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* Set the invitestate to INV_CALLING when we send a connected line reinvite.
This prevents us from potentially rapid-firing reinvites to a single peer.

* Use the astdb to store a peer's allowed methods. This prevents us from sending
an UPDATE during the interval between startup and the peer's first registration
if the peer does not support the UPDATE method.

* Handle Polycom's method of indicating allowed methods in REGISTER. Instead of
using an Allow header, they place the allowed methods in a methods= parameter
in the Contact header.

ABE-1873



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I still need to set up tests to make sure my examples are completely correct,
but I ran out of time tonight and felt that they at least would give an idea as
to how to use calendaring. I will try to test the examples and do some cleanup
on the docs tomorrow night.


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Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize away the Local
channel when using this option.

(closes issue #14829)
Reported by: licedey
Tested by: mmichelson, licedey, lmadsen

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Ensure that accidental calls to ast_string_field_free_memory() on embedded stringfield pools are safe.

It is possible for a stringfield manager structure (and pool) structure to be allocated
as part of a larger structure allocation (using ast_calloc_with_strinfields()); when
this is done, the stringfield pool cannot be separately freed, but users of the tructure
may not be aware (and shouldn't have to be aware) of whether the pool was embedded.
This patch modifies the behavior so that they can always call ast_string_field_free_memory()
and the function will do the right thing for both embedded and non-embedded situations.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197775 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This makes sure that we mark a method as being unallowed if we
receive a 405 response so that we don't continue to try to 
send that same type of message.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197740 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS
Exchange calendars. Exchange support has only been tested on Exchange Server 2k3
and does not support forms-based authentication at this time (patches *very*
welcome). Exchange support is also currently missing the ability to return a
list of a meting's attendees (again, patches are very, very welcome).

Features include:
  Querying a calendar for events over a specific time range
  Checking a calendar's busy status via the dialplan
  Writing calendar events via the dialplan (CalDAV and Exchange only)
  Handling calendar event notifications through the dialplan

(closes issue #14771)
Tested by: lmadsen, twilson, Shivaprakash

Review: https://reviewboard.asterisk.org/r/58


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Fix a bug where the trunkmtu setting was not set to the default value of 1240 on load but was on reload.


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines
  
  Use the address we already know when reloading a peer with nat=yes.
  
  If we already have an address for a peer, and we are reloading the sip
  configuration, try to use that address to contact the peer, instead of
  getting it from the Contact.
  
  (closes issue #15194)
  Reported by: ibc
  Patches:
        sip.patch uploaded by eliel (license 64)
        Tested by: manwe
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197621 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14833)
 Reported by: contactmayankjain
 Patches: 
       all_changes.patch uploaded by contactmayankjain (license 740)
       slightly modified by me


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
  
  Allow for media to arrive from an alternate source when responding to a reinvite with 491.
  
  When we receive a SIP reinvite, it is possible that we may not be able to process the
  reinvite immediately since we have also sent a reinvite out ourselves. The problem is
  that whoever sent us the reinvite may have also sent a reinvite out to another party,
  and that reinvite may have succeeded.
  
  As a result, even though we are not going to accept the reinvite we just received, it
  is important for us to not have problems if we suddenly start receiving RTP from a new
  source. The fix for this is to grab the media source information from the SDP of the
  reinvite that we receive. This information is passed to the RTP layer so that it will
  know about the alternate source for media.
  
  Review: https://reviewboard.asterisk.org/r/252
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197606 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since the message structure is allocated using ast_calloc_with_stringfields we do not need to
free the memory used for the stringfields as it will get freed when the message structure is.


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
  
  Add flags to chanspy audiohook so that audio stays in sync.
  
  There are two flags being added to the chanspy audiohook here. One
  is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
  we ensure that the read and write slinfactories on the audiohook do
  not skew beyond a certain tolerance.
  
  In addition, there is a new audiohook flag added here,
  AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
  a slinfactory to build up a substantial amount of audio before 
  flushing it. For this particular issue, this means that the person 
  spying on the call will hear the conversations in real time with very 
  little delay in the audio.
  
  (closes issue #13745)
  Reported by: geoffs
  Patches:
        13745.patch uploaded by mmichelson (license 60)
  Tested by: snblitz
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15074)
Reported by: pj


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197538 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Let's try that again, this time removing trailing whitespace and not leading
whitespace.  I can't believe no one noticed.


