asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin P. Fleming	ed2a3cedd1	Merged revisions 209759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209759 \| kpfleming \| 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) \| 7 lines 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/trunk@209760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	2df5b70b16	Improve chan_sip's ability to determine what methods should and should not be used in a dialog. The previous effort here was to store what a peer is capable of receiving by parsing REGISTER requests from the peer and keeping that information for as long as the registration was active. The problem with this is that there are a great number of SIP devices which give no indication of the methods allowed in their REGISTER requests, and it is unreasonable to try to guess what the device may or may not support. In addition, some SIP devices have been found to claim support for a specific method, but their handling the method is less than ideal, or they are actually lying. With this patch, we now determine what methods a device supports by parsing the Allow header we receive from them, and we do this with each new dialog. In addition, a configuration option has been added so that an administrator can essentially blacklist certain methods from being used with certain peers if the admin knows that support for a specific method is dodgy or nonexistent. ABE-1822 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	bab2f57316	Add missing ifdef-s for service maintenance message functionality (closes issue #15614) Reported by: fabled git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	48363c16e1	Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	192e2be596	Fix a crash that can result if text codecs are allowed but textsupport is disabled. (closes issue #15596) Reported by: fabled Patches: sip-red.patch uploaded by fabled (license 448) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ba020fc390	Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	d81d6d3415	Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	0f31e6c26c	Merged revisions 208923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 \| jpeeler \| 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) \| 2 lines Fix logic errors from 208746 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b7cfe90404	Merged revisions 208746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 \| jpeeler \| 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) \| 7 lines 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/trunk@208749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	554c5e62d0	Merged revisions 208587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 \| mmichelson \| 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) \| 10 lines 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/trunk@208588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	17e2d9fdbc	Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	0a6e06c7ff	Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum IFP size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending our MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	88f1d14766	Merged revisions 208386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208386 \| mmichelson \| 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) \| 17 lines 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/trunk@208388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	dcd6227f6c	Merged revisions 208380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208380 \| jpeeler \| 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) \| 6 lines 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/trunk@208383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bacf6ab51e	Merged revisions 208312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208312 \| mmichelson \| 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) \| 3 lines Remove inaccurate XXX comment. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	980db1601a	Fix sending of interface identifier unconditionally in sig_pri The wrong logic was being used in chan_dahdi to convert a sig_pri_chan to the proper libpri channel number. The most significant bit must only be set only when trunk groups are being used. (closes issue #15452) Reported by: alecdavis Patches: bug15452.patch uploaded by jpeeler (license 325) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	98b4bdc1b9	Merged revisions 208262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208262 \| mmichelson \| 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) \| 8 lines 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/trunk@208263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	3843480b8f	Fix potential crash if p->owner is NULL. Problem was observed when a call-forwarding loop was accidentally configured. ABE-1906 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	58699809a5	Reset the fax buffers back to default settings regardless of signaling in use - Pointed out by Matt F. Also in the case of not using a signaling module, set the law back to the default as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	16328efb78	Do not dial digits when none were specified for sig_pri based calls (closes issue #15524) Reported by: elguero Patches: pri-sig-no-dest-set.patch uploaded by elguero (license 37) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	56c59985de	whitespace fix only git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	7466e00663	Fix my_is_off_hook to check rxbits only for FXS signaling git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6ac23c3eca	Merged revisions 207827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207827 \| jpeeler \| 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) \| 9 lines 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/trunk@207854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	96e4e31eeb	Merged revisions 207647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 \| kpfleming \| 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) \| 12 lines 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/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3f8059f87d	reg->username is parsed only once on sip reload The registration string can contain an expanded user portion of the form user@domain. This expanded user portion was stored in reg->username and parsed each time there is a registration refresh. Now, the domain portion of the user is parsed and stored separately in the regdomain field. (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c.domainparse3.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bec894cbe5	Merged revisions 207423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207423 \| mmichelson \| 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) \| 33 lines 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/trunk@207424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	bcff592839	Merged 207316 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r207316 \| rmudgett \| 2009-07-17 23:05:05 -0500 (Fri, 17 Jul 2009) \| 20 lines Fixed incoming calls being matched to MSNs without type-of-number prefix added. For an incoming ISDN call the dialed.number is incorrectly matched against the configured MSNs in misdn.conf. The numbers passed to the dialplan include the configured prefix for the dialed.number_type, whereas the check against the configured MSNs (to decide if the call is accepted at all), is executed without the configured prefix. e.g., dialed.number = 241168020, TON = national, configured national prefix is "0". (This is the TON which is used by ISDN providers in the Netherlands.) In chan_misdn.c:cb_events() in case EVENT_SETUP the call to misdn_cfg_is_msn_valid() uses the unnormalized number 241168020, but 57 lines later the call to read_config() adds the prefix, and the dialed.number is now 0241168020, which is then used in the dialplan. misdn_cfg_is_msn_valid() must use the normalized number, too. JIRA ABE-1912 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	090066be3b	fixes an error in r203638 CEL commit (closes issue #15525) Reported by: elguero Patches: iax2-double-unlock.patch uploaded by elguero (license 37) 15525.diff uploaded by dvossel (license 671) Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	74de8256bd	Merged revisions 207155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207155 \| jpeeler \| 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) \| 7 lines 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/trunk@207156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	65388d4e21	sip option flags handled incorrectly (closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	8270339965	Fix segfault in sig_analog when using callwaiting, respect callwaiting options Sig_analog handles allocating the sub channel for callwaiting, so no longer try to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as allocated upon success of the alloc_sub callback, which was responsible for the segfault. Also, the callwaiting and callwaitingcallerid options were being unconditionally set to true. Now, the options are properly set from chan_dahdi.conf. (closes issue #15508) Reported by: elguero Tested by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	0ce3fa1c22	Merged revisions 206938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 \| dvossel \| 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) \| 14 lines 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/trunk@206939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f91bc197cd	Session timer were not activated if Supported header field in INVITE had both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license 38) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	646cd02c09	The dialing flag was mistakingly removed from sig_pri. This readds the proper setting of the flag and is really a continuation of r205731. The flag was being set properly in sig_analog, but use of the newly added set_dialing callback allowed for some simplification in chan_dahdi. (closes issue #15486) Reported by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	e9e753d6f3	Merged revisions 206706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 \| rmudgett \| 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) \| 26 lines 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/trunk@206707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3402f34e9b	callerid(num) is wrong when username is missing A domain only sip uri <sip:123.123.123.123> would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b9e898017e	Restore some missing functionality to sig_analog. The main purpose of this commit is to restore missing functionality present in the ss_thread before all the sig related work was done. Two of the biggest missing things were distinctive ring detection and cid handling for V23. fxsoffhookstate and associated mwi variables have been moved inside sig_analog as they were not being set properly as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	58b440bc29	Merged revisions 206487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 \| rmudgett \| 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) \| 28 lines 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/trunk@206489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	e55d1b11b9	Merged revisions 206385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206385 \| russell \| 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) \| 13 lines Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 \| russell \| 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) \| 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	c90a8c0921	Merged revisions 206284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 \| rmudgett \| 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) \| 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6891ccad28	dns lookup of peername rather than peer's host in transmit_register() (closes issue #15052) Reported by: fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by fsantulli (license 818) Tested by: fsantulli git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c01286976a	SIP register not using peer's outbound proxy If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	5aab96f0b7	Merged revisions 205877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 \| mmichelson \| 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) \| 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 \| mmichelson \| 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) \| 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 \| mmichelson \| 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) \| 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	fe493cf85e	Merged revisions 205804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 \| dvossel \| 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) \| 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	aafa57cf4b	Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 \| mmichelson \| 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) \| 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	67d1957e60	Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ba2a8457b8	Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 \| dvossel \| 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) \| 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e76a0e92d2	Permit setting custom headers from the peer definition. (closes issue #14059) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	cf8395002d	Fix a deadlock in sig_analog git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	5e2a5d16b6	Add CEL transfer events to analog (chan_dahdi) transfers. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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6092 Commits (104974cdb4c57f697e78e4073e68ecb88ecea12c)