Commit Graph

5687 Commits

Author SHA1 Message Date
Mark Michelson
64e003be29 Merged revisions 183117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines
  
  Merged revisions 183115 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines
    
    Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use."
    
    A user was having an issue where if an outgoing SIP call was canceled, the SIP device
    would remain in use if we had not received any response to the initial INVITE we sent out.
    The SIP device would remain in use until the autocongestion timer was exhausted.
    
    I tracked down the cause of this to be the section of code I am removing here. I asked several
    people what the purpose of this code was meant to be, but no one could give me any sort of
    answer as to why this was here. The person who was having this issue has been using this patch
    for several months and it has stopped the problems they have had.
    
    AST-196
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@183121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 16:09:41 +00:00
Joshua Colp
37c533bcf7 Merged revisions 183108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
  
  Improve our triggering of a T38 switchover internally when triggered by a received reinvite.
  
  Previously we reached across the channel bridge to get the other party's SIP dialog
  structure in order to trigger an outgoing reinvite. This is extremely dangerous to do
  and only works if bridged to another SIP channel. This patch changes this to use the
  T38 control frame method of requesting a switchover. This change also causes the SIP
  channel driver to propogate back whether the switchover worked or not instead of blindly
  accepting the incoming T38 reinvite.
  
  Review: http://reviewboard.digium.com/r/200/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@183110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 15:43:16 +00:00
Jeff Peeler
74ceddaeea Merged revisions 183028 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines
  
  Add some code removed by mistake from commit 182722 that works around a file
  descriptor leak in versions of PWLib prior to 1.12.0.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@183030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 21:19:27 +00:00
Russell Bryant
baab6e74b9 Merged revisions 182847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines

Merged revisions 182810 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines

Fix cases where the internal poll() was not being used when it needed to be.

We have seen a number of problems caused by poll() not working properly on 
Mac OSX.  If you search around, you'll find a number of references to using 
select() instead of poll() to work around these issues.  In Asterisk, we've 
had poll.c which implements poll() using select() internally.  However, we 
were still getting reports of problems.

vadim investigated a bit and realized that at least on his system, even 
though we were compiling in poll.o, the system poll() was still being used.  
So, the primary purpose of this patch is to ensure that we're using the 
internal poll() when we want it to be used.

The changes are:

1) Remove logic for when internal poll should be used from the Makefile.  
   Instead, put it in the configure script.  The logic in the configure 
   script is the same as it was in the Makefile.  Ideally, we would have 
   a functionality test for the problem, but that's not actually possible, 
   since we would have to be able to run an application on the _target_ 
   system to test poll() behavior.

2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
   is not defined.

3) Change uses of poll() throughout the source tree to ast_poll().  I feel 
   that it is good practice to give the API call a new name when we are 
   changing its behavior and not using the system version directly in all cases.
   So, normally, ast_poll() is just redefined to poll().  On systems where 
   AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().

4) Change poll() in main/poll.c to be ast_internal_poll().

It's worth noting that any code that still uses poll() directly will work fine 
(if they worked fine before).  So, for example, out of tree modules that are 
using poll() will not stop working or anything.  However, for modules to work 
properly on Mac OSX, ast_poll() needs to be used.

(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim

http://reviewboard.digium.com/r/198/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 14:32:47 +00:00
Jeff Peeler
90907a186f Merged revisions 182722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
  
  Allow H.323 Plus library to be used in addition to the OpenH323 library
  
  Chan_h323 can now be compiled against both the previously supported versions of
  OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure
  script has been modified to look in the default install location of h323 to
  hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR.
  Also, the CLI command "h323 show version" has been added which indicates which
  version of h323 is in use.
  
  (closes issue #11261)
  Reported by: vhatz
  Patches:
        asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 20:52:54 +00:00
David Vossel
0267ce94e5 Merged revisions 182282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines
  
  Merged revisions 182281 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines
    
    Randomize IAX2 encryption padding
    
    The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all.  This patch calls ast_random to fill the padding buffer with random data.  The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame.
    
    Review: http://reviewboard.digium.com/r/193/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-16 17:53:54 +00:00
Tilghman Lesher
37db651ba8 Merged revisions 182211,182278 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009) | 14 lines
  
  Merged revisions 182208 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines
    
    Fixup glare detection, to fix a memory leak of a local pvt structure.
    (closes issue #14656)
     Reported by: caspy
     Patches: 
           20090313__bug14656__2.diff.txt uploaded by tilghman (license 14)
     Tested by: caspy
  ........
................
  r182278 | tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines
  
  Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution.
  Previously, FILE() returned one less character than specified, due to the
  terminating NULL.  Both the offset and length parameters now behave
  identically to the way variable substitution offsets and lengths also work.
  (closes issue #14670)
   Reported by: BMC
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-16 17:38:50 +00:00
Joshua Colp
90cf0f4c6e Merged revisions 182022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines
  
  Fix an issue with requesting a T38 reinvite before the call is answered.
  
  The code responsible for sending the T38 reinvite did not check if an INVITE was
  already being handled. This caused things to get confused and the call to fail.
  The code now defers sending the T38 reinvite until the current INVITE is done being
  handled.

  (issue AST-191)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-13 17:29:26 +00:00
Kevin P. Fleming
65ed9947f7 Merged revisions 181985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181985 | kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line
  
  improve a bit of suboptimal code
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-13 16:58:00 +00:00
Mark Michelson
6dd8307b7f Merged revisions 181769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines
  
  Merged revisions 181768 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines
    
    Properly send a 487 on an INVITE we have not responded to if we receive a BYE.
    
    If we receive an INVITE from an endpoint and then later receive a BYE from that
    same endpoint before we have sent a final response for the INVITE, then we need
    to respond to the INVITE with a 487. 
    
    There was logic in the code prior to this commit which seemed to exist solely to 
    handle this situation, but there was one condition in an if statement which 
    was incorrect. The only way we would send a 487 was if the sip_pvt had no owner
    channel. This made no sense since we created the owner channel when we received
    the INVITE, meaning that the majority of the time we would never send the 487.
    The 487 being sent should not rely on whether we have created a channel. Its
    delivery should be dependent on the current state of the initial INVITE transaction.
    With this commit, that logic is now correctly in place.
    
    (closes issue #14149)
    Reported by: legranjl
    Patches:
          14149.patch uploaded by mmichelson (license 60)
    Tested by: legranjl
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-12 18:36:34 +00:00
David Vossel
cb9f721fcd Merged revisions 181371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines
  
  Merged revisions 181340 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines
    
    encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames
    
    If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted.  This causes the entire frame to be corrupted.  When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense.  When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop.  To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted.  Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct.
    
    (closes issue #14607)
    Reported by: stevenla
    Tested by: dvossel
    
    Review: http://reviewboard.digium.com/r/192/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 17:40:10 +00:00
Joshua Colp
a58f23f79a Merged revisions 181345 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines
  
  Merged revisions 181328 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines
    
    Fix issue where an attended transfer could not be completed under a rare scenario.
    
    When completing an attended transfer chan_sip does a check to make sure the extension
    in the URI portion of the Refer-To header is a local valid extension. We don't actually
    need to check this since we know for sure the other channel is already up and talking to
    the extension. Some devices do not put the extension in the Refer-To header either, which
    can cause the extension check to fail. We now no longer do this check if it is an attended
    transfer.
    
    (closes issue #14628)
    Reported by: sverre
    Patches:
          14628.diff uploaded by file (license 11)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 17:29:45 +00:00
Joshua Colp
75c9cd1128 Merged revisions 181296 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines
  
  Merged revisions 181295 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines
    
    Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto.
    
    When dtmfmode was set to auto the inband DTMF detector was not setup
    on outgoing SIP calls. This caused inband DTMF detection to fail.
    The inband DTMF detector is now setup for both dtmfmode inband and auto.
    
    (closes issue #13713)
    Reported by: makoto
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 16:43:51 +00:00
Jeff Peeler
43ed1b3888 add missing header file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 15:54:38 +00:00
Jeff Peeler
efc87a9c86 Merged revisions 181135 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines
  
  Fix malloc debug macros to work properly with h323.
  
  The main problem here was that cstdlib was undefining free thereby causing the
  proper debug macros to not be used. ast_h323.cxx has been changed to call
  ast_free instead to avoid the issue. 
  
  A few other issues were addressed:
  - There were a few instances of functions improperly passing ast_free instead
  of ast_free_ptr.
  - Some clean up was done to avoid the debug macros intentionally being redefined.
  (copied below from Kevin's commit, appreciate the help)
  - disable astmm.h from doing anything when STANDALONE is defined, which is used
  by the tools in the utils/ directory that use parts of Asterisk header files in
  hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are
  compiled with STANDALONE defined.
  
  (closes issue #13593)
  Reported by: pj
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 04:33:29 +00:00
Mark Michelson
23e19b9227 Merged revisions 181032-181033 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181032 | mmichelson | 2009-03-10 19:46:47 -0500 (Tue, 10 Mar 2009) | 19 lines
  
  Merged revisions 181029,181031 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines
    
    Fix incorrect tag checking on transfers when pedantic=yes is enabled.
    
    (closes issue #14611)
    Reported by: klaus3000
    Patches:
          patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65)
    Tested by: klaus3000
  ........
    r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines
    
    Remove unused variables.
  ........
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  r181033 | mmichelson | 2009-03-10 19:49:00 -0500 (Tue, 10 Mar 2009) | 3 lines
  
  Add missing comment that quotes RFC 3891
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 01:04:04 +00:00
Russell Bryant
49b3688d42 Merged revisions 180261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r180261 | russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines

Resolve object matching issues related to the removal of the sip_user object.

Previously, chan_sip had both sip_peer and sip_user objects in memory.  A
patch went in to remove sip_user to simplify the code, since everything
could be done with just sip_peer.  This patch resolves some regressions
found that were introduced by those changes.

This code comes from svn/asterisk/team/group/sip-object-matching/.

Here is a list of the changes that have been made:

1) When doing a match by name with the find_peer() function, make it much
   easier to specify which objects should be matched by having a parameter
   that specifies exactly which object types should be considered.  Also,
   update find_by_name() to handle this parameter.  Finally, update all
   code to use the new option values.

2) When looking up an object for an outbound request by name, consider
   peers only.  (create_addr())

3) Only match peers on an incoming registration request.

4) When doing authentication (except for SUBSCRIBE), look up users
   by name, instead of all objects by name.
   
5) When doing authentication (except for SUBSCRIBE), after looking for
   a user by name, look for a peer by IP address, instead of all objects
   by IP address.

6) When handling the SIP qualify CLI command or manager action, look for
   a peer by name, instead of any object by name.

7) When handling the SIP unregister CLI command, look for a peer by name,
   instead of any object by name.

9) In sip_do_debug_peer(), search for a peer by name, instead of any object
   by name.

9) When handling the SIPPEER() dialplan function, search for a peer by name,
   instead of any object by name.

10) In the following session timer related functions, st_get_se(),
    st_get_refresher(), and st_get_mode(), when looking for an object for a
    given sip_pvt using pvt->peername, look for a peer by name, instead of any
    object by name.

11) Fix build_peer() to properly handle the case where separate type=peer and
    type=user entries were specified in sip.conf.

(closes issue #14505)
Reported by: lmadsen

Review: http://reviewboard.digium.com/r/172/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@180263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04 21:09:13 +00:00
Mark Michelson
22e08ba056 Merged revisions 179219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines
  
  Properly free memory and remove scheduler entries when a transmission failure occurs.
  
  Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit 
  was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called,
  this inevitably resulted in the reading and writing of freed memory.
  
  XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet
  at all. The proper action to take is to remove the scheduler entry we just created,
  free the packet's data as well as the packet itself, and unlink it from the list of
  packets on the sip_pvt structure.
  
