16 Nov
2005
16 Nov
'05
10:59 p.m.
We could determine who sent the BYE from the value of
ftag parameter in
Route header field and swap the values, the question is whether this
makes sense. The billing system could take all info from start requests
I think it would make sense just from a uniformity stand point.
and use the stop request to calculate the session
duration only.
I don't know if that's the case, but this would probably vary from billing
system to billing system. I think if we keep it uniform, then it would never
be a problem. Specially, for billing systems that only support stop-only
records and billing systems that authenticate by using PIN prefixes,
specially when one of the parties is not a user but a PSTN number. On my
billing platform it wont work, it complains about the discrepancy, and the
fact that when the callee hangs up it doesn't recognize that the CDR is for
a PSTN user.
Lenir
-----Original Message-----
From: 'Jan Janak' [mailto:jan@iptel.org]
Sent: Wednesday, November 16, 2005 3:45 PM
To: Lenir
Cc: serdev(a)lists.iptel.org; serusers(a)lists.iptel.org; users(a)openser.org;
devel(a)openser.org
Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius
Accounting
Yes, this can happen if the callee sends BYE. In this case the URIs in
From and To header fields are swapped.
The only SIP RADIUS related document (which has expired a long time ago)
says that Calling-Station-Id is the URI from From header field of any
SIP request. Thus -- according this I-D it is correct. But use of RADIUS
with SIP is underspecified so there is no real standard for this.
I don't think it matters that much, because other attributes, namely
Acct-Session-Id are used to match start and stop radius requests.
We could determine who sent the BYE from the value of ftag parameter in
Route header field and swap the values, the question is whether this
makes sense. The billing system could take all info from start requests
and use the stop request to calculate the session duration only.
Jan.
On 16-11-2005 15:04, Lenir wrote:
Ok, that put the right things in place. However,
here's a twist:
Either way, wit switching Sip-Translated-Request-URI and Called-Station-ID
around, or not, the CDRs are sent differently depending on who "hands up"
the call. For example, notice the User-Name, Calling-Station-ID and
Called-Station-ID:
User 1000 calls 15615551212, and user 1000 hangs up:
=====================================================
rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39228, id=206,
length=306
Acct-Status-Type = Start
Service-Type = Sip-Session
Acct-Terminate-Cause = 200
Error-Cause = Invite
User-Name = "1000(a)xx.xx.xx.xx"
Calling-Station-Id = "sip:1000@xx.xx.xx.xx"
Sip-Translated-Request-URI =
"sip:15615551212@xx.xx.xx.xx;user=phone"
Called-Station-Id =
"sip:949449852150#15615551212@gateway;user=phone"
Acct-Session-Id =
"000d2890-d47f01f5-6410e8bd-4baf0435(a)66.232.8.116"
Sip-To-Tag = "33C5A3C-76A"
Sip-From-Tag = "000d2890d47f24f85e6c0b91-0092ab53"
Sip-CSeq = "102"
NAS-IP-Address = xx.xx.xx.xx
NAS-Port = 5060
Acct-Delay-Time = 0
rlm_sql (sql): Reserving sql socket id: 15
rlm_sql (sql): Released sql socket id: 15
Sending Accounting-Response of id 206 to xx.xx.xx.xx:39228
rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39228, id=207,
length=300
Acct-Status-Type = Stop
Service-Type = Sip-Session
Acct-Terminate-Cause = 200
Error-Cause = 8
User-Name = "1000(a)xx.xx.xx.xx"
Calling-Station-Id = "sip:1000@xx.xx.xx.xx"
Sip-Translated-Request-URI =
"sip:15615551212@xx.xx.xx.xx;user=phone"
Called-Station-Id =
"sip:949449852150#15615551212@gateway:5060"
Acct-Session-Id =
"000d2890-d47f01f5-6410e8bd-4baf0435(a)66.232.8.116"
Sip-To-Tag = "33C5A3C-76A"
Sip-From-Tag = "000d2890d47f24f85e6c0b91-0092ab53"
Sip-CSeq = "103"
NAS-IP-Address = xx.xx.xx.xx
NAS-Port = 5060
Acct-Delay-Time = 0
User 1000 calls 15615551212, and user 15615551212 hangs up:
===========================================================
rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39227, id=200,
length=306
Acct-Status-Type = Start
Service-Type = Sip-Session
Acct-Terminate-Cause = 200
Error-Cause = Invite
User-Name = "1000(a)xx.xx.xx.xx"
Calling-Station-Id = "sip:1000@xx.xx.xx.xx"
Sip-Translated-Request-URI =
"sip:15615551212@xx.xx.xx.xx;user=phone"
Called-Station-Id =
"sip:949449852150#15615551212@gateway;user=phone"
Acct-Session-Id =
"000d2890-d47f01f4-372dff4b-793d4acc(a)66.232.8.116"
Sip-To-Tag = "33B37E4-77B"
Sip-From-Tag = "000d2890d47f24f639ccef77-4ef08b02"
Sip-CSeq = "102"
NAS-IP-Address = xx.xx.xx.xx
NAS-Port = 5060
Acct-Delay-Time = 0
rlm_sql (sql): Reserving sql socket id: 1
rlm_sql (sql): Released sql socket id: 1
Sending Accounting-Response of id 200 to xx.xx.xx.xx:39227
rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39227, id=201,
length=286
Acct-Status-Type = Stop
Service-Type = Sip-Session
Acct-Terminate-Cause = 200
Error-Cause = 8
User-Name = "15615551212(a)xx.xx.xx.xx"
Calling-Station-Id = "sip:15615551212@xx.xx.xx.xx;user=phone"
Sip-Translated-Request-URI = "sip:1000@xx.xx.xx.xx"
Called-Station-Id = "sip:1000@66.232.8.116:5061"
Acct-Session-Id =
"000d2890-d47f01f4-372dff4b-793d4acc(a)66.232.8.116"
Sip-To-Tag =
"000d2890d47f24f639ccef77-4ef08b02"
Sip-From-Tag = "33B37E4-77B"
Sip-CSeq = "101"
NAS-IP-Address = xx.xx.xx.xx
NAS-Port = 5060
Acct-Delay-Time = 0
Shouldn't it always be that the User-Name attribute is the user who
initiated the call?
