search for: nocdr

Hi, When I use NoCDR application I obtain this warning in console log: Mar 1 11:16:08 WARNING[3513]: cdr.c:114 ast_cdr_free: CDR on channel 'SIP/492-7371' not posted Mar 1 11:16:08 WARNING[3513]: cdr.c:116 ast_cdr_free: CDR on channel 'SIP/492-7371' lacks end Can someone explain to me what is due?...

I am running 1.4.10.1. I have a macro that is called from default for a certain extension (both below). I added NoCDR to s to try and stop extra CDR records, but I am still getting them. Any idea how to stop them? extensions.conf: [macro-STDEXT] exten =s,1,NoCDR() exten =s,2,Dial(${ARG1},30,Tt) exten =s,3,Goto(s-${DIALSTATUS},1) exten =s-NOANSWER,1,Voicemail(${ARG2}|u) exten =s-NOANSWER,n,Hangup exten =s-BUSY,1...

Hi All, I'm using asterisk 1.2.15 and call forwarding doesnt work for me. from my extensions.conf: ; Unconditional Call Forward exten => _*21*X.,1,NoCDR exten => _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) exten => _*21*X.,3,Playback(vm-saved) exten => _*21*X.,4,Hangup exten => #21#,1,NoCDR exten => #21#,2,DBdel(CFIM/${CALLERID(NUM)}) exten => #21#,3,Playback(auth-thankyou) exten => #21#,4,Hangup debug from as...

...0-07-10 14:18:29","2020-07-10 14:18:31","2",""+blabla" <CID>","NO ANSWER","Callee DID","","DOCUMENTATION" 1. How it's possible tha CDR billsec appears in NO ANSWER for an answered call ? 2. How to set NOCDR for the phones who didn't answer the call ? In the h extension we set NOCDR for all DIALSTATUS except ANSWER but we still get those CDRs Is it possible to start CDR when a call is bridged ? Other solutions ? Thanks for any hint -- Daniel

...sterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1 was answered. > Launching NoCDR() on Local/s at tc-maint-000002a4;1 [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread: Call completed to Local/s at tc-maint [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread: Call completed to Local/s at tc-maint == Spawn extension (tc-maint, s, 5) exited non-...

What would it do if you exten => h,1,ResetCDR(w) exten => h,2,NoCDR() exten => h,3,DEADAGI(get-unqiueid.php) I have not tried it but in theory it should write the first CDR and then kill the write of the second NO ANSWER CDR. Let me know if it works for you as I may need to do it on some of my h exten code as well. Bryant ----------------------------------...

Hi everyone: The following code dials our prefix, sends a beep, and sends a DTMF "c" tone, then dials the phone number. I need to send the DTMF only if the phone is answered. [voip] exten=>i,1,NoCDR() exten=>i,2,Hangup() exten=>s,1,Wait(2) exten=>s,2,Background(beep||) exten=>s,3,DigitTimeout(6) exten=>s,4,ResponseTimeout(10) exten=>s,5,SendDTMF(c) exten=>t,1,NoCDR() exten=>t,2,Hangup() exten=>_009[13456789].,1,Dial(SIP/operador/${EXTEN},60,tr) exten=>_009[2].,1,D...

Hi, okay heres what I want to do .. simple ivr, we take a call, answer it, play a menu, dial out based on options. No problems so far. The CDR always shows the call as answered as I answer the 1st leg to play the prompts, I am actually more interested in if the 2nd leg - the outbound part - has been answered or not before the call is hungup. How can I get this and record the information in

...l application that users can listen to messages and leave messages. I am looking for a way to play a beep tone to a user when a new message is received when they are on the phone. Here is what I have come up with: in extensions.conf: [beepvoicemail] exten => 1000,1,answer() exten => 1000,2,NoCDR() exten => 1000,3,wait(2) exten => 1000,4,Set(TIMEOUT(absolute)=5) exten => 1000,5,playback(voicemail/beeps) exten => 1000,7,SendDTMF(9) exten => 1000,8,hangup() exten => 2000,1,Set(TIMEOUT(absolute)=5) exten => 2000,2,NoCDR() exten => 2000,3,extenspy(,g(${mailbox})WqX) ext...

