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20 authors
  1. Aug 09, 2012
    • Alexandr Anikin's avatar
      change opening h323 logfile with append mode instead of overwrite · 0d82844c
      Alexandr Anikin authored 
      ........

Merged revisions 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370989 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0d82844c
    • Kinsey Moore's avatar
      Correct documentation for the MeetMe x flag · 609061a8
      Kinsey Moore authored 
      The documentation for the x flag for MeetMe incorrectly described its
function as closing down the conference when the last marked user left.
It actually causes the users with that flag to leave the conference
when the last marked user exits. The functionality of this flag is not
changing.
........

Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370986 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      609061a8
    • Mark Michelson's avatar
      Extend extension state callbacks to have more information. · 9ee8b3c0
      Mark Michelson authored 
      Quote from review board:

This patch extends the extension state callbacks so that monitoring channels
(as chan_sip) get more information of the devices which are responsible for
an extension state change. The additional information is needed by chan_sip
to present names/numbers of the caller and callee in an early-state SIP
notification. Users of extenstion state callback not interested in the
additional information are not affected by the changes.

Motivation: to present the involved party's name/number in an early-state
nofification (used by the notified device as a pickup offer) one after another
so that a user can see which call he will pick up in an undirected pickup.
Such a pickup offer to a user shall indicate the same call (number/name-A calls
number/name-B) as the call which would be picked up when an undirected pickup
is executed.


Users interested in additional state info must use the new functions
ast_extension_state_add_extended() resp.
ast_extension_state_add_destroy_extended() to register an extended state
callback. When the callback is registered this way, an extra member
device_state_info of struct ast_state_cb_info is passed to the callback in
addition to the aggregated extension state. This container holds an object for
every device of the monitored extension hint consisting of the device name, the
device state and a channel reference to the channel which (presumably) caused
the device state.

The information is used by chan_sip for early-state notifications. When the
state of a device changes and the new state contains AST_EVENT_RINGING, an
early-state notification is sent to the subscribed devices with the
caller/callee names/numbers of the oldest ringing channel of the monitored
extension. The notified user may then invoke a direct pickup, which will pickup
exactly this channel.

Users of the old non-extended callbacks will only be called when the aggregated
state did change (same behavior as before). Users of the extended callback will
also be called when the state is unchanged but does contain AST_EVENT_RINGING.
That could be the case if two channels are ringing at one device and one of
them hangs up, so the aggregated state does not change. This way the monitoring
channel can create a new early-state notification with the now ringing
party-ids.

Review: https://reviewboard.asterisk.org/r/2048

This contribution comes from Guenther Kelleter



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9ee8b3c0
    • Jonathan Rose's avatar
      DUNDi: Add CLI commands DUNDi show cache and DUNDi show hints · 10672940
      Jonathan Rose authored 
      (closes issue ASTERISK-18390)
Reported by: Peter Racz
Patches:
	dundi_cli_cache.patch.v2 uploaded by Peter Racz (license #6290)
	ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by Jonathan Rose (license #6182)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      10672940
  3. Aug 08, 2012
  5. Aug 07, 2012
  7. Aug 06, 2012
  9. Aug 03, 2012
    • Mark Michelson's avatar
      Multiple revisions 370769-370771 · 9f0127f0
      Mark Michelson authored 
      ........
  r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri, 03 Aug 2012) | 24 lines
  
  Fix error in the "IPorHost" section of a SIP dialstring.
  
  This is based on the review request posted by Walter Doekes
  (referenced lower in the commit message)
  
  The main fix here is to treat the IPorHost portion of the dial
  string as a temporary outbound proxy. This ensures requests
  get sent to the proper location.
  
  Due to the age of the request, some parts were no longer relevant.
  For instance, the request moved outbound proxy parsing code into
  a single method. This is done in a previous commit, so it was not
  necessary to do again.
  
  Also, the review request fixed some errors with regards to request
  routing for CANCEL and ACK requests. This has also been fixed in
  more recent commits.
  
  (closes issue ASTERISK-19677)
  reported by Walter Doekes
  
  Review https://reviewboard.asterisk.org/r/1859
........
  r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug 2012) | 3 lines
  
  Remove unused variable.
........
  r370771 | mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5 lines
  
  Seriously? Another compilation error fixed.
  
  Somebody beat me.
........

Merged revisions 370769-370771 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370772 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9f0127f0
  11. Aug 02, 2012
  13. Aug 01, 2012
  15. Jul 31, 2012
    • Mark Michelson's avatar
      Add headers from SIPAddHeader to outbound REFER requests. · 4377d511
      Mark Michelson authored 
      This is a patch from kkm from review board.

This is useful for adding headers to REFER requests that
emanate from a Transfer() dialplan application call.

This also fixes some uses of the Referred-by header, removing
an extra set of angle brackets.

I've modified the reporter's original patch to not require
any additions to the sip_refer header and to just remove the
referred_by_name from sip_refer since it is no longer needed
or used.

(closes Issue ASTERISK-17639)
reported by Kirill Katsnelson
Patches:
	019059-sip-refer-addheaders-trunk-353549.diff
	uploaded by Kirill Katsnelson (license #5845)

Review: https://reviewboard.asterisk.org/r/1159



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4377d511
    • Mark Michelson's avatar
      Add "setvar" option to manager.conf. · 58f281a6
      Mark Michelson authored 
      With this option set, channel variables can be set on
every manager originate. The Variable header can still
be used to set additional channel variables for individual
calls if desired.

This work was completed by Olle Johansson on review board.
I have applied the review feedback and am committing it in
order to get this into trunk before Asterisk 11 is branched.

