The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and res_feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The response message (either Error or Success) needs an extra trailing \r\n
after the fields to inform the client that the message is complete.
(closes issue #14876)
Reported by: srt
Patches:
05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71)
asterisk_14876.patch uploaded by srt (license 378)
trunk-14876-2.diff uploaded by phsultan (license 73)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
(closes issue #12946)
Reported by: meral
Patches:
null-cdr2.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, dbrooks
(closes issue #15122)
Reported by: sum
Tested by: sum
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The frames here should have always been freed. However, out of luck, there was
never any memory leaked. However, after file streams became reference counted,
this code would leak the file stream for the file being read.
(closes issue #15181)
Reported by: jkroon
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@196826 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected.
This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times.
(closes issue #13797)
Reported by: sh0t
Tested by: sh0t
(closes issue #14744)
Reported by: deepesh
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Just changed park_exec to always return non-zero. I really wasn't entirely sure
at first if this was a bug. Decided it was since it would be surprising when
not using ParkedCall in the dialplan to hang up and have dialplan execution
continue.
(closes issue #14555)
Reported by: francesco_r
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CDR code involved with bridges wrongly assumed that the currently executing application and data
values will always exist. It is possible for this to be false when call forwarding is involved.
(closes issue #14984)
Reported by: gincantalupo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When chan_dahdi goes to get an SMDI message, it provides no search criteria.
It just grabs the next message that arrives. This code was written with the
SMDI dialplan functions in mind, since that is now the preferred method of
using SMDI. However, this broke support of it being used from chan_dahdi.
(closes AST-212)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@190661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We were unconditionally incrementing the number of mohclasses
registered. However, we should actually only increment if the
call to moh_register was successful.
While this probably has never caused problems, I noticed it
and decided to fix it anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The only way that this leak would occur is if Monitor were started
using the Manager interface and no File: header were given. Discovered
while reviewing the ast_channel_ao2 review request.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect.
issue #11583
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With these changes, for a module to export symbols into the global namespace, it must have *both* the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With these changes, for a module to export symbols into the global namespace, it must have *both* the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dynamic features defined in the applicationmap section of features.conf allow
one to specify whether the caller, callee, or both have the ability to use the
feature. The documentation in the features.conf.sample file could be interpreted
to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the
calling channel in order to allow for the callee to be able to use the features
which he should have permission to use. However, the DYNAMIC_FEATURES variable
would only be read from the channel of the participant that pressed the DTMF
sequence to activate the feature. The result of this was that the callee was
unable to use dynamic features unless the dialplan writer had taken measures
to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel.
This commit changes the behavior of ast_feature_interpret to concatenate the
values of DYNAMIC_FEATURES from both parties involved in the bridge. The features
themselves determine who has permission to use them, so there is no reason to believe
that one side of the bridge could gain the ability to perform an action that they
should not have the ability to perform.
Kevin Fleming pointed out on the asterisk-users list that the typical way that this
was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel
so that the value would be inherited by the called channel. While this works, the
documentation alone is not enough to figure out why this is necessary for the callee
to be able to use dynamic features. In this particular case, changing the code to match
the documentation is safe, easy, and will generally make things easier for people for
future installations.
This bug was originally reported on the asterisk-users list by David Ruggles.
(closes issue #14657)
Reported by: mmichelson
Patches:
14657.patch uploaded by mmichelson (license 60)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For custom commands we don't know whether the audio is coming from a stream or not
so we are going to have to read the data despite no channels.
(closes issue #14416)
Reported by: caspy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a music class is setup to actually provide music on hold
from a stream we need to constantly read audio from it since it
will constantly be providing audio. This is now done despite there
being no channels listening to it.
(closes issue #14416)
Reported by: caspy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue. Because using the ast prefix calls are
a better choice, ast_free_ptr is the new wrapper for free to pass to functions.
Also, a little bit of clean up was done to avoid the debug macros intentionally
being redefined.
(closes issue #13593)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves
the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to.
We can not safely modify it afterwards because of this, so don't even try.
