git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175512 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)

This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.

Along the way, some related work was done:

1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.

2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.

3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).

Review: http://reviewboard.digium.com/r/158/



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When the 'faxdetect' configuration option is used, one may also want to use
the 'faxbuffers' configuration option in chan_dahdi.conf.  This option will
dynamically use the configured 'faxbuffers' buffer policy on a channel for
the life of the call following the detection of fax tones.  The faxbuffers
buffer policy will be reverted during call teardown.

An example use of 'faxbuffers' is below.  This example would switch to using
6 buffers with a full buffer policy.

faxbuffers=>6,full


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175411 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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  r175407 | mmichelson | 2009-02-12 17:22:44 -0600 (Thu, 12 Feb 2009) | 12 lines
  
  Fix a place where filestreams were not refcounted properly
  
  This section was already present in trunk and other branches,
  but did not exist in 1.4.
  
  (closes issue #14395)
  Reported by: ZX81
  Patches:
        14395.patch uploaded by putnopvut (license 60)
  Tested by: ZX81
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This patch adds forceencryption=yes as an iax.conf option.  When force encryption is enabled, no unencrypted connections are allowed.  This insures all connections are encrypted.  This is a new feature, so CHANGES and iax.conf.sample are updated as well.   

(closes issue #13285)
Reported by: sgofferj
Tested by: russell
Review: http://reviewboard.digium.com/r/150/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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  r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines
  
  Fix crashes when receiving certain T.38 packets.  Also, increase the maximum
  size of T.38 packets and warn users when they try to set the limits above those
  maximums.
  (closes issue #13050)
   Reported by: schern
   Patches: 
         20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
   Tested by: schern
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines
  
  Fix ParkedCall event information for From field in the case of a blind transfer
  
  If the parker information can not be obtained from the peer, try and see if
  the BLINDTRANSFER channel variable has been set. Previously, a blind transfer
  to the ParkAndAnnounce app would return nothing for the From.
  
  Closes AST-189
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menuselect was not happy with this.


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

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  r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines
  
  Fix crash in event of failed attempt to transfer to parking
  
  The peer may not necessarily exist, such as in the case of a transfer to 
  ParkAndAnnounce. In this case don't try to play a sound to it.
........


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

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r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines

Don't send DTMF for infinite time if we do not receive an END event.

I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf, 
dtmftimeout, that was intended to handle this situation.  However, in between 
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.

The default timeout is 3 seconds.  However, it is worth noting that having
this be configurable at all is not really the recommended behavior in RFC 2833.
From Section 3.5 of RFC 2833:

      Limiting the time period of extending the tone is necessary
      to avoid that a tone "gets stuck". Regardless of the
      algorithm used, the tone SHOULD NOT be extended by more than
      three packet interarrival times. A slight extension of tone
      durations and shortening of pauses is generally harmless.

Three seconds will pretty much _always_ be far more than three packet 
interarrival times.  However, that behavior is not required, so I'm going to
leave it with our legacy behavior for now.

Code from svn/asterisk/team/russell/issue_14460

(closes issue #14460)
Reported by: moliveras

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Core show locks information involving an ao2_trylock did not
show the function that called ao2_trylock, but would instead
show ao2_trylock as the source of the lock. This is not useful
when trying to debug locking issues.

One bizarre note is that this logic is already in 1.4 but somehow
did not get merged to trunk or the 1.6.X branches.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175121 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13985)
Reported by: jcovert
Tested by: phsultan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines

Set the initiator attribute to lowercase in our replies when receiving calls.

This attribute contains a JID that identifies the initiator of the GoogleTalk
voice session. The GoogleTalk client discards Asterisk's replies if the 
initiator attribute contains uppercase characters.

(closes issue #13984)
Reported by: jcovert
Patches:
      chan_gtalk.2.patch uploaded by jcovert (license 551)
Tested by: jcovert

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174951 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If someone has configured the queue to play an position or holdtime
announcement, then it is odd and potentially unexpected to hear a 
"Thank you for your patience" sound when no position or holdtime
was actually announced.

This fixes the announcement so that the "thanks" sound is only played
in the case that a position or holdtime was actually announced.

There is a way that the "thank you" sound can be played without a
position or holdtime, and that is to set announce-frequency to a value
but keep announce-position and announce-holdtime both turned off.

(closes issue #14227)
Reported by: caspy
Patches:
      14227_v3.patch uploaded by putnopvut (license 60)
Tested by: caspy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174948 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The 'd' option would not work for channel types which use RTP to transport
DTMF digits. The only way to allow for this to work was to answer the channel
if we saw that this option was enabled.

