https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) | 2 lines
  
  Change some variable names to make pickup code easier to understand.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) | 25 lines
  
  Crash when using directed pickup applications.
  
  The directed pickup applications can cause a crash if the pickup was
  successful because the dialplan keeps executing.
  
  This patch does the following:
  
  * Completes the channel masquerade on a successful pickup before the
  application returns.  The channel is now guaranteed a zombie and must not
  continue executing the dialplan.
  
  * Changes the return value of the directed pickup applications to return
  zero if the pickup failed and nonzero(-1) if the pickup succeeded.
  
  * Made some code optimizations that no longer require re-checking the
  pickup channel to see if it is still available to pickup.
  
  (closes issue #19310)
  Reported by: remiq
  Patches:
        issue19310_v1.8_v2.patch uploaded by rmudgett (license 664)
  Tested by: alecdavis, remiq, rmudgett
  
  Review: https://reviewboard.asterisk.org/r/1221/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319998 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines
  
  Adds legacy_useroption_parsing to address interoperability concerns.
  
  With the new option engaged, Asterisk should interpret user fields with useroptions
  contained within the userfield of the uri by stripping them out of the original message
  whenever a semicolon is encountered in the userfield string.
  
  (closes issue #18344)
  Reported by: danimal
  Tested by: jrose
  
  Review: https://reviewboard.asterisk.org/r/1223/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319866 | jrose | 2011-05-19 13:32:38 -0500 (Thu, 19 May 2011) | 11 lines
  
  Fix Randomize option on Park()
  
  The randomize option was generally not working like it should have at all on Park().
  This patch restores intended functionality.
  
  (closes issue #18862)
  Reported by: davidw
  Tested by: jrose
  
  Review: https://reviewboard.asterisk.org/r/1222/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319867 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319812 | markm | 2011-05-19 13:59:01 -0400 (Thu, 19 May 2011) | 9 lines
  
  In cel_odbc, an uninitialized RWLIST is attempted to be locked.
  
  Added INIT and DESTROY for the RWLIST odbc_tables
  
  (closes issue #19331)
  Reported by: kobaz
  Patches: 
        odbc_cel.patch uploaded by kobaz (license 834)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319813 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) | 21 lines
  
  CCSS generic agent with POTS and ISDN phones fail caller busy call-back test.
  
  If the following is true after a CCSS activation:
  * The generic agent is for an analog phone or ISDN phone.  (Caller party)
  * The called party becomes available.
  * The caller party is not available.
  
  When the caller party becomes available, the caller is not alerted to the
  called party being available.  The generic agent still thinks the caller
  is busy.
  
  * Fixed the generic agent device state event subscription to look for all
  device states that are considered available.
  
  * Encapsulated the device state test for CCSS generic device available in
  cc_generic_is_device_available().  Made the generic agent and monitor use
  the new function instead of the manually coded inline equivalent.
  
  JIRA AST-559
  JIRA SWP-3462
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319759 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines
  
  Merged revisions 319653 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines
    
    Merged revisions 319652 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines
      
      Make sure everyone gets an unhold when a transfer succeeds
      
      Some phones, like the Snom phones, send a hold to the transfer target after
      before sending the REFER. We need to make sure that we unhold the parties
      that are being connected after the masquerade. If Local channels with the /nm
      option are used when dialing the parties, hold music would still be playing on
      the transfer target, even after being connected with the transferee.
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319661 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines
  
  Unbreak the storing of registrations for restart
  
  The fix for issue 18882 broke retrieving non-realtime peers from the ast_db
  on restart/reload. This patch tries to unbreak things while leaving the intent
  of the original fix intact.
  (closes issue #19318)
  Reported by: remiq
  Patches: 
        diff.txt uploaded by twilson (license 396)
  Tested by: lmadsen, remiq
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319564 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r319529 | twilson | 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) | 24 lines
  
  Merged revisions 319528 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines
    
    Merged revisions 319527 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines
      
      Fix app_dial ring groups
      
      Revert part of r315643. We need to remove the datastore here as well.
      The code in bridging code will catch anything that app_dial might miss.
      
      (closes issue #19311)
      Reported by: mspuhler
      Patches: 
            issue_19311_no_answer.diff uploaded by elguero (license 37)
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319530 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines
  
  Merged revision 319468 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
  
  ..........
    r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines
  
    The mISDN HDLC mode is prevented on dialed channels.
  
    The use of mISDN HDLC mode is prevented if the mISDN dial technology
    option 'h1' is used when config option astdtmf=yes.
  
    There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC
    mode.  Instead of setting the channel to HDLC mode it is set to
    transparent(no dsp, no hdlc), although hdlc is not "no hdlc".  I.e the
    logging message is correct, but the if condition is not.
  
    Make check the nodsp and hdlc flags.
  
    JIRA ABE-2787
    JIRA SWP-3437
  ..........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The vars were either explicitly or implicitly not used.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG.

