock it. 2). stop_stuff() which calls iax2_destroy_helper() Assumes the lock on the pvt is already held, when iax2_destroy_helper() is called. Thus we need to lock the bridgecallno pvt before we call stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When evaluating the state of 'callno->transferring' of the current leg, we can't change it to READY unless the bridgecallno is locked. Why, if we are interrupted by the other call leg before 'transferring = TRANSFER_RELEASED', the interrupt will find that it is READY and that the bridgecallno is also READY so Releases the legs. (closes issue ASTERISK-21409) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2594/ ........ Merged revisions 391062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	97ca159774	Fix several problems caused by multiple line usage with i2004 phones.	12 years ago
Michael L. Young	eec46f56f4	Fix Crash Caused By One-way Audio With auto_* NAT Settings Fix The prior code committed, r385473, failed to take into consideration that not all outgoing calls will be to a peer. My fault. This patch does the following: * Check if there is a related peer involved. If there is, check and set NAT settings according to the peer's settings. * Fix a problem with realtime peers. If the global setting has auto_force_rport set and we issued a "sip reload" while a peer is still registered, the peer's flags for NAT are reset to off. When this happens, we were always setting the contact address of the peer to that of the full contact info that we had. (closes issue ASTERISK-21374) Reported by: jmls Tested by: Michael L. Young Patches: asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2524/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f296671ec5	Allow mISDN to send PROGRESS messsage. * Made isdn_msg_parser.c build a progress message with the mandatory progress indicator IE. (The mISDNuser NT state machine rejected sending the incomplete message.) Note: The associated mISDN and mISDNuser patches respectively are viewable here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200 http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes issue AST-1153) Reported by: Guenther Kelleter Patches: progress-chan_misdn.diff (license #6372) patch uploaded by Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch uploaded by Guenther Kelleter progress-misdnuser.diff (license #6372) mISDNuser patch uploaded by Guenther Kelleter ........ Merged revisions 388425 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	771ce9e1e7	Fix copy/paste error in one-touch-recording implementation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	527a611c80	chan_sip: NOTIFYs for BLF start queuing up and fail to be sent out after retries fail RFC6665 4.2.2: ... after a failed State NOTIFY transaction remove the subscription The problem is that the State Notify requests rely on the 200OK reponse for pacing control and to not confuse the notify susbsystem. The issue is, the pendinginvite isn't cleared if a response isn't received, thus further notify's are never sent. The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the subscription after failure. (closes issue ASTERISK-21677) Reported by: Dan Martens Tested by: Dan Martens, David Brillert, alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2475/ ........ Merged revisions 387875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	aec4d2f239	chan_sip: Session-Expires: Set timer to correctly expire at (~2/3) of the interval when not the refresher RFC 4028 Section 10 if the side not performing refreshes does not receive a session refresh request before the session expiration, it SHOULD send a BYE to terminate the session, slightly before the session expiration. The minimum of 32 seconds and one third of the session interval is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the Session-Expires interval, or if the remote device was the refresher, asterisk would timeout at interval end. Now, when not refresher, timeout as per RFC noted above. (closes issue ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2488/ ........ Merged revisions 387344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	2846881045	chan_sip: Honor Session-Expires in 200OK response when it's a RE-INVITE when asterisk is the refresher. RFC 4028 Section 7.2 "UACs MUST be prepared to receive a Session-Expires header field in a response, even if none were present in the request." What changed After ASTERISK-20787, inbound calls to asterisk with no Session-Expires in the INVITE are now are offered a Session-Expires (1800 asterisk default) in the response, with asterisk as the refresher. Symptom: After 900 seconds (asterisk default refresher period 1800), asterisk RE-INVITEs the device, the device may respond with a much lower Session-Expires (180 in our case) value that it is now using. Asterisk ignores this response, as it's deemed both an INBOUND CALL, and a RE-INVITE. After 180 seconds the device times out and sends BYE (hangs up), asterisk is still working with the refresher period of 1800 as it ignored the 'Session Expires: 180' in the previous 200OK response. Fix: handle_response_invite() when 200OK, remove check for outbound and reinvite. (closes issue ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2463/ ........ Merged revisions 387312 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	a08c0c7e5d	chan_dahdi: fix lower bound check with -ve integer conversion from a float Lower bound of a 16bit signed int is -32768 not -32767 (closes issue ASTERISK-21744) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) ........ Merged revisions 387297 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	95dcae4aa6	Prevent crash in 'sip show peers' when the number of peers on a system is large When you have lots of SIP peers (according to the issue reporter, around 3500), the 'sip show peers' CLI command or AMI action can crash due to a poorly placed string duplication that occurs on the stack. This patch refactors the command to not allocate the string on the stack, and handles the formatting of a single peer in a separate function call. (closes issue ASTERISK-21466) Reported by: Guillaume Knispel patches: fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch uploaded by gknispel (License 6492) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	9d809c0f42	Fix Displaying Symmetric RTP Global Setting * Use comedia_string() to display correctly the symmetric rtp setting when running "sip show settings" git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	99f3a897fb	Change Case On Forcerport For Consistency * Change "ForcerPort" to "Forcerport" to match everywhere else it is displayed ........ Merged revisions 386483 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9c315f85c1	Don't attempt to create a voice frame on a read error Prior to this patch, a read error in snd_pcm_readi would still be treated as a nominal result when constructing a voice frame from the expected data. Since the value returned is negative, as opposed to the number of samples read, this could result in a crash. With this patch, we now return a null frame when a read error is detected. Note that the patch on ASTERISK-21329 was modified slightly for this commit, in that we bail immediately on detecting the read error, rather than bypassing the construction of the voice frame. (closes issue ASTERISK-21329) Reported by: Keiichiro Kawasaki patches: chan_alsa.diff uploaded by kawasaki (License 6489) ........ Merged revisions 385633 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	f07cccecfd	Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX When we reload Asterisk or chan_sip, the flags force_rport and comedia that are turned on and off when using the auto_force_rport and auto_comedia nat settings go back to the default setting off. These flags are turned on when needed or off when not needed at the time that a peer registers, re-registers or initiates a call. This would apply even when only the default global setting "nat=auto_force_rport" is being used, which in this case would only affect the force_rport flag. Everything is good except for the following: The nat setting is set to auto_force_rport and auto_comedia. We reload Asterisk and the peer's registration has not expired. We load in the settings for the peer which turns force_rport and comedia back to off. Since the peer has not re-registered or placed a call yet, those flags remain off. We then initiate a call to the peer from the PBX. The force_rport and comedia flags stay off. If NAT is involved, we end up with one-way audio since we never checked to see if the peer is behind NAT or not. This patch does the following: * Moves the checking of whether a peer is behind NAT into its own function * Create a function to set the peer's NAT flags if they are using the auto_* NAT settings * Adds calls in sip_request_call() to these new functions in order to setup the dialog according to the peer's settings (closes issue ASTERISK-21374) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2421/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	82e70b2128	IAX2 defer_full_frames fail to get sent Ensure iax2_process_thread is signalled when a deferred frame is queued to it. (issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2426/ ........ Merged revisions 385429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	4a06abfee4	IAX2, prevent network thread starting before all helper threads are ready On startup, it's possible for a frame to arrive before the processing threads were ready. In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive before we are at ast_cond_wait, the signal will be ignored. The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued. Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled after each thread is started. (issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2427/ ........ Merged revisions 385402 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9511761e81	Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE When a BYE request is processed in chan_sip, the current SIP dialog is detached from its associated Asterisk channel structure. The tech_pvt pointer in the channel object is set to NULL, and the dialog persists for an RFC mandated period of time to handle re-transmits. While this process occurs, the channel is locked (which is good). Unfortunately, operations that are initiated externally have no way of knowing that the channel they've just obtained (which is still valid) and that they are attempting to lock is about to have its tech_pvt pointer removed. By the time they obtain the channel lock and call the channel technology callback, the tech_pvt is NULL. This patch adds a few checks to some channel callbacks that make sure the tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit callbacks, which would crash if AMI initiated a DTMF on the channel at the same time as a BYE was received from the UA. This patch also adds checks on sip_transfer (as AMI can also cause a callback into this function), as well as sip_indicate (as lots of things can queue an indication onto a channel). Review: https://reviewboard.asterisk.org/r/2434/ (closes issue ASTERISK-20225) Reported by: Jeff Hoppe ........ Merged revisions 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	74c57919a4	Fix For Not Overriding The Default Settings In chan_sip The initial report was that the "nat" setting in the [general] section was not having any effect in overriding the default setting. Upon confirming that this was happening and looking into what was causing this, it was discovered that other default settings would not be overriden as well. This patch works similar to what occurs in build_peer(). We create a temporary ast_flags structure and using a mask, we override the default settings with whatever is set in the [general] section. In the bug report, the reporter who helped to test this patch noted that the directmedia settings were being overriden properly as well as the nat settings. This issue is also present in Asterisk 1.8 and a separate patch will be applied to it. (issue ASTERISK-21225) Reported by: Alexandre Vezina Tested by: Alexandre Vezina, Michael L. Young Patches: asterisk-21225-handle-options-default-prob_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2385/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384827 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	fe8c92adc8	chan_dahdi: Add inband_on_proceeding compatibility option. The new inband_on_proceeding option causes Asterisk to assume inband audio may be present when a PROCEEDING message is received. Q.931 Section 5.1.2 says the network cannot assume that the CPE side has attached to the B channel at this time without explicitly sending the progress indicator ie informing the CPE side to attach to the B channel for audio. However, some non-compliant ISDN switches send a PROCEEDING without the progress indicator ie indicating inband audio is available and assume that the CPE device has connected the media path for listening to ringback and other messages. ASTERISK-17834 which causes this issue was dealing with a non-compliant network switch. (closes issue ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett ........ Merged revisions 384685 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	ef79c00991	Address uninitialized conditional that valgrind found ........ Merged revisions 384162 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b984d78c5c	AST-2013-003: Prevent username disclosure in SIP channel driver When authenticating a SIP request with alwaysauthreject enabled, allowguest disabled, and autocreatepeer disabled, Asterisk discloses whether a user exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The information is disclosed when: * A "407 Proxy Authentication Required" response is sent instead of a "401 Unauthorized" response * The presence or absence of additional tags occurs at the end of "403 Forbidden" (such as "(Bad Auth)") * A "401 Unauthorized" response is sent instead of "403 Forbidden" response after a retransmission * Retransmission are sent when a matching peer did not exist, but not when a matching peer did exist. This patch resolves these various vectors by ensuring that the responses sent in all scenarios is the same, regardless of the presence of a matching peer. This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of the testing and the solution to this problem was done by Walter as well - a huge thanks to his tireless efforts in finding all the ways in which this setting didn't work, providing automated tests, and working with Kinsey on getting this fixed. (closes issue ASTERISK-21013) Reported by: wdoekes Tested by: wdoekes, kmoore patches: AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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