[asterisk-commits] russell: tag 1.2.19 r69597 - in /tags/1.2.19: .lastclean .version ChangeLog

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Jun 15 16:00:18 CDT 2007


Author: russell
Date: Fri Jun 15 16:00:17 2007
New Revision: 69597

URL: http://svn.digium.com/view/asterisk?view=rev&rev=69597
Log:
importing files for 1.2.19 release

Added:
    tags/1.2.19/.lastclean   (with props)
    tags/1.2.19/.version   (with props)
    tags/1.2.19/ChangeLog   (with props)

Added: tags/1.2.19/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.2.19/.lastclean?view=auto&rev=69597
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--- tags/1.2.19/ChangeLog (added)
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@@ -1,0 +1,5881 @@
+2007-06-15  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.2.19 released
+
+2007-06-14 23:21 +0000 [r69469]  Jason Parker <jparker at digium.com>
+
+	* config.c: Fix an issue where the line number in an unterminated
+	  comment block error message would show the wrong line number.
+	  "Reported" to me on #asterisk (somebody posted an error message,
+	  and I happened to catch it)
+
+2007-06-14 20:56 +0000 [r69347]  Russell Bryant <russell at digium.com>
+
+	* channel.c: Backport rev 69010 from the 1.4 branch ... In
+	  ast_channel_make_compatible(), just return if the channels' read
+	  and write formats already match up. There are code paths that
+	  call this function on a pair of channels multiple times. This
+	  made calls fail that were using g729 in some cases. The reason is
+	  that codec_g729a will unregister itself from the list of
+	  available translators will all licenses are in use. So, the first
+	  time the function got called, the right translation path was
+	  allocated. However, the second time it got called, the code would
+	  not find a translation path to/from g729 and make the call fail,
+	  even if the channel actually already had a g729 translation path
+	  allocated. (SPD-32)
+
+2007-06-14 15:15 +0000 [r69258]  Jason Parker <jparker at digium.com>
+
+	* funcs/func_groupcount.c: Change a quite broken while loop to a
+	  for loop, so "continue;" works as expected instead of eating 99%
+	  CPU... Issue 9966, patch by me.
+
+2007-06-13 18:12 +0000 [r69127]  Joshua Colp <jcolp at digium.com>
+
+	* app.c: Return group counting to previous behavior where you could
+	  only have one group per category. (issue #9711 reported by
+	  irroot)
+
+2007-06-13 09:55 +0000 [r69053]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/misdn/isdn_msg_parser.c: restart indicator 0x80 is
+	  correct, at least that's what libpri does.
+
+2007-06-12 14:18 +0000 [r68921]  Joshua Colp <jcolp at digium.com>
+
+	* rtp.c: Bring RTP back to Asterisk at the end of a native bridge
+	  no matter what.
+
+2007-06-12 08:35 +0000 [r68732-68887]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c: if the bridged partner is mISDN too we
+	  should not send dtmf tones, they are transmitted inband always
+
+	* channels/chan_misdn.c: if we have already some digits, we just
+	  stop the tones.
+
+	* channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
+	  check for NULL Pointer when calling misdn_new. Asterisk does not
+	  allow us to create channels anymore when stop gracefully is used
+	  :). also modified the restart_indicator to 0
+
+2007-06-11 14:29 +0000 [r68682]  Joshua Colp <jcolp at digium.com>
+
+	* channel.c: Improve deadlock handling of the channel list. (issue
+	  #8376 reported by one47)
+
+2007-06-11 09:18 +0000 [r68631]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+	  channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
+	  channels/misdn/isdn_lib.c: fixed problem that the dummybc chanels
+	  had no lock, checking for the lock now. Also fixed the channel
+	  restart stuff, we can now specify and restart particular channels
+	  too.
+
+2007-06-08 22:22 +0000 [r68526]  Russell Bryant <russell at digium.com>
+
+	* apps/app_dictate.c: Don't automatically hang up after running
+	  Dictate so that callers can exit cleanly using '#' (closes issue
+	  #9577, patch from Thomas Andrews)
+
+2007-06-08 00:15 +0000 [r68368-68397]  Joshua Colp <jcolp at digium.com>
+
+	* say.c: Don't call ast_waitstream_full when the control file
+	  descriptor and audio file descriptor are not set, simply call
+	  ast_waitstream! (issue #8530 reported by rickead2000)
+
+	* dnsmgr.c: Do a DNS lookup immediately upon calling the dnsmgr
+	  function, don't wait until a refresh happens. (issue #9097
+	  reported by plack)
+
+2007-06-07 23:13 +0000 [r68351]  Russell Bryant <russell at digium.com>
+
+	* say.c: Fix a problem where saying a character wouldn't properly
+	  break out when the caller pressed '#' (issue #8113, reported by
+	  patbaker82, patch from jamesgolovich (hey, long time no see!) and
+	  patbaker82)
+
+2007-06-07 20:02 +0000 [r68204]  Jason Parker <jparker at digium.com>
+
+	* apps/app_voicemail.c: Don't try to save voicemail greetings
+	  unless the user presses '1' to accept/save. Issue 9904, patch by
+	  me.
+
+2007-06-07 14:19 +0000 [r67938-68070]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_dial.c: Allow the 'g' option to work if used with the
+	  'S' option. (issue #9888 reported by gasparz)
+
+	* channels/chan_sip.c: Only notify the devicestate system of a peer
+	  state change when the peer is built from the config file. (issue
+	  #9900 reported by arkadia)
+
+2007-06-06 16:40 +0000 [r67715]  Russell Bryant <russell at digium.com>
+
+	* channel.c: We have some bug reports showing crashes due to a
+	  double free of a channel. Add a sanity check to
+	  ast_channel_free() to make sure we don't go on trying to free a
+	  channel that wasn't found in the channel list. (issue #8850, and
+	  others...)
+
+2007-06-06 13:28 +0000 [r67593-67649]  Joshua Colp <jcolp at digium.com>
+
+	* rtp.c: Reinvite the RTP back to the Asterisk machine when the
+	  timeout happens. (issue #9888 reported by gasparz)
+
+	* devicestate.c: Revert channel name splitting fix for Zap. The
+	  moral of the story is don't use - in your user/peer names. (issue
+	  #9668 reported by stevedavies)
+
+2007-06-05 15:42 +0000 [r67306-67307]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/misdn/chan_misdn_config.h: briding is a bool, fixed copy
+	  and paste issue.
+
+	* channels/chan_misdn.c: simplified the EVENT_SETUP handling in the
+	  cb_events function a lot. Commented the different possibilities a
+	  bit and made functions of shared code. When the dialed extension
+	  does not exist in the extensions.conf we'll jump into the 'i'
+	  extension if this does exist, else we disconnect the call with
+	  the cause:1 = No Route to Destination.
+
+2007-06-05 11:18 +0000 [r67239]  Nadi Sarrar <ns at beronet.com>
+
+	* channels/misdn_config.c, channels/chan_misdn.c,
+	  channels/misdn/chan_misdn_config.h: Backport of the overlap_dial
+	  functionality from asterisk-1.4's chan_misdn.
+
+2007-06-05 10:05 +0000 [r67209]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/misdn_config.c, channels/chan_misdn.c,
+	  channels/misdn/chan_misdn_config.h: added possibility to
+	  deactivate bridging per port
+
+2007-06-04 23:41 +0000 [r67060-67161]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* funcs/func_math.c: According to MATH, 0+1181000386 = 1181000448.
+	  Oops.
+
+	* contrib/init.d/rc.debian.asterisk,
+	  contrib/init.d/rc.mandrake.asterisk,
+	  contrib/init.d/rc.redhat.asterisk,
+	  contrib/init.d/rc.gentoo.asterisk,
+	  contrib/init.d/rc.mandrake.zaptel,
+	  contrib/init.d/rc.slackware.asterisk: Add revision Id tags (by
+	  request of tzafrir)
+
+2007-05-31 16:12 +0000 [r66764]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: It is now possible for this path of
+	  execution to have the frame pointer be NULL, therefore we need to
+	  check for it before trying to access it. (issue #9836 reported by
+	  barthpbx)
+
+2007-05-31 15:58 +0000 [r66744]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_macro.c: Issue 9818 - Fix for issue 8329 breaks
+	  pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime,
+	  but only because we lack core API to do it.
+
+2007-05-29 21:49 +0000 [r66537]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* funcs/func_strings.c: If the value of a variable passed to
+	  FIELDQTY is blank, then FIELDQTY should return 0, not 1.
+
+2007-05-29 07:53 +0000 [r66349]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Issue #9802 - Change inuse counter on CANCEL
+
+2007-05-25 13:46 +0000 [r66127]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* say.c: Issue 9791 - Fix pronunciation of seconds in Dutch
+
+2007-05-24 14:40 +0000 [r65837]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Allow RFC2833 to be negotiated when an
+	  INVITE comes in without SDP and is not matched to a user or peer.
+	  (issue #9546 reported by mcrawford)
+
+2007-05-24 09:19 +0000 [r65767]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c: we should only activate the generator in
+	  chan_misdn, when asterisk hask not yet taken the call
+	  (WAITING4DIGS state). Alerting audio will be generated fomr
+	  asterisk for example.
+
+2007-05-23 20:46 +0000 [r65676-65682]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: ensure that variables are set on a newly
+	  created channel before we start a PBX on it
+
+	* channels/chan_iax2.c: if we are going to set variables on a newly
+	  created channel, it should be done *before* we start the PBX on
+	  it
+
+2007-05-23 13:06 +0000 [r65389-65588]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_zap.c: Revert revision 62417 as someone reported
+	  problems with it to Mark. This was related to issue #9588.
+
+	* apps/app_queue.c: Fix a memory leak that I just noticed in the
+	  device state handling in app_queue. On most device state changes,
+	  it would leak roughly 8 to 64 bytes (the length of the name of
+	  the device).
+
+2007-05-22 07:46 +0000 [r65328]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c: we stop the tones only when we're in the
+	  pre-call phase, otherwise e.g. when in CONNECTED state we should
+	  not stop tones when we receive an Information Message
+
+2007-05-18 20:56 +0000 [r65172]  Steve Murphy <murf at digium.com>
+
+	* apps/app_dial.c, include/asterisk/cdr.h, cdr.c: This update will
+	  fix the situation that occurs as described by 9717, where when
+	  several targets are specified for a dial, if any one them reports
+	  FAIL, the whole call gets FAIL, even though others were ringing
+	  OK. I rearranged the priorities, so that a new disposition, NULL,
+	  is at the lowest level, and the disposition get init'd to NULL.
+	  Then, next up is FAIL, and next up is BUSY, then NOANSWER, then
+	  ANSWERED. All the related set routines will only do so if the
+	  disposition value to be set to is greater than what's already
+	  there. This gives the intended effect. So, if all the targets are
+	  busy, you'd get BUSY for the call disposition. If all get BUSY,
+	  but one, and that one rings is not answered, you get NOANSWER. If
+	  by some freak of nature, the NULL value doesn't get overridden,
+	  then the disp2str routine will report NOANSWER as before.
+
+2007-05-18 18:10 +0000 [r65075-65122]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Not getting an ACK to a 200 OK in the
+	  initial invite is critical to the call.
+
+	* channels/chan_sip.c: Issue 9235 - part of the problem, maybe not
+	  all. Please retry with this patch (and no other patch) if you
+	  have problems with hanging SIP channels. Thank you. A special
+	  Thank You to WeBRainstorm that gave me access to his system.
+
+2007-05-18 11:23 +0000 [r64902-65007]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: fixed a
+	  warning regarding Keypad encoding. encode the IE sending_complete
+	  at the right position.
+
+	* channels/chan_misdn.c: we *need* to send a PROCEEDING when
+	  sending_complete is set, even if need_more_infos is requested.
+
+2007-05-17 21:14 +0000 [r64819]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* include/asterisk/linkedlists.h: How is it that we never caught
+	  that this is returning the opposite of our documentation, until
+	  now?
+
+2007-05-17 16:52 +0000 [r64758]  Jason Parker <jparker at digium.com>
+
+	* apps/app_voicemail.c: If we have a negative current message, we
+	  shouldn't go back even further... Issue 9727.
+
+2007-05-16 10:55 +0000 [r64514-64603]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Fixing possible bug in auth of BYE
+
+	* channels/chan_sip.c: Support SIP uri's starting with SIP: and
+	  sip: (reported by Tony Mountfield on the mailing list. Thanks!)
+
+	* channels/chan_sip.c: Issue #9726 - rlister - Better logging for
+	  ACL denials While at it, also added better logging and handling
+	  of peers that are not supposed to register. My patch, stole the
+	  issue report from Russell. My apologies, Russell :-)
+
+2007-05-16 08:23 +0000 [r64513]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c: in the case immediate=yes, we directly
+	  jump into the dialplan, where people can use PlayTones to
+	  indicate a Dialtone, so we don't need to to that by ourself. also
+	  we should not do a dialtone_indicate for incoming calls on a TE
+	  port in overlapdialmode.
+
+2007-05-14 18:34 +0000 [r64275]  Joshua Colp <jcolp at digium.com>
+
+	* devicestate.c: Only perform stripping of - strings from the
+	  channel name for Zap channels. Anywhere else we might remove a
+	  legitimate part of a device name. (issue #9668 reported by
+	  stevedavies)
+
+2007-05-11 16:31 +0000 [r63903]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* Makefile, contrib/scripts/safe_asterisk: Issue 9121 - fixups for
+	  safe_asterisk script
+
+2007-05-10 23:14 +0000 [r63828]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_iax2.c: Fix an issue with trying to kill a thread
+	  before it gets created. Issue 9709, patch by nic_bellamy.
+
+2007-05-10 20:38 +0000 [r63748]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Do not allocate SIP pvt's for PEERs we can
+	  not reach. This was seen as a lot of dialogs being created then
+	  immediately destroyed at reload/restart of the SIP channel.
