I've been playing with ser for a while checking it in different
scenarios. All this time I used it with
fork=no
log_stderror=yes
settings and it worked.
Now I've come up with some draft ser.cfg and wanted to
run it as daemon. I've changed it to
fork=yes
log_stderror=no
and now ser doesn't start. Actually "log_stderror=no" is the setting
that causes problem. If it's set to "yes", ser starts fine even in forked
mode.
Also I narrowed down the problem to postgres module. If I remove it
from ser.cfg then ser starts as it should.
With postgres module enabled it doesn't throw a word of complaint to
a log and it looks fine from console:
root@pg1old:~/src/ser/ser-0.9.3# /usr/local/ser/sbin/ser -f /etc/ser/ser.cfg
Listening on
udp: 192.168.0.61 [192.168.0.61]:5060
tcp: 192.168.0.61 [192.168.0.61]:5060
Aliases:
tcp: pg1old:5060
tcp: pg1old.acecape.com:5060
udp: pg1old:5060
udp: pg1old.acecape.com:5060
*: 192.168.0.61:*
but
root@pg1old:~/src/ser/ser-0.9.3# ps -auxw | grep ser
root 18322 0.0 0.3 38836 1824 ? S 15:05 0:00 /usr/local/ser/sbin/ser -f /etc/ser/ser.cfg
shows that there's just one ser process running, netstat shows that it's not listening on any port and
also no FIFO file is created.
Again everything works fine if "log_stderror" set to "yes".
Looks to me as some strange bug. Any clues?
Thanks,
Michael Ulitskiy
Hello,
No I have ser-0.9.3 and two different versions of rtpproxy.
Is tried all combination without results.
25(664) WARNING: rtpp_test: can't get version of the RTP proxy
25(664) WARNING: rtpp_test: support for RTP proxy has been disabled
temporarily
26(665) ERROR: send_rtpp_command: can't connect to RTP proxy
Has anyone of you an idea, please?
Thanks!
Hello.
Mmmhh, are you sure you modified the de www_challenge for the
proxy_challenge in the ser.cfg file?. I use RADIATOR as my Radius Server so
i'm not very familiarized with freeRadius. But for the debug it seems to be
an error maybe with the configuration from the Radius Server?
For example , is normal this : Invalid operator for item Suffix: reverting
to '==' ?
Maybe somone that uses freeRadius could give you more details.
To accounting i use Radiator but working together with an Oracle Database, i
use the Start and Stop message from SER to bill the call.
Regards,
Ricardo Martinez.-
-----Mensaje original-----
De: Naresh Parmar [mailto:naresh_parmar14@yahoo.com]
Enviado el: Miércoles, 20 de Julio de 2005 13:09
Para: Ricardo Martinez; serusers(a)lists.iptel.org
Asunto: RE: [Serusers] Problem authorizing with radius
Hi Ricardo,
Tried it. It still gives me the same error. Please let me know the version
of the radius server you are using.?? Also can you please let me know wht
did u do to make the accounting work...??
Best Regards,
Naresh
Ricardo Martinez <rmartinez(a)redvoiss.net> wrote:
Hello Naresh.
I guess there is an error in the way you call the authorization for the
INVITE. As far as i know for the REGISTER message (authentication) you need
the statement :
radius_www_authorize
But for the INVITE you need to call "radius_proxy_authorize". This is
what i have in my ser.cfg
if (method=="INVITE") {
if (!radius_proxy_authorize("")) {
proxy_challenge("","1");
break;
};
};
maybe you can try this and tell me how it works.
Good luck
Ricardo Martinez.-
-----Mensaje original-----
De: Naresh Parmar [mailto:naresh_parmar14@yahoo.com]
Enviado el: Miércoles, 20 de Julio de 2005 12:10
Para: Ricardo Martinez; serusers(a)lists.iptel.org
Asunto: RE: [Serusers] Problem authorizing with radius
Hi Ricardo,
We are using freeradius server 0.9.1 and SER 0.9.3. The version of radius
client is radiusclient-ng-0.5.1. The users file in the radius server looks
like as below:
test(a)sip2.zone <mailto:test@sip2.zone> Auth-Type := Digest, User-Password
== "cisco1234"
Reply-Message = "Authenticated",
Sip-Rpid = "1970"
test(a)sip2.zone <mailto:test@sip2.zone> Auth-Type := Accept
Reply-Message = "Authorized",
Sip-Group == "ld"
The radius authentication and authorization parts in the ser.cfg file are
given below:
if (uri=~"^sip:9[0-9]*@") {
if (method=="INVITE"){
if (!radius_www_authorize("")) {
www_challenge("", "1");
break;
}else{
if (radius_is_user_in("Credentials",
"ld")){
forward(192.168.2.101,5060);
break;
}else{
break;
};
};
};
};
And finally the error is as below:
Invalid operator for item Suffix: reverting to '=='
modcall[authorize]: module "preprocess" returns ok
modcall[authorize]: module "chap" returns noop
rlm_eap: No EAP-Message, not doing EAP
modcall[authorize]: module "eap" returns noop
rlm_digest: Converting Digest-Attributes to something sane...