Hello ; how can i enable billing only while using specific trunk (ex:zap) but internal sip calls will not be counted specifically how to make all outbound is counted i am using asterisk mysql cdr enabled -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050626/0faf0974/attachment.htm

Hello I use asterisk ver 0.7.2 Can I play any wave file into the client riciever without billing count ? I call from A IAX client to B IAX client. B client is not available and I would like to play some file with the message user_is_unavailable.gsm But when I look into my CDR table, this call is billed. I don't want to bill these messages. Is it possible ? thank you -- Vit Bohacek

...digs - ${EXTEN:1}) exten => _0NXXXXXX,2,Set(ZAP01=$[${ZAP01} + 1]|g) exten => _0NXXXXXX,3,Set(UPDATED=true) exten => _0NXXXXXX,4,Dial(${TRUNK}/${EXTEN},60) exten => _0NXXXXXX,6,Busy exten => _0NXXXXXX,7,Playback(thank-you) include => hangupcontext [hangupcontext] exten => h,1,NoCDR() exten => h,2,GotoIf($["${UPDATED}" != "true"]?5) exten => h,3,Set(UPDATED="") exten => h,4,Set(ZAP01=$[${ZAP01} - 1]|g) exten => h,5,Hangup I am not sure about how to use NoCDR, ForkCDR and ResetCDR. If don't use any of them, every call generates an...

Hi guys I've just read this about the upcoming release of * 1.6: ?Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing.? That

...gger.c: -- Starting simple switch on 'Zap/3-1' [Mar 30 10:51:30] VERBOSE[1944] logger.c: -- Executing [18585300400 at from-internal:1] ResetCDR("Zap/3-1", "") in new stack [Mar 30 10:51:30] VERBOSE[1944] logger.c: -- Executing [18585300400 at from-internal:2] NoCDR("Zap/3-1", "") in new stack [Mar 30 10:51:30] VERBOSE[1944] logger.c: -- Executing [18585300400 at from-internal:3] Wait("Zap/3-1", "1") in new stack [Mar 30 10:51:31] VERBOSE[1944] logger.c: -- Executing [18585300400 at from-internal:4] Playback(&quo...

...before GotoIf. Here are the test context and macro: [test] exten => _NXXNXXXXXX,1,Dial(IAX2/XXXXX/${EXTEN},30) exten => h,1,Macro(test-hangup) [macro-test-hangup] exten => s,1,GotoIf($["1"="1"]?2:3) exten => s,n,NoOp(A) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() exten => s,n,Hangup() -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090715/5fd42eda/attachment-0001.htm

...dy, Happy New Year ! I'm trying to detect if a call was answered by a machine (linke voicemail systems) or a human. I would like to use AMD (Answering Machine Detect) command, but with my configuration it was not possible get there. Follow my dialplan: exten => _[789].,1,NoCDR exten => _[789].,n,Dial(SIP/${EXTEN}@111,60) exten => _[789].,n,AMD exten => _[789].,n,NoOp(AMD Status is: ${AMDSTATUS}) exten => _[789].,n,Hangup What is happening is when the call is answered by the other part, Asterisk doesn't go to the next level (exten =...

...Playback("Zap/7-1", "vm-goodbye") in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Macro("Zap/7-1", "hangupcall") in new stack -- Executing ResetCDR("Zap/7-1", "w") in new stack -- Executing NoCDR("Zap/7-1", "") in new stack -- Executing Wait("Zap/7-1", "5") in new stack -- Executing Playback("Zap/14-1", "vm-goodbye") in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Hangup("Zap/7...

...= 3500 greeting = 1500 after_greeting_silence = 300 total_analysis_time = 5000 min_word_length = 120 between_words_silence = 50 maximum_number_of_words = 5 silence_threshold = 256 In extensions.conf [outboundmsg1] exten => s,1,NoCDR exten => s,n,AMD exten => s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach) exten => s,n(mach),WaitForSilence(2500) exten => s,n,Playback(outboundmsgs/msg1) exten => s,n,Hangup exten => s,n(humn),WaitForSilence(500) exten => s,n,Playback(outboundmsgs/msg1) exten => s,n,Hangup...

...Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve these values in my MySQL CDR table in order to calculate a MOS value: "ssrc=592614191;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=20734;rlp=0;rtt=0.094000" codec used: g711a -- -- -- Marc LEURENT lftsy at leurent.eu

...angs up first. [dialout] exten => _NXXNXXXXXX,s,1,Dial(SIP/XXXXXX/${EXTEN},30) exten => h,1,Macro(hangupcall,${EXTEN},${CDR(accountcode)}) [macro-hangupcall] NoOp(${CDR(dst)}) Set(dialout_num=${ARG1}) Set(user_id=${ARG2}) ResetCDR(vw); NoCDR(); Hangup(); -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090713/0dfcfea4/attachment.htm