During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
Review: http://reviewboard.digium.com/r/184/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-15 are not defined, these bits are never turned off. In trunk, bits 13-15 are defined, which means 1.4 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities.
(closes issue #14283)
Reported by: jcovert
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177696 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
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
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
whether autoconf detected a static initializer value. This fixes rwlocks
on all such platforms (mainly, Mac OS X).
(closes issue #13767)
Reported by: jcovert
Patches:
20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
Tested by: jcovert, Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@166093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: tzafrir
Replace a bunch of if defined checks for Zaptel/DAHDI through several new defines in dahdi_compat.h. This removes a lot of code duplication. Example from bug:
#ifdef HAVE_ZAPTEL
fd = open("/dev/zap/pseudo", O_RDWR);
#else
fd = open("/dev/dahdi/pseudo", O_RDWR);
#endif
is replaced with:
fd = open(DAHDI_FILE_PSEUDO, O_RDRW);
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@165991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
One issue was that the ast_mutex_* API was being used within the context of the
thread local data destructors. We would go off and allocate more thread local data
while the pthread lib was in the middle of destroying it all. This led to a memory
leak.
Another issue was an invalid argument being provided to the the object_add
API call.
(closes issue #13678)
Reported by: ys
Tested by: russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@164736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These changes come from team/russell/issue_12658
1) Change autoservice to put digits on the head of the channel's frame readq
instead of the tail. If there were frames on the readq that autoservice
had not yet read, the previous code would have resulted in out of order
processing. This required a new API call to queue a frame to the head
of the queue instead of the tail.
2) Change up the processing of DTMF in ast_read(). Some of the problems
were the result of having two sources of pending DTMF frames. There
was the dtmfq and the more generic readq. Both were used for pending
DTMF in various scenarios. Simplifying things to only use the frame
readq avoids some of the problems.
3) Fix a bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation,
and a digit arrived before emulation was complete, digits would get
processed out of order.
(closes issue #12658)
Reported by: dimas
Tested by: russell, file
Review: http://reviewboard.digium.com/r/85/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@163448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
parameters are not evaluated multiple times.
An example where this caused a problem was in chan_sip.c, with
the line
ast_string_field_set(p, fromdomain, ++fromdomain);
This patch was originally uploaded to issue #13783 by
jamessan. While the issue was closed for other reasons, this
patch is valid and fixes a separate problem, and is thus
being committed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The test is not valid. Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ckjohnsonme
Patches:
14019.diff uploaded by murf (license 17)
Tested by: ckjohnsonme, murf
This crash was the result of a few small errors that
would combine in 64-bit land to result in a crash.
32-bit land might have seen these combine to mysteriously
drop the args to an application call, in certain
circumstances.
Also, in trying to find this bug, I spotted
a situation in the flex input, where, in passing
back a 'word' to the parser, it would allocate
a buffer larger than necessary. I changed the
usage in such situations, so that strdup was
not used, but rather, an ast_malloc, followed
by ast_copy_string.
I removed a field from the pval struct, in
u2, that was never getting used, and set in
one spot in the code. I believe it was an
artifact of a previous fix to make switch
cases work invisibly with extens.
And, for goto's I removed a '!' from
before a strcmp, that has been there
since the initial merging of AEL2, that
might prevent the proper target of a
goto from being found. This was pretty
harmless on its own, as it would just
louse up a consistency check for users.
Many thanks to ckjohnsonme for providing
a simplified and complete set of information
about the bug, that helped considerably in
finding and fixing the problem.
Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state,
so I can run the regression suite!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: tzafrir
When compiling against Zaptel dahdi_compat will now only define all the DAHDI defines if the Zaptel define is present. Also, there is no such thing as DAHDI_PRI.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
frame in a freed ast_filestream. This patch makes use of the
ao2 functions to make sure that we do not free an ast_filestream
structure until the embedded ast_frame has been "freed" as well.
(closes issue #13496)
Reported by: fst-onge
Patches:
filestream_frame_1_4.diff uploaded by putnopvut (license 60)
Tested by: putnopvut
Closes AST-89
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.
This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@157305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.
Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@149204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose
- astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled
- simplify the usage of some of these headers in the AEL-related stuff in the utils directory
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from
the ground up!
This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.
Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.
While I dearly hope that this patch overcomes all problems, and
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@142675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: korihor
Wow, this one was a challenge!
I regrouped and ran a new strategy for
setting the ~~MACRO~~ value; I set it once
per extension, up near the top. It is only
set if there is a switch in the extension.
So, I had to put in a chunk of code to detect
a switch in the pval tree.
I moved the code to insert the set of ~~exten~~
up to the beginning of the gen_prios routine,
instead of down in the switch code.
I learned that I have to push the detection
of the switches down into the code, so everywhere
I create a new exten in gen_prios, I make sure
to pass onto it the values of the mother_exten
first, and the exten next.
I had to add a couple fields to the exten
struct to accomplish this, in the ael_structs.h
file. The checked field makes it so we don't
repeat the switch search if it's been done.
I also updated the regressions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@136726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
