git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222614 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://reviewboard.asterisk.org/r/115/

(closes issue #15820)
Reported by: kshumard


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

........
  r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) | 8 lines
  
  crash on transfer
  
  handle_invite_replaces() attempts to uplock a pvt's
  owner channel without first verifing that it exists.
  
  (issue #16027)
........


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

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  r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) | 8 lines
  
  Add missing unlock(s) in dahdi_read
  
  (two cases in trunk)
  
  (closes issue #15683)
  Reported by: alecdavis
........


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222399 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds support for TCP/TLS in the Contact header when using
NAT, specifically externip or externhost. The original issue was that
Asterisk sent 5060 as the port in the contact header whether TLS was
used or not. Additionally, this patch adds 2 config options to sip.conf,
specifically externtcpport and externtlsport. This allows a user to
specify different external ports for TCP and TLS other than those used
internally, this is especially useful in in a PAT/port redirection setup.
Thanks to ebroad for reporting the issue and providing the patch!

(closes issue #15880)
Reported by: ebroad
Patches:
      portmap.patch uploaded by ebroad (license 878)
      externtXXport_v2.patch uploaded by ebroad (license 878)
Tested by: ebroad

Review: https://reviewboard.asterisk.org/r/392/



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........
  r222393 | jpeeler | 2009-10-06 17:27:13 -0500 (Tue, 06 Oct 2009) | 11 lines
  
  Fix potential crash when entire span request is received.
  
  The variable index used in this scenario for accessing the dahdi_pvts was
  wrong and was most likely copied from the several other places it is used
  correctly.
  
  (closes issue #15998)
  Reported by: tsearle
  Patches: 
        dahdi_reset_crash.patch uploaded by tsearle (license 373)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222394 65c4cc65-6c06-0410-ace0-fbb531ad65f3

setvar).

I mistakenly reasoned that setvar would be used on all channels. Since it can
be set per channel, give each dahdi channel a copy of the variable.

(related to #15899)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222351 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change is done in such a way as to allow the driver to continue to
function with older databases which don't have these features.
(closes issue #16000)
 Reported by: jamicque
 Patches: 
       20091002__issue16000.diff.txt uploaded by tilghman (license 14)
       20091002__issue16000__1.6.1.diff.txt uploaded by tilghman (license 14)
 Tested by: jamicque


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222309 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The setvar line in chan_dahdi.conf is shared among all the channels, so make
sure to only free the resources only when the last channel is destroyed.

(closes issue #15899)
Reported by: tzafrir



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222298 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This affected the ~~EXTEN~~ hack, where a subroutine might have changed the
value before it was used in the caller.
Patch by myself, tested by ebroad on #asterisk


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222273 65c4cc65-6c06-0410-ace0-fbb531ad65f3

dahdievent_to_analogevent used a simple switch statement to convert DAHDI
event numbers to "ANALOG_*" event numbers. However "digit" events
(DAHDI_EVENT_PULSEDIGIT, DAHDI_EVENT_DTMFDOWN, DAHDI_EVENT_DTMFUP)
are accompannied by the digit in the low word of the event number.

This fix makes dahdievent_to_analogevent() return the event number as-is
for such an event.

This is also required to fix #15924 (in addition to r222108).  


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

........
  r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
  
  Fix ao2_iterator API to hold references to containers being iterated.
  
  See Mantis issue for details of what prompted this change.
  
  Additional notes:
  
  This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
  has become an enum instead of a macro, with a name that fits our
  naming policy; also, it is now necessary to call
  ao2_iterator_destroy() on any iterator that has been
  created. Currently this only releases the reference to the container
  being iterated, but in the future this could also release other
  resources used by the iterator, if the iterator implementation changes
  to use additional resources.
  
  (closes issue #15987)
  Reported by: kpfleming
  
  Review: https://reviewboard.asterisk.org/r/383/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.

This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.

In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).

(issue #15586)
Reported by: globalnetinc


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The reported bug was actually only for pulsedigit, dtmfup, and dtmfdown
handling. Also added recognition for fax events (just some verbose output) and
fixed handling for the ec_disabled_event. In order to make comparing the analog
version of events to the DAHDI events easier, the ordering has been changed to
follow that of the DAHDI events.

