mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
2637 Commits (f311539733fed595cfc2fd2fdfead06fcb6e69b7)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
f311539733 |
Truncate userevents at the end of a line, when the command exceeds the buffer.
(closes issue #14278) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
d671bb1404 |
Fix the conjugation of Russian and Ukrainian languages.
(related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168828 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
26658ead16 |
Fix some crashes from bad datastore handling in app_queue.c
* The queue_transfer_fixup function was searching for and removing the datastore from the incorrect channel, so this was fixed. * Most datastore operations regarding the queue_transfer datastore were being done without the channel locked, so proper channel locking was added, too. (closes issue #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by putnopvut (license 60) Tested by: ZX81, festr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168628 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
7c1d2de899 |
app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168608 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
b3ea953313 |
Don't overflow when paging more than 128 extensions
The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbitrary limit. This issue would have special importance to anyone who is dynamically creating the argument passed to the Page application and allowing more than 128 extensions to be added by an outside user via some external interface. The patch posted by a_villacis was slightly modified for some coding guidelines and other cleanups. Thanks, a_villacis! (closes issue #14217) Reported by: a_villacis Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660) Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
9161b7fc87 |
Revert unnecessary indications API change from rev 122314
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168561 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
e3700a13a4 |
This merges the masqpark branch into 1.4
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
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18 years ago |
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e0bec5d67d |
(closes issue #13480)
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
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18 years ago |
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5de6848d0c |
Ensure that the chanspy datastore is fully initialized.
This patch resolved some random crash issues observed by a user on a BSD system (closes issue #14111) Reported by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@165889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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139be9556b |
Add mutexes around accesses to the IMAP library interface. This prevents
certain crashes, especially when shared mailboxes are used. (closes issue #13653) Reported by: howardwilkinson Patches: asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590) Tested by: jpeeler git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@165767 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3a0e7aec6a |
Do not crash if we are not passed in a followme id.
(closes issue #14106) Reported by: ys Patches: app_followme.c.2.diff uploaded by ys (license 281) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@165537 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8468d435e8 |
Reverse the fix from issue #6176 and add proper handling for that issue.
(Closes issue #13962, closes issue #13363) Fixed by myself (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@165317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a391fd9b1c |
Fix some memory leaks found while looking at how realtime
configs are handled. Also cleaned up some coding guidelines violations in app_realtime.c, mostly related to spacing git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@165255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c703960acc |
Do not dereference the channel if AST_PBX_KEEPALIVE has been returned.
This is a bug I noticed while looking at the code for app_macro. This return code means that another thread has assumed ownership of the channel and it can no longer be touched. (I hate this return code with a passion, by the way.) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@164876 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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dfa5f7c4b4 |
Can we try not to assign an unsigned int to -1?
(closes issue #14074) Reported by: wetwired git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@164204 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d8ff8d169d |
Change the default calldurationlimit from the special value 0 to -1, so we
can better detect an exceptional case. This follows on to the changes made in revision 156386. Related to issue #13851. (closes issue #13974) Reported by: paradise Patches: 20081208__bug13974.diff.txt uploaded by Corydon76 (license 14) Tested by: file, blitzrage, ZX81 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@164082 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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60c9cae6d6 |
Revert this cast to long. Using time_t here causes build failures on a
FreeBSD 32-bit build. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@163084 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ef56f0dbd4 |
Fix a potential crash due to unsafe datastore handling.
This patch also contains a conversion from using long to time_t for representing times for a queue, as well as some whitespace fixes. (closes issue #14060) Reported by: nivek Patches: datastore_fixup.patch.corrected uploaded by nivek (license 636) with slight modification from me Tested by: nivek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@163080 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8a7982c1ff |
Oops, should be "tz", not "zonetag".
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5134bb74a9 |
We appear to have documented tz= in the [general] section of voicemail.conf,
without actually having implemented it. Oops. (Reported by Olivier on the -users list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162348 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a7520ee4bf |
Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing.
