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  1. Feb 26, 2010
  2. Feb 25, 2010
    • Jeff Peeler's avatar
      Merged revisions 248860 via svnmerge from · 406bb181
      Jeff Peeler authored
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
      ........
        r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines
        
        Ensure that monitor recordings are written to the correct location (again)
        
        This is an extension to 248757. As such the dialplan test has been extended:
        
        exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b)
        exten => 5040, n, dial(sip/5001)
        exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b)
        exten => 5041, n, dial(sip/5001)
        exten => 5042, 1, monitor(wav,monitor_test3,b)
        exten => 5042, n, dial(sip/5001)
        exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m)
        exten => 5043, n, changemonitor(monitor_test4)
        exten => 5043, n, dial(sip/5001)
        exten => 5044, 1, monitor(wav,monitor_test4,m)
        exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning
        exten => 5044, n, dial(sip/5001)
      ........
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      406bb181
    • Mark Michelson's avatar
      Fix incorrect ACL behavior when CIDR notation of "/0" is used. · a1af5f4f
      Mark Michelson authored
      AST-2010-003
      
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a1af5f4f
    • Tilghman Lesher's avatar
      Merged revisions 248859 via svnmerge from · a0af2cff
      Tilghman Lesher authored
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
      ........
        r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) | 15 lines
        
        Some platforms clear /var/run at boot, which makes connecting a remote console... difficult.
        
        Previously, we only created the default /var/run/asterisk directory at install
        time.  While we could create it in the init script, that would not work for
        those who start asterisk manually from the command line.  So the safest thing
        to do is to create it as part of the Asterisk boot process.  This also changes
        the ownership of the directory, because the pid and ctl files are created after
        we setuid/setgid.
        
        (closes issue #16802)
         Reported by: Brian
         Patches: 
               20100224__issue16802.diff.txt uploaded by tilghman (license 14)
         Tested by: tzafrir
      ........
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a0af2cff
    • Jeff Peeler's avatar
      Merged revisions 248757 via svnmerge from · d64987f8
      Jeff Peeler authored
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
      ........
        r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines
        
        Ensure that monitor recordings are written to the correct location.
        
        Recordings should be placed in the monitor directory when a non-absolute path
        is used.
        
        Exact dialplan used for testing:
        exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b)
        exten => 5040, n, dial(sip/5001)
        exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b)
        exten => 5041, n, dial(sip/5001)
        exten => 5042, 1, monitor(wav,monitor_test3,b)
        exten => 5042, n, dial(sip/5001)
        
        ABE-2101
      ........
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d64987f8
  3. Feb 24, 2010
  4. Feb 23, 2010
  5. Feb 22, 2010
  6. Feb 21, 2010
  7. Feb 20, 2010
  8. Feb 19, 2010
    • Moises Silva's avatar
      mfcr2 issue 0016844 - Fix portability bit fields and make... · 0d838691
      Moises Silva authored
      mfcr2 issue 0016844 - Fix portability bit fields and make mfcr2_immediate_accept work again, reported and patched by korihor
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0d838691
    • David Vossel's avatar
      handle_request_invite revise comment, fix coding guideline issues · fc0cb53a
      David Vossel authored
      I'm working with this code right now trying to analyze a deadlock.
      This change is just to clean up a few things before I make a more
      complex patch.
      
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      fc0cb53a
    • Richard Mudgett's avatar
      Merged revisions 247910 via svnmerge from · 57ee669d
      Richard Mudgett authored
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
      ................
        r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines
        
        Merged revision 247904 from
        https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
        
        ..........
        r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines
        
        Make chan_misdn DTMF processing consistent with other channel technologies.
        
        The processing of DTMF tones on the receiving side of an ISDN channel is
        inconsistent with the way it is handled in other channels, especially
        DAHDI analog.  This causes DTMF tones sent from an ISDN phone to be
        doubled at the connected party.
        
        We are using the following 2 options of misdn.conf
        1) astdtmf=yes
        2) senddtmf=yes
        
        Option one is necessary because the asterisk DSP DTMF detection is better
        than mISDN's internal DSP.  Not as many false positives.
        
        Option two is necessary to transmit DTMF tones end to end when mISDN
        channels are connected to SIP channels with out of band DTMF for example.
        
        The symptom is that DTMF tones sent by an ISDN phone are doubled on the
        way through asterisk when two mISDN channels are connected with a Local
        channel in between or if it is bridged to an analog channel.
        
        The doubling of DTMF tones is because DTMF is passed inband to asterisk by
        the mISDN channel and passed out of band once again after the release of
        the DTMF tone.  Passing it inband is wrong.  Neither an analog channel nor
        SIP channel passes DTMF inband if configured to inband DTMF.  Analog and
        SIP channels filter out the DTMF tones because they use the voice frames
        returned by ast_dsp_process.  But chan_misdn passes the unfiltered input
        voice frames instead.
        
        To overcome one aspect of the problem, the doubling of DTMF tones when two
        mISDN channels are directly bridged, someone made an 'optimization', where
        in that case the DTMF tone passed out-of-band to the peer channel is not
        translated to an inband tone at the transmit side.  This optimization is
        bad because it does not work in general.  For example, analog channels or
        mISDN channels when bridged through an intermediary local channel will
        generate DTMF tones from out-of-band information.  Also, of course, it
        must not be done when there is no inband DTMF available.
        
        This patch fixes the issue.  Now chan_misdn will filter the received
        inband DTMF signal the same as other channel types.
        
        Another change included: No need to build an extra translation path
        because ast_process_dsp does it if required.
        
        Patches:
        	misdn-dtmf.patch
        
        JIRA ABE-2080
      ................
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      57ee669d
  9. Feb 18, 2010
  10. Feb 17, 2010
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