+
+2007-05-09 17:20 +0000 [r63653]  Matthew Fredrickson <creslin at digium.com>
+
+	* channels/chan_zap.c: Make sure we only create a DSP if it's
+	  requested on SUB_REAL
+
+2007-05-09 16:51 +0000 [r63610]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Properly handle hints that point to multiple
+	  devices in chan_sip. Why chan_sip is even doing this I have no
+	  idea but I would rather not go into a rant. (issue #9536 reported
+	  by rlister)
+
+2007-05-09 14:48 +0000 [r63565]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_directory.c: Replicate fix from 51158 (app_voicemail) to
+	  app_directory (Issue 9224)
+
+2007-05-09 11:26 +0000 [r63519]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c: release_chan frees ch, so we should never
+	  touch ch after release_chan, this may cause segfaults.
+
+2007-05-08 22:19 +0000 [r63477]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_macro.c: Issue 9602 - segfault in app_macro
+
+2007-05-08 15:07 +0000 [r63402]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: added
+	  application misdn_check_l2l1 which tries to pull up the L1/L2 on
+	  all ports that have the layers down in a group. It waits then for
+	  a timeout. This helps for scenarios where multiple PMP BRIs are
+	  grouped together, or where a provider has a faulty PTP
+	  Implementation, that looses the L2 after a while.
+
+2007-05-08 06:20 +0000 [r63359]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Issue 9527 - upon entering a folder, no
+	  message is selected (curmsg == -1), so deleting causes memory
+	  corruption (beyond bounds)
+
+2007-05-07 21:39 +0000 [r63285]  Joshua Colp <jcolp at digium.com>
+
+	* include/asterisk/app.h, app.c, channel.c: Properly handle what
+	  happens during a masquerade in relation to group counting. (issue
+	  #9657 reported by ramonpeek)
+
+2007-05-07 14:06 +0000 [r63195]  Olle Johansson <oej at edvina.net>
+
+	* config.c: Releasing the whole in-memory configuration while
+	  you're adding to it is not a good thing. Please review this
+	  change. Caused by additions (+) to non-existing contexts.
+
+2007-05-03 16:42 +0000 [r62987]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: When a peer is seeded or built tell the
+	  devicestate core to update it's status. This is easier then
+	  having chan_sip load before pbx_config. (issue #9658 reported by
+	  dlynes)
+
+2007-05-03 15:39 +0000 [r62885-62945]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: when
+	  we're in state WAITING4DIGS, we use the asterisk tone-generator
+	  which prods us, so we can't just return -1 in misdn_write in this
+	  case. Added a MISDN_KEYPAD channel variable, and fixed the
+	  sending of keypad. this enables us to modify the call forward
+	  parameters in the switch.
+
+	* channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: fixed
+	  the problem that misdn_write did not return -1 when called with 0
+	  samples in a frame this resultet in a deadlock in some
+	  circumstances, when the call ended because of a busy extension.
+	  added encoding of keypad.
+
+2007-05-03 00:23 +0000 [r62796-62841]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* res/res_config_odbc.c: doh... initializing the pointer variable
+	  will work just a bit better
+
+	* res/res_config_odbc.c: increase reliability and efficiency of
+	  static Realtime config loading via ODBC: don't request fields we
+	  aren't going to use don't request sorting on fields that are
+	  pointless to sort on explicitly request the fields we want,
+	  because we can't expect the database to always return them in the
+	  order they were created (reported by blitzrage in person (!),
+	  patch by me)
+
+2007-05-02 20:10 +0000 [r62737]  Steve Murphy <murf at digium.com>
+
+	* cdr.c, pbx.c: Some tweaks to satisfy CDR bug 8796, where being in
+	  'h' extension louses up the dst field
+
+2007-05-02 17:38 +0000 [r62691]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* channels/chan_iax2.c: Issue 9638 - if a text frame is sent with
+	  no terminating NULL through a bridged IAX connection, the remote
+	  end will receive garbage characters tacked onto the end.
+
+2007-05-01 21:55 +0000 [r62496-62547]  Russell Bryant <russell at digium.com>
+
+	* res/res_features.c: Remove an unnecessary check that makes it so
+	  if you hang up after doing an attended transfer before the target
+	  extension answers the channel, the transfer is not successful.
+	  (issue #9338, patch by svanlund)
+
+	* configs/indications.conf.sample: Add indications.conf information
+	  for the Philippines. (issue #9525, reported and patched by
+	  loloski)
+
+2007-04-30 15:57 +0000 [r62417]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_zap.c: This patch fixes an issue where depending on
+	  the cause code, when the network sends a PRI disconnect, the call
+	  may not be properly hung up. (issue #9588, reported and patched
+	  by softins)
+
+2007-04-30 14:34 +0000 [r62368]  Joshua Colp <jcolp at digium.com>
+
+	* asterisk.c: Update copyright notice. It's now the year 2007!
