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2010-06-22Merged revisions 271761 via svnmerge from Matthew Nicholson
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
2010-06-22Merged revisions 271689 via svnmerge from Matthew Nicholson
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
2010-06-21Conflict kqueue on OS X, since it doesn't work there yet, anyway.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271657 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-21add speex 16khz sample frame so codec cost can be calculatedDavid Vossel
(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
2010-06-21Merged revisions 271552 via svnmerge from Jeff Peeler
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
2010-06-21fixes crash when From header URI is missing "sip:"David Vossel
(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
2010-06-21fixes logic error introduced by slin16 sip supportDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-21Add new application for declining counting words in multiple languages.Tilghman Lesher
(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
2010-06-18Merged revisions 271399 via svnmerge from Jeff Peeler
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
2010-06-18file.c was truncating audio file formats to the lower 32bits.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-18Recorded merge of revisions 271335 via svnmerge from Jeff Peeler
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
2010-06-17fixes some coding guideline issueDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-17retransmit response to BYE requests until timer J expiresDavid Vossel
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
2010-06-17adds support for slin16 in sipDavid Vossel
(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
2010-06-17adds speex 16khz audio supportDavid Vossel
(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
2010-06-17Change expected operation from error to debug messageJeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-17Blocked revisions 271123 via svnmergeMatthew Nicholson
........ 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
2010-06-17option w[(secs)] incorrectly capitalized in xmldocPaul Belanger
(closes issue #17516) Reported by: karlfife git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16addition of more parse_uri test casesDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16Merged revisions 270979 via svnmerge from Paul Belanger
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
2010-06-16Fix the actual place that was pointed out, for previous commit.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16Merged revisions 270980 via svnmerge from Jason Parker
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
2010-06-16Set sin_family to AF_INET when doing lookups, also reset sin_port the first ↵Matthew Nicholson
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
2010-06-16addition of G.719 pass-through supportDavid Vossel
(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
2010-06-16MSG_OOB flag on HANGUP packet removed.Paul Belanger
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
2010-06-16Merged revisions 270866 via svnmerge from David Vossel
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
2010-06-16Fix no call waiting caller IDJeff Peeler
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
2010-06-16Update formatting for channelvariables.texPaul Belanger
(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
2010-06-15Don't blow up if an ast_channel doesn't get allocated.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-15Don't continue sending the file when there has been an errorTerry Wilson
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
2010-06-15Don't send files twice and remove extra \r\n from headerTerry Wilson
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
2010-06-15Make contactdeny apply to src ip when nat=yesTerry Wilson
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
2010-06-15Merged revisions 270583 via svnmerge from Tilghman Lesher
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
2010-06-15Argh, mixed declarations and code.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-15Add distributed devicestate via the XMPP protocol.Tilghman Lesher
(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
2010-06-15Merged revisions 270442 via svnmerge from Leif Madsen
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
2010-06-14Merged revisions 270331 via svnmerge from Paul Belanger
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
2010-06-14Extract sig_ss7_init_linkset() to sig_ss7.Richard Mudgett
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
2010-06-14Add option to get untruncated channel name from AGENT function.Jason Parker
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
2010-06-14Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.Richard Mudgett
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
2010-06-13bashism in configure script Tzafrir Cohen
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
2010-06-13Reverting patch and reopening issue #16155, as patch breaksPaul Belanger
FreeBSD / OSX builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-12Merged revisions 270078 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun 2010) | 2 lines Fix typo in example ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-11Use pkg-config to find gmime librariesPaul Belanger
This way the libraries can be found even if they are in non-standard locations. (closes issue #16155) Reported by: jcollie Patches: 0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch uploaded by jcollie (license 412) Tested by: jsmith, tilghman, pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-11Merged revisions 269960 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010) | 8 lines For SpeeX, 0 bits remaining is valid and does not need an emitted warning. (closes issue #15762) Reported by: nblasgen Patches: issue15672.patch uploaded by pabelanger (license 224) Tested by: nblasgen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-11Add DBGetComplete event after a DBGetResponse.Tilghman Lesher
(closes issue #16965) Reported by: rrb3942 Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-11Remove lines from the output related to the backtrace itself.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-10Remove ASTBINDIR variablePaul Belanger
(closes issue #17031) Reported by: pabelanger Patches: Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224) Tested by: pabelanger, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-10Merged revisions 269821 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun 2010) | 19 lines Fix potential crash when writing raw SLIN audio on a PLC-enabled channel. The issue here was that the frame created when adjusting for PLC had no offset to its audio data. If this frame were translated to another format prior to being sent out an RTP socket, all went well because the translation code would put an appropriate offset into the frame. However, if the SLIN audio were not translated before being sent out the RTP socket, bad things would happen. Specifically, the ast_rtp_raw_write makes the assumption that the frame has at least enough of an offset that it can accommodate an RTP header. This was not the case. As such, data was being written prior to the allocation, likely corrupting the data the memory allocator had written. Thus when the time came to free the data, all hell broke loose. ....Well, Asterisk crashed at least. The fix was just what one would expect. Offset the data in the frame by a reasonable amount. The method I used is a bit odd since the data in the frame is 16 bit integers and not bytes. I left a big ol' comment about it. This can be improved on if someone is interested. I was more interested in getting the crash resolved. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-10Add documentation explaining PLC in Asterisk.Mark Michelson
Review: https://reviewboard.asterisk.org/r/688/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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