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Author	SHA1	Message	Date
Richard Mudgett	00de2431be	Fix uninitialized variable. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@212430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	b35312b15b	This patch adds additional checking when generating queue log TRANSFER events. ... The additional checks prevent generation of false TRANSFER events in certain situations. (closes issue #14536) Reported by: aragon Patches: queue-log-xfer-fix1.diff uploaded by mnicholson (license 96) Tested by: aragon, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ad76c40551	Backport fix so that outbound CANCEL requests have same branch as challenged INVITEs. ... There already was code present to be sure that a CANCEL will contain the same branch-id as the INVITE it is cancelling. However, for INVITES which are challenged downstream, this mechanism did not work properly. Now this is taken care of. This is a backport of a fix already present in all 1.6.X branches and in trunk. It also fixes ABE-1907. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	90f3605f6f	Conversion specifiers, not format specifiers ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	63cc189747	AST-2009-005 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	541976843e	Small oops. Clear the flags which have been checked. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a56006702e	Resolve a deadlock involving app_chanspy and masquerades. ... (ABE-1936) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	17656694c3	QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername. ... This is a partial revert of revision 82590, which was an attempted cleanup, but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended as a method by which component interfaces could be queried from the queue. Membername isn't useful here, because that field cannot be used to obtain further information about the member. See the documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various AMI commands which take a member argument for further justification. (closes issue #15664) Reported by: rain Patches: app_queue-queue_member_list.diff uploaded by rain (license 327) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e87d76cb94	Because channel information can be accessed outside of the channel thread, we must lock the channel prior to modifying it. ... (closes issue #15397) Reported by: caspy Patches: 20090714__issue15397.diff.txt uploaded by tilghman (license 14) Tested by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	20d63bd1c0	Dialplan starts execution before the channel setup is complete. ... * Issue 15655: For the case where dialing is complete for an incoming call, dahdi_new() was asked to start the PBX and then the code set more channel variables. If the dialplan hungup before these channel variables got set, asterisk would likely crash. * Fixed potential for overlap incoming call to erroneously set channel variables as global dialplan variables if the ast_channel structure failed to get allocated. * Added missing set of CALLINGSUBADDR in the dialing is complete case. (closes issue #15655) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	01416248d5	Update imapstorage.txt documentation. ... Updated the imapstorage.txt documentation to reflect that issues with c-client versions older than 2007 seem to cause crashing issues that are not seen with more recent versions. Documentation has been updated to reflect this. (closes issue #14496) Reported by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	22140c486f	Eliminate spurious compiler warnings from system headers on *BSD platforms. ... Ensure that system headers located in /usr/local/include are actually treated as system headers by the compiler, and not as local headers which are subject to warnings from the -Wundef compiler option and others. (closes issue #15606) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	29f865ad17	Fixes dialplan wildcard extension taking precedence over call pickup code. ... Prior to this patch, a wildcard extension in the dialplan (for example, _*.) would take precedence over picking up a call in the channel's pickup group. This patch simply moves the block of code handling pickup group matching to above the extension matching code. (closes issue #14735) Reported by: stevedavies Review: https://reviewboard.asterisk.org/r/319/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ca0f026f41	Reverting index() fix, applying a different methodology, based upon developer discussions. ... (related to issue #15639) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f5a5763ee9	Helps if we export the index() function. ... (Related to issue #15639) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a70128e190	Apparently, some platforms don't have the index() function. ... (closes issue #15639) Reported by: nmav git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ffe395f410	Resolve a valgrind warning about a read from uninitialized memory. ... (issue #15396) Reported by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a687e8c53f	Modify how Playtones() is used in Milliwatt() to resolve gain issue. ... When Milliwatt() was changed internally to use Playtones() so that the proper tone was used, it introduced a drop in gain in the output signal. So, use the playtones API directly and specify a volume argument such that the output matches the gain of the original Milliwatt() code. (closes issue #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff uploaded by russell (license 2) Tested by: rue_mohr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b5bea3704c	Minor changes inspired by testing with latest GCC. ... The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e1226e2411	Publish French extra sounds ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	361c9a99e1	Allow for UDPTL to use only even-numbered ports if desired. ... There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	0f05b9b9de	backport rev 205532 from trunk: ... pthread_self returns a pthread_t which is not an unsigned int on all pthread implementations. Casting it to an unsigned int fixes compiler warnings. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	f622e06bbe	Fix logic errors from 208746 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	fc5db2b241	Fix compiling under dev-mode with gcc 4.4.0. ... Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0660bfbe74	Don't impose an arbitrary limit on member lines in queues.conf ... I know what some of you are thinking: "UGH! Mark, why are you using ast_strdup and ast_free for the string when you can just use ast_strdupa and let the memory free itself?! Have the bats been chewing on your brain again?" Based on past experiences, I don't like using ast_strdupa inside a loop. It's a good way to potentially exhaust stack space. Also, since this only happens when reloading queues, I don't think that heap allocations and frees are going to be a huge problem. (closes issue #15559) Reported by: amorsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	55d9c2ecaf	Do not log an ERROR if autoservice_stop() returns -1. ... This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	38e98f42bc	Only send a BYE when hanging up a channel that is up. ... For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	1c46ba9635	Fix a problem where a 491 response could be sent out of dialog. ... This generalizes the fix for issue 13849. The initial fix corrected the problem that Asterisk would reply with a 491 if a reinvite were received from an endpoint and we had not yet received an ACK from that endpoint for the initial INVITE it had sent us. This expansion also allows Asterisk to appropriately handle an INVITE with authorization credentials if Asterisk had not received an ACK from the previous transaction in which Asterisk had responded to an unauthorized INVITE with a 407. (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch uploaded by mmichelson (license 60) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	594a236e12	Only set the priindication setting when not performing a reload ... (closes issue #14696) Reported by: fdecher git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	94bc859e81	Remove inaccurate XXX comment. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	eb5f3170fc	Properly handle 183 responses which do not contain an SDP. ... (closes issue #15442) Reported by: ffloimair Patches: 15442.patch uploaded by mmichelson (license 60) Tested by: tkarl, ffloimair git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	98dcd8946e	Export symbols for functions included in our compatibility headers. ... (closes issue #15556) Reported by: smw1218 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5dbbf21212	Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional). ... This change makes URIENCODE and QUOTE behave similarly, since the documentation states that the argument is not optional, for both. (closes issue #15439) Reported by: pkempgen Patches: 20090706__issue15439.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e07afa4876	Wait for wink before dialing when using E&M wink signaling ... There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simply add SIG_EMWINK to the list. (closes issue #14434) Reported by: araasch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207827 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	dca651b85d	Revert r207573, this approach could potentially block for an unacceptable ... amount of time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e0827ae778	Document default timeout for AMI originations. ... AST-224 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	75f1eaf2a1	Ensure that user-provided CFLAGS and LDFLAGS are honored. ... This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either* in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	8b940dbeb7	Wait for wink before dialing when using E&M wink signaling ... This patch adds a new dahdi_wait function to specifically wait for the wink event. If the wink is not eventually received the channel is hung up. (closes issue #14434) Reported by: araasch Patches: emwinkmod uploaded by araasch (license 693) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	423a444c0b	Answer video SDP offers properly when videosupport is not enabled. ... Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch *only* helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8b67a33369	Only do the chan->fdno check in ast_read() in a developer build. ... I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	d162e4b055	Fix format specifier to print out an unsigned long long. ... Yep, it's even ifdefed out code. But it made it to the RR list... (closes issue #14726) Reported by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	1e30dcf61c	Enhance configuration option for overlapdial allowing direction choice ... Previously overlap dialing could only be turned on or off for both incoming and outgoing calls. New parameters incoming, outgoing, and both have been added to allow further control. There is no change in default behavior with these new options and allows in band DTMF to be accepted in one direction if required. (closes issue #14471) Reported by: eboscani git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	98a6820737	sip option flags handled incorrectly ... (issue #15376) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	7c82de7d7e	SIP incorrect From: header information when callpres is prohib ... Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	5510a1c74e	error in iax.conf related IP-based access control ... (closes issue #15518) Reported by: pkempgen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	b1fe655954	avoid segfault caused by user error ... If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ed177d72d4	Fix a memory leak. ... (closes issue #15517) Reported by: adomjan Patches: func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7782df0963	Merged revision 206700 from ... https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	455ccbae20	Only print debug info in codec_dahdi if we are asking for it. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	6db6a73b8d	Fixes several call transfer issues with chan_misdn. ... * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago