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20 authors
  1. Nov 08, 2010
    • Jeff Peeler's avatar
      Merged revisions 294278 via svnmerge from · 12a40275
      Jeff Peeler authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines
  
  Merged revisions 294277 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines
    
    Fix playback failure when using IAX with the timerfd module.
    
    To fix this issue the alert pipe will now be used when the timerfd module is
    in use. There appeared to be a race that was not solved by adding locking in the
    timerfd module, but needed to be there anyway. The race was between the timer
    being put in non-continuous mode in ast_read on the channel thread and the IAX 
    frame scheduler queuing a frame which would enable continuous mode before the
    non-continuous mode event was read. This race for now is simply avoided.
    
    (closes issue #18110)
    Reported by: tpanton
    Tested by: tpanton
    
    I put tested by tpanton because it was tested on his hardware. Thanks for the
    remote access to debug this issue!
  ........
................


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      12a40275
    • Matthew Nicholson's avatar
      Merged revisions 294243 via svnmerge from · 2df9e23e
      Matthew Nicholson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r294243 | mnicholson | 2010-11-08 14:56:30 -0600 (Mon, 08 Nov 2010) | 15 lines
  
  Merged revisions 294242 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) | 8 lines
    
    Go off hold when we get an empty reinvite telling us to.
    
    (closes issue 0014448)
    Reported by: frawd
    
    (closes issue #17878)
    Reported by: frawd
  ........
................


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      2df9e23e
    • Terry Wilson's avatar
      Merged revisions 294207 via svnmerge from · aa0f407b
      Terry Wilson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r294207 | twilson | 2010-11-08 13:56:10 -0600 (Mon, 08 Nov 2010) | 2 lines
  
  Set a default waittime, and make sure to convert it to milliseconds
........


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      aa0f407b
    • Richard Mudgett's avatar
      Merged revisions 294125 via svnmerge from · 18553bb8
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines
  
  valgrind reported references to freed memory during a mISDN hangup collision.
  
  Bad things have been happening in chan_misdn because the chan_misdn
  channel private struct chan_list is not protected from reentrancy.  Hangup
  collisions have be causing read and write accesses to freed memory.
  
  Converted chan_misdn struct chan_list to an ao2 object for its reference
  counting feature.
  
  **********
  Removed an impediment to converting chan_list to an ao2 object.
  
  The use of the other_ch member in chan_list is shaky at best.  It is set
  if the incoming and outgoing call legs are mISDN.  The use of the other_ch
  member goes against the Asterisk architecture and can even cause problems.
  
  1) It is used to disable echo cancellation.  This could be bad if the call
  is forked and the winning call leg is not mISDN or the winning call leg is
  not the last mISDN channel called by the fork.  The other_ch would become
  a dangling pointer.
  
  2) It is used when the far end is alerting to hear the far end's inband
  audio instead of Asterisk's generated ringback tone.  This is bad if the
  call is forked.  You would only hear the last forked mISDN channel and it
  may not be ringing yet.
  
  The other_ch would become a dangling pointer if the call is later
  transferred.
  **********
  
  JIRA SWP-2423
  JIRA ABE-2614
........


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      18553bb8
  3. Nov 05, 2010
  5. Nov 04, 2010
  7. Nov 03, 2010
    • Terry Wilson's avatar
      Merged revisions 293803 via svnmerge from · abc94089
      Terry Wilson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines
  
  Avoid valgrind warnings for ast_rtp_instance_get_xxx_address
  
  The documentation for ast_rtp_instance_get_(local/remote)_address stated that
  they returned 0 for success and -1 on failure. Instead, they returned 0 if the
  address structure passed in was already equivalent to the address instance
  local/remote address or 1 otherwise. 90% of the calls to these functions
  completely ignored the return address and passed in an uninitialized struct,
  which would make valgrind complain even though the operation was technically
  safe.
  
  This patch fixes the documentation and converts the get_xxx_address functions
  to void since all they really do is copy the address and cannot fail.
  Additionally two new functions
  (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3
  times where the return value was actually checked. The
  get_and_cmp_local_address function is currently unused, but exists for the sake
  of symmetry.
  
  The only functional change as a result of this change is that we will not do an
  ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the
  ast_sockaddr_copy() in the get_*_address functions. So, even though it is an
  API change, it shouldn't have a noticeable change in behavior.
  
  Review: https://reviewboard.asterisk.org/r/995/
........


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      abc94089
    • Richard Mudgett's avatar
      Merged revisions 293807 via svnmerge from · cbd42ce6
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r293807 | rmudgett | 2010-11-03 13:35:19 -0500 (Wed, 03 Nov 2010) | 34 lines
  
  Merged revisions 293806 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r293806 | rmudgett | 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) | 27 lines
    
    Merged revisions 293805 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) | 20 lines
      
      Party A in an analog 3-way call would continue to hear ringback after party C answers.
      
      All parties are analog FXS ports.
      1) A calls B.
      2) A flash hooks to call C.
      3) A flash hooks to bring C into 3-way call before C answers.  (A and B hear ringback)
      4) C answers
      5) A continues to hear ringback during the 3-way call. (All parties can hear each other.)
      
      * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on
      the wrong subchannel.
      
      * Made several debug messages have more information.
      
      A similar issue happens if B and C are SIP channels.  B continues to hear
      ringback.  For some reason this only affects v1.8 and trunk.
      
      * Don't start ringback on the real and 3-way subchannels when creating the
      3-way conference.  Removing this code is benign on v1.6.2 and earlier.
    ........
  ................
................


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      cbd42ce6
  9. Nov 02, 2010
  11. Nov 01, 2010
  13. Oct 30, 2010
  15. Oct 29, 2010
  17. Oct 28, 2010
  19. Oct 26, 2010
  21. Oct 25, 2010
    • Tilghman Lesher's avatar
      Merged revisions 292969 via svnmerge from · f96d27b9
      Tilghman Lesher authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r292969 | tilghman | 2010-10-25 16:15:19 -0500 (Mon, 25 Oct 2010) | 2 lines
  
  Several more defines that need to be altered for compiling against an older version of libpri
........


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      f96d27b9
    • Tilghman Lesher's avatar
      Merged revisions 292906 via svnmerge from · 7bc278bd
      Tilghman Lesher authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r292906 | tilghman | 2010-10-25 14:28:35 -0500 (Mon, 25 Oct 2010) | 4 lines
  
  Allow the DAHDI driver to compile, even with a sufficiently older version of libpri.
  
  Fixes our Bamboo builds.
........


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      7bc278bd
    • David Vossel's avatar
      Merged revisions 292868 via svnmerge from · 7189a944
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r292868 | dvossel | 2010-10-25 14:07:50 -0500 (Mon, 25 Oct 2010) | 39 lines
  
  Merged revisions 292867 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r292867 | dvossel | 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) | 32 lines
    
    Merged revisions 292866 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) | 27 lines
      
      This patch turns chan_local pvts into astobj2 objects.
      
      chan_local does some dangerous things involving deadlock avoidance.
      tech_pvt functions like hangup and queue_frame are provided with a
      locked channel upon entry.  Those functions are completely safe as
      long as you don't attempt to give up that channel lock, but that is
      impossible to guarantee due to the required deadlock avoidance necessary
      to lock both the tech_pvt and both channels involved.
      
      In the past, we have tried to account for this by doing things like
      setting a "glare" flag that indicates what function should destroy the
      pvt.  This was used in local_hangup and local_queue_frame to decided
      who should destroy the pvt if they collided in separate threads.  I
      have removed the need to do this by converting all chan_local tech_pvts
      to astobj2.  This means we can ref a pvt before deadlock avoidance
      and not have to worry about that pvt possibly getting destroyed under
      us.  It also cleans up where we destroy the tech_pvt.  The only unlink
      from the tech_pvt container occurs in local_hangup now, which is where
      it should occur.
      
      Since there still may be thread collisions on some functions like
      local_hangup after deadlock avoidance, I have added some checks to detect
      those collisions and exit appropriately.  I think this patch is going to
      solve quite a bit of weirdness we have had with local channels in the past.
    ........
  ................
................


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      7189a944
  23. Oct 22, 2010
    • Terry Wilson's avatar
      Merged revisions 292825 via svnmerge from · 8ea287b0
      Terry Wilson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r292825 | twilson | 2010-10-22 15:35:29 -0700 (Fri, 22 Oct 2010) | 4 lines
  
  Don't create directories without at least o+x
  
  Also, making files that you are going to modify read-only is dumb.
........


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      8ea287b0
    • Terry Wilson's avatar
      Merged revisions 292794 via svnmerge from · 3d658dce
      Terry Wilson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r292794 | twilson | 2010-10-22 15:18:36 -0700 (Fri, 22 Oct 2010) | 2 lines
  
  Make files readable only by the owner
........


