similar to: Failed to record Radius CDR record!

hi , all when i do the command "module load cdr_radius.so" ,error happens. i have installed radiusclient-ng , what's wrong with it? thanks! error message as follow: ZHANGSHUKUN*CLI> module load cdr_radius.so Unable to load module cdr_radius.so Command 'module load cdr_radius.so' failed. [Dec 23 17:55:41] WARNING[31072]: loader.c:380 load_dynamic_module: Error

I did radius client status testing with radius server, able to access the radius server. However, still getting radius CDR issue after setting debug level 8 even granting 666 access to radiusclient-ng config files. message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR not recorded! Please advise if I missed out anything. Date: Mon, 26 Sep 2016 12:09:34 +0200 > From:

Hi Andrew and Willy, Thanks for sharing the info. As for enabling radius server debugging 'radiusd -X', made some test calls don't see the radiusclient sending data to radius server. However, using radtest or radiusclient testing, able to send data to radius server (after enabling debug). For further testing, on my other server using OpenSIPs, setup the radiusclient and data was

Hello list. I'm having some problems with the CDR Radius in my Asterisk 1.4. I'm using two TC400B cards for transcoding. When I reach nearly 100 simmultaneous calls, the CDR radius packets are being duplicated and I'm getting this message in the asterisk console : cdr_radius.c:227 radius_log: Failed to record Radius CDR record! I'm also using the radiusclient-ng 0.5.6

Hi, I needed my CDR's to be stored using a RADIUS server. I found cdr_radius in the src directory. Looked in /docs for how to install it and I got it to work. Just want to say thanks to those who helped write this. Has anybody else used this, any comments, cause I found nothing using google, even voip-info has nothing on this module? -- thanks, Yusuf

Hello, I'm trying to configure Asterisk with Radius cdr support. Asterisk version 1.6.2.13 Server Radius: Freeradius version 1.X Radius client: radiusclient-ng version 0.5.5 With the Asterisk core debug on 1 when a call terminate, on the console appear this error: Unable to create RADIUS record. CDR not recorded! My cdr.conf is: [radius] usegmtime=yes ; log date/time in GMT

hi, guys, when i create a manager account used for freepbx, the follow info produce all the time? do you know that's the reason? == Manager 'bitzsk' logged off from 127.0.0.1 == Manager 'bitzsk' logged on from 127.0.0.1 == Manager 'bitzsk' logged off from 127.0.0.1 == Manager 'bitzsk' logged on from 127.0.0.1 == Manager 'bitzsk' logged

hi, All one thing confused me a long time. when i change the extensions.conf file. why not take effects after restart the asterisk? details as follow: my dailplan is : [95040] exten => _95040XXXXX,1,Set(CALLINNUM=${CALLERID(dnid)}) exten => _95040XXXXX,n(start),Answer exten => _95040XXXXX,n(welcome),Background(${welcomefile},,123) ... exten => i,1,Playback(invalid) exten =>

hi, list i want to know what is the best OS for install Asterisk 1.6.2.9, which should work properly on working system. i want to use CentOS5.2 or CentOS 5.4. Which is better and stable? Thanks for your help. -- Thanks for your supporting, have a nice day. Sucan

hi, all i want to use g729 codec for set up a call. so i donwloaded the so file from web site: http://asterisk.hosting.lv/#bin and install it properly. *CLI> *CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin

hi, Does A2Billing has mial list? -- Thanks, Sucan

Hi,all i want to hear the voicemail recorded, but when hear "if you want to play message , press 3", after i press 3 i only hear that that's the time the file recorded. not the content. do you know how to hear content of voicemail fle? debug message: == Parsing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001.txt': Found -- <SIP/1003-00000058>

hello,all there is a AudioCodes Mediant 2000 out there. i want to realise ip to PSTN and PSTN to ip connection. after some configuration on AudioCodes Mediant 2000, PSTN to ip connecttion works. next ,i want to dial from asterisk to PSTN now. i have see the sample in the extensions.conf relevent to PSTN as follow: ; If you are freely delivering calls to the PSTN, list them here ; ;exten =>

hi,list Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ? after i make and make install. i cant find the .so file. is this mean it can't install on 64bit Cent-OS. ps: it works fine on the 32 bit Cent-OS Thanks very much! -- Thanks for your supporting, have a nice day. Sucan

hi,all for a long time, i cant understand the difference between ${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)} i know ${CDR(start)} mean when a call is start. and ${CDR(answer)} means when a call was pick up. but what's ${CDR(calldate)} mean? Could you help me ? Thansk a lot! -- Thanks for your supporting, have a nice day. Sucan

Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several times and ring another? ; A strategy may be specified. Valid strategies include: ; ; ringall - ring all available channels until one answers (default) ; roundrobin - take turns ringing each available interface ;

Hai all. I'm currently developing a softswitch with asterisk server. But i have a few problems here. Anyone can help me? 1. Is there any add on for asterisk 1.2.0 to connect it to Freeradius server? 2. As far as i know, i need a file named cdr_radius.c and compile my asterisk again, but when i did this, my asterisk suddenly error. Any time i start the service, it will terminate after a few

Hi All, I found app_radius and cdr_radius, but I'm not sure how to use them. They are poorly documented. There are a few open source billing systems (trabas) out there that use radius as the cdr method and I'm wondering how this can be integrated with *. Just adding cdr_radius, like cdr_mysql doesn't work.. Does anyone have any ideas? Anyone know of a billing system that can just use

hi, in my test, i noticed that sip connection will hangup automaticlly when no speaks between the channel. about half a minute. is this the asterisk inner mechanism or is my configuration error? Thanks! messages on the cli as follow: -- SIP/1003-0000001d is ringing -- SIP/1003-0000001d answered SIP/1004-0000001c -- Stopped music on hold on SIP/1004-0000001c [Jan 18 10:08:42]

Hello there, I have successfully installed and configured asterisk for use as an office PBX using SIP trucks and Voip handsets (using g.729 codec) which works great. Now I wish to try and configure asterisk to do a HTTP request and submit callerID to an external website when a call is missed. eg Someone calls PBX and rings extension 100 -> Call is not answered -> HTTP request is initiated