  (closes issue #14455)
  Reported by: Nick_Lewis
  Patches:
        14455.patch uploaded by mmichelson (license 60)
  Tested by: Nick_Lewis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@179221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-01 21:57:18 +00:00
David Vossel
7a3aaf4f7e Merged revisions 178871 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) | 6 lines
  
  IAX2 prune realtime, minor tweak to last fix
  
  A return statement was missing which caused unexpected cli output.
  
  issue #14479
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@178875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26 17:50:51 +00:00
David Vossel
44cbe73882 Merged revisions 178767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines
  
  IAX2 prune realtime fix
  
  Iax2 prune realtime had issues.  If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened.  This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing.  If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine.  Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime.  These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend.  For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.  
  
  (closes issue #14479)
  Reported by: mousepad99
  Review: http://reviewboard.digium.com/r/176/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@178769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26 16:07:08 +00:00
Joshua Colp
69602d500a Merged revisions 178213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines
  
  Merged revisions 178205 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines
    
    Skip check for extension when subscribing for MWI.
    
    Since the remote side is not actually subscribing to a specific extension when
    subscribing for MWI just skip the check to see if the extension exists. They can't use it
    to specify the mailbox either since we require configuration of that in sip.conf
    
    (closes issue #14531)
    Reported by: festr
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@178232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24 15:22:25 +00:00
Tilghman Lesher
23111db55e Merged revisions 177944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177944 | tilghman | 2009-02-21 09:59:49 -0600 (Sat, 21 Feb 2009) | 2 lines
  
  On update, test against the existence of sipregs.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@177945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21 16:04:09 +00:00
Michiel van Baak
7c7f6266fa Merged revisions 177849 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177849 | mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines
  
  make chan_sip.c compile on OpenBSD again.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@177851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21 12:51:33 +00:00
David Vossel
f9fa2e07ab Fixes issue with undefined audio codecs in chan_iax2
During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec.  In 1.6 only audio codec bits 0-12 are defined, leaving bits 13-14 undefined.  By default all bits are enabled unless specified otherwise.  Since its a 2 byte field and 13-14 are not defined, these bits are never turned off.  In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk.  I fixed this by adding #define for undefined audio codec bits.  These bits are then removed from iax2's full bandwidth capabilities.   

(closes issue #14283)
Reported by: jcovert



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@177700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-20 20:34:11 +00:00
Jeff Peeler
765b86befa Merged revisions 177624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177624 | jpeeler | 2009-02-19 18:35:53 -0600 (Thu, 19 Feb 2009) | 7 lines
  
  Set sip_request ast_str data to NULL so ast_str_copy allocates space properly
  in copy_request
  
  (issue #14478)
  Reported by: erik_dedecker
........


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2009-02-20 00:38:19 +00:00
Jeff Peeler
10f903a9de Merged revisions 177162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177162 | jpeeler | 2009-02-18 14:11:57 -0600 (Wed, 18 Feb 2009) | 14 lines
  
  Modify h323 to build against PTLib as well as the older PWLib
  
  Several changes in PTLib have occurred requiring build time detection. Changes
  accounted for include the library name change, config option change, install
  location change, and a boolean type change which is handled by ast_ptlib.h.
  Also, the sed check has been modified to properly work with autoconf >= 2.62.
  
  (closes issue #14224)
  Reported by: bergolth
  Patches:
        asterisk-autoconf-sed.patch uploaded by bergolth (license 661)
        asterisk-pwlib-v3.patch uploaded by bergolth (license 661)
  Tested by: jpeeler
........


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2009-02-18 20:16:31 +00:00
Joshua Colp
f08c2a683b Merged revisions 177005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177005 | file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines
  
  Fix ordering of output for a ChannelUpdate manager event.
  (closes issue #14497)
  Reported by: vinsik
  Patches:
        chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@177007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18 17:14:16 +00:00
Dwayne M. Hubbard
e51c85db06 Merged revisions 176705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176705 | dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11 lines
  
  create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38
  
  This is required to create a UDPTL structure in create_addr_from_peer() to handle the
  scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but 
  is defined the peer's context.  I tested this patch by enabling t38pt_udptl in the 
  [general] section on one system and only enabling t38pt_udptl in a peer's context on
  the system sending a fax.  Without the patch, the sending system will fail to initiate
  T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure".
  When this patch is applied the sending side will successfully initiate T38 negotiation.
........


