(issue ASTERISK-18680)
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(closes issue ASTERISK-18696)
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whether modules are embedded or not; using just the bare category name led to
accidentally enabling these options when users used the wrong "--enable"
operation on the menuselect command line.

Now the internal option names are prefixed with "EMBED_", so they won't be
the same as the name of the category containing the modules they control
the embedding of.
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If a simple switch was started on a device and then a specific call
made (such as redial or speed dial), on timeout of the simple switch
the call would be attempted again. This patch only allows the simple
switch to make a call if the substate is still in the collecting
digits mode.

Also added small debug message to dialAndAactivate sub. 

Tested by snuff and myself.



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  r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines
  
  Merged revisions 340970 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines
    
    Quiet RTCP Receiver Reports during fax transmission
    
    RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions.
    The ability to disable RTCP streams in res_rtp_asterisk was missing, so this
    code was added to support the bug fix.
    
    (closes issue ASTERISK-18400)
  ........
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(issue ASTERISK-18268)
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Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
displaying a WARNING message.

(closes issue ASTERISK-18610)
 Patch by: Kristijan_Vrban
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Party A calls Party B.
Party A DTMF blind transfers Party B to Party C.
Party A channel continues to execute dialplan.

* Fixed the return value of builtin_blindtransfer() to return the correct
value after a transfer so the dialplan will not keep executing.

* Removed unnecessary connected line update that did not really do
anything.

* Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer().

* Fixed leak of xferchan for failure cases in check_goto_on_transfer().

* Updated debug messages in builtin_blindtransfer() and
check_goto_on_transfer().

(closes issue ASTERISK-18275)
Reported by: rmudgett
Tested by: rmudgett
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  r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | 9 lines
  
  Only change the capabilities on the gateway when
  the session is been destroyed there is still
  a race condition that ends in a segfault.
  
  if the caps are changed the logic in res_fax_spandsp
  will run T30 code not gateway code to end the session.
  this has been experienced on a "slower" under spec system.
........


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  r340718 | schmidts | 2011-10-13 06:59:50 +0000 (Thu, 13 Oct 2011) | 9 lines
  
  Merged revisions 340717 via svnmerge from 
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  ........
    r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) | 3 lines
    
    storing the route-set also on a 181 response not only on 180,182 or 183.
  ........
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Avoid possible jump based on unitialized value
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There is no documented reason to not add the query field to the varlist
returned by a realtime multi query, despite the config category being
set to its value. Of course, there is no documentation that the category
should be set to the value either. There is lots of no documentation
when it comes to realtime. But, other engines do not skip this field so
I am forcing this backend to follow the convention, because not doing so
is very silly.
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  r340577 | schmidts | 2011-10-12 20:33:37 +0000 (Mit, 12 Okt 2011) | 9 lines
  
  Merged revisions 340576 via svnmerge from 
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  ........
    r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) | 3 lines
    
    Store route-set from provisional SIP responses so early-dialog requests can be routed properly
  ........
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  r340578 | twilson | 2011-10-12 13:57:19 -0700 (Wed, 12 Oct 2011) | 16 lines
  
  Merged revisions 340534 via svnmerge from 
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  ........
    r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) | 9 lines
    
    Update SIP realtime fullcontact regardless of caching
    
    We should update the fullcontact field in the realtime table whether or
    not rtcachefriends is set. There is no reason to treat a non-cached
    realtime entity differently than a cached in this regard.
    
    (closes issue ASTERISK-18446)
     Reported by: wdoekes
  ........
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The PRI channel alarms were initialized with an inverted sense.

(closes issue ASTERISK-18710)
Reported by: Tzafrir Cohen
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ASTERISK-12175 changed the p and X options to not interfere with the s
option when they are used together.  It makes more sense for the s option
to have priority for the DTMF '*' key since it cannot change its
activation code.  Otherwise, you could not use option s with the p or X
options.

JIRA AST-671
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(closes issue ASTERISK-18612)
Reported by: Tim Osman
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* Added a CLI "ss7 show channels" command that might prove useful for
future debugging.

* Made the incoming SS7 channel event check and gripe message uniform.

* Made sure that the DNID string for an incoming call is always
initialized.

(issue ASTERISK-17966)
Reported by: Kenneth Van Velthoven
Patches:
      jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett
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* Fixed deadlock potential calling dialog_unlink_all() in
__sip_autodestruct().  Found by helgrind.

