similar to: NoCDR Warning

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 =>

I'm trying get this working. I've looked through the list, and can't see how to get AMD to print out more. I have it call and say Hello like I normally would. I've tried to say more and less doesn't seem to matter. After I hangup it does recognize hangup. Here's logging during an attempt where I make outbound call and answer, but then hangup after 1-2 seconds: Jan 24

Hi. Asterisk 1.4.12.1. I get a lot of message like this. Someone knows what this message mean? Do i have to worry about it? [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Local/152 at local-f137,1' not posted [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Agent/152' not posted [Nov 7 15:24:25] NOTICE[31247]: cdr.c:434

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

Hey gang, Currently I have this dialplan: exten => _9.,1,Dial(blah/blah) exten => h,1,ResetCDR(w) exten => h,2,NoCDR() exten => h,3,DeadAGI(rate_call.php) The AGI script takes the completed call, determines all that NPNAXX crap, finds the cost and then updates the CDR with the cost. Problem is, I keep getting these messages: Jan 4 13:25:36 WARNING[13689]: cdr.c:114

Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion.

I?m looking to install a couple of SIP phones into a small/medium company. The easiest way would be to simply add the phones on the LAN network. But what would happened if someone make a huge file transfer: will it make trouble on the Sip connections ? I think so, that?s why I?m asking you if there is a good (better) way? Maybe connect SIP phones on a separate network and setup a linux box as

I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico. This is really the first server I have used with PRI in Mexico as we normally use MFC/R2. Everything seems to be working except that some numbers always seem to be busy when you dial them. All these numbers belong to different phone companies. I know that with R2 this problem is present if you have a "#define

I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though

Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was

I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten => 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [0 at default:1] Dial("SIP/5166-b76004f8",

Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work fine. To avoid problems with the phone line (german E1) I called the provider, he did a 45 min. route

I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on a CentOS 5 server. The server has a single TE110 card connected to a provider called Alestra in Monterrey, Mexico. Since we installed everything we have been having problems dialing certain numbers, those numbers always fail when dialed from Asterisk but if you dial from your cell phone they always go through. I once has a

Hi, in dialplan -Asterisk 16.2 from Debian Buster- we have  same = n,Dial(PJSIP/101&PJSIP/102&PJSIP/103,15,tT) If thew call is not answered after 20 seconds, we launch a new dial with same and/or other extensions  same = n,Dial(PJSIP/101&PJSIP/104&PJSIP/110,20,tT) Looking in CDR we have at the end of the call (here we called 3 extensions which where ringing, let say 110

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

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

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk 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

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

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

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)