Add option to specify if and how much of the current time is put in
Q931_IE_TIME_DATE.
* Send date/time ie never.
* Send date/time ie date only.
* Send date/time ie date and hour.
* Send date/time ie date, hour, and minute.
* Send date/time ie date, hour, minute, and second.
* Send date/time ie default: Libpri will send date and hhmm only when in
NT PTMP mode to support ISDN phones.

(closes issue #19221)
Reported by: kenner

JIRA SWP-3396


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319367 | lmadsen | 2011-05-17 07:53:50 -0500 (Tue, 17 May 2011) | 10 lines
  
  Don't create [general] voicemail context when using users.conf
  
  Prior to this patch, app_voicemail would create a [general] context when parsing users.conf.
  
  (closes issue #18891)
  Reported by: pdugas
  Patches: 
        app_voicemail-ignore-general.patch uploaded by pdugas (license 1222)
        app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71)
  Tested by: pdugas
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319368 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319365 | lmadsen | 2011-05-17 07:39:37 -0500 (Tue, 17 May 2011) | 6 lines
  
  Make Debian init script lsb compliant
  
  (closes issue #18896)
  Reported by: manwe
  Patches: 
        debian_init_lsb.patch uploaded by manwe (license 1223)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319366 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Probably haven't been working for a couple of years. May still need
some more love, but they are now working, both as a hint device and
monitoring a hint. Changes centre around the long ago change
to remove the requirement for a device name in a skinny line, and 
changes to the transmit_* functions.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | 2 lines
  
  Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319262 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) | 13 lines
  
  Deadlock between generic CCSS agent and native ISDN CCSS.
  
  Deadlock can occur when the generic CCSS agent is deleting duplicate CC
  offers and the native ISDN CC driver is processing an incoming CC message.
  The cc_core_instances container lock cannot be held when an agent or
  monitor callback is invoked without the possibility of a deadlock.
  
  * Make kill_duplicate_offers() remove the reference in cc_core_instances
  outside of the container lock.
  
  JIRA AST-566
  JIRA SWP-3469
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319260 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r319204 | twilson | 2011-05-16 13:17:43 -0500 (Mon, 16 May 2011) | 11 lines
  
  Merged revisions 319202 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) | 4 lines
    
    Unlink a peer from peers_by_ip when expiring a registration
    
    Review: https://reviewboard.asterisk.org/r/1218/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319212 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r319145 | dvossel | 2011-05-16 10:57:26 -0500 (Mon, 16 May 2011) | 9 lines
  
  Merged revisions 319144 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) | 2 lines
    
    Fixes issue with peer ref-counting during handle_request_subscribe.
    (closes issue #19293)
    Reported by: irroot
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319146 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319142 | mnicholson | 2011-05-16 10:53:26 -0500 (Mon, 16 May 2011) | 8 lines
  
  Make sure tcptls_session exists before dereferencing it.
  
  (closes issue #19192)
  Reported by: stknob
  Patches:
        10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723)
  Tested by: vois, Chainsaw
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 state of the channel reverts to unknown this should be rejected.
 
 this is important for negotiating T.38 gateway see #13405

 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected.

 Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states.

 (closes issue #18889)
 Reported by: irroot
 Tested by: irroot, darkbasic, 	mnicholson

 Review: https://reviewboard.asterisk.org/r/1115



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319085 | pabelanger | 2011-05-16 10:35:21 -0400 (Mon, 16 May 2011) | 10 lines
  
  Support gmime-2.4
  
  (closes issue #18863)
  Reported by: tzafrir
  Patches:
        gmime-2.4-18.diff uploaded by tzafrir (license 46)
        Tested by: tzafrir
  
  Review: https://reviewboard.asterisk.org/r/1213/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319086 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319083 | dvossel | 2011-05-16 09:26:33 -0500 (Mon, 16 May 2011) | 5 lines
  
  Fixes Big Endian build issue.
  
  (closes issue #19298)
  Reported by: tzafrir
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319084 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When called, activatesub first cleans up the active sub and then
handles the sub passed. dialandactivatesub first sets sub->exten
and then calls activatesub. Revise handle_offhook to utilise the
callid sent to chan_skinny. Some other minor fixes especially around
d->hookstate (which still needs some more work).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319024 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318921 | bbryant | 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) | 8 lines
  
  Fixes a segmentation fault in dynamic hints when a channel technology isn't
  loaded for a hint.
  
  (closes issue #18495)
  Reported by: bertrand
  Tested by: bertrand
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318922 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) | 10 lines
  
  This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too
  much time has passed between sending audio.
  
  (closes issue #18206)
  Reported by: bernhardsi
  Patches: 
        res_srtp_unhold.patch uploaded by bernhards (license 1138)
  Tested by: bernhards, notthematrix
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318920 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318917 | bbryant | 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) | 11 lines
  
  This patch allows TCP peers into the ast_db where they were previously
  restricted.
  
  (closes issue #18882)
  Reported by: cmaj
  Patches: 
        patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt
        uploaded by cmaj (license 830)
  Tested by: cmaj
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318918 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) | 19 lines
  
  CDR's are being written immediately on caller hangup.
  
  CDR's are being written immediately on caller hangup.  The dialplan is not
  able to modify it in the h exten.  The h exten in the initial context is
  not run before closing CDR's when the bridge is unlinked if a macro is
  active and does not have an h exten.
  
  * Make ast_bridge_call() check for an h exten in the current context and
  if a macro is active then the initial context.  The first h exten found is
  then run before closing the CDR.
  
  (closes issue #18212)
  Reported by: leearcher
  Patches:
        issue18212_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: rmudgett
  
  Review: https://reviewboard.asterisk.org/r/1206/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318869 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There were some issues where if a simple switch was cancelled and a
new switch started before the first had timed out where the d->exten
would be used for both subchannels. This was bad leading to possible
invalid extensions if some digits had been entered in the abandoned
simple switch and the second one was completed before the first timed
out, or the second would be cancelled because d->exten would be set to
nothing on the time out of the first.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318833 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318720 | mnicholson | 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) | 4 lines
  
  Handle ipv6 addresses in the sent-by Via: field.
  
  This change fixes a regression in via header parsing and ipv6 handling.

  (closes issue #18951)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318785 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) | 14 lines
  
  PRI early media won't ring.
  
  And another way to pass early media.  Don't indicate that there is inband
  information present, just assume that the B channel is connected.
  
  * Restore clearing the dialing flag Rx squelch unconditionally when a
  PROCEEDING message comes in.
  
  (closes issue #19268)
  Reported by: tbsky
  Patches:
        issue19268_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: tbsky
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318784 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines
  
  Fix directed group pickup feature code *8 with pickupsounds enabled 
  
  Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues.
  
  1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.
  2). dialplan applications for directed_pickups shouldn't beep.
  3). feature code for directed pickup should beep on success/failure if configured.
  
  Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.
  
  Moved app_directed:pickup_do() to features:ast_do_pickup().
  
  Functions below, all now use the new ast_do_pickup()
  app_directed_pickup.c:
     pickup_by_channel()
     pickup_by_exten()
     pickup_by_mark()
     pickup_by_part()
  features.c:
     ast_pickup_call()
  
  (closes issue #18654)
  Reported by: Docent
  Patches: 
        ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585)
  Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett
  
  Review: https://reviewboard.asterisk.org/r/1185/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318672 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Consolidate the functions and add some debugging info. Allows to be
able to set a substate without explicitly knowing what the state is. 


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318635 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add the setsubstate_onhook to complete the initial substate handling
procedures. Added dumpsub(sub, forcehangup) which is the common way of
calling setsubstate_onhook. Dumpsub attempts to activate another sub
after setting the current one onhook.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318550 | twilson | 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) | 2 lines
  
  Comment out the REF_DEBUG that slipped in during debugging
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318552 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r318549 | twilson | 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) | 27 lines
  
  Merged revisions 318548 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) | 19 lines
    
    Clean up several chan_sip reference leaks
    
    Several situations in the code could lead to peers or sip_pvt references
    being leaked. This would cause RTP ports to never be destroyed (leading
    to exhaustion of all available RTP ports) and memory leaks.
    
    The original patch for this issue from rgagnon was the result of an
    obscene amount of testing and hard work, for which I am very grateful. I
    did some cleanup and added a few additional refcount fixes that I found.
    
    (closes issue #17255)
    Reported by: kvveltho
    Patches: 
          tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202)
    Tested by: rgagnon, twilson, wdoekes, loloski
    
    Review: https://reviewboard.asterisk.org/r/1101/
    Review: https://reviewboard.asterisk.org/r/1207/
    Review: https://reviewboard.asterisk.org/r/1210/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) | 15 lines
  
  Unable to pickup DAHDI/PRI call because call state is reported as DIALING.
  
  The channel state is not updated to RINGING when an ALERTING message is
  received.  Regression caused when sig_pri.c (also sig_ss7.c) extracted
  from chan_dahdi.c.
  
  * Added missing channel state update to RINGING when the
  AST_CONTROL_RINGING frame is queued for ISDN and SS7.
  
  (closes issue #19257)
  Reported by: alecdavis
  Patches:
        issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
  Tested by: alecdavis, rmudgett
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318500 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318436 | russell | 2011-05-10 10:13:16 -0500 (Tue, 10 May 2011) | 2 lines
  
  chan_iax2: change LOG_NOTICE to LOG_DEBUG in iax2_read().
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318437 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r318337 | twilson | 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) | 18 lines
  
  Merged revisions 318331 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) | 12 lines
    
    Don't offer video to directmedia callee unless caller offered it as well
    
    Make sure that when directmedia is enabled, that video is not offered to the
    callee even if it supports it. p->vrtp will not exist since the caller didn't
    offer video.
    
    (closes issue #19195)
    Reported by: one47
    Patches: 
          sip_cant_add_video_rtp uploaded by one47 (license 23)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318400 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) | 6 lines
  
  Remove references to res_features and its export file.
  
  The contents of res/res_features.c was moved to into main/features.c
  awhile ago.  There is no longer any need for the res/Makefile to reference
  res_features or the res_features linker exports file to exist.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318352 65c4cc65-6c06-0410-ace0-fbb531ad65f3