Hello Klaus,
Thursday, April 1, 2004, 4:46:36 PM, you wrote:
you where right, I was stupid :) TNX !
now I have byes logged, but strangely I get 2 radius entry for each
invite ... perfectly equal !
an example of call logs:
hu Apr 1 15:59:46 2004
Acct-Status-Type = Failed
Service-Type = Sip-Session
Sip-Response-Code = 0
Sip-Method = 1
User-Name = "300300(a)mydomain.com"
Calling-Station-Id = "sip:300300@mydomain.com"
Called-Station-Id = "sip:0190@mydomain.com"
Sip-Translated-Req-ID = "sip:0190@mydomain.com"
Acct-Session-Id = "3274C566-B0AC-44C5-B5DB-D7BA8A08902C(a)192.168.0.157"
Sip-To-Tag = "n/a"
Sip-From-Tag = "3523280348"
Sip-Cseq = "31534"
NAS-IP-Address = 192.168.0.230
NAS-Port = 5060
Acct-Delay-Time = 0
Client-IP-Address = 192.168.0.230
Acct-Unique-Session-Id = "e98c50261d9120cf"
Timestamp = 1080827986
Thu Apr 1 15:59:46 2004
Acct-Status-Type = Failed
Service-Type = Sip-Session
Sip-Response-Code = 0
Sip-Method = 1
User-Name = "300300(a)mydomain.com"
Calling-Station-Id = "sip:300300@mydomain.com"
Called-Station-Id = "sip:0190@mydomain.com"
Sip-Translated-Req-ID = "sip:0190@mydomain.com"
Acct-Session-Id = "3274C566-B0AC-44C5-B5DB-D7BA8A08902C(a)192.168.0.157"
Sip-To-Tag = "n/a"
Sip-From-Tag = "3523280348"
Sip-Cseq = "31535"
NAS-IP-Address = 192.168.0.230
NAS-Port = 5060
Acct-Delay-Time = 0
Client-IP-Address = 192.168.0.230
Acct-Unique-Session-Id = "e98c50261d9120cf"
Timestamp = 1080827986
Thu Apr 1 15:59:57 2004
Acct-Status-Type = Failed
Service-Type = Sip-Session
Sip-Response-Code = 0
Sip-Method = 8
User-Name = "300300(a)mydomain.com"
Calling-Station-Id = "sip:300300@mydomain.com"
Called-Station-Id = "sip:0190@mydomain.com"
Sip-Translated-Req-ID = "sip:0190@192.168.0.230:5090"
Acct-Session-Id = "3274C566-B0AC-44C5-B5DB-D7BA8A08902C(a)192.168.0.157"
Sip-To-Tag = "as60ec7890"
Sip-From-Tag = "3523280348"
Sip-Cseq = "31536"
NAS-IP-Address = 192.168.0.230
NAS-Port = 5060
Acct-Delay-Time = 0
Client-IP-Address = 192.168.0.230
Acct-Unique-Session-Id = "e98c50261d9120cf"
Timestamp = 1080827997
my cfg
if (uri==myself) {
if (method=="INVITE") {
setflag(1);
acc_rad_request("Registrazione
INVITE");
record_route();
if (uri=~"sip:0[0-9]+@"){
if (!proxy_authorize("mydomain.com",
"subscriber")) {proxy_challenge("mydomain.com",
"0");sl_send_reply("403", "That's not your
home");break;}; #fine proxy challenge
if (!is_user_in("credentials",
"local")){sl_send_reply("403", "No permission for local
calls");break;}; #fine invite
rewritehostport("sip.mydomain.com:5090");
t_relay();
break;
}; #fine if uri sip:0
if (!lookup("location")) {sl_send_reply("404", "Not
Found");break;}; #fine if lookup location
}; #fine invite
}; #fine myself
KD> you have to set the accounting flag also for the bye message.
KD> if (method=="BYE") {
KD> setflag(1);
KD> acc_rad_request("Some comment");
KD> };
KD> regards,
KD> klaus
KD> Alessio Focardi wrote:
> Hi friends,
>
> we recently added an isdn card to our asterisk/ser server.
>
> to keep track of outbound calls I have set up a radius server, I
> would like to log every pstn call.
>
> Problem is that I get start logged, but no call end.
>
> this is ser.cfg part of the trick
>
> if (method=="INVITE") {
>
> if (uri==myself) {
>
>
> if (uri=~"sip:0[0-9]+@"){
>
> setflag(1);
> acc_rad_request("Some comment");
> record_route();
> rewritehostport("sip.mydomain.com:5090");
> t_relay_to_udp("127.0.0.1", "5090");
> break;
>
> }; #fine if uri sip:0
>
> as you can see there is an asterisk listening on port 5090 (same
> machine) that strips the leading 0 from every call and then forward to
> the pstn ....
>
> I suppose maybe t_relay_to_udp is not the right forward type ... do I
> have to append brach or something like ?
>
> Tnx for any help !
>
--
Best regards,
Alessio mailto:afoc@interconnessioni.it