7564 Commits (ec68a156191e9d8d6c752b9232b46a29f90a5cbb)

Author	SHA1	Message	Date
Automerge script	096540c2c0	Merged revisions 376219 via svnmerge from ... file:///srv/subversion/repos/asterisk/trunk ........ r376219 | jrose | 2012-11-13 13:42:13 -0600 (Tue, 13 Nov 2012) | 12 lines chan_sip: Add SubscribeContext field to SIPshowpeer AMI response The new field is will show up within the response if the requested peer has a subscribe context set. (closes issue ASTERISK-20626) Reported by: Jaco Kroon Patches: asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671) -with modifications by jrose to conform to style guidelines Review: https://reviewboard.asterisk.org/r/2195/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	0874e3c825	Merged revisions 376131 via svnmerge from ... file:///srv/subversion/repos/asterisk/trunk ................ r376131 | file | 2012-11-11 11:15:47 -0600 (Sun, 11 Nov 2012) | 16 lines Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default. With ICE support enabled in chan_sip and a large number of interfaces on the system it was possible for the produced SDP to be truncated due to some fixed size buffers. These buffers have now been changed so they will dynamically grow as needed. ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is no longer enabled by default there. (closes issue ASTERISK-20643) Reported by: coopvr ........ Merged revisions 376130 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	bf369f2018	Merged revisions 376061 via svnmerge from ... file:///srv/subversion/repos/asterisk/trunk ................ r376061 | rmudgett | 2012-11-08 15:12:35 -0600 (Thu, 08 Nov 2012) | 22 lines chan_dahdi/SS7: Made reject incoming call for an in-alarm or blocked channel. If a SS7 call comes in requesting a CIC that is in-alarm, the call is accepted and connects if the extension exists in the dialplan. The call does not have any audio. * Made release the call immediately with circuit congestion cause. (closes issue ASTERISK-20204) Reported by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376059 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376060 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	f2bb9afe17	Multiple revisions 375993-375994 ... ........ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	82dc21e0e1	Fix a bug where our Motif ICE candidates were not quite proper, and make us more forgiving. ... An issue was reported on the mailing list where calling would result in an "Incomplete ICE-UDP candidate received on session" error message. This is the result of the ICE-UDP candidate code not placing a "network" attribute within the candidates. This is now done. To increase compatibility though I have removed the requirement for the "network" attribute to exist within ICE-UDP candidates that are received since we don't actually require the value. Reported on the mailing list by Jean-Denis Girard. ........ Merged revisions 375925 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a0c363e227	Refactor ast_timer_ack to return an error and handle the error in timer users ... Currently, if an acknowledgement of a timer fails Asterisk will not realize that a serious error occurred and will continue attempting to use the timer's file descriptor. This can lead to situations where errors stream to the CLI/log file. This consumes significant resources, masks the actual problem that occurred (whatever caused the timer to fail in the first place), and can leave channels in odd states. This patch propagates the errors in the timing resource modules up through the timer core, and makes users of these timers handle acknowledgement failures. It also adds some defensive coding around the use of timers to prevent using bad file descriptors in off nominal code paths. Note that the patch created by the issue reporter was modified slightly for this commit and backported to 1.8, as it was originally written for Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches: jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 375893 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375894 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375895 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	732767f230	Fix for chan_skinny leaving RTP ports open ... Skinny wasn't closing RTP sockets. This patch includes ast_rtp_instance_stop before ast_rtp_instance_destroy which fixes the problem. Also add destroy for VRTP (which I believe is unused, but exists). Review: https://reviewboard.asterisk.org/r/2176/ ........ Merged revisions 375660 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f85db0e34d	Things don't need to be that const. ... ........ Merged revisions 375658 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375659 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375661 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e950086daf	Multiple revisions 375519-375524 ... ........ r375519 | rmudgett | 2012-10-30 16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer primitives must be handled first. The frm->addr is a different "address space" than the stack/instance address of other Lx primitives. The test for B channel instance address could fail. Patches: patch01_timers.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett | 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines chan_misdn: Free memory in error paths and other memory leaks. The one line commented with BUG is not easily fixable because there is no de-init function one can call. Patches: patch02_memory.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30 16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT L2 de-establish/establish * An NT-PTMP cannot de/establish L2 since it doesn't know the TEIs. * On NT-PTP L2 is started when L1 is finally active in handle_l1. * L2 deactivation logging cleanup. * L2 aggregate link status is unknown for NT-PTMP, show as "UNKN". * Removed unused functions and code for L2 handling. Patches: patch03_L2estab.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 | rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22 lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH prim via lower_id layer (3 or 1) simply does not work. For TE (3) it returns an error (len=-6) which is not evaluated by handle_l1(), so the L1 layer status ends up wrong. Instead PH must be sent via L4, only then does it reach L1 without an error message. And NT PH prims only reach L1 when they are sent to layer 2 id. --> use upper_id to send PH primitives. * Check for errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are improved. * The lower_id is now not used for anything, except: Why is lower_id layer deleted when it wasn't created? I removed this code since it looks very wrong. Patches: patch04_l1activation.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett | 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines chan_misdn: Fix loss of B channels if L1 is down. If you make 2 calls out an NT PTMP port which is not connected to any phone, the B channel associated with that call becomes unusable until Asterisk is restarted. The problem is the EVENT_SETUP is queued when L1 is not up in misdn_lib_send_event(). If L1 cannot be activated the event won't be dequeued. It gets even worse when the call is hung up. The queued EVENT_SETUP will be overwritten by an EVENT_DISCONNECT. The reserved B channel then will never be freed. If later someone connects a phone to the port, L1 will eventually activate and the queued EVENT_DISCONNECT is sent down the stack. However, it is ignored because it is the wrong call state. The real fix would be that activation and queueing for a new SETUP is done by the NT stack. But since it doesn't, the workaround must be removed because it doesn't always work. Fix: The event is no longer queued but immediately sent to the stack. If L1 cannot be activated, the L3 state machine that was started by the EVENT_SETUP will do its work, i.e. a timeout will release the B channel properly. The SETUP possibly cannot be sent the first time but is resent by T303 in case L1 could be activated. Patches: patch05_bchan-loss.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 | rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13 lines chan_misdn: Remove some calls to exit(). Try proper cleanup when something goes wrong in misdn_lib_init(). Especially do not call exit()! * Fix memory leak because stack_destroy() does not free the stack struct. Patches: patch06_cleanup-init.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........ Merged revisions 375519-375524 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 375625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375626 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375627 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	01526b2c3c	Fix Wrong Result In Debug Message For SDP Origin Processing ... While looking at some debug logs, I noticed that it was being reported that the SDP origin line was unsupported or failed. Upon looking into this on my local machine, I found that I too was getting this debug message yet everything seemed to be getting processed properly. What was discovered is, that, the variable to determine what is displayed in the debug message for the SDP line that was processed, was not being set for the origin line when the result was successful. This patch fixes this and was tested on local machine. ........ Merged revisions 375594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375601 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375613 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	d4a357b82f	chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry ... A regression was introduced in chan_sip by changes to sip reload introduced by r349097. That patch moved peer purging from the beginning of the reload to after the general configuration was finished. This patch fixes that by undoing the repositioning of the original peer purging code and using a similar function after performing general configuration that purges only autocreated peers that were created when persist mode isn't enabled. (closes issue ASTERISK-20611) Reported by: Alisher Review: https://reviewboard.asterisk.org/r/2171/ ........ Merged revisions 375575 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	5f3f32c494	Prevent resetting of NATted realtime peer address on reload. ... If a "sip reload" is issued for a SIP peer, then his IP address will be cleared, thus resulting in forgetting the public IP address. Asterisk will then attempt to route SIP traffic to the private IP address. The fix here is to make "sip reload" ignore realtime peers when "host = dynamic" is spotted. Realtime peers can now only have their IP address reset if they have gone from being not dynamic to being dynamic. (closes issue ASTERISK-18203) reported by daren ferreira (closes issue ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff uploaded by JoshE (license #6075) ........ Merged revisions 375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375417 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375437 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	da85f8489f	Make evaluation of channel variables consistently case-sensitive. ... Due to inconsistencies in how variable names were evaluated, the decision was made to make all evaluations case-sensitive. See the UPGRADE.txt file or https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity for more details. (closes issue ASTERISK-20163) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2160 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e2702177a4	chan_dahdi: Fix segfault dereferencing a NULL tech_pvt. ... The tech support customer was using the AMI Redirect action shortly after a call was placed. While the channel tried to do an ast_read(), the masquerade resulting from the channel redirect took place. The masquerade in the middle of the ast_read() resulted in the segfault. (closes issue AST-1025) Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 375361 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375362 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375363 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	6d57ecd48c	Change a few warnings to debug and the inverse. ... Remove the "RTP Read too short" warning for RTP keepalives. Remove the the warning about the application delimiter switch from pipe to comma. (You should've done this by now.) Make cdr_odbc report more when an insert fails. Make chan_sip warn less when the peer wants SRTP (and we don't) or sends a zero port to disable a media type. Review: https://reviewboard.asterisk.org/r/2167 (closes issue ASTERISK-20538) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	1a0646aec1	Fixes to the fd-oriented SIP TCP reads. ... Don't crash on large user input. Allow SIP headers without space. Optimize code a bit. Review: https://reviewboard.asterisk.org/r/2162 ........ Merged revisions 375111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375112 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375113 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	8a65f47e88	Don't do SIP contact/route DNS if we're not using the result. ... In many cases (for peers behind NAT or for TCP sockets) we do not need to look up any hostname in the Contact (or Route) when sending an in-dialog request. This should reduce netsock2.c: getaddrinfo errors in certain scenarios. Review: https://reviewboard.asterisk.org/r/2156 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	2142fc3bc7	Update sip_request_call SIP dial string documentation. ... This was missed when merging review r1859. ........ Merged revisions 375074 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375078 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375079 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	c4df9778cb	Remove a log message that was left in accidentally from call-id logging development. ... ........ Merged revisions 375051 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	e9ab568f88	Fix some potential misuses of ast_str in the code. ... Passing an ast_str pointer by value that then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or ast_str_append_va() can result in the pointer originally passed by value being invalidated if the ast_str had to be reallocated. This fixes places in the code that do this. Only the example in ccss.c could result in pointer invalidation though since the other cases use a stack-allocated ast_str and cannot be reallocated. I've also updated the doxygen in strings.h to include notes about potential misuse of the functions mentioned previously. Review: https://reviewboard.asterisk.org/r/2161 ........ Merged revisions 375025 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375026 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375027 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	e41a591dfc	Fix underscreen buttons warnings apeared while transfer process ...	13 years ago
Andrew Latham	3820f1586e	Doxygen Updates - Title update ... Update and extend the configuration_file group and enable linking. Update title that was left behind many years ago. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	c7b23cbb0a	Do not use a FILE handle when doing SIP TCP reads. ... This is used to solve an issue where a poll on a file descriptor does not necessarily correspond to the readiness of a FILE handle to be read. This change makes it so that for TCP connections, we do a recv() on the file descriptor instead. Because TCP does not guarantee that an entire message or even just one single message will arrive during a read, a loop has been introduced to ensure that we only attempt to handle a single message at a time. The tcptls_session_instance structure has also had an overflow buffer added to it so that if more than one TCP message arrives in one go, there is a place to throw the excess. Huge thanks goes out to Walter Doekes for doing extensive review on this change and finding edge cases where code could fail. (closes issue ASTERISK-20212) reported by Phil Ciccone Review: https://reviewboard.asterisk.org/r/2123 ........ Merged revisions 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374906 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374914 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	ccb7b3a1b5	Fix a bug where audio on Google Voice would not work due to ignoring candidates. ... Instead of ignoring parts of the message that are not known just ignore the ones we know may be present and that would cause a problem. ........ Merged revisions 374877 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	cd9745be1b	Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures. ... This change removes the requirement for ufrag and pwd in the transport stanza and also makes us the controlling agent. (closes issue ASTERISK-20554) Reported by: mmichelson ........ Merged revisions 374850 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	825607e09b	Don't make chan_sip export global symbols. ... During testing, it was discovered that having chan_sip export global symbols was problematic. The biggest problem was that load order was affected. Trying to use realtime could be problematic since in all likelihood the necessary realtime driver(s) would not be loaded before chan_sip. In addition, it was found that it was impossible to use the Digium Phone Module for Asterisk since it must be loaded before chan_sip since it must hook into chan_sip's configuration parsing. The solution is to use a virtual table in the same manner that other modules in Asterisk do, like app_voicemail. (closes issue ASTERISK-20545) Reported by: kmoore ........ Merged revisions 374842 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374849 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	755c2b8708	Consider the Google Talk content stanza name (jin:content) valid. ... ........ Merged revisions 374833 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	766d133c62	Improve logging for DTLS-SRTP failure situations. ... (closes issue ASTERISK-20487) Reported by: mjordan ........ Merged revisions 374756 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	79baef5bbd	Merged revisions 374515-374535 from ... https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	d78f7f92b2	Add support for applying direct media ACLs between differing channel technologies. ... Review: https://reviewboard.asterisk.org/r/2122/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a094707d51	Fix a variety of ref counting issues ... This patch resolves a number of ref leaks that occur primarily on Asterisk shutdown. It adds a variety of shutdown routines to core portions of Asterisk such that they can reclaim resources allocate duringd initialization. Review: https://reviewboard.asterisk.org/r/2137 ........ Merged revisions 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374178 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374196 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	99e1174bfa	Doxygen Cleanup ... Start adding configuration file linking and pages. Add module loading doxygen block. Breaking up commits to keep it easy to track (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c3c317433f	Fix ref leak when adding ICE candidates to an SDP ... There was a missing decrement to the reference count for the current ICE candidate when local candidates are being added to an outbound SDP. This patch corrects that. ........ Merged revisions 374085 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b5138fccf4	Add pause one second W dial modifier. ... * The following dialplan applications now recognize 'W' to pause sending DTMF for one second in addition to the previously existing 'w' that paused sending DTMF for half a second. Dial, ExternalIVR, and SendDTMF. * The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now distinguishes between 'w' and 'W'. The 'w' pauses dialing for half a second. The 'W' pauses dialing for one second. * Created dahdi_dial_str() in chan_dahdi that eliminated a lot of duplicated dialing code and diagnostic messages for the channel driver. (closes issue ASTERISK-20039) Reported by: Jeremiah Gowdy Patches: jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy Expanded patch to add support in chan_dahdi. Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	9f55e5e928	Make res_http_websocket an optional dependency on supported platforms for chan_sip. ... (closes issue ASTERISK-20439) Reported by: sruffell Patches: 0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded by sruffell (license 5417) ........ Merged revisions 373914 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	10eb78d213	Fix an issue where Local channels dialed by app_queue are considered in use immediately. ... The chan_local channel driver returns a device state of in use even if a created Local channel has not yet been dialed. This fix changes the logic to return a state of not in use until the channel itself has been dialed. (closes issue ASTERISK-20390) Reported by: tim_ringenbach Review: https://reviewboard.asterisk.org/r/2116/ ........ Merged revisions 373878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373879 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373880 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	b6a780b923	Move handling of 408 response so there is no misleading warning message. ... (closes issue ASTERISK-20060) Reported by: Walter Doekes ........ Merged revisions 373848 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373849 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373850 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	2b56626b43	Remove dead code and documentation for nonexistent feature. ... multiplelogin was removed from chan_agent back in 1.6.0 when AgentCallbackLogin() was removed. (closes issue AST-948) reported by Steve Pitts ........ Merged revisions 373768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373769 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373770 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	318c7bea44	Fix T.38 support when used with chan_local in between. ... Users of the T.38 API can indicate AST_T38_REQUEST_PARMS on a channel to request that the channel indicate a T.38 negotiation with the parameters present on the channel. The return value of this indication is expected to be AST_T38_REQUEST_PARMS upon success but with chan_local involved this could never occur. This fix changes chan_local to always return AST_T38_REQUEST_PARMS for this situation. If the underlying channel technology on the other side does not support T.38 this would have been determined ahead of time using ast_channel_get_t38_state and an indication would not occur. (closes issue ASTERISK-20229) Reported by: wdoekes Patches: ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review: https://reviewboard.asterisk.org/r/2070/ ........ Merged revisions 373705 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373706 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373707 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	fdfb3ae5fa	Allow for redirecting reasons to be set to arbitrary strings. ... This allows for the REDIRECTING dialplan function to be used to set the reason to any string. The SIP channel driver has been modified to set the redirecting reason string to the value received in a Diversion header. In addition, SIP 480 response reason text will set the redirecting reason as well. (closes issue AST-942) reported by Malcolm Davenport (closes issue AST-943) reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/2101 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	b7233b18eb	Properly handle UAC/UAS roles for SIP session timers ... The SIP session timer mechanism contains a mandatory 'refresher' parameter (included in the Session-Expires header) which is used in the session timer offer/answer signaling within a SIP Invite dialog. It looks like asterisk is interpreting the uac resp. uas role only as the initial role of client and server (caller is uac, callee is uas). The standard rfc 4028 however assigns the client role to the ((RE)-Invite) requester, the server role to the ((RE)-Invite) responder. This patch has Asterisk track the actual refresher as "us" or "them" as opposed to relying on just the configured "uas" or "uac" properties. (closes issue AST-922) Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged revisions 373652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373665 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373690 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	c7850a198b	chan_sip: Set Quality of Service for video rtp instance ... (closes issue ASTERISK-20201) Reported by: ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license 6008) ........ Merged revisions 373617 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373631 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373632 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	da8c22fe45	Be consistent, send From: "Anonymous" <sip:anonymous@anonymous.invalid> ... When setting CALLERID(pres)=unavailable in the dialplan, the From header in the SIP message contains "Anonymous" <sip:Anonymous@anonymous.invalid>. For consistency, Asterisk should use a lowercase a in the userpart of the URI. * Make the From header use a lowercase A in the userpart of the anonymous URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383) patch uploaded by Antti Yrjola ........ Merged revisions 373500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373501 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373502 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	bc090677bc	Fix potential reentrancy problems in chan_sip. ... Asterisk v1.8 and later was not as vulnerable to this issue. * Made find_call() lock each private as it processes the found dialogs. (Primary cause of ABE-2876) * Made the other functions that traverse the dialogs container lock each private as it examines them. * Fix race condition in sip_call() if the thread that sent the INVITE is held up long enough for a response to be processed. The p->initid for the INVITE retransmission could be added after it was canceled by the response processing. * Made __sip_destroy() clean up resource pointers after freeing. This is primarily defensive in case someone has a stale private pointer. * Removed redundant memset() in reqprep(). The call to init_req() already does the memset() and is the first reference to req in reqprep(). * Removed useless set of req.method in transmit_invite(). The calls to initreqprep() and reqprep() have to do this because they memset() the req. JIRA ABE-2876 .......... Merged -r373423 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 373424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373466 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373469 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	f6e0406239	Fix a deadlock caused by a race condition between removing a hint and reloading the dialplan and subscribing to the removed hint. ... If conditions were right it was possible for both the PBX core and chan_sip to deadlock by both having a lock that the other wants. In the case of the PBX core it had the contexts lock and wanted a SIP dialog lock, while in the case of chan_sip it had the SIP dialog lock and wanted the contexts lock. This fix unlocks the SIP dialog before getting the extension state so that the other thread will not block on trying to lock it. Once the extension state is retrieved the SIP dialog is locked again and life carries on. As the SIP dialog is reference counted it is not possible for it to go away after unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins ........ Merged revisions 373438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373440 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373454 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	ad3e51bf4c	Fix an issue with H.264 format attribute comparison and fix an issue with improper SDP being produced. ... The H.264 format attribute module compares two format attribute structures to determine if they are compatible or not. In some instances it was possible for this check to determine that both structures were incompatible when they actually should be considered compatible. This check has now been made even more permissive by assuming that if no attribute information is available the two structures are compatible. If both structures contain attribute information a base level comparison of the H.264 IDC value is done to see if they are compatible or not. The above issue uncovered a secondary issue in chan_sip where the SDP being produced would be incorrect if the formats were considered incompatible. This has now been fixed by checking that all information required to produce the SDP is available instead of assuming it is. (closes issue ASTERISK-20464) Reported by: Leif Madsen ........ Merged revisions 373413 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	fd98835f1f	Doxygen Updates Janitor Work ... * Whitespace, doc-blocks, spelling, case, missing and incorrect tags. * Add cleanup to Makefile for the Doxygen configuration update * Start updating Doxygen configuration for cleaner output * Enable inclusion of configuration files into documentation * remove mantisworkflow... * update documentation README * Add markup to Tilghman's email and talk with him about updating his email, he knows... * no code changes on this commit other than the mentioned Makefile change (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	ca8aeeef1b	iax2-provision: Fix improper return on failed cache retrieval ... (closes issue ASTERISK-20337) reported by: John Covert Patches: iax2-provision.c.patch uploaded by John Covert (license 5512) ........ Merged revisions 373342 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373343 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373368 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	6f61cb50c5	Doxygen Updates - janitor work ... Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style. Some missing txt file links are removed but their content or essense will be included in some later updates. A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen. Further updates coming. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e8380afc8a	Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip. ... As mentioned on the review for this, WebRTC has moved towards choosing DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds support for this but makes it available for normal SIP clients as well. Testing has been done to ensure that this introduces no regressions with existing behavior and also that it functions as expected. Review: https://reviewboard.asterisk.org/r/2113/ ........ Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago