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 !
you have to set the accounting flag also for the bye message.
if (method=="BYE") { setflag(1); acc_rad_request("Some comment"); };
regards, klaus
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:0as 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 !
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@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@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@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@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@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@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:0as 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 !
Hi,
I'm testing click to dial service in serweb, and I there are some differences with draft-ietf-sipping-service-examples-06, are there intentioned changes?
http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-06.t xt
I think the draft solution is simpler to implement in terminals and application servers.
G.
serweb implements exactly that (REFER based) scenario.
Jan.
On 01-04 17:40, Gustavo Garcia Bernardo wrote:
Hi,
I'm testing click to dial service in serweb, and I there are some differences with draft-ietf-sipping-service-examples-06, are there intentioned changes?
http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-06.t xt
I think the draft solution is simpler to implement in terminals and application servers.
G.
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Klaus Darilion