+
+2007-04-27 16:16 +0000 [r62173]  Jason Parker <jparker at digium.com>
+
+	* codecs/codec_zap.c: This transcoder message needn't be a NOTICE.
+	  I've seen it cause confusion more than a few times.
+
+2007-04-27 13:57 +0000 [r62126]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Issue #7351 - SIP Cancel fails due to the
+	  wrong contact uri. Reported by PPYY, failed to fix by OEJ final
+	  fix by wojtekka - THANKS!!!! THis was a hard one to catch.
+
+2007-04-26 16:30 +0000 [r61958-62037]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_iax2.c: Revert previous fix for when the IAX2
+	  channel goes funky (that's the technical term). This is causing
+	  legit calls to be prematurely hung up. (issue #9600 reported by
+	  justdave)
+
+	* config.c: Don't count failed include attempts against the
+	  configuration include level. (issue #9593 reported by mostyn)
+
+2007-04-25 22:24 +0000 [r61913]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_zap.c: handle a very bizarre race condition with
+	  channels being redirected before a simple switch can be started
+	  on them (issue #9286)
+
+2007-04-25 21:55 +0000 [r61862-61866]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: If the callerid= option is specified, but
+	  empty, clear any previous data.
+
+	* channels/chan_iax2.c: Ensure that callerid settings are reset on
+	  a reload.
+
+2007-04-25 18:52 +0000 [r61804]  Joshua Colp <jcolp at digium.com>
+
+	* include/asterisk/app.h, funcs/func_groupcount.c, app.c,
+	  apps/app_groupcount.c, channel.c: Merge rewritten group counting
+	  support. No more storing data on the variable list of the
+	  channels. That was bad, mmmk? (issue #7497 reported by sabbathbh)
+
+2007-04-25 16:20 +0000 [r61798]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_zap.c: Fix a typo where cid_num got copied instead
+	  of cid_ani. (issue #9587, reported and patched by xrg)
+
+2007-04-24  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.2.18 released
+
+2007-04-24 21:33 +0000 [r61786]  Russell Bryant <russell at digium.com>
+
+	* manager.c: Don't crash if a manager connection provides a
+	  username that exists in manager.conf but does not have a
+	  password, and also requests MD5 authentication. (ASA-2007-012)
+
+2007-04-24 18:20 +0000 [r61776-61777]  Dwayne M. Hubbard <dhubbard at digium.com>
+
+	* channels/chan_phone.c, channels/chan_zap.c,
+	  channels/chan_modem.c: removed #if 0 block from chan_phone,
+	  chan_zap, and chan_modem restart_monitor()
+
+	* channels/chan_modem.c: removed pthread_join in restart_monitor()
+	  to make it like chan_zap and other channel drivers
+
+2007-04-24 16:05 +0000 [r61771]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Allow RFC2833 to be sent in the response SDP
+	  when an INVITE comes in without SDP. (issue #9546 reported by
+	  mcrawford)
+
+2007-04-24 13:50 +0000 [r61770]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
+	  channels/misdn/isdn_lib.c: added lock for sending messages to
+	  avoid double sending. shuffled some empty_chans after the
+	  cb_event calls, this avoids that a release_complete from a quite
+	  different call releases a fresh created setup by accident.
+
+2007-04-20 21:14 +0000 [r61692-61704]  Jason Parker <jparker at digium.com>
+
+	* loader.c: Fix an issue that I noticed while looking over issue
+	  9571. The reload timestamp was getting set after reloading the
+	  built-in stuff, and before the modules.
+
+	* apps/app_queue.c: If the '* to hangup' option is not enabled, we
+	  don't need to disable * as a valid exit key. If it was enabled,
+	  this statement would've never been checked in the first place.
+	  Issue #9552
+
+2007-04-19 07:56 +0000 [r61685]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Send NOTIFY to Contact: in SUBSCRIBE - as
+	  reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp.
+
+2007-04-19 02:30 +0000 [r61680]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* funcs/func_language.c, funcs/func_moh.c, funcs/func_groupcount.c,
+	  funcs/func_timeout.c, funcs/func_cdr.c, funcs/func_callerid.c:
+	  Bug 9557 - Specifying the GetVar AMI action without a Channel
+	  parameter can cause Asterisk to crash. The reason this needs to
+	  be fixed in the functions instead of in AMI is because Channel
+	  can legitimately be NULL, such as when retrieving global
+	  variables.
+
+2007-04-16 14:08 +0000 [r61663]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Don't stop RTP on errors on INFO messages.
+	  Disclaimer: This patch was needed for Edvina AstHoloApp and was
+	  meant to be included in 1.2, but never made it in time so I felt
+	  I could add it now. No, just joking, patching error found while
+	  testing T.140 with Omnitor earlier this spring.
+
+2007-04-13 19:15 +0000 [r61655]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_dial.c: Add OUTBOUND_GROUP_ONCE variable to app_dial.
+	  This behaves the same as OUTBOUND_GROUP except it will get unset
+	  after use so it won't get accidentally inherited. (issue #BE-140)
+
+2007-04-11 16:01 +0000 [r61376-61476]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: If someone sets the "useragent" option in
+	  sip.conf to be empty, then don't add the User-Agent header at
+	  all. It is an optional header, anyway. Also, the bug report says
+	  that some of Japan's SIP providers don't allow it for some weird
+	  reason. (issue #9488, reported by makoto, fixed by me)
+
+	* channels/chan_sip.c: Fix a bug with switching between
+	  host=dynamic and using specific hosts for peers. The code would
+	  only reset the peer's address when it is dynamic if it was a new
+	  peer structure. Now, it will also reset the address if it was
+	  already in the peer list, but before the reload, it was not
+	  dynamic. (issue #9515, reported by caio1982, fixed by me)
+
+	* channels/chan_sip.c: Remove the attempt at reporting
+	  configuration errors in sip.conf. This can cause a bunch of
+	  improper messages when using realtime. I give up. As oej tried to
+	  convince me when I put this in, there is just no easy way to do
+	  it. (inspired by a message on the -dev list)
+
+2007-04-11 12:05 +0000 [r61357]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: some fixes for
+	  PMP Hold/Retrieve, it should work now, when briding=no
+
+2007-04-10 12:31 +0000 [r61170]  Nadi Sarrar <ns at beronet.com>
+
+	* channels/misdn_config.c: msns config parameter defaults to '*'
+
+2007-04-09 19:38 +0000 [r61038]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: - Don't send ActionID before Response:
+	  header. - Don't use a blank in an AMI header
+
+2007-04-09 17:22 +0000 [r60935]  Jason Parker <jparker at digium.com>
+
+	* apps/app_directory.c: Allow matching on names shorter than 3
+	  chars. This also fixes the case where somebody wants to match on
+	  less then 3 chars. Issue 9071
+
+2007-04-09 02:49 +0000 [r60846-60849]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* include/asterisk.h, asterisk.c: Don't check for error when
+	  lowering priority (according to the manpage, it should never
+	  happen anyway). It might could happen, though, if another thread
+	  messed with the priority, so safeguard against that (reported via
+	  -dev list).
+
+	* channels/chan_local.c: Bug 9505 - If the return value for
+	  local_queue_frame is set, then p->lock is no longer valid.
+
+2007-04-09 00:59 +0000 [r60797]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_dial.c: When calling a device that then forwards us
+	  elsewhere... we have to make our channels compatible if it is the
+	  only channel being dialed. (issue #9445 reported by
+	  marcelbarbulescu)
+
+2007-04-08 14:00 +0000 [r60660-60711]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_macro.c: Gosub called within a Macro resets the
+	  arguments improperly and causes general weirdness. (Issue 8329)
+
+	* file.c: Bug 9486 - memory leak when opening a filestream
+
+2007-04-06 17:03 +0000 [r60456]  Matt Frederickson <creslin at digium.com>
+
+	* channels/chan_zap.c: There should only be one code path for doing
+	  DTMF conditionals on channels. This fixes it.
+
+2007-04-06 14:41 +0000 [r60398]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* codecs/codec_zap.c: remove undocumented 'cardsmode' parameter and
+	  stop searching for transcoders during reload()
+
+2007-04-05 16:09 +0000 [r60267]  Jason Parker <jparker at digium.com>
+
+	* apps/app_voicemail.c: Just because we can't find the voicemail
+	  configuration file, doesn't mean that the module failed to load.
+	  The user could be using realtime. Issue #9473
+
+2007-04-05 12:52 +0000 [r60213]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Only unlock our pvt and net locks if we are
+	  actually going to try to lock the owner again. (issue #9472
+	  reported by zoa)
+
+2007-04-04 17:38 +0000 [r59886-60134]  Russell Bryant <russell at digium.com>
+
+	* manager.c: It is valid to redirect channels via the manager
+	  interface that are not in the UP state. Instead of checking for
+	  that to prevent to ensure a dead channel doesn't get redirected,
+	  just use the ast_check_hangup() API call. (issue #9457, reported
+	  by Callmewind, patch by me) (related to issue #8977)
+
+	* channels/chan_sip.c: Fix the return value of
+	  handle_common_options() so that it always properly indicates
+	  whether it handled the option or not. (issue #9455, reported by
+	  Netview, fixed by me)
+
+	* channels/chan_sip.c: Add a missing "\r\n" in the body of the
+	  NOTIFY that is sent to indicate the status of a transfer. (issue
+	  #9388, reported by rarritt)
+
+	* mkpkgconfig: Use the more generic check for "sed -r" support that
+	  was already present in 1.4. (related to issue #9399)
+
+	* mkpkgconfig: On Darwin, the -r argument to sed is not valid. It
+	  has to be -E. (issue #9399, reported by jcovert)
+
+	* channels/chan_sip.c: Don't attempt to report configuration errors
+	  in build_user(). oej pointed out that for a "friend" entry, this
+	  won't work, because all user options are valid for peers, but not
+	  the other way around.
+
+	* channels/chan_sip.c: Make chan_sip report when it encounters an
+	  unknown option. (issue #9440, reported by nightcrawler)
+
+	* app.c: When doing a built-in blind or attended transfer, restore
+	  the ability to use '#' to terminate the number and immediately do
+	  the transfer instead of having to dial the number and just wait
+	  for the feature digit timeout. (issue #8366, xueliangliang)
+
+2007-04-03 10:40 +0000 [r59788-59803]  Nadi Sarrar <ns at beronet.com>
+
+	* channels/misdn_config.c: ptp is the 5th bit, not the 4th.
+
+	* channels/misdn_config.c, channels/misdn/chan_misdn_config.h: Use
+	  the new sysfs way of mISDN 1.2 to check if a port is NT or not.
+
+2007-04-02 18:55 +0000 [r59723]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_voicemail.c: Increase the maximum size for a string of
+	  mailboxes to 1024. (issue #9270 reported by rtucker)
+
+2007-04-02 12:08 +0000 [r59623-59639]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/misdn_config.c, channels/chan_misdn.c,
+	  channels/misdn/chan_misdn_config.h: added option which allows us
+	  to accept incoming SETUP Messages without automatically sending
+	  Proceeding or Setup Acknowledge, this is useful with some broken
+	  switches and if you want to Release incoming calls without
+	  previously having acknowledged them. The new option is
+	  noautorespond_on_setup=yes|no default is no, so we don't break
+	  the existing behaviour
+
+	* channels/chan_misdn.c: don't be verbose if no need
+
+	* channels/misdn/isdn_lib.c: we can now make 30 channels on a PRI
+	  (before we forgot chan 31..)
+
+2007-04-01 22:35 +0000 [r59608]  Russell Bryant <russell at digium.com>
+
+	* netsock.c: Add the SO_REUSEADDR flag to sockets handled by
+	  netsock. This is needed by the patch that went in for issue 7874.
+	  chan_iax2 needs to be able to create socket that is lisetning on
+	  INADDR_ANY, but also be able to bind sockets to specific
+	  addresses. (Thanks to Stevenson on the asterisk-dev mailing list
+	  for explaining why this flag was needed.)
+
+2007-03-29 17:33 +0000 [r59360]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_meetme.c: Keep a global array of variables indicating
+	  whether certain conference rooms are in use. This ensures that
+	  two people going into a new dynamic conference when the 'e'
+	  option is set don't go into the same conference room. (issue
+	  #8835 reported by eliel)
+
+2007-03-29 17:14 +0000 [r59355-59357]  Russell Bryant <russell at digium.com>
+
+	* rtp.c: If an error occurs when reading from an RTP socket, and
+	  the error code does not indicate that we should try again, then
+	  return NULL instead of a "null frame". This will prevent Asterisk
+	  from trying over and over again, and eventually causing the
+	  system to crash. (issue #8285, john)
+
+	* channels/chan_iax2.c: Backport the change to chan_iax2 to return
+	  NULL instead of a "null frame" from its read callback. See
+	  revision 59341 to the 1.4 branch for more info.
+
+2007-03-29 16:04 +0000 [r59299-59301]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* cdr/cdr_odbc.c: Issue 9415 - No point to getting a diagnostic
+	  field if we aren't doing anything with the information. (Plus, it
+	  tends to crash the Postgres ODBC driver.)
+
+	* res/res_odbc.c: Change ENV section to use setenv, instead of
+	  putenv (Alexandru Pirvulescu <sigxcpu at gmail.com>, reported via
+	  -dev list)
+
+2007-03-27 23:36 +0000 [r59280-59283]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Oops
+
+	* apps/app_voicemail.c: Fix a few remaining bad mmap(2) return
+	  values
+
+2007-03-27 23:19 +0000 [r59258-59277]  Russell Bryant <russell at digium.com>
+
+	* apps/app_directory.c: Fix the check of the return value from
+	  mmap(). Thanks to Corydon for catching this one.
+
+	* channels/chan_iax2.c: Fix the use of the "sourceaddress" option
+	  when "bindaddr" is set to 0.0.0.0 instead of having each
+	  interface explicitly listed. (issue #7874, patch by stevens)
+
+2007-03-27 13:56 +0000 [r59252]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c: fixed #9355
+
+2007-03-26 10:21 +0000 [r59199]  Nadi Sarrar <ns at beronet.com>
+
+	* channels/misdn_config.c, channels/misdn/isdn_lib.h,
+	  channels/chan_misdn.c, channels/misdn/Makefile,
+	  channels/misdn/chan_misdn_config.h, channels/Makefile,
+	  channels/misdn/isdn_lib.c: mISDN >= 1.2 provides a dsp pipeline
+	  for i.e. echo cancellation modules, make chan_misdn use it.
+
+2007-03-24 01:35 +0000 [r59194]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Only try to handle a response if it has a
+	  response code. (ASA-2007-011)
+
+2007-03-23 16:07 +0000 [r59186-59187]  Steve Murphy <murf at digium.com>
+
+	* apps/app_macro.c: Ugh. that was dumb. Fixed an error.
+
+	* apps/app_macro.c: Added a few words in the Macro doc strings
+	  about the behavior of macros with hangups (et al.), as per 9337
+
+2007-03-21 18:03 +0000 [r59086]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_monitor.c: Indicate the filename changed when it is
+	  changed. (issue #9311 reported by jsmith)
+
+2007-03-20 09:23 +0000 [r59062-59063]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c: modified a loglevel
+
+	* channels/chan_misdn.c: avoid sending a disconnect when we already
+	  received one.
+
+2007-03-19  Jason Parker <jparker at digium.com>
+
+	* Asterisk 1.2.17 released
+
+2007-03-14 16:38 +0000 [r58896]  Russell Bryant <russell at digium.com>
+
+	* SECURITY: Add a note to the security file that the Asterisk CLI
+	  and log files may contain sensitive information, and that people
+	  should keep this in mind.
+
+2007-03-13 12:58 +0000 [r58849-58850]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c: fixed the crypt_keys stuff
+
+	* channels/misdn_config.c, channels/misdn/isdn_lib.h,
+	  channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
+	  channels/misdn/isdn_lib.c: added method standard_dec for dialing
+	  out on groups, to avoid conflicts, which caused issues with some
+	  ISDN providers
+
+2007-03-13 11:45 +0000 [r58847]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Issue #9229 - No port in request URI on
+	  register to non default SIP ports (neelakantan)
+
+2007-03-12 16:49 +0000 [r58832]  Joshua Colp <jcolp at digium.com>
+
+	* include/asterisk/lock.h: We can't use the assembler version of
+	  fetchadd_int under Intel Macs. (issue #9254 reported by darrell
+	  budic)
+
+2007-03-09 20:46 +0000 [r58579]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: If we are unable to lookup the host in a c
+	  line we have to abort, otherwise the previous data is gone and we
+	  will (potentially) have no data when all is said and done.
+
+2007-03-09 14:43 +0000 [r58558]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/misdn/isdn_lib.c: we can free channel 31 as well, since
+	  we can occupy it
+
+2007-03-08 16:04 +0000 [r58388]  Joshua Colp <jcolp at digium.com>
+
+	* dsp.c: Only print out debug message if the definition that makes
+	  the variables shows up was actually defined. (issue #9233
+	  reported by serginuez)
+
+2007-03-07 18:17 +0000 [r58164-58242]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix a problem where the Asterisk channel
+	  name could be that of the wrong IAX2 user for a call. This is
+	  because the first step of choosing this name is to look for an
+	  IAX2 peer that happens to have the same IP/port number that this
+	  call is coming from and assuming that is it. However, this is not
+	  always correct. So, I have made it change this name after
+	  authentication happens since at that point, we have an exact
+	  match.
+
+	* manager.c: Fix a misplaced block of code in the 1.2 version of
+	  the patch to fix issue #8977
+
+	* manager.c: If the channels acquired using the manager Redirect
+	  action are not up, then don't attempt to do anything with them.
+	  It could lead to weird behavior, including crashes. (issue #8977)
+
+2007-03-06 22:52 +0000 [r58115]  Steve Murphy <murf at digium.com>
+
+	* channels/chan_sip.c: Fix for 9220: Eyebeam cannot renew
+	  subscriptions for presence info. Reason: re-SUBSCRIBE requests
+	  don't include Accept headers, which the rfc says are optional (to
+	  put it tersely), (it uses MAY), and luckily, the sip_pvt struct
+	  has the format info stored, so we simply leave it if the format
+	  is set, and the accept header null.
+
+2007-03-06 20:33 +0000 [r58052]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Change error message to proper message
+
+2007-03-06 15:17 +0000 [r58008]  BJ Weschke <bweschke at btwtech.com>
+

[... 5008 lines stripped ...]



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