similar to: CDR on channel not posted

Hello. Just installed the Beronet BN4S0 card. But i can't connect to my ISDN Line. misdnportinfo gives (what does ":Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib" mean?): best regards and thanks t. asterix asterisk # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> Layer 4 protocol 0x04000001

Hello. I have problems to integrate an isdn phone. I don't know why but the isdn phone rings only once and than it looses its connection to his base station. I can make a call from the isdn phone to an VoIP Phone inside my network but when i pick up the phone the isdn phone also crashes. misdn.conf: [ntport1] ports=5 context=isdn-telefon msns=* extensions.conf: exten =>

Hello. Someone knows what cause 127 mean. The phone that i'm calling rings once and than the connection interrupts: P[ 5] --> l3id:10040 P[ 5] --> cause:127 P[ 5] --> out_cause:127 P[ 5] --> state:ALERTING P[ 5] --> Channel: mISDN/5-1 hanguped new state:CLEANING P[ 5] $$$ CLEANUP CALLED pid:3 best regards -- Thomas Stein knowledgeTools? ....damit Sie sehen, was Sie

Hello. Is it possible to "mix" musiconhold music and playback voices? What i want to do is something like this: A person calls a number, gets a playback voice while in background music is playing. The configuration i use at the moment don't do what i want. Someone knows how to do it? Thanks in advance. exten => 18,1,Answer exten => 18,n,Background() exten =>

Hello. I have a problem recieving fax without a callerid. Somehow the script i'm using fails and i don't know how to fix it. Does anyone have an idea how to solve this? Here an example of a working fax transmission: >fax2mail v2.0 > Triggered on Tuesday, May 29 2007, at 10:38 AM > $1 = CallerID number of fax sender = 02365207150 > $2 = CallerID name of fax sender = >

Hello. Is it possible to change the defined moh sound file within an extension? I have: exten => 18,1,Answer exten => 18,n,Wait(3) exten => 18,n,SetMusicOnHold(durchwahl) exten => 18,n,Dial(SIP/118,15,m) exten => 18,n,Hangup Now i have the situation someone calls and my phone is ringing while moh (durchwahl) is playing. When i pickup the call and press the hold button during

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 =>

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? Thanks.

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.

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log

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

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

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

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

1.6.2: -- Executing [s at incoming-pstn-line:4] VoiceMail("DAHDI/4-1", "100 at default,u") in new stack -- <DAHDI/4-1> Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]:

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

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

All, I am using Asterisk 1.4.4 and it is not playing any MOH. I think the underlying problem is the following error: [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/moh/asterisk' [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player Now it does not matter what I change in the