I realized that this may cause issues with CDRs, specifically with giving false
dispositions and answer times. I therefore modified ast_answer to take another
parameter which would tell if the CDR should be marked answered.

I also extended this to the Answer application so that the channel may be answered
but not CDRified if desired.

I also modified app_dictate and app_waitforsilence to only answer the channel if it
is not already up, to help not allow for faulty CDR answer times.

All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
the changes except for the change to the Answer application will go in since we do
not introduce new features into stable branches

(closes issue #14164)
Reported by: DennisD
Patches:
      14164.patch uploaded by putnopvut (license 60)
Tested by: putnopvut

Review: http://reviewboard.digium.com/r/145



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........
  r174885 | tilghman | 2009-02-11 14:54:18 -0600 (Wed, 11 Feb 2009) | 13 lines
  
  Restore a behavior that was recently changed, when we fixed issue #13962 and
  issue #13363 (related to issue #6176).  When a hangup occurs during a Macro
  execution in earlier 1.4, the h extension would execute within the Macro
  context, whereas it was always supposed to execute only within the main context
  (where Macro was called).  So this fix checks for an "h" extension in the
  deepest macro context where a hangup occurred; if it exists, that "h" extension
  executes, otherwise the main context "h" is executed.
  (closes issue #14122)
   Reported by: wetwired
   Patches: 
         20090210__bug14122.diff.txt uploaded by Corydon76 (license 14)
   Tested by: andrew
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174886 65c4cc65-6c06-0410-ace0-fbb531ad65f3

We need to do this because while we know that the freeing of the channel may cause something to become
not in use we do not know this for sure. There may be another channel that is still up which would cause
it to be in use.
(closes issue #13238)
Reported by: kowalma
Patches:
      20090121__bug13238.diff.txt uploaded by Corydon76 (license 14)
Tested by: alecdavis


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When using the 'g' or 'e' options, the stack allocations that
were used could cause a stack overflow if a spyer stayed on the
line long enough without actually successfully spying on anyone.

The problem has been corrected by using static buffers and copying
the contents of the appropriate strings into them instead of using
functions like alloca or ast_strdupa



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174805 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The explanation behind this fix is a bit complicated, and I've already
typed it up in the code as a huge comment inside of manager.c, so I'll
give the abridged version here.

We needed a way to separate action-specific data from session-specific data.
Unfortunately, the only way to maintain API compatibility and to not have to
change every single manager action was to rename the current mansession structure
and wrap it inside a new mansession structure which actually contains action-
specific data.

(closes issue #14364)
Reported by: awk
Patches:
      14364_better.patch uploaded by putnopvut (license 60)
Tested by: putnopvut

Review: http://reviewboard.digium.com/r/148/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue #13238)
Reported by: kowalma


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174710 65c4cc65-6c06-0410-ace0-fbb531ad65f3

improve slinfactory API to remove implicit sample rate and require explicit sample rate selection by creator of the slinfactory



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........
  r174644 | file | 2009-02-10 14:50:50 -0400 (Tue, 10 Feb 2009) | 6 lines
  
  Go off hold when we get an empty reinvite telling us to.
  (closes issue #14448)
  Reported by: frawd
  Patches:
        hold_invite_nosdp.patch uploaded by frawd (license 610)
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines
  
  Improve behavior of jitterbuffer when maxjitterbuffer is set.
  
  This change improves the way the jitterbuffer handles maxjitterbuffer and
  dramatically reduces the number of frames dropped when maxjitterbuffer is
  exceeded.  In the previous jitterbuffer, when maxjitterbuffer was exceeded, all
  new frames were dropped until the jitterbuffer is empty.  This change modifies
  the code to only drop frames until maxjitterbuffer is no longer exceeded.
  
  Also, previously when maxjitterbuffer was exceeded, dropped frames were not
  tracked causing stats for dropped frames to be incorrect, this change also
  addresses that problem.
  
  (closes issue #14044)
  Patches:
        bug14044-1.diff uploaded by mnicholson (license 96)
  Tested by: mnicholson
  Review: http://reviewboard.digium.com/r/144/
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(closes issue #14447)
Reported by: triccyx


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174580 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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from app_rpt.c.


(closes issue #14435)
Reported by: D_McNaul



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

........
  r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines
  
  This patch solves some compiler complaints
  in both 32 and 64-bit environments.
........


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The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued.  If the external script expects to be notified of a hangup and never receives one, it runs indefinitely. 

(closes issue #14251)
Reported by: chris-mac
Tested by: dvossel



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174325 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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  r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines
  
  Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off.
  (closes issue #14407)
  Reported by: mostyn
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