mirror of https://github.com/asterisk/asterisk
master
20
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${ noResults }
332 Commits (cd054ab3b4e1d58d69fa06f0ba730f26f80a905f)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
fa79e11a91 |
Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial
(closes issue #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275027 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
1d9be78f12 |
Add documentation clarifying when 't' and 'T' can be used.
(closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@255503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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43393bd1a9 |
Resolve a number of FreeBSD build issues.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@253631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c79e242ada |
fixes solaris segfault on dial with verbosity >= 3
(closes issue #16193) Reported by: asgaroth Patches: bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@231235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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3c256882d6 |
This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements.
(closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@227827 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ed413ec76c |
Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up
while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@226889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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926a033bf9 |
Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it.
(closes issue #14763) Reported by: cupotka git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@224565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e3464ac40a |
Ensure ringing continues for branched calls after progress is received
While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a9317f6cbe |
Fix potential memory leak in app_dial.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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63cc189747 |
AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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55d9c2ecaf |
Do not log an ERROR if autoservice_stop() returns -1.
This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208592 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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48253ef901 |
Treat an empty FORWARD_CONTEXT the same way we treat a missing one.
(closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198251 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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59ee389e31 |
Update CDR appropriately when AST_CAUSE_NO_ANSWER is set
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189465 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ef9ef40c19 |
Don't treat a NOANSWER like a CHANUNAVAIL
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8e31a331b4 |
Fix a crash due to too few arguments to RetryDial.
(closes issue #14852) Reported by: junky Patches: retry_fix.diff uploaded by junky (license 177) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187135 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2da89b3022 |
Add missing datastore inherit (exists in all other branches)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183481 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f42e9eb6bf |
Cleaning up a few things in detect disconnect patch
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 |
17 years ago |
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dd17912d68 |
Allow disconnect feature before a call is bridged
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 |
17 years ago |
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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 |
17 years ago |
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8a1faf0a58 |
When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself.
(closes issue #14310) Reported by: RadicAlish git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170568 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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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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17 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 |
17 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 |
17 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 |
17 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 |
17 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 |
17 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 |
17 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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17 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 |
17 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 |
17 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 |
17 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 |
17 years ago |
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eccd14d7f0 |
Tested by: sergee, murf, chris-mac, andrew, KNK
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 |
18 years ago |
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ee8adb313e |
(closes issue #12982)
Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@135799 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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cb503c4cb4 |
Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels involved. This fixed a one-way audio situation I had in my test setup. I couldn't find any open issues that suggested one-way audio with regards to mixmonitor (or other audiohook) usage, though. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@130792 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e46bb5f5bc |
Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered. (closes issue #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14) Tested by: ibc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@129149 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d432d18723 |
Remove extra option from previous solution attempt
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@122617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e128c2a314 |
This should fix the behavior of the 'T' dial feature being passed incorrectly to the transferee when builtin_atxfers are used.
Also, doing a builtin_atxfer to parking was broken and is fixed here as well. (closes issue #11898) Reported by: sergee Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@122589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c340004d76 |
Backport fix for 11520--for some reason I didn't do this back in February when I patched for trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@121992 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c6c26f7c34 |
Fix another typo in documentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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383a1ece6b |
small typo fix 'retires' => 'retries'
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119478 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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f1683753b8 |
If the datastore has been moved to another channel due to a masquerade, then
freeing the datastore here causes an eventual double free when the new channel hangs up. We should only free the datastore if we were able to successfully remove it from the channel we are referencing (i.e. the datastore was not moved). (closes issue #12359) Reported by: pguido git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2bf8f9fca3 |
Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial.
(closes issue #11516) Reported by: ys Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: anest, jcapp, dartvader git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107016 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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cd703523db |
Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148) Reported by: jcomellas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3d434f14c3 |
Add dependency on chan_local to app_dial.
Dial still runs without chan_local, but will be missing forwarding functionality. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@99592 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3e913c706f |
* Add channel locking around datastore operations that expect the channel
to be locked. * Document why we don't record Local channels in the dialed interfaces list. * Remove the dialed variable as it isn't needed. * Restructure some code for clarity and coding guidelines stuff git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91783 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c13539cb13 |
Don't unlock the dialed_interfaces list until we're done messing with the iterator.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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85e0e1919d |
Allow dialing local channels from Queue() and Dial() again. There was a slight
flaw in the code to prevent call forwards from looping that caused this problem. (related to issue #11486) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a1a592f3f0 |
The 'G' option for Dial() did not properly handle the case where only a label was
provided. This was due to the fact that the answering channel did not have an extension set, so ast_parseable_goto would fail. This fix eliminates the call to ast_parseable_goto on the answering channel since it is a wasteful call. The answering channel and the calling channel are both directed to the same extension and context, just different priorities, so we can just copy the values from the calling channel to the answering channel and increment the answering channel's priority. (closes issue #11382, reported by jon, patch by me with correction by jon) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91273 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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335d271dda |
Fix build issue on the build cluster.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90798 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |