configs/samples/minivm.conf.sample contains invalid UTF-8, but that
appears to be intentional.

Change-Id: I7b1e0d332f3380fd0425962a3c9c55f9b200c8cc

Change-Id: I8f3d0a6c3f1075a1f7d8308593394611a96749de

Implements the following cli commands:
pjsip list aors
pjsip list auths
pjsip list channels
pjsip list contacts
pjsip list endpoints
pjsip show aor(s)
pjsip show auth(s)
pjsip show channels
pjsip show endpoint(s)

Also...
Minor modifications made to the AMI command implementations to facilitate
reuse.

New function ast_variable_list_sort added to config.c and config.h to implement
variable list sorting.

(issue ASTERISK-22610)
patches:
  pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
........

Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12


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This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.

A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.

Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/

(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
  path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
  oolong-path-support-trunk in team branch by oej (License 5267)



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Update Jean-Denis and add myself

(issue ASTERISK-20259)


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As discussed online http://lists.digium.com/pipermail/asterisk-dev/2012-October/057245.html the credits file needs some cleaning.  This is 95% whitespace with a few additions found in file headers.  Further additions should be added here instead of in the file being updated.

(issue ASTERISK-20259)


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https://origsvn.digium.com/svn/asterisk/branches/10

........
  r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines
  
  Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.
  
  When using Meetme as a modular call bridge from third party applications, it's handy to make
  it behave like a normal call bridge. When the second to last person exists, the last person
  will be kicked out of the conference when this option is enabled.
  
  (closes issue ASTERISK-18234)
  
  Review: https://reviewboard.asterisk.org/r/1376/
  
  Patch by oej, sponsored by ClearIT, Solna, Sweden
........


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Review: https://reviewboard.asterisk.org/r/1429/

(closes issue ASTERISK-18497)

Thanks to russellb for peer review.


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Add Asterisk Device State information and callbacks to the Call Completion
Supplemental Services for generic agents.

There are currently not many devices that have native support for CCSS.
Even as the devices become available there may be other reasons why one
may choose to not take advantage of the native abilities and stick with
the generic implementation.  The generic implementation is quite capable
and could be greatly enhanced by adding device state capabilities.  A
phone could then subscribe to the device state with a BLF key in
conjunction with Asterisk hints.

The advantages of the device state information would allow a single button
to: request CCSS, cancel a CCSS request, and display the current state of
a CCSS request.

For example, you may have a single button that when not lit, there is no
active CCSS request.  When you press that button, the dialplan can query
the DEVICE_STATE() associated with that caller to determine whether they
should be calling CallCompletionRequest() or CallCompletionCancel().  If
there is currently a pending request, then the dialplan would cancel it.
This also has the advantage of showing the true state of a request, which
is an asynchronous call, even when CallCompletionRequest() thinks it was
successful.  The actual request could ultimately fail.  Once lit, further
feedback can be provided to the caller about the current state of their
request since it will be updated by the CCSS State Machine as appropriate.

The DEVICE_STATE mapping is configurable since the BLF being used on a
given phone type may vary.  The idea is to allow some level of
customization as to the phone's behavior.

As an example, you may want the BLF key to go solid once you have
requested a callback.  You may then want the LED to blink (typically
ringing) when either the callback is in process, which is a visual
indication that the incoming call is the desired callback.  You may want
it to blink when the callee is ready but you are busy, giving you a visual
indication that the target is available as you may want to get off the
line so that the callback can be successful.

Device state information is sent back via the ast_devstate_prov_add()
callback for any generic CCSS device as it traverses through the state
machine.  You simply provide a map between CC_STATE values and the
corresponding AST_DEVICE state values.

You could then generate hints against these states similar to what is
possible today with Custom Devstates or MeetMe states.  For example, you
may have an extension 3000 that is currently associated with device
SIP/3000.  You could then create a feature code for that extension that
may look something like:

exten => *823000,hint,ccss:sip/3000

You would then subscribe a BLF button to *823000 which would point to the
dialplan that handled CCSS requests/cancels using the available
DEVICE_STATE() information about ccss:sip/3000 to make the decision about
what to do.

(closes issue #18788)
Reported by: p_lindheimer
Patches:
      ccss.trunk.18788.patch uploaded by p lindheimer (license 558)
      Modified with final reviewboard comments.
Tested by: p_lindheimer, loloski

Review: https://reviewboard.asterisk.org/r/1105/


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(closes issue #17600)
Reported by: minaguib
Patches:
      ast_unistim_height_v2.patch uploaded by minaguib (license 1078)
Tested by: minaguib


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This commit adds support for calendaring with Exchange Server 2007+ via
Exchange Web Services. Full write support and for querying attendees. Many
thanks to Jan Kaláb for the feature.

(closes issue #17022)
Reported by: pitel
Patches: 
      res_calendar_ews.c uploaded by pitel (license 1008)
Tested by: pitel, twilson

Review: https://reviewboard.asterisk.org/r/557/
Review: https://reviewboard.asterisk.org/r/668/


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For more details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/



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This commit introduces official support for R2 signaling in chan_dahdi.  The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.

Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1) 
are using it in each of the following countries: Colombia, Nepal, Thailand, 
Venezuela, Perú, and probably others.

To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.

The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.

I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message.  These are the names that I
could find in the mantis issue.

(closes issue #12509)
Reported by: moy
Patches:
      chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen

Review: http://reviewboard.digium.com/r/40/


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Update changes for SIPRemoveHeader()


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Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se)



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This set of changes introduces TCP and TLS support for chan_sip.  There are various
new options in configs/sip.conf.sample that are used to enable these features.  Also,
there is a document, doc/siptls.txt that describes some things in more detail.

This code was implemented by Brett Bryant and James Golovich.  It was reviewed
by Joshua Colp and myself.  A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names.  If you were one of them, thanks a lot for the help!

(closes issue #4903, but with completely different code that what exists there.)


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This set of changes introduces SIP session timers support (RFC 4028).  In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.

To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."

(closes issue #10665)
Reported by: rjain
Patches:
      chan_sip.c.1.diff uploaded by rjain (license 226)
      chan_sip.c.diff uploaded by rjain (license 226)
      sip.conf.sample.diff uploaded by rjain (license 226)
      proc_422_rsp_comment.diff uploaded by rjain (license 226)
      chan_sip.c.cache.diff uploaded by rjain (license 226)
      chan_sip.memalloc uploaded by rjain (license 226)
      chan_sip.memalloc.bugfix uploaded by rjain (license 226)

      Patches tracked in team/group/sip_session_timers, with some additional fixes
      by russell and oej.

Tested by: jtodd, rjain, loloski


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This introduces a new channel driver, chan_unistim, that supports the Unistim
VoIP protocol for Nortel phones.  The following models have been confirmed 
to work: i2002, i2004 and i2050.

(closes issue #8864)
Reported by: c_hans
Patches: 
      chan_unistim.patch uploaded by c (license 304)
      ustm_no_conf.diff uploaded by junky (license 177)
Tested by: c_hans, dbowerman, math, junky, loloski


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Asterisk), and obfuscate email addresses enough to stop most spam harvesters.


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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired. 
Feels very much like the old Unix talk application.

This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.

A big thank you to everyone involved in this.


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when the number of channels fill the MTU on a given link.

In the future, this needs to be configurable per peer with trunking enabled.



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https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r40692 | tilghman | 2006-08-20 17:09:57 -0500 (Sun, 20 Aug 2006) | 2 lines

Reformat to match the contribution style of other contributors

........


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credits fixed.


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