Commit Graph

13266 Commits

Author SHA1 Message Date
Tilghman Lesher
e891f2a92d After a 'sip reload', qualifies for realtime peers weren't immediately
restarted, instead waiting until the next registration.  We're now
caching the qualify across a reload/restart and starting the qualify
immediately upon loading the peer.
(closes issue #14196)
 Reported by: pdf
 Patches: 
       20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
 Tested by: pdf


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 00:49:22 +00:00
David Vossel
03dd54be23 Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging
This should have been committed with rev176247, but I missed it.  srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either.  This fixs that.

issue #13749




git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 23:30:52 +00:00
Kevin P. Fleming
4d808232cf correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 21:41:46 +00:00
Mark Michelson
b30800adfb Remove unused variable and make dev-mode compilation happy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 21:39:21 +00:00
Mark Michelson
5ae664d5aa Open the DAHDI pseudo device and set it to be nonblocking atomically
Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately
from opening the file was causing an "inappropriate ioctl for device" error.
While I cannot fathom why this would be happening, I certainly am not opposed
to making the code a bit more compact/efficient if it also fixes a bug.

(closes issue #14482)
Reported by: ys
Patches:
      meetme.patch uploaded by ys (license 281)
Tested by: ys



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 21:34:27 +00:00
David Vossel
1a00cbbf1d Fixes issue with AST_CONTROL_SRCUPDATE breaking out of native bridge
In iax2, when a AST_CONTROL_SRCUPDATE is received it breaks from the native bridge, but since there is no code path to handle srcupdate it just goes to be beginning of the loop.  This was causing packet storms of srcupdate frames between servers.  Now srcupdate frames do not break the native bridge for processing.    

(closes issue #13749)
Reported by: adiemus



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 21:28:20 +00:00
Kevin P. Fleming
1b5b3efcae fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 21:10:38 +00:00
Joshua Colp
22734e39dc Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog.
This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value
is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the
pool was used for the value while the old was left untouched/unused. If the current pool was full a new
pool was created. This would cause memory usage to increase steadily.

(issue #AA50-2332)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 15:33:53 +00:00
Michiel van Baak
db4dc67740 fix mis-spelling of the word registered.
Reported by De_Mon on #asterisk-dev.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175921 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-15 23:37:03 +00:00
Olle Johansson
4c3b9ccf3b format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-15 20:33:17 +00:00
Olle Johansson
beaf6760c4 Disable format_ilbc.so by default, like codec_ilbc.so
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-15 20:20:21 +00:00
Olle Johansson
ada21a8039 Make sure that the debug line is not printed on debug level 0
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-15 19:48:38 +00:00
Jason Parker
3cc3863d28 Zaptel is not DAHDI. Rather, Zaptel is actually Zaptel. (in case you're confused, DAHDI is still DAHDI)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 21:53:16 +00:00
Mark Michelson
3e0b36d9d0 Fix a potential crash situation when using IMAP voicemail
If calling into VoiceMailMain when using IMAP storage, it was
possible to crash Asterisk by hanging up the phone when prompted
for a voicemail mailbox. This patch fixes the issue.

While it may appear that this patch is superficial, it allows code
execution to continue to the failure case just below the IMAP_STORAGE
code block where this patch has been applied

(closes issue #14473)
Reported by: dwpaul
Patches:
      voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 19:47:48 +00:00
Mark Michelson
7accd1ec46 Fix a place where filestreams were not refcounted properly
This section was already present in trunk and other branches,
but did not exist in 1.4.

(closes issue #14395)
Reported by: ZX81
Patches:
      14395.patch uploaded by putnopvut (license 60)
Tested by: ZX81



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 23:22:44 +00:00
Tilghman Lesher
a13deff994 Fix crashes when receiving certain T.38 packets. Also, increase the maximum
size of T.38 packets and warn users when they try to set the limits above those
maximums.
(closes issue #13050)
 Reported by: schern
 Patches: 
       20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
 Tested by: schern


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 21:19:40 +00:00
Jeff Peeler
46963bc8b5 Fix ParkedCall event information for From field in the case of a blind transfer
If the parker information can not be obtained from the peer, try and see if
the BLINDTRANSFER channel variable has been set. Previously, a blind transfer
to the ParkAndAnnounce app would return nothing for the From.

Closes AST-189



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 20:34:36 +00:00
Jeff Peeler
3e396d458a Fix crash in event of failed attempt to transfer to parking
The peer may not necessarily exist, such as in the case of a transfer to
ParkAndAnnounce. In this case don't try to play a sound to it.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 17:57:10 +00:00
Russell Bryant
bfaa341f58 Don't send DTMF for infinite time if we do not receive an END event.
I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf, 
dtmftimeout, that was intended to handle this situation.  However, in between 
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.

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

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

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

Code from svn/asterisk/team/russell/issue_14460

(closes issue #14460)
Reported by: moliveras


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 16:51:13 +00:00
Philippe Sultan
d045d36561 Set the initiator attribute to lowercase in our replies when receiving calls.
This attribute contains a JID that identifies the initiator of the GoogleTalk
voice session. The GoogleTalk client discards Asterisk's replies if the 
initiator attribute contains uppercase characters.

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 10:16:21 +00:00
Joshua Colp
70f7c7e9cb Revert RTP changes for continuation of DTMF. Proxy commit by russell via SMS.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 00:19:30 +00:00
Russell Bryant
1d4e4ff3d1 Clear out the current event after forcing the end of a digit
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 00:01:02 +00:00
Russell Bryant
9dff8995b4 Fixify infinite DTMF in the case that no RFC2833 END event is ever received
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-11 23:56:37 +00:00
Tilghman Lesher
9e38dd5427 Restore a behavior that was recently changed, when we fixed issue #13962 and
issue #13363 (related to issue #6176).  When a hangup occurs during a Macro
execution in earlier 1.4, the h extension would execute within the Macro
context, whereas it was always supposed to execute only within the main context
(where Macro was called).  So this fix checks for an "h" extension in the
deepest macro context where a hangup occurred; if it exists, that "h" extension
executes, otherwise the main context "h" is executed.
(closes issue #14122)
 Reported by: wetwired
 Patches: 
       20090210__bug14122.diff.txt uploaded by Corydon76 (license 14)
 Tested by: andrew


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-11 20:54:18 +00:00
Joshua Colp
9fa3324845 Go off hold when we get an empty reinvite telling us to.
(closes issue #14448)
Reported by: frawd
Patches:
      hold_invite_nosdp.patch uploaded by frawd (license 610)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 18:50:50 +00:00
Matthew Nicholson
b60ec2baa1 Improve behavior of jitterbuffer when maxjitterbuffer is set.
This change improves the way the jitterbuffer handles maxjitterbuffer and
dramatically reduces the number of frames dropped when maxjitterbuffer is
exceeded.  In the previous jitterbuffer, when maxjitterbuffer was exceeded, all
new frames were dropped until the jitterbuffer is empty.  This change modifies
the code to only drop frames until maxjitterbuffer is no longer exceeded.

Also, previously when maxjitterbuffer was exceeded, dropped frames were not
tracked causing stats for dropped frames to be incorrect, this change also
addresses that problem.

(closes issue #14044)
Patches:
      bug14044-1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
Review: http://reviewboard.digium.com/r/144/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 17:52:42 +00:00
Steve Murphy
680cc35607 This patch solves some compiler complaints
in both 32 and 64-bit environments.




git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 02:27:40 +00:00
Mark Michelson
7f20e5ffab Don't do an SRV lookup if a port is specified
RFC 3263 says to do A record lookups on a hostname
if a port has been specified, so that's what we're
going to do. See section 4.2.

(closes issue #14419)
Reported by: klaus3000
Patches:
      patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09 17:11:05 +00:00
Joshua Colp
e2fd8852db Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off.
(closes issue #14407)
Reported by: mostyn


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09 14:48:21 +00:00
Russell Bryant
c347a43c9f Fix a race condition that could cause a crash.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-07 16:15:07 +00:00
Dwayne M. Hubbard
d29a99cb89 check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp()
and sip_uri_params_cmp()

The reporter didn't actually upload a properly-formed patch, instead a 
modified chan_sip.c file was uploaded.  I created a patch to determine the
changes, then modified the suggested changes to create a proper fix.  The
summary above is a complete description of the changes.

(closes issue #13547)
Reported by: tecnoxarxa
Patches:
      chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 23:36:03 +00:00
Joshua Colp
c80b2b93b5 Remove a debug message I put in by accident.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 17:15:01 +00:00
Joshua Colp
6cda579f17 Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string.
(closes issue #14350)
Reported by: fhackenberger


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 17:14:15 +00:00
Matthew Nicholson
5edf9d8a59 Limit the addition of the Contact header in SIP responses according to various
SIP RFCs.

(closes issue #13602)
Reported by: hjourdain
Tested by: mnicholson


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 16:20:23 +00:00
Tilghman Lesher
76af76c5f4 Backport OS X fix from trunk
(AGAIN, closes issue #14360)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 15:43:32 +00:00
Mark Michelson
f70845aa24 Fix logic regarding when to perform an SRV lookup for outgoing REGISTER requests
With this fix, we only will perform an SRV lookup at the following times:

* The first time we register with a remote registrar
* If we send a REGISTER but do not receive a response
* If the sendto() function returns an error

While I wrote the patch that fixes this issue, a huge amount of credit is due
to Brett Bryant, who wrote the initial patch on which I based this one.

