search for: end_sound

...eats 79 times> 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...

...-- 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' to use the usual beep.gsm file same results :(...

...XX|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 'Zap/29-1' == Spawn extension (default, +8621XXXXXX...

...P/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: file:Class.A2Billing.php - line:634 - [CARD STATUS UPDATE : U...

...1] > 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 warning is played to late it used to be exactly "you have 30 seconds left" but now its [Jun 10 16:15:20] VERBOSE[12201] logger.c: [Jun 10 16:15:20] -- <SIP/OB-0847a3c8>...

...his 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[1020]: app_directed_pickup.c:313 pickup_exec: No target channel foun...

...3 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: 880178034593 Host: xx.xx.xx.xx:1720 2005-06-28 20:26:13 DEBUG[19094] chan_h323.c: Could not find peer xx.xx.xx.xx by name or address...

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

...0\000\200a@\000\000\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\...