(closes issue #14005) Reported by: ddl git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162341 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5d985ce022 |
Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.
We need to make sure that we don't start writing audio to the trunk channel until we're actually ready to answer it. Otherwise, the channel driver will treat it as inband progress, even though all they are getting is silence. (closes issue #12471) Reported by: mthomasslo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7eeb7f7493 |
Fix double declaration of 'x' on the PPC platform.
(closes issue #14038) Reported by: ffloimair git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162273 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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97bb6d4c65 |
Allow DISA to handle extensions that start with #.
(closes issue #13330) Reported by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162014 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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16f94acd53 |
Some compilers warn on null format strings; some don't (caught by buildbot)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1653a9ef65 |
Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc
and glibc. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160207 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e8f9274ca7 |
Add missing variable declaration in the PPC code
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159096 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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929c14e432 |
System call ioperm is non-portable, so check for its existence in autoconf.
(Closes issue #13863) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159025 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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72a8661580 |
This change had somehow gotten reverted due to a
completely unrelated commit. Thanks to Theo Belder on the Asterisk-dev list for pointing this out. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3a1a981e2e |
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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161a05a3d1 |
(closes issue #13899)
Reported by: akkornel This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@157365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3429c9de5e |
Fix a crash in the end_bridge_callback of app_dial and
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 |
18 years ago |
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b753654cbd |
If the prompt to reenter a voicemail password timed out, it
resulted in the password not being saved, even if the input matched what you gave when first prompted to enter a new password. This is because the return value of ast_readstring was checked, but not checked properly. This bug was discovered by Jared Smith during an Asterisk training course. Thanks for reporting it! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156816 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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bc208d63c2 |
ast_waitfordigit() requires that the channel be up, for no good logical
reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156755 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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382459fac8 |
When using call limits under 1 second, infinite call lengths are allowed,
instead. (closes issue #13851) Reported by: ruddy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156386 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7e353eeaa1 |
If the SLA thread is not started, then reload causes a memory leak.
(closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156294 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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738ec1fb71 |
For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c9cec47bf0 |
(closes issue #13173)
Reported by: pep This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference. Reviewed by Russell on Review Board: http://reviewboard.digium.com/r/25/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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bd9001e16b |
When doing some tests, I was having a crash at the end of every call
if an attended transfer occurred during the call. I traced the cause to the CDR on one of the channels being NULL. murf suggested a check in the end bridge callback to be sure the CDR is non-NULL before proceeding, so that's what I'm adding. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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f2ecc4c80e |
Use static functions here instead of nested ones. This requires a small
change to the ast_bridge_config struct as well. To understand the reason
for this change, see the following post:
http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@155553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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18 years ago |
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799fe53401 |
Attempting to expunge a mailbox when the mailstream is NULL will crash Asterisk.
(Closes issue #13829) Reported by: jaroth Patch by: me (modified jaroth's patch) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@154066 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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add5ff5b05 |
fix a bunch of potential problems found by gcc 4.3.x, primarily bare strings being passed to printf()-like functions and ignored results from read()/write() and friends
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@153337 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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705d6f3742 |
Add end_bridge_callback for app_follome and add AUTOLOOP flag to res_features
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@153270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6280e04736 |
Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call
(closes issue #13793) Reported by: greenfieldtech git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@153095 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fffb7722be |
A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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961bc7e758 |
The magic trick to avoid this crash is not to
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 |
18 years ago |
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fd9de775c5 |
Quoting in the wrong direction
(Fixes AST-107) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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785f27a29a |
Reset all DIAL variables back to blank, in case Dial is called multiple times
per call (which could otherwise lead to inconsistent status reports). (closes issue #13216) Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 (license 14) Tested by: ruddy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3c205d2d3f |
Update the queue with the correct number of calls and
whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@149200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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124e45e739 |
Check correct values in the return of ast_waitfor(); also, get rid of a
possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@149061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |