20248 Commits (3b5d63fd2b323c5b10daec876bd85d032d88177c)

 

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Author	SHA1	Message	Date
David Vossel	1509737580	ignore CANCEL request after having already received final response to INVITE ... RFC 3261 section 9 states that a CANCEL has no effect on a request to a UAS that has already given a final response. This patch checks to make sure there is a pending invite before allowing a CANCEL request to be processed, otherwise it responds to the CANCEL with a "481 Call/Transaction Does Not Exist". Review: https://reviewboard.asterisk.org/r/697/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3a875d8524	minor fixes for white/black event filters ... This fixes a ref count leak in event filters and checks for a filter container allocation failure during session creation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	5f45ca4d50	Merged revisions 271902 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct. (closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	42c24b585a	Add regular expression filtering for manager events. ... This patch as documented in the sample config allows one to optionally apply white, black, or both types of filtering to manager events. The new 'eventfilter' option is set per user. (closes issue #14861) Reported by: fnordian Patches: eventfilter3.patch uploaded by fnordian (license 110), modified by me Review: https://reviewboard.asterisk.org/r/673/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	294c78a27e	Resolve some errors that occur on a graceful shutdown. ... Don't Finalize() if Initialize() did not succeed. This resulted in an error about trying to Finalize() an invalid handle. Also trim some trailing whitespace while in the area. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	9cc7c55578	Change the method of retrieving the Asterisk version string. ... Using this method makes it so res_fax doesn't have to be rebuilt on every svn update. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6d82dbb905	fixes attended transfer behavior when both transferee and transferer hung up ... If both the transferer and transferee of a attended transfer hangup before the new channel picks up, the new channel should be hung up as well as it has no endpoint to talk to. This mirrors the expected behavior used in 1.4. (closes issue #17444) Reported by: corruptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	519b5a09e4	Updated the CHANGES file documenting the addition of a configurable port in the dundi config file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	01507f4ab7	Merged revisions 271761 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines Allow users to specify a port for dundi peers. (closes issue #17056) Reported by: klaus3000 Patches: dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	9bbeb945e8	Merged revisions 271689 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct. (closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ ........ This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e3873889b8	Conflict kqueue on OS X, since it doesn't work there yet, anyway. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d4bbf88e96	add speex 16khz sample frame so codec cost can be calculated ... (closes issue #17534) Reported by: fabled Patches: speex-wb-sample.diff uploaded by fabled (license 448) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b3281ac725	Merged revisions 271552 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines Do not use sizeof to calculate size of a heap allocated character array. Change left out from 271399. (closes issue #16053) Reported by: diLLec ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	462da0585e	fixes crash when From header URI is missing "sip:" ... (closes issue #17437) Reported by: klaus3000 Patches: sip_crash uploaded by dvossel (license 671) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	1a7e1aee5e	fixes logic error introduced by slin16 sip support ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	63fd368411	Add new application for declining counting words in multiple languages. ... (closes issue #16869) Reported by: chappell Patches: app_say_counted-20100317.c uploaded by chappell (license 8) Tested by: chappell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	54f2dfc91c	Merged revisions 271399 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines Fix crash when parsing some heavily nested statements in AEL on reload. Due to the recursion used when compiling AEL in gen_prios, all the stack space was being consumed when parsing some AEL that contained nesting 13 levels deep. Changing a few large buffers to be heap allocated fixed the crash, although I did not test how many more levels can now be safely used. (closes issue #16053) Reported by: diLLec Tested by: jpeeler ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271483 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3f9c6bb3bc	file.c was truncating audio file formats to the lower 32bits. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	0effac1e66	Recorded merge of revisions 271335 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010) | 13 lines Eliminate deadlock potential in dahdi_fixup(). (This is a backport of 269307, committed to trunk by rmudgett.) Calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. (closes issue #17261) Reported by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	846050f698	fixes some coding guideline issue ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a1fe641a38	retransmit response to BYE requests until timer J expires ... According to RFC 3261 section 17.2.2, which describes non-INVITE server transaction, when a dialog enters the Completed state it must destroy the dialog after Timer J (T1*64) fires. For a BYE transaction Asterisk terminates the dialog immediately during sip_hangup() when it should be waiting T1*64 ms. This results in some odd behavior. For instance if Asterisk receives a BYE and transmits a 200ok in response, if the endpoint never receives the 200ok it will retransmit the BYE to which Asterisk responds with a "481 Call leg/transaction does not exist" because the dialog is already gone. To resolve this I made a function called sip_scheddestroy_final(). This differs slightly from sip_schedestroy() in that it enables a flag that will prevent the destruction from ever being rescheduled or canceled afterwards. It also prevents the pvt's needdestroy flag from being set which triggers the destruction of the dialog within the do_monitor thread(). By using this function we are guaranteed destruction will not occur until the scheduled time. This allows Asterisk to respond to any possible retransmits for a dialog after we process the initial BYE request for T1*64 ms. Other changes: I removed two instances where sip_cancel_destroy is used right before calling sip_scheddestroy. sip_scheddestroy always calls sip_cancel_destroy before scheduling the new destruction so it is completely unnecessary. Review: https://reviewboard.asterisk.org/r/694/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ba3d1ad680	adds support for slin16 in sip ... (closes issue #16153) Reported by: kfister Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested by: kfister, malcolmd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b00f58da25	adds speex 16khz audio support ... (closes issue #17501) Reported by: fabled Patches: asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448) Tested by: malcolmd, fabled, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	0ef5550742	Change expected operation from error to debug message ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	a087394ae4	Blocked revisions 271123 via svnmerge ... ........ r271123 | mnicholson | 2010-06-17 10:11:27 -0500 (Thu, 17 Jun 2010) | 7 lines Set sin_family in ast_get_ip_or_srv() and removed the 'last' member of the ast_dnsmgr_entry struct. (closes issue #15827) Reported by: DennisD Patches: (modified) dnsmgr_15827.patch uploaded by chappell (license 8) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	531290385c	option w[(secs)] incorrectly capitalized in xmldoc ... (closes issue #17516) Reported by: karlfife git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	2112418032	addition of more parse_uri test cases ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	91bb18f5e8	Merged revisions 270979 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun 2010) | 4 lines Fixed typo in macro-page Reported to #asterisk-dev by a student of jsmith. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	01039c0465	Fix the actual place that was pointed out, for previous commit. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	8ef2c3100a	Merged revisions 270980 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines Need to lock the agent chan before access its internal bits. Pointed out by russellb on asterisk-dev mailing list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	9f1136143b	Set sin_family to AF_INET when doing lookups, also reset sin_port the first time the ip address changes. ... (closes issue #17496) Reported by: ManChicken (closes issue #15827) Reported by: DennisD Patches: dnsmgr_15827.patch uploaded by chappell (license 8) Tested by: DennisD, gentlec, damage, wimpy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	fcb055fb4e	addition of G.719 pass-through support ... (closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	0bf94685fd	MSG_OOB flag on HANGUP packet removed. ... Per Tilghman's request on IRC (#asterisk-bugs). (closes issue #17506) Reported by: brycebaril Tested by: pabelanger, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	1da8159aa6	Merged revisions 270866 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines fixes chan_iax2 race condition There is code in chan_iax2.c that attempts to guarantee that only a single active thread will handle a call number at a time. This code works once the thread is added to an active_list of threads, but we are not currently guaranteed that a newly activated thread will enter the active_list immediately because it is left up to the thread to add itself after frames have been queued to it. This means that if two frames come in for the same call number at the same time, it is possible for them to grab two separate threads because the first thread did not add itself to the active_list fast enough. This causes some pretty complex problems. This patch resolves this race condition by immediately adding an activated thread to the active_list within the network thread and only depending on the thread to remove itself once it is done processing the frames queued to it. By doing this we are guaranteed that if another frame for the same call number comes in at the same time, that this thread will immediately be found in the active_list of threads. Review: https://reviewboard.asterisk.org/r/720/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	00433d60e6	Fix no call waiting caller ID ... Clearing the callwaitcas flag in analog_call was causing the incoming D digit to be ignored which triggers sending the caller ID. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	5d2bbe86ad	Update formatting for channelvariables.tex ... (closes issue #17511) Reported by: klaus3000 Patches: channelvariables.tex-patch.txt uploaded by klaus3000 (license 65) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	3194c061a7	Don't blow up if an ast_channel doesn't get allocated. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	de18661bee	Don't continue sending the file when there has been an error ... If there is a problem with a firmware file, Polycom phones will close the connection. We were continuing to send the file anyway. There should be no reason to continue sending a file if there is an error writing it. (closes issue #16682) Reported by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	fb71a38a41	Don't send files twice and remove extra \r\n from header ... After the manager http auth changes, we forgot to remove the manual sending of the file. Also, ast_http_send adds two \r\n to the header that is passed to it, so a trailing \r\n is removed from the Content-type header. It might be better to change ast_http_send, but I don't like changing the behavior of an API function. (closes issue #17239) Reported by: cjacobsen Patches: patch2.diff uploaded by cjacobsen (license 1029) Tested by: lathama, cjacobsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	480f412858	Make contactdeny apply to src ip when nat=yes ... chan_sip's "contactdeny" feature screens the "to be registered contact". In case of nat=yes it should not use the address information from the Contact header (which is not used at all for routing), but the source IP address of the request. Thus, if nat=yes and a client sends a request from a denied IP address (e.g. by spoofing the src-IP address) it can bypass the screening. This commit makes contactdeny apply to the src ip when nat=yes instead. (closes issue #17276) Reported by: klaus3000 Patches: patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7037dd6680	Merged revisions 270583 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010) | 5 lines Variables have always been case-sensitive, so we should not be removing case-insensitive matches. Bug reported via the -dev list. See http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a0d8a35659	Argh, mixed declarations and code. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	81c15adfa2	Add distributed devicestate via the XMPP protocol. ... (closes issue #15757) Reported by: Marquis Patches: distributed_devstate-XMPP.txt uploaded by lmadsen (license 10) Tested by: Marquis, lmadsen, marcelloceschia Review: https://reviewboard.asterisk.org/r/351/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	be0249427e	Merged revisions 270442 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010) | 1 line Move information about zonemessages into the [zonemessages] section. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	41d7b51bf7	Merged revisions 270331 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon, 14 Jun 2010) | 14 lines Properly play first file in sort list. When using sort=alpha we would always skip the first file in the list first time through. We now check for that properly. (closes issue #17470) Reported by: pabelanger Patches: sort.aplha.patch uploaded by pabelanger (license 224) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/703/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	fb5fddd987	Extract sig_ss7_init_linkset() to sig_ss7. ... Also found a place where sig_pri_init_pri() was inlined and called it instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	a806196f79	Add option to get untruncated channel name from AGENT function. ... The "channel" option would chop the channel name at the last '-', which made it useless for something like a channel transfer from the dialplan. The "fullchannel" option will return the channel name as-is. ABE-2218 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	93a5e74e37	Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn. ... Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn. Review: https://reviewboard.asterisk.org/r/696/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	2743e9b2ac	bashism in configure script ... Theoretically the ./configure script is a pure bourne-shell script. Practically it may be run by bash if /bin/sh is not good enough. But we should not count on it. See bug report for the gory details. (closes issue #17485) Patches: 0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	41821558c7	Reverting patch and reopening issue #16155, as patch breaks ... FreeBSD / OSX builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago