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2010-06-08Fix some doxygen warnings.Leif Madsen
(closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Release list lock before returning on error.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Fix trunk build on Mac OS X.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Add SRTP support for AsteriskTerry Wilson
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Make SIP tests compile again.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Use the mailbox destructor function, instead.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Mailbox list would previously grow at each reload, containing duplicates.Tilghman Lesher
Also, optimize the allocation of mailboxes to avoid additional memory structures. (closes issue #16320) Reported by: Marquis Patches: 20100525__issue16320.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Extract sig_ss7 out of chan_dahdi.Richard Mudgett
Extract the SS7 specific code out of chan_dahdi like what was done to ISDN/PRI and analog signaling. The new SS7 structures were modeled on sig_pri. The changes to sig_pri are an enhancement and a bug fix made possible because SS7 was extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable should have been set unconditionally in sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of SS7 extraction. 3) Module ref count error in dahdi_new() if startpbx failed to start the PBX for some reason. Review: https://reviewboard.asterisk.org/r/661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Seems strange (and the code backs up) that if the max and min of a statistic ↵Tilghman Lesher
is expressed as a double, the last value would not also need to be a double. (closes issue #15807) Reported by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Moved AOC request code out of the middle of code parsing the dialed number.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Event well was going dry.Tilghman Lesher
(issue #17234) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Set threshold for silence detection defaults to 256Paul Belanger
(closes issue #15685) Reported by: david_s5 Patches: dsp-silence-threshold-init.diff uploaded by dant (license 670) issue15685.patch.v5 uploaded by pabelanger (license 224) Tested by: danti Review: https://reviewboard.asterisk.org/r/670/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Avoid unloading res_smdi twice.Tilghman Lesher
(closes issue #17237) Reported by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Suppress warning in waitstream_core().Richard Mudgett
Suppress the warning about unexpected control subclass frames for AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and AST_CONTROL_AOC in file.c:waitstream_core(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-06Take advantage of variable substitution already in the Makefile to specify ↵Tilghman Lesher
the correct location for files in init.d. (closes issue #16979) Reported by: jw-asterisk (issue #15691) Reported by: itamarjp git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-06Finally track down and eliminate the "FRACK! warnings from chan_iax2".Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-05Fix crash in DTMF detection.Tilghman Lesher
What I did not originally see in my previous commit was that even though the next digit could be detected before the previous was considered ended, the detection of the next digit effectively ends the detection of the previous. Therefore, the length moves in lockstep with the digit, and no separate counter is needed for the length alone. (closes issue #17371) Reported by: alecdavis (closes issue #17474) Reported by: kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-05Verify event is not NULL before attempting to lower its usecount.Tilghman Lesher
(closes issue #17234) Reported by: mav3rick git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-05Typo fix.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-05Grammatical error fix.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-05Merged revisions 268320 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010) | 3 lines Rest In Peace http://www.outandaboutnewspaper.com/article/4061 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04fixes compile error from uninitialized variableDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04RFC3261 compliant sip unreliable retransmit timing + 'registerattempts' ↵David Vossel
option tweak Changes. 1. RFC 3261 states in section 17.1.2.2 and 17.1.1.2 that retransmission timers should initially be set to timer T1. T1 by default is 500ms. Asterisk was starting the retransmission timers at T1*2 which shouldn't cause any problems, but is not RFC compliant. 2. RFC 3261 states in section 17.1.2.2 that for a non-INVITE client transaction, if the retransmit timer fires while in the proceeding state that the request must be retransmitted. Asterisk currently ack's requests for both INVITE and non-INVITE transactions when a 1XX response is received, this patch changes this for non-INVITE requests. 3. The 'registerattempts' option in sip.conf is supposed to set how many registry attempts will be made before giving up. When this option is set to 1, I would expect only one registry attempt to be made before stopping because of a failure, but instead two are made. In my opinion this is not expected behavior. This option does not indicate that these are re-attempts. The logic behind this option has been changed to only attempt registers the exact number of times this option is set to. If this option is 0, it still continues to re-attempt the registration forever. Review: https://reviewboard.asterisk.org/r/687/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04Merged revisions 268126 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r268126 | tilghman | 2010-06-04 15:41:24 -0500 (Fri, 04 Jun 2010) | 2 lines AC_CONFIG_SUBDIRS has a bad side-effect on cross-compiles. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04Merged revisions 268050 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r268050 | tilghman | 2010-06-04 14:38:57 -0500 (Fri, 04 Jun 2010) | 6 lines Build menuselect with the build environment's compiler, not the host (target)'s compiler. (closes issue #17464) Reported by: pprindeville Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04Merged revisions 267971 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010) | 2 lines As-fixiate the build process ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04Incoming overlap dialing no longer works after sig_pri extraction.Richard Mudgett
The problem would manifest itself if your dialplan matching could accept more digits to match than were actually dialed. The time out waiting for overlap digits disconnected the call instead of matching any accumulated digits to the dialplan. Accidental conversion of a break out of loop as a break out of switch. (closes issue #17401) Reported by: avalentin Patches: issue17401_digit_timeout.patch uploaded by rmudgett (license 664) Tested by: avalentin, rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04As signed linear audio data is accessed as 16-bit values, certain processors ↵Tilghman Lesher
require the values to be aligned in memory. (closes issue #16912) Reported by: michaelevdokimov Patches: asterisk.patch uploaded by michaelevdokimov (license 997) Tested by: michaelevdokimov git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04Send an ACK for every final response received for an INVITETerry Wilson
From issue ABE-2247. RFC 3261 compliance for sections 13.2.24 and 17.1.1.2. Review: https://reviewboard.asterisk.org/r/692/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04As signed linear audio data is accessed as 16-bit values, certain processors ↵Tilghman Lesher
require the values to be aligned in memory. (closes issue #16912) Reported by: michaelevdokimov git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04If there's a default, turn it on, even when the option isn't specified.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04Merged revisions 267759 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010) | 7 lines Make the default install path appear to be /usr on Linux, instead of /usr/local. Also, reorganize the options, so that they're more alphabetical. (closes issue #17013) Reported by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Remove a LOG_WARNING.Russell Bryant
This came up when using the sample configs, and just indicates expected behavior. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Handle OOM errors more gracefully.Tilghman Lesher
(closes issue #17084) Reported by: falves11 Patches: issue17084_162_A.diff uploaded by falves11 (license 374) Tested by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Update UPGRADE.txt and CHANGE for CDR functionality changes.Leif Madsen
Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity written unless cdr.conf exists and is configured. (closes issue #17373) Reported by: wdoekes Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Make compile again.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Don't stop Asterisk if chan_usbradio isn't configured.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Remove unnecessary code relating to PLC.Mark Michelson
The logic for handling generic PLC is now handled in ast_write in channel.c instead of in translation code. Review: https://reviewboard.asterisk.org/r/683/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Remove a line that was killing Asterisk on startup.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Comment out a rule that likes to run implicitly unnecessarily, breaking buildsRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Add ETSI Message Waiting Indication (MWI) support.Richard Mudgett
Add the ability to report waiting messages to ISDN endpoints (phones). Relevant specification: EN 300 650 and EN 300 745 Review: https://reviewboard.asterisk.org/r/599/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02try to fix some random chan_h323 compilation failuresRussell Bryant
After some debugging, the random chan_h323 build failures appear to be due to complications introduced by some chan_h323 specific build stuff getting triggered during a clean. Simplify this by moving the h323 clean commands down into channels/makefile. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add ETSI Malicious Call ID support.Richard Mudgett
Add the ability to report malicious callers as an AMI event in the call event class. Relevant specification: EN 300 180 Review: https://reviewboard.asterisk.org/r/576/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Fix a build error on mac.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Ensure the -Wno-strict-aliasing flag makes it, even if ASTCFLAGS has been ↵Russell Bryant
specified. When ASTCFLAGS was specified with the make command, Makefile.rules was using the specified value from the command line and not the one here, making it so this flag would go missing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add ETSI Call Waiting support.Richard Mudgett
Add the ability to announce a call to an endpoint when there are no B channels available. A call waiting call is a SETUP message with no B channel selected. Relevant specification: EN 300 056, EN 300 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the "no_media_path" option. * Returns "0" if there is a B channel associated with the call. * Returns "1" if no B channel is associated with the call. The call is either on hold or is a call waiting call. If you are going to allow incoming call waiting calls then you need to use CHANNEL(no_media_path) do determine if you must drop a call to accept the new call. Review: https://reviewboard.asterisk.org/r/568/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Update CHANGES and aoc help doc to reflect AOC additionsDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add a CLI command that blocks until Asterisk has fully booted.Russell Bryant
Review: https://reviewboard.asterisk.org/r/684/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Prevent use of uninitialized values.Mark Michelson
Two struct sockaddr_ins are created when applying directmedia host access rules. The addresses of these are passed to the RTP engine to be filled in. However, the RTP engine inspects the fields of the structs before actually taking action. This inspection caused valgrind to be a bit unhappy. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Generic Advice of Charge.Richard Mudgett
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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