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2009-02-17 22:21:59 +00:00
Tilghman Lesher
75ff7a609d Merged revisions 176592,176642 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176592 | tilghman | 2009-02-17 12:49:20 -0600 (Tue, 17 Feb 2009) | 4 lines
  
  Add assertions in the quest to track down a refcount leak.
  (closes issue #14485)
   Reported by: davevg
........
  r176642 | tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines
  
  Prior to masquerade, move the group definitions to the channel performing the
  masq, so that the group count lingers past the bridge.
  (closes issue #14275)
   Reported by: kowalma
   Patches: 
         20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
   Tested by: kowalma
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@176644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 21:16:53 +00:00
Tilghman Lesher
19515dc39f Merged revisions 176501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176501 | tilghman | 2009-02-17 08:39:36 -0600 (Tue, 17 Feb 2009) | 3 lines
  
  In this version, we can combine the queries, because we support dropping
  nonexistent columns.
........


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2009-02-17 14:48:01 +00:00
Tilghman Lesher
e97be790c0 Merged revisions 176459 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009) | 17 lines
  
  Merged revisions 176426 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines
    
    After a 'sip reload', qualifies for realtime peers weren't immediately
    restarted, instead waiting until the next registration.  We're now
    caching the qualify across a reload/restart and starting the qualify
    immediately upon loading the peer.
    (closes issue #14196)
     Reported by: pdf
     Patches: 
           20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
     Tested by: pdf
  ........
................


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2009-02-17 02:07:18 +00:00
David Vossel
1ed4960ea5 Merged revisions 176355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176355 | dvossel | 2009-02-16 17:33:55 -0600 (Mon, 16 Feb 2009) | 13 lines
  
  Merged revisions 176354 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines
    
    Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging
    
    This should have been committed with rev176247, but I missed it.  srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either.  This fixs that.
    
    issue #13749
  ........
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2009-02-16 23:57:11 +00:00
Tilghman Lesher
891769666b Merged revisions 176320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r176320 | tilghman | 2009-02-16 17:14:08 -0600 (Mon, 16 Feb 2009) | 7 lines
  
  Use the correct list macros for deleting an item from the middle of a list.
  (issue #13777)
   Reported by: pj
   Patches: 
         20090203__bug13777.diff.txt uploaded by Corydon76 (license 14)
   Tested by: pj
........


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2009-02-16 23:17:01 +00:00
David Vossel
d9d35e070b Merged revisions 176248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176248 | dvossel | 2009-02-16 15:30:17 -0600 (Mon, 16 Feb 2009) | 11 lines
  
  Merged revisions 175597 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/trunk
  
  ........
    r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
    
    Fixed iax2 key rotation backwards compatibility
    
    Turns key rotation back on by default.  Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility.  This eliminates the need for the keyrotate option in iax.conf, so it has been removed.  
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@176251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 21:36:07 +00:00
Russell Bryant
fbec79a2f5 Merged revisions 176100 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r176100 | russell | 2009-02-16 11:09:24 -0600 (Mon, 16 Feb 2009) | 4 lines

Remove chan_features.

Review: http://reviewboard.digium.com/r/161/

........


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2009-02-16 17:10:16 +00:00
Joshua Colp
800c99a36c Merged revisions 176030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) | 16 lines
  
  Merged revisions 176029 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines
    
    Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog.
    
    This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value
    is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the
    pool was used for the value while the old was left untouched/unused. If the current pool was full a new
    pool was created. This would cause memory usage to increase steadily.
    