(closes issue #15924)
Reported by: tzafrir



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222108 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 Oct 2009) | 3 lines
  
  Removes unnecessary unlock, clarifies a memcpy.
........


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

........
  r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines
  
  Ensure the result of the hash function is positive.  Negative array offsets suck.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221971 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15973)
 Reported by: atis


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221881 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221880 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines
  
  Occasionally losing use of B channels in chan_misdn.
  
  I have not been able to reproduce the problem of losing channels.
  However, I have seen in the code a reentrancy problem that might give
  these symptoms.
  
  The reentrancy patch does several things:
  1) Guards B channel and B channel structure allocation.
  2) Makes the B channel structure find routines more precise in locating records.
  3) Never leave a B channel allocated if we received cause 44.
  
  The last item may cause temporary outgoing call problems, but they should
  clear when the line becomes idle.
  
  (closes issue #15490)
  Reported by: slutec18
  Patches:
        issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
  Tested by: rmudgett, slutec18
  
  (closes issue #15458)
  Reported by: FabienToune
  Patches:
        issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
  Tested by: FabienToune, rmudgett, slutec18
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221781 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines
  
  Fix a bunch of off-by-one errors
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221777 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221709 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221705 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The PRI channels can no longer change the channel name if a different B
channel is selected during call negotiation.  To prevent using the channel
name to infer what B channel a call is using and to avoid name collisions,
the channel name format is changed.

The new channel naming for PRI channels is:
DAHDI/ISDN-<span>-<sequence-number>


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221701 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15854)
Reported by: dvossel
Patches:
      sip_port_config_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel

Review: https://reviewboard.asterisk.org/r/357/


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221627 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221592 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines
  
  Use unsigned ints for portinuri flags.
........


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221554 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221484 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221436 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines
  
  Fix SRV lookup and Request-URI generation in chan_sip.
  
  This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct.  That field is used during RURI generation to determine if the port should be included in the RURI.  It is also used in some places to determine if an SRV lookup should occur.
  
  (closes issue #14418)
  Reported by: klaus3000
  Tested by: klaus3000, mnicholson
  
  Review: https://reviewboard.asterisk.org/r/369/
........


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

........
  r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines
  
  check bounds - prevents for buffer overflow
........
  r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines
  
  added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf
  
  (closes issue #15471)
  Reported by: dkerr
  Patches:
        csv_quote_14.txt uploaded by mnick (license )
  Tested by: mnick
........
  r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines
  
  changed the prototype definition of csv_quote
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221368 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Thanks, Josh.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
  
  Change the SSRC by default when our media stream changes
  
  Be default, change SSRC when doing an audio stream changes Asterisk doesn't
  honor marker bit when reinvited to already-bridged RTP streams,resulting in
  far-end stack discarding packets with "old" timestamps that areactually part of
  a new stream.  This patch sends AST_CONTROL_SRCUPDATE whenever there is a
  reinvite, unless the 'constantssrc' is set to true in sip.conf.
  
  The original issue reported to Digium support detailed the following situation:
  ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
  fromITSP, Asterisk dials the app server which sends a re-invite back
  toAsterisk--not to negotiate to send media directly to the ITSP, but to
  indicatethat it's changing the stream it's sending to Asterisk.  The app
  servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
  bit on the new stream.  Asterisk passes through the teimstamp of the new stream,
  butdoes not reset the SSRC, sequence numbers, or set the marker bit.
  
  When the timestamp on the new stream is older than the timestamp on the
  originalstream, the ITSP (which doesn't know there has been any change) discards
  the newframes because it thinks they are too old.  This patch addresses this by
  changing the SSRC on a stream update unless constantssrc=true is set in
  sip.conf.
  
  Review: https://reviewboard.asterisk.org/r/374/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines
  
  Avoid a potential NULL dereference.
  (closes issue #15865)
   Reported by: kobaz
   Patches: 
         20090915__issue15865.diff.txt uploaded by tilghman (license 14)
   Tested by: kobaz
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221201 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14740)
Reported by: pj
Patches:
      issue14740_09022009.diff uploaded by seanbright (license 71)
Tested by: seanbright, lmadsen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221090 65c4cc65-6c06-0410-ace0-fbb531ad65f3