Lenir
-----Original Message-----
From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On
Behalf
Of 'Jan Janak'
Sent: Wednesday, November 16, 2005 1:53 PM
To: lsantiago(a)globalgatewaycom.com
Cc: serdev(a)lists.iptel.org; serusers(a)lists.iptel.org; users(a)openser.org;
devel(a)openser.org
Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius
Accounting
Either reconfigure the billing application to use
Sip-Translated-Request-URI (recommended) or change the ID of the
attribute in the radiusclient dictionary as suggested by Bogdan.
In this particular example you should set Called-Station-Id to 107.
Jan.
On 16-11-2005 13:09, 'Lenir' wrote:
> Ok, so how do I tell acc.so to send Sip-Translated-Request-URI as the
> Called-Station-ID instead?
>
> Lenir Santiago, Partner
> Global Gateway Communications, Inc.
> 2500 Quantum Lakes Drive, Ste. 203
> Boynton Beach, FL 33426
> Office: 561-853-2213
> Cell: 561-722-0966
> Fax: 877-875-0060
> http://www.globalgatewaycom.com
>
> -----Original Message-----
> From: 'Jan Janak' [mailto:jan@iptel.org]
> Sent: Wednesday, November 16, 2005 11:29 AM
> To: Bogdan-Andrei Iancu
> Cc: Lenir; 'Lenir Santiago'; serdev(a)lists.iptel.org;
serusers(a)lists.iptel.org;
> devel(a)openser.org; users(a)openser.org
> Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius
> Accounting
>
> Why would you do something like this ? All the info that is needed is in
> Sip-Translated-Ruquest-ID below. You can see it in the RADIUS dump:
>
> Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060"
>
> Jan.
>
> On 16-11-2005 16:50, Bogdan-Andrei Iancu wrote:
> > Hi Lenir,
> >
> > you can keep the modified To value into an AVP and use the extra
> > accounting feature to log it. See
> > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN115
> > and
> > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN350
> >
> > the problem is how to set it as Called-Station-ID attribute in order
to
> > avoid overlapping. You can do a trick by
playing with the radius
> > dictionary (on proxy side) :
> > changed the Called-Station-ID to a different ID attribute
> > set extra accounting with the original attribute ID of
> Called-Station-ID
> > in this way the acc will set de default TO as something else and you
AVP
> > as Called-Station-ID ;)
> >
> > regards,
> > bogdan
> >
> >
> > Lenir wrote:
> >
> > >Well, if userA dials to:30, 30 gets translated to:15615551212 and SER
> dials
> > >15615551212, how can I make the acc.so module send the modified
> > >to:15615551212 as the Called-Station-ID attribute?
> > >
> > >This is important for billing, specially if someone forwards to a
long
> > >distance or international long distance
number.
> > >
> > >-----Original Message-----
> > >From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
On
> > >Behalf Of Jan Janak
> > >Sent: Wednesday, November 16, 2005 3:15 AM
> > >To: Lenir Santiago
> > >Cc: serdev(a)lists.iptel.org; serusers(a)lists.iptel.org; users(a)openser.org;
> > >devel(a)openser.org
> > >Subject: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
> > >
> > >Hello,
> > >
> > >Called-Station-Id attribute always contains the To URI (not
> > >Request-URI). You can find the outbound Request-URI in
> > >Sip-Translated-Request-URI attribute.
> > >
> > >Even if you rewrite To header field, it will never be accounted in
> > >Called-Station-Id.
> > >
> > > Jan.
> > >
> > >On 15-11-2005 21:59, Lenir Santiago wrote:
> > >
> > >
> > >>Hey guys,
> > >>
> > >>
> > >>
> > >>I'm using Radius to do pretty much everything regarding my users,
i.e.
> > >>registrations, authorization,
features and preferences. I
implemented
my
> > >>
> > >>
> > >own
> > >
> > >
> > >>way of doing speeddial by using AVPs provided by radius and AVPOPs
> module,
> > >>Here is the snippet:
> > >>
> > >>
> > >>
> > >>AVP-SIP = speedial:30:15615551212
> > >>
> > >>
> > >>
> > >> avp_subst("s:caller_speedial/i:99",
"/(.*):(.*)/\2/");
> > >>#15615551212
> > >>
> > >> avp_printf("i:98","$rU:*"); #30
> > >>
> > >>
avp_printf("i:97","sip:$avp($temp2)@$td"); #
> > >>sip:15615551212@xx.xx.xx.xx
> > >>
> > >>
avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); #
> > >><sip:15615551212@xx.xx.xx.xx>.
> > >>
> > >> xlog("L_N"," SPEEDIALING: $fU is
speed-dialing
> > >>$hdr(To)->$avp($temp2)\n");
> > >>
> > >> if (avp_check("s:caller_speedial",
"fm/$temp1/g")) {
> > >>
> > >>
> > >>
> > >> #avp_pushto("$To","i:96");
> > >>
> > >>
#avp_pushto("$ruri","i:97");
> > >>
> > >>
#avp_pushto("$duri","i:97");
> > >>
> > >> #subst('/^To: (.*)(a)(.*)/To:
"$avp($temp2)"
> > >><$avp($temp2)@\2/');
> > >>
> > >> #append_hf("To: \"5615551212\"
> > >><15615551212(a)xx.xx.xx.xx>;user=phone")user=phone");
> > >>
> > >>
> > >>#subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i');
> > >>
> > >> #subst_user('/(.*)/$avp($temp2)/');
> > >>
> > >> #remove_hf("To");
> > >>
> > >> rewriteuser("15615551212");
> > >>
> > >> xlog("L_N"," SPEEDIALING: $fU
is dialing
> > >>
> > >>
> > >$hdr(To)
> > >
> > >
> > >>$avp($temp2)\n");
> > >>
> > >> avp_delete("$temp1/g");
> > >>
> > >> avp_delete("$temp2/g");
> > >>
> > >>
> > >>
> > >> };
> > >>
> > >>
> > >>
> > >>In this scenario, im forcing it instead of getting the value from
AVPs
>>
>>
>just
>
>
>>for ease of showing my point. The problem that I'm having is that no
>>
>>
>matter
>
>
>>what I do to change the To header or change the R-URI, radius ALWAYS
sends
>>the accounting records with the original
R-URI, in my case
30(a)xx.xx.xx.xx,
> >>how can I get acc.so to send the final resulting URI
> >>(15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP?
> >>
> >>
> >>
> >>
> >>
> >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, > >>length=298
> >>
> >> Acct-Status-Type = Start
> >>
> >> Service-Type = Sip-Session
> >>
> >> Acct-Terminate-Cause = 200
> >>
> >> Error-Cause = Invite
> >>
> >> User-Name = "1000(a)xx.xx.xx.xx"
> >>
> >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx"
> >>
> >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone"
> >>
> >> Sip-Translated-Request-URI =
> >>"sip:949449852150#15615551212@gateway;user=phone"
> >>
> >> Acct-Session-Id =
> >>
> >>
> >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy"
> >
> >
> >> Sip-To-Tag = "230383F4-2AF"
> >>
> >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33"
> >>
> >> Sip-CSeq = "102"
> >>
> >> NAS-IP-Address = xx.xx.xx.xx
> >>
> >> NAS-Port = 5060
> >>
> >> Acct-Delay-Time = 0
> >>
> >>rlm_sql (sql): Reserving sql socket id: 12
> >>
> >>rlm_sql (sql): Released sql socket id: 12
> >>
> >>Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216
> >>
> >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216,
id=98,
>>length=292
>>
>> Acct-Status-Type = Stop
>>
>> Service-Type = Sip-Session
>>
>> Acct-Terminate-Cause = 200
>>
>> Error-Cause = 8
>>
>> User-Name = "1000(a)xx.xx.xx.xx"
>>
>> Calling-Station-Id = "sip:1000@xx.xx.xx.xx"
>>
>> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone"
>>
>> Sip-Translated-Request-URI =
>>"sip:949449852150#15615551212@gateway:5060"
>>
>> Acct-Session-Id =
>>
>>
>"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy"
>
>
>> Sip-To-Tag = "230383F4-2AF"
>>
>> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33"
>>
>> Sip-CSeq = "103"
>>
>> NAS-IP-Address = xx.xx.xx.xx
>>
>> NAS-Port = 5060
>>
>> Acct-Delay-Time = 0
>>
>>
>>
>>
>>
>>Thanks in advance!
>>
>>
>>
>>
>>
>>Lenir
>>_______________________________________________
>>Users mailing list
>>Users(a)openser.org
>>http://openser.org/cgi-bin/mailman/listinfo/users
>>
>>
>>