    (issue #AA50-2332)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@176032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 15:38:20 +00:00
Michiel van Baak
9dab42b819 Merged revisions 175952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines
  
  Merged revisions 175921 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines
    
    fix mis-spelling of the word registered.
    Reported by De_Mon on #asterisk-dev.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@176023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 09:42:46 +00:00
Russell Bryant
3e2b976fc0 Merged revisions 175829 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r175829 | russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines

Fix a number of problems with ast_sched_report().

1) It had numerous coding guidelines violations with regards to formatting.

2) It allocated memory using ast_calloc() that was never freed.

3) It didn't check for failure from the allocation.

4) It used sprintf() and strcat() to build the result, doing zero checking to
   prevent writing past the end of the provided buffer.

The function also lacks API documentation, but that has not been addressed in
this commit.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@175831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-15 21:00:58 +00:00
David Vossel
972e704692 Merged revisions 175597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
  
  Fixed iax2 key rotation backwards compatibility
  
  Turns key rotation back on by default.  Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility.  This eliminates the need for the keyrotate option in iax.conf, so it has been removed.  
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@175662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 20:48:47 +00:00
Russell Bryant
30e5135fec Merged revisions 175368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines

Remove useless string copy, and make sscanf safe again

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2009-02-12 21:41:36 +00:00
Russell Bryant
b792800b01 Merged revisions 175295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines

Avoid using ast_strdupa() in a loop.

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2009-02-12 20:48:13 +00:00
Kevin P. Fleming
63b7cfb593 Merged revisions 175250 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb 2009) | 1 line
  
  correct warning message to not refer specifically to DAHDI
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2009-02-12 18:50:04 +00:00
David Vossel
7827a31bfe Merged revisions 175127 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175127 | dvossel | 2009-02-12 11:07:17 -0600 (Thu, 12 Feb 2009) | 4 lines
  
  Setting key rotation to be off by default
  
  Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0).  As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off. 
........


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2009-02-12 17:09:55 +00:00
Joshua Colp
8fc9ea25ab Merged revisions 174710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174710 | file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines
  
  Only decrease inringing count if above zero.
  (issue #13238)
  Reported by: kowalma
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2009-02-10 20:17:43 +00:00
Joshua Colp
d14fe9fcee Merged revisions 174580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174580 | file | 2009-02-10 13:48:29 -0400 (Tue, 10 Feb 2009) | 4 lines
  
  Set the type for the peer structure to be a peer as the default.
  (closes issue #14447)
  Reported by: triccyx
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2009-02-10 17:49:45 +00:00
Joshua Colp
5ed074591b Merged revisions 174543 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174543 | file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines
  
  Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported.
  (closes issue #14399)
  Reported by: caspy
  (issue #13238)
  Reported by: kowalma
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2009-02-10 15:39:53 +00:00
Mark Michelson
1e4f1aff0c Merged revisions 174327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174327 | mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 lines

Fix something I messed up in the merge I just did


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2009-02-09 17:30:33 +00:00
Mark Michelson
b9b73e763b Merged revisions 174301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb 2009) | 20 lines

Merged revisions 174282 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines

Don't do an SRV lookup if a port is specified

RFC 3263 says to do A record lookups on a hostname
if a port has been specified, so that's what we're
going to do. See section 4.2.

(closes issue #14419)
Reported by: klaus3000
Patches:
      patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65)


........

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2009-02-09 17:26:39 +00:00
Dwayne M. Hubbard
697dd6060d Merged revisions 174084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) | 13 lines

Merged revisions 174082 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines

check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp()
and sip_uri_params_cmp()

The reporter didn't actually upload a properly-formed patch, instead a 
modified chan_sip.c file was uploaded.  I created a patch to determine the
changes, then modified the suggested changes to create a proper fix.  The
summary above is a complete description of the changes.

(closes issue #13547)
Reported by: tecnoxarxa
Patches:
      chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa

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2009-02-07 00:09:08 +00:00
Joshua Colp
4484e4e6d6 Merged revisions 174041 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 lines
  
  Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription.
  (closes issue #14322)
  Reported by: amessina
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