(closes issue #12312)
Reported by: jrast
Patches:
      12312.patch uploaded by putnopvut (license 60)
Tested by: blitzrage

Review: http://reviewboard.digium.com/r/132/



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 23:19:16 +00:00
Jeff Peeler
0b026ce04f Add new configuration option to make shared IMAP mailboxes function as expected.
The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
using the same IMAP storage location to function as one mailbox. This allows
all messages to be retrieved for any user in the group. The patch alters the
'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
for a given user.

(closes issue #13673)
Reported by: howardwilkinson



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 20:47:51 +00:00
Mark Michelson
e7478c7b15 Fix situations where queue members could be autopaused unexpectedly
Specifically, this patch prevents us from autopausing members when
we receive a busy or congestion frame from them.

(closes issue #14376)
Reported by: fiddur
Patches:
      14376.patch uploaded by putnopvut (license 60)
Tested by: fiddur



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 20:29:09 +00:00
Mark Michelson
d914df762a Add some missing cleanup to app_mixmonitor
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 18:47:24 +00:00
Mark Michelson
8e764c583d Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up.
app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed
audio to a file. Since this thread runs independently of the channel, it is possible that
the mixmonitor thread's channel pointer will point to freed memory when the channel either
is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the
cases slightly differently).

The solution for this is to employ a datastore, which has the nice benefit of allowing us 
to hook into channel masquerades and hangups and update our pointer as necessary. If this
looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more
involved since it does a lot more operations on the channel that is being spied upon.

app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there
is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em-
ploy a condition-and-boolean combination to ensure that the channel thread finishes with
our structure before the mixmonitor thread attempts to free it. No crashes!

(closes issue #14374)
Reported by: aragon
Patches:
	  14374.patch uploaded by putnopvut (license 60)
Tested by: aragon, putnopvut



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 17:34:33 +00:00
Mark Michelson
82937be553 Revert my previous change because it was stupid
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 17:44:48 +00:00
Mark Michelson
fed7d2308b Add a missing unlock. Extremely unlikely to ever matter, but it's needed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 17:40:29 +00:00
David Vossel
28056ffc94 Fixes issue with IAX2 transfer not handing off calls.
Fixes issue with IAX2 transfers not taking place.  As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table.  The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required.  This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. 

(issue #13468)
Review: http://reviewboard.digium.com/r/140/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-03 23:35:55 +00:00
Jeff Peeler
87c921a68e Parking attempts made to one end of a bridge no longer will hang up due to a
parking failure.

Parking attempts made using either one-touch, or doing either a blind or 
assisted transfer to the parking extension now keep up the bridge instead of
hanging up the attempted parked party. Normal causes for the parking attempt
to fail includes the specific specified extension (via PARKINGEXTEN) not being 
available or if all the parking spaces are currently in use. To avoid having
to reverse a masquerade park_space_reserve was made to provide foresight if
a parking attempt will succeed and if so reserve the parking space.

(closes issue #13494)
Reported by: mdu113

Reviewed by Russell: http://reviewboard.digium.com/r/133/



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-03 21:57:01 +00:00
Tilghman Lesher
13138151e1 Add warning to standard config, that globals may be overridden by other
dialplan configuration files.
(closes issue #14388)
 Reported by: macli


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-03 00:15:59 +00:00
Terry Wilson
1a9018e799 Fix a feature inheritance bug I added after code review
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02 23:48:06 +00:00
Richard Mudgett
cefe4f025d channels/chan_dahdi.c
*  Added doxygen comments to the major dahdi structures.
*  Fixed PRI using an incorrect string value if the extension
delimiter is not present in the Dial() function.
*  Fixed some uninitialized string variables on FXS ports.

configs/chan_dahdi.conf.sample
*  Updated some documentation.

These changes are already in trunk -r172400


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02 20:28:54 +00:00
Terry Wilson
2b340eab54 Rename new parkedcallparking option to parkedcallreparking
Since this option actually already existed in 1.6.0+, use the same name so as
not to confuse people when they upgrade


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-31 00:15:09 +00:00
Terry Wilson
4e069885ce Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback.  Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.

This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it.  This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel.  The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.

2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.

3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone

4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches: 
      fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy

Review http://reviewboard.digium.com/r/138/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-30 17:47:41 +00:00
Tilghman Lesher
c257ffeed0 Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if
Asterisk runs as a non-root user and the administrator does a 'restart now',
Asterisk loses the ability to set QOS on packets.
(closes issue #14004)
 Reported by: nemo
 Patches: 
       20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29 22:54:29 +00:00