(closes issue #14564)
Reported by: meric
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As per bug 14515, a dev discussion arrived at a "mediated concensus"
of a default feature digit timeout of 1.0 sec. Some voted for 1300;
ctooley thought 1500 for distracted phone users in phone booths;
kpfleming put his foot down at 1.0 sec.
Users who found the previous default max delay of 250 msec perfect,
are welcome to override the new default. Notice that I said that
250 msec was the default; wait a minute, you might say, the config
file said it was 500 msec!; well, because of the bug fix for 14515,
we found that 500 msec was actually enforcing a max of 250. The bug
fix would restore 500 msec, but we felt even that was a bit tight
for most users... 2000 msec was pushed earlier by mmichelson, so
that reduces to 1000 msec after the bug fix. Enjoy!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Because the ast_channel_bridge() call will return 0 and pass
a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer
field in hte config struct is getting decremented twice, which
effectively cuts the digittimeout in half. I added conditions
to the if statement to only let DTMF_END frames to flow thru,
which solved the problem. Also, when the frame pointer is null,
let control flow thru-- this usually happens on timeouts. I added
a comment to the code to explain what's going on and why.
Many thanks to sodom for reporting this problem. Personnally, it always seemed
like something was wrong with the featuredigittimeout, but I never
could quite decide what... and was too busy to investigate.
This bug forced the issue, and now we know.
Sodom had other issues in 14515, but I couldn't reproduce them. If
he still has problems, and wants to get them solved, he is welcome
to reopen 14515.
(closes issue #14515)
Reported by: sodom
Patches:
14515.patch uploaded by murf (license 17)
Tested by: murf, sodom
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The main problem is currently if the Dial flag L is used with a warning sound,
DTMF is not evaluated after the first warning sound. To fix this, a flag has
been added in ast_generic_bridge for playing the warning which ensures that if
a scheduled warning is missed, multiple warrnings are not played back (due to a
feature evaluation or waiting for digits). ast_channel_bridge was modified to
store the nexteventts in the ast_bridge_config structure as that information
was lost every time ast_channel_bridge was reentered, causing a hangup due to
incorrect time calculations.
(closes issue #14315)
Reported by: tim_ringenbach
Reviewed on reviewboard:
http://reviewboard.digium.com/r/163/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
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
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
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
situations.
What it does:
1. A new Flag value is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop
level instead.
2. In the manager Redirect code, I set this flag on the channel
if the channel has a non-null pbx pointer. I did the same for the
second (chan2) channel, which gets run if name2 is set...
and the first succeeds.
3. I restored the ending of the cdr for the pbx loop h-exten
running code. Don't know why it was removed in the first place.
4. The first attempt at the fix for this bug was to place code
directly in the async_goto routine, which was called from a
large number of places, and could affect a large number of
cases, so I tested that fix against a fair number of transfer
scenarios, both with and without the patch. In the process,
I saw that putting the fix in async_goto seemed not to affect
any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to
its current scope, and jmls tested both. In the process, tho,
I saw that blind xfers in one situation, when the one-touch
blind-xfer feature is used by the peer, we got strange
h-exten behavior. So, I inserted code to swap CDRs and
to set the HANGUP_DONT field, to get uniform behavior.
5. I added code to the bridge to obey the HANGUP_DONT flag,
skipping both publishing the bridge CDR, and running
the h-exten; they will be done at the pbx-loop (higher)
level instead.
6. I removed all the debug logs from the patch before committing.
7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
so it's only done if the h-exten is going to be run. A very
minor performance improvement, but technically correct.
(closes issue #14241)
Reported by: jmls
Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
Tested by: murf, jmls
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Blindtransfer also was calling back the wrong channel when a timeout occurred.
This should be the last of these, as there are no more masq_park_announce calls
that aren't passing orig_chan_name
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded
parking, otherwise we will try to call ast_hangup() in __pbx_run() and in
do_parking_thread() and then promptly crash.
(closes issue #14215)
Reported by: waverly360
Tested by: otherwiseguy
(closes issue #14228)
Reported by: kobaz
Tested by: otherwiseguy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
