search for: play_to_caller

Displaying 9 results from an estimated 9 matches for "play_to_caller".

Did you mean: play_to_callee
2004 May 20
0
Time Limit Warning File
...e warning message. I?m using a Cisco 7960 as the caller and a Polycom 500 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...
2007 Feb 24
0
Call was hangup when LIMIT_WARNING_FILE was playing
...AX2/24012100-2", "LIMIT_TIMEOUT_FILE=beep") in new stack -- Executing 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...
2007 Jun 20
1
Asterisk RealTime
...813 OK (20 ms) 1 sip peers [1 online , 0 offline] Error message while receiving the 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 desti...
2009 Jun 10
0
Dial option limit call duration
...-- Limit Data for this call: [Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] > timelimit = 60000 [Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] > play_warning = 30000 [Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] > 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:...
2011 Jun 09
0
Change to pickups in Asterisk 1.8 - not working on local channels?
...-- AGI Script Executing Application: (Dial) Options: (Local/1000103 at product-pickup /n,60,M(product-answered^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 pr...
2005 Jul 10
0
Time out not working from php agi...
...23/880178034593@xx.xx.xx.xx:1720|40|HL(585000:61000:30000)) 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- Limit Data: 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- timelimit=585000 2005-06-28 20:26:13 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 VERBO...
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
2006 May 03
1
my asterisk crashed
...warning_freq = 0 warning_sound = 0x0 end_sound = 0x0 start_sound = 0x0 dtmfcalled = 0x0 dtmfcalling = 0x0 var = 0x0 ---Type <return> to continue, or q <return> to quit--- status = '\0' <repeats 255 times> play_to_caller = 0 play_to_callee = 0 sentringing = 0 moh = 0 outbound_group = 0x0 macro_result = 0x0 macro_transfer_dest = 0x0 digit = 0 result = 0 start_time = 0 answer_time = 0 end_time = 0 app = (struct ast_app *)...
2007 Oct 31
1
segfault - asterisk crash and restart
...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@\tzA\000\000\000\000?LR\000\000\000\000\000?LR\000\000\000\000\000\f\000\000\000\000\000\000\000?4\004?7\000\000\000s\000\000\000\001\000\...