Review: https://reviewboard.asterisk.org/r/1412



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      58f281a6
    • Matthew Jordan's avatar
      Schedule pokes of registered SIP peers within a given timespan after SIP reload · d5d41741
      Matthew Jordan authored 
      With a large number of SIP peers registered, performing a SIP reload causes a
flood of SIP OPTIONS request packets.  These are immediately sent out, and, as
responses come back, can cause peers to be flagged as 'lagged' due to handling
of the many response messages.

This fix prevents this "packet storm" and schedules the pokes for a random
time.  That time varies between 1 ms and the peer's qualify time, or, if
the qualify time is unknown, the global qualifyfreq setting.

The committed patch has some very small modifications to the patch schmidts
wrote for the review.

(closes issue ASTERISK-19154)
Reported by: Nicolo Mazzon
patches:
  issue19154.patch license #6034 uploaded by schmidts

Review: https://reviewboard.asterisk.org/r/1652
........

Merged revisions 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370672 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d5d41741
    • Russell Bryant's avatar
      Move event cache updates into event processing thread. · 733b4602
      Russell Bryant authored 
      Prior to this patch, updating the device state cache was done by the thread
that originated the event.  It would update the cache and then queue the event
up for another thread to dispatch.  This thread moves the cache updating part
to be in the same thread as event dispatching.

I was working with someone on a heavily loaded Asterisk system and while
reviewing backtraces of the system while it was having problems, I noticed that
there were a lot of threads contending for the lock on the event cache.  By
simply moving this into a single thread, this helped performance *a lot* and
alleviated some deadlock-like symptoms.

Review: https://reviewboard.asterisk.org/r/2066/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      733b4602
    • Kinsey Moore's avatar
      Clean up and ensure proper usage of alloca() · 9b16c8b0
      Kinsey Moore authored 
      This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().

(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
........

Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9b16c8b0
    • Mark Michelson's avatar
      Add "dialplan remove context" and modify "dialplan add include" · 6c23a60f
      Mark Michelson authored 
      From corruptor's review board posting:

"I've noticed that we can remove particular extension from context with
dialplan remove extension command but in order to remove all extensions
in the context we should delete them on by one. I've created dialplan
remove context command which uses ast_context_destroy to destroy the
whole context with all extensions. I've created to functions for in
pbx_config.c: handle_cli_dialplan_remove_context which actually removes
context and complete_dialplan_remove_context which completes input.
They are based on other similar functions and pretty trivial but I can be
mistaken somewhere.

"I've also modified dialplan add include <context2> into <context1>. I've
made it similar dialplan add extension ... command. It creates <context1>
if it doesn't exist and I've also modified complete_dialplan_add_include
and removed check for existance of <context2> because we can include
non-existent context into another one. (I usually include empty
(non-existent) contexts in advance). Should we raise warning in this case
as it's raised while reading extensions.conf?

"I use those functions with AMI. I think manager commands should be created
in addition to those CLI commands."

I've addressed the latest comments on review board and have made some other
coding guidelines-related cleanup. I also have modified the CHANGES file to
mention these new commands.

(closes issue ASTERISK-19292)
reported by Andrey Solovyev

Patches:
	dialplan_add_include.patch
    uploaded by Andrey Solovyev (license #5214)
    dialplan_remove_context.patch
    uploaded by Andrey Solovyev (license #5214)

Review: https://reviewboard.asterisk.org/r/2042



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6c23a60f
    • Kinsey Moore's avatar
      Clean up chan_sip · e5210366
      Kinsey Moore authored 
      This clean up was broken out from
https://reviewboard.asterisk.org/r/1976/ and addresses the following:
 - struct sip_refer converted to use the stringfields API.
 - sip_{refer|notify}_allocate -> sip_{notify|refer}_alloc to match
   other *alloc functions.
 - Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not
   get_pidf_msg_text_body3 but get_content, to match add_content.
 - get_body doesn't get the request body, renamed to get_content_line.
 - get_body_by_line doesn't get the body line, and is just a simple if
   test. Moved code inline and removed function.
 - Remove camelCase in struct sip_peer peer state variables,
   onHold -> onhold, inUse -> inuse, inRinging -> ringing.
 - Remove camelCase in struct sip_request rlPart1 -> rlpart1,
   rlPart2 -> rlpart2.
 - Rename instances of pvt->randdata to pvt->nonce because that is what
   it is, no need to update struct sip_pvt because _it already has a
   nonce field_.
 - Removed struct sip_pvt randdata stringfield.
 - Remove useless (and inconsistent) 'header' suffix on variables in
   handle_request_subscribe.
 - Use ast_strdupa on Event header in handle_request_subscribe to avoid
   overly complicated strncmp calls to find the event package.
 - Move get_destination check in handle_request_subscribe to avoid
   duplicate checking for packages that don't need it.
 - Move extension state callback management in handle_request_subscribe
   to avoid duplicate checking for packages that don't need it.
 - Remove duplicate append_date prototype.
 - Rename append_date -> add_date to match other add_xxx functions.
 - Added add_expires helper function, removed code that manually added
   expires header.
 - Remove _header suffix on add_diversion_header (no other header adding
   functions have this).
 - Don't pass req->debug to request handle_request_XXXXX handlers if req
   is also being passed.
 - Don't pass req->ignore to check_auth as req is already being passed.
 - Don't create a subscription in handle_request_subscribe if
   p->expiry == 0.
 - Don't walk of the back of referred_by_name when splitting string in
   get_refer_info
 - Remove duplicate check for no dialog in handle_incoming when
   sipmethod == SIP_REFER, handle_request_refer checks for that.

Review: https://reviewboard.asterisk.org/r/1993/
Patch-by: gareth


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e5210366
  17. Jul 30, 2012