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines
  
  Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting.
  
  The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated
  (or it passes through unauthenticated) the proper nat flag is set.
  
  (closes issue #13823)
  Reported by: dimas
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197467 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added AstVoicemailContext

(closes issue #15155)
Reported by: scramatte
Tested by: suretec



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197431 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added new ObjectClass AsteriskVoiceMail, and AstAccountCallLimit attribute 
and cleaned up formatting and tested with OpenLDAP

(closes issue #15155)
Reported by: scramatte
Patches:
      asterisk.schema uploaded by scramatte (license 796)
Tested by: suretec

Review: [full review board URL with trailing slash]



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Previously they always set hangupcause 0, which is generally wrong. With this
change, we're setting some generic hangup causes. For 5xx errors, which indicate
some sort of problem with the remote server, we're now setting CONGESTION.

EDVX002


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........
  r197264 | seanbright | 2009-05-27 16:12:06 -0400 (Wed, 27 May 2009) | 6 lines
  
  Since we use bashisms in build_tools/mkpkgconfig, we should call on bash
  explicitly when running from the Makefile.
  
  (closes issue #15209)
  Reported by: seandarcy
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197265 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since we use bashisms in build_tools/mkpkgconfig, we should call on bash
explicitly when running from the Makefile, otherwise we get errors during a
'make install.'

(closes issue #15209)
Reported by: seandarcy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197260 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines
  
  Use a different determinator on whether to print the delimiter, since leading fields may be blank.
  (closes issue #15208)
   Reported by: ramonpeek
   Patch by me, though inspired in part by a patch from ramonpeek
........


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........
  r197124 | jpeeler | 2009-05-27 11:49:38 -0500 (Wed, 27 May 2009) | 12 lines
  
  Fix broken attended transfers
  
  The bridge was terminating immediately after the attended transfer was 
  completed. The problem was because upon reentering ast_channel_bridge
  nexteventts was checked to see if it was set and if so could possibly
  return AST_BRIDGE_COMPLETE.
  
  (closes issue #15183)
  Reported by: andrebarbosa
  Tested by: andrebarbosa, tootai, loloski
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197125 65c4cc65-6c06-0410-ace0-fbb531ad65f3

the sample configuration files.

(closes issue #15207)
Reported by: seandarcy


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........
  r197024 | seanbright | 2009-05-27 09:54:35 -0400 (Wed, 27 May 2009) | 17 lines
  
  Fix handling of the 'state_interface' option of the 'queue add member' CLI
  command.
  
  This change relates to r184980, which was a backport of the state interface
  changes to app_queue from trunk.  trunk and all of the 1.6.x branches are not
  affected.
  
  'queue add member' allows for specifying an interface to use for device state
  when adding a queue member via CLI, but the validation code was not properly
  updated to reflect this optional argument.
  
  (closes issue #15198)
  Reported by: loloski
  Patches:
        05272009_app_queue.diff uploaded by seanbright (license 71)
  Tested by: loloski
........


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or inaccessible configuration file.

Before this change, when chan_alsa failed to load due to a missing or
inaccessible configuration file, no message would be displayed.  With this
change, when chan_alsa fails to load due to a missing or inaccessible
configuration file, a message will be displayed.

(closes issue #14760)
Reported by: Nick_Lewis
Patches:
      chan_alsa.c-confload.patch uploaded by Nick (license 657)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15200)
Reported by: ajohnson
Patches:
      05262009_xmldoc.patch uploaded by seanbright (license 71)
Tested by: ajohnson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196948 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14988)
Reported by: tzafrir
Patches:
      configure.ac.diff uploaded by homesick (license 91)
      new_ast_check_osptk.m4 uploaded by homesick (license 91)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196946 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There is inconsistency in how we handle manager responses that are lists of
items and, unfortunately, third parties have come to rely on ActionID being on
every event within those lists instead of just keeping track of the ActionID for
the current response.  This change makes CoreShowChannels include the ActionID
with each CoreShowChannel event generated as a result of it being called.

(closes issue #15001)
Reported by: sum
Patches:
      patchactionid2.patch uploaded by sum (license 766)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196945 65c4cc65-6c06-0410-ace0-fbb531ad65f3