* Fixed deadlock potential in handle_request_invite() after calling
sip_new().  Found by helgrind.

* The sip_new() function now returns with the created channel already
locked.

* Removed the dead code that starts a PBX in in sip_new().  No sip_new()
callers caused that code to be executed and it was a bad thing to do
anyway.

* Removed unused parameters and return value from dialog_unlink_all().

* Made dialog_unlink_all() and __sip_autodestruct() safely obtain the
owner and private channel locks without a deadlock avoidance loop.
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RFC 6234 is an update to RFC 3174 from which the code was originally taken.
It has a slightly better code, and a better phrased license (simple 3-clause
BSD).

* main/sha1.c is sha1.c from RFC 6234 with formatting changes only.
* include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234.
* Removed unused include of asterisk/sha1.h from main/channels.c

Review: https://reviewboard.asterisk.org/r/1503/

Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263

Merge-From: http://svn.asterisk.org/svn/asterisk/branches/10@340280


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* Fixed race between calling an AMI action callback and unregistering that
action.  Refixes ASTERISK-13784 broken by ASTERISK-17785 change.

* Fixed potential memory leak if an AMI action failed to get registered
because is already was registered.  Part of the ao2 conversion.

* Fixed AMI ListCommands action not walking the actions list with a lock
held.

* Fix usage of ast_strdupa() and alloca() in loops.  Excess stack usage.

* Fix AMI Originate action Variable header requiring a space after the
header colon.  Reported by Yaroslav Panych on the asterisk-dev list.

* Increased the number of listed variables allowed per AMI Originate
action Variable header to 64.

* Fixed AMI GetConfigJSON action output format.

* Fixed usage of res contents outside of scope in append_channel_vars().

* Fixed inconsistency of config file channelvars option.  The values no
longer accumulate with every channelvars option in the config file.  Only
the last value is kept to be consistent with the CLI "manager show
settings" command.

(closes issue ASTERISK-18479)
Reported by: Jaco Kroon
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(closes issue AST-654)


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  r340222 | twilson | 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011) | 8 lines
  
  On astdb conversion, also warn about permissions requirements
  
  The user running Asterisk must have permission to the directory
  the Asterisk database resides in since SQLite 3 needs to be able
  to create a journal file.
  
  (closes issue ASTERISK-18174)
........


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  r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10 Oct 2011) | 8 lines
  
  Add astdb conversion utility for Berkeley to SQLite 3
  
  If someone wants to backtrack from Asterisk 1.8 to 10 they can use the
  astdb2bdb utility to convert the database back to the Berkeley format
  that Asterisk 1.8 uses.
  
  Review: https://reviewboard.asterisk.org/r/1502/
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  r340220 | twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines
  
  Add a missing file for the astdb2bdb conversion utility
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  r340165 | mjordan | 2011-10-10 15:30:18 -0500 (Mon, 10 Oct 2011) | 20 lines
  
  Merged revisions 340164 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) | 13 lines
    
    Updated chan_sip to place calls on hold if SDP address in INVITE is ANY
    
    This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::.
    In this case, the call should be placed on hold.  Previously, we checked for
    the address being null; this patch keeps that behavior but also checks for
    the ANY IP addresses.
    
    Review: https://reviewboard.asterisk.org/r/1504/
    
    (closes issue ASTERISK-18086)
    Reported by: James Bottomley
    Tested by: Matt Jordan
  ........
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  r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines
  
  Merged revisions 340108 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines
    
    Load the proper XML documentation when multiple modules document the same application.
    
    This patch adds an optional "module" attribute to the XML documentation spec
    that allows the documentation processor to match apps with identical names from
    different modules to their documentation. This patch also fixes a number of
    bugs with the documentation processor and should make it a little more
    efficient. Support for multiple languages has also been properly implemented.
    
    ASTERISK-18130
    Review: https://reviewboard.asterisk.org/r/1485/
  ........
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Added some data to skinny packet structures to make compatible
with v17. Added protocolversion to device, set on registration
based on the version provided by device.

v17 includes some increased ip space for ip6. This patch increases
ip space in the packets but still only uses ip4. Some packet
structures duplicated (ip4 and ip6 types). ip4 type used unless
version is greater or equal to 17.

Tested by snuff and myself on 7961 with recent 8.5 firmware. Also
tested compatible with old 7960 and older 30VIPs.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340071 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Increase SKINNY_MAX_PACKET to 2000 bytes to handle some messages
in v17 that are greater than the old 1000 bytes. Also add some
useful logging regarding packet and session handling.

A device (with protocol v17) was sending a packet with length 
greater than 1000 which resulted in the TCP session being
destroyed and registration being retryed.



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  r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011) | 8 lines
  
  Return -1 to skinny_session if register rejected.
  
  If device registration is rejected, return -1 so that the session is
  destroyed immediately. Previously, a segfault would occur on a 
  graceful shutdown if a register is rejected and the skinny_session
  has not yet timed out.
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  r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011) | 9 lines
  
  Remove log message on traverse session list.
  
  On destroying a session, a list of sessions is traversed to find the 
  matching session. For each session not matching, skinny erroneously
  logged that the session was not matched. While technically correct
  the message was misleading, and tended to indicate errors that 
  were not there.
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  r339942 | igorg | 2011-10-09 08:18:02 +0700 (Вск, 09 Окт 2011) | 12 lines
  
  Merged revisions 339938 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) | 6 lines
    
    Fix compilation issue, caused by missed session structure
    
    (closes issue ASTERISK-18694)
    Reported by: alex70
  ........
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  r339885 | igorg | 2011-10-08 22:46:27 +0700 (Сбт, 08 Окт 2011) | 13 lines
  
  Merged revisions 339884 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | 7 lines
    
    
    Fix segfault in Unistim channel
    
    (closes issue ASTERISK-18638)
    Reported by: jonnt
  ........
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  r339831 | igorg | 2011-10-08 22:01:35 +0700 (Сбт, 08 Окт 2011) | 14 lines
  
  Merged revisions 339830 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | 8 lines
    
    
    Fix char array cast as short array in send_client() function (for ARM
    platform)
    
    (closes issue ASTERISK-17314)
    Reported by: jjoshua
  ........
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  r339777 | rmudgett | 2011-10-07 14:36:24 -0500 (Fri, 07 Oct 2011) | 12 lines
  
  Merged revisions 339776 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011) | 5 lines
    
    Initialize option flags for SendURL application.
    
    (closes issue ASTERISK-18574)
    Reported by: marcelloceschia
  ........
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  r339681 | wedhorn | 2011-10-06 15:47:08 -0500 (Thu, 06 Oct 2011) | 10 lines
  
  Fixed segfault on core stop gracefully.
  
  There was an issue that the cap and confcap pointers for each line and device
  were being memcpy'd so they all pointed to the same ast_format_cap. On
  destroying, a segfault occured on the second call to the same struct.
  
  skinny reload now works again as well.
  
  Tested by snuff (in trunk) and myself. 
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  r339720 | rmudgett | 2011-10-06 17:58:40 -0500 (Thu, 06 Oct 2011) | 27 lines
  
  Merged revisions 339719 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines
    
    Fix regression in configure script for libpri capability checks.
    
    JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer
    2 persistence issues with some telcos.  ASTERISK-18535 attempted to fix
    the unexpected requirement that libpri *must* have that feature to work
    with Asterisk.  The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI
    optional features required.  Unfortunately, I thought
    AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and
    deleted those lines for libpri.  The result was the HAVE_PRI_xxx defines
    that control the ability to use optional libpri features were also
    deleted.
    
    * Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
    features in a library that the source code could take advantage of if the
    code supports the feature.
    
    (closes issue ASTERISK-18687)
    Reported by: Norbert
    Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339721 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There was an issue that the cap and confcap pointers for each line and device
were being memcpy'd so they all pointed to the same ast_format_cap. On
destroying, a segfault occured on the second call to the same struct.

skinny reload now works again as well.

Tested by snuff and myself. 


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339680 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

................
  r339626 | rmudgett | 2011-10-06 12:53:00 -0500 (Thu, 06 Oct 2011) | 25 lines
  
  Merged revisions 339625 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) | 18 lines
    
    Fix debugging messages generated by 'udptl debug'.
    
    * Makes chan_sip set the tag to the channel name.
    
    * Fixes received debug message sequence number.
    
    * Removed tx/rx debug message type since it was hard coded to 0.
    
    * Made udptl.c logged message header consistent if possible: "UDPTL (%s): ".
    
    * Removed unused rx_expected_seq_no from struct ast_udptl.
    
    (closes issue ASTERISK-18401)
    Reported by: Kevin P. Fleming
    Patches:
          jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: Matthew Nicholson
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339627 65c4cc65-6c06-0410-ace0-fbb531ad65f3