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      3d658dce
    • Leif Madsen's avatar
      Merged revisions 292787 via svnmerge from · 8de8e4a1
      Leif Madsen authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r292787 | lmadsen | 2010-10-22 16:28:43 -0500 (Fri, 22 Oct 2010) | 21 lines
  
  Merged revisions 292786 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) | 13 lines
    
    Update the LDIF file for LDAP.
    The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've
    now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems
    where I was doing an ldapadd to import the schema into the LDAP database, and the existing file
    would cause problems and ERROR messages when registering.
    
    Additional documention has been added based on feedback in the issue I'm closing.
    
    (closes issue #13861)
    Reported by: scramatte
    Patches:
          ldap-update.txt uploaded by lmadsen (license 10)
    Tested by: lmadsen, jcovert, suretec, rgenthner
  ........
................


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      8de8e4a1
    • Terry Wilson's avatar
      Merged revisions 292740 via svnmerge from · 89a16f9c
      Terry Wilson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r292740 | twilson | 2010-10-22 09:49:34 -0700 (Fri, 22 Oct 2010) | 45 lines
  
  Add TLS cert helper script
  
  This script is useful for quickly generating self-signed CA, server, and client
  certificates for use with Asterisk. It is still recommended to obtain
  certificates from a recognized Certificate Authority and to develop an
  understanding how SSL certificates work. Real security is hard work.
  
  OPTIONS:
    -h  Show this message
    -m  Type of cert "client" or "server". Defaults to server.
    -f  Config filename (openssl config file format)
    -c  CA cert filename (creates new CA cert/key as ca.crt/ca.key if not passed)
    -k  CA key filename
    -C  Common name (cert field)
          For a server cert, this should be the same address that clients
          attempt to connect to. Usually this will be the Fully Qualified
          Domain Name, but might be the IP of the server. For a CA or client
          cert, it is merely informational. Make sure your certs have unique
          common names.
    -O  Org name (cert field)
          An informational string (company name)
    -o  Output filename base (defaults to asterisk) 
    -d  Output directory (defaults to the current directory)
  
  Example:
  
  To create a CA and a server (pbx.mycompany.com) cert with output in /tmp:
    ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp
  
  This will create a CA cert and key as well as asterisk.pem and the the two
  files that it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem
  and ca.crt somewhere (like /etc/asterisk) and set tlscertfile=/etc/asterisk.pem
  and tlscafile=/etc/ca.crt. Since this is a self-signed key, many devices will
  require you to import the ca.crt file as a trusted cert.
  
  To create a client cert using the CA cert created by the example above:
    ast_tls_cert -m client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \
      "My Company" -d /tmp -o joe_user
  
  This will create client.crt/key/pem in /tmp. Use this if your device supports
  a client certificate. Make sure that you have the ca.crt file set up as
  a tlscafile in the necessary Asterisk configs. Make backups of all .key files
  in case you need them later.
........


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      89a16f9c
    • Mark Michelson's avatar
      Merged revisions 292741 via svnmerge from · 3964b078
      Mark Michelson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r292741 | mmichelson | 2010-10-22 12:09:52 -0500 (Fri, 22 Oct 2010) | 12 lines
  
  Prevent multiple runs of event_sub_test from producing false failure results.
  
  The array of test subscriptions was declared "static," meaning that the
  data.count field would retain its value between runs of the test. After the
  first test run, this would result in false reports of test failures.
  
  I chose to just remove the "static" keyword from the structure since it's not
  a huge deal to construct this structure during each run of the test. Another
  alternative would have been to zero out the data.count fields of each test
  subscription instead.
........


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      3964b078
    • Richard Mudgett's avatar
      Merged revisions 292704 via svnmerge from · 64845d73
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) | 19 lines
  
  Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call.
  
  When a call is transfered by ECT or implicitly by disconnect in sig_pri or
  implicitly by disconnect in chan_misdn, the connected line information is
  not exchanged.  The connected line interception macros also need to be
  executed if defined.
  
  The CALLER interception macro is executed for the held call.
  The CALLEE interception macro is executed for the active/ringing call.
  
  JIRA ABE-2589
  JIRA SWP-2296
  
  Patches:
        abe_2589_c3bier.patch uploaded by rmudgett (license 664)
        abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664)
  
  Review: https://reviewboard.asterisk.org/r/958/
........


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      64845d73
  25. Oct 21, 2010