Displaying 9 results from an estimated 9 matches for "start_sound".
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start_send
2006 May 03
1
my asterisk crashed
...l = 0x0
privdb_val = 0
calldurationlimit = 0
config = {features_caller = {flags = 0}, features_callee = {flags =
0}, start_time = {tv_sec = 0, tv_usec = 0},
feature_timer = 0, timelimit = 0, play_warning = 0, warning_freq = 0,
warning_sound = 0x0, end_sound = 0x0,
start_sound = 0x0, firstpass = 0, flags = 0}
timelimit = 0
play_warning = 0
warning_freq = 0
warning_sound = 0x0
end_sound = 0x0
start_sound = 0x0
dtmfcalled = 0x0
dtmfcalling = 0x0
var = 0x0
---Type <return> to continue, or q <re...
2004 May 20
0
Time Limit Warning File
...as the callee. The audio is passing through Asterisk:
-- Executing Dial("SIP/8992-9712", "SIP/8988|20|L(10000:2000)") in new
stack
-- Limit Data:
-- timelimit=10000
-- play_warning=2000
-- play_to_caller=yes
-- play_to_callee=no
-- warning_freq=0
-- start_sound=UNDEF
-- warning_sound=timeleft
-- end_sound=UNDEF
-- Called 8988
-- SIP/8988-6922 is ringing
-- SIP/8988-6922 answered SIP/8992-9712
== Spawn extension (local, 8988, 1) exited non-zero on 'SIP/8992-9712'
If I change the LIMIT_WARNING_FILE to something like 'beep...
2007 Feb 24
0
Call was hangup when LIMIT_WARNING_FILE was playing
...Dial("IAX2/24012100-2",
"zap/g1/0028621XXXXXXXX|60|L(65000:60000:30000)") in new stack
-- Limit Data for this call:
-- - timelimit = 65000
-- - play_warning = 60000
-- - play_to_caller= yes
-- - play_to_callee= no
-- - warning_freq = 30000
-- - start_sound = UNDEF
-- - warning_sound = beep
-- - end_sound = beep
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/0028621XXXXXXXX
-- Zap/29-1 is proceeding passing it to IAX2/24012100-2
-- Zap/29-1 is ringing
-- Zap/29-1 answered IAX2/24012100-2
-- Hungup ...
2007 Jun 20
1
Asterisk RealTime
...call: -
-- AGI Script Executing Application: (DIAL) Options:
(SIP/2486543210|60|HL(3600000:61000:30000))
-- Limit Data for this call:
-- - timelimit = 3600000
-- - play_warning = 61000
-- - play_to_caller= yes
-- - play_to_callee= no
-- - warning_freq = 30000
-- - start_sound = UNDEF
-- - warning_sound = timeleft
-- - end_sound = UNDEF
Jun 20 09:49:58 NOTICE[24952]: app_dial.c:1069 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
a2billing.php|1|did:...
2009 Jun 10
0
Dial option limit call duration
...41] >
play_to_caller = yes
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] >
play_to_callee = yes
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] >
warning_freq = 0
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] >
start_sound =
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] >
warning_sound = timeleft
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] >
end_sound =
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] == Using
SIP RTP CoS mark 5
but the wa...
2011 Jun 09
0
Change to pickups in Asterisk 1.8 - not working on local channels?
...ed^0^1306286740.11)orL(3600000:60000))
> Limit Data for this call:
> timelimit = 3600000 ms (3600.000 s)
> play_warning = 60000 ms (60.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 0 ms (0.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
-- Called 1000103 at product-pickup/n
-- Executing [1000103 at product-pickup:1]
Pickup("Local/1000103 at product-pickup-db70;2", "1000103 at product-phone") in
new stack
[May 25 11:25:40] NOTICE[102...
2005 Jul 10
0
Time out not working from php agi...
...6:13 VERBOSE[19094] logger.c: -- play_warning=61000
2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- play_to_caller=yes
2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- play_to_callee=no
2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- warning_freq=30000
2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- start_sound=UNDEF
2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- warning_sound=timeleft
2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- end_sound=UNDEF
2005-06-28 20:26:13 DEBUG[19094] chan_h323.c: type=H323, format=4,
data=880178034593@xx.xx.xx.xx:1720.
2005-06-28 20:26:13 DEBUG[19094] chan_h323.c: Extension...
2005 Jan 08
3
ASTCC questions
Hello.
I have set up ASTCC properly, calling it like this:
DeadAGI(${ACCOUNTCODE},${EXTEN})
It seems to be working correctly, but I have two questions:
- Although the cards' credit seems to be maintained correctly, I cannot
see the call details in astcc-admin. When I try to view information on
the card, it's just blank. Any ideas?
- When does the 2nd, 3rd and 4th trunk get used? I have
2007 Oct 31
1
segfault - asterisk crash and restart
...00\000\000??\004?7\000\000\0000\000\000\0000\000\000\000?\tzA\000\000"...
cidname = '\0' <repeats 79 times>
privdb_val = 0
calldurationlimit = 0
timelimit = 1800000
play_warning = 120000
warning_freq = 0
warning_sound = 0x2aaabf53cd0a "timeleft"
end_sound = 0x0
start_sound = 0x0
dtmfcalled = 0x0
dtmfcalling = 0x0
status = "BUSY\000WER\000GS", '\0' <repeats 244 times>
play_to_caller = 1
play_to_callee = 0
sentringing = 0
moh = 0
outbound_group = 0x0
result = 0
start_time = 1193798657
privintro = "\001\000\000\000\000\000\000\000...