Digest-User-Name = "test"
Digest-Realm = "sip2.zone"
Digest-Nonce = "42de75b2e9e39194a286e8ccd284646ffa14bcc2"
Digest-URI = "sip:94161000@sip2.zone"
Digest-Method = "INVITE"
Digest-QOP = "auth"
Digest-Nonce-Count = "0000000a"
Digest-CNonce = "753F926DB8F5415D8D56EE7816410E33"
rlm_digest: Adding Auth-Type = DIGEST
modcall[authorize]: module "digest" returns ok
rlm_realm: Looking up realm "sip2.zone" for User-Name = " test(a)sip2.zone
<mailto:test@sip2.zone> "
rlm_realm: No such realm "sip2.zone"
modcall[authorize]: module "suffix" returns noop
users: Matched entry test(a)sip2.zone <mailto:test@sip2.zone> at line 226
modcall[authorize]: module "files" returns ok
modcall[authorize]: module "mschap" returns noop
modcall: group authorize returns ok
rad_check_password: Found Auth-Type Digest
auth: type "digest"
modcall: entering group authenticate
A1 = test:sip2.zone:cisco1234
A2 = INVITE:sip:94161000@sip2.zone
KD =
53d3b82970bada131a062103f553b8b8:42de75b2e9e39194a286e8ccd284646ffa14bcc2:00
00000a:753F926DB8F5415D8D56EE7816410E33:auth:18227b358ffe96049a3745eeb449fae
2
modcall[authenticate]: module "digest" returns ok
modcall: group authenticate returns ok
radius_xlat: 'Authenticated'
Login OK: [test(a)sip2.zone/<no User-Password attribute>] (from client proxy
port 5060)
Sending Access-Accept of id 203 to 192.168.2.1:32831
Reply-Message = "Authenticated"
Sip-Rpid = "1970"
Finished request 6
Going to the next request
--- Walking the entire request list ---
Waking up in 6 seconds...
rad_recv: Access-Request packet from host 192.168.2.1:32831, id=204,
length=53
User-Name = "test"
Sip-Group = "ld"
Service-Type = Group-Check
NAS-IP-Address = 192.168.2.1
NAS-Port = 0
modcall: entering group authorize
Invalid operator for item Suffix: reverting to '=='
Invalid operator for item Suffix: reverting to '=='
Invalid operator for item Suffix: reverting to '=='
modcall[authorize]: module "preprocess" returns ok
modcall[authorize]: module "chap" returns noop
rlm_eap: No EAP-Message, not doing EAP
modcall[authorize]: module "eap" returns noop
modcall[authorize]: module "digest" returns noop
rlm_realm: No '@' <mailto:'@'> in User-Name = "test", looking up realm
NULL
rlm_realm: No such realm "NULL"
modcall[authorize]: module "suffix" returns noop
modcall[authorize]: module "files" returns notfound
modcall[authorize]: module "mschap" returns noop
modcall: group authorize returns ok
auth: No authenticate method (Auth-Type) configuration found for the
request: Rejecting the user
auth: Failed to validate the user.
Login incorrect: [test/<no User-Password attribute>] (from client proxy port
0)
Delaying request 7 for 1 seconds
Finished request 7
Going to the next request
Waking up in 6 seconds...
As you can see from the above configuration, the authentication works
perfect, its only in the authorization where it fails. Also can you please
let me know about the accounting configuration??
Thanks a lot..
Naresh
Ricardo Martinez <rmartinez(a)redvoiss.net> wrote:
Hello Naresh
I have authentication, authorization and accounting (AAA) through radius
working fine. What radius server are you using?, can you send us more
information about the configuration?
Cheers,
Ricardo.-
-----Mensaje original-----
De: Naresh Parmar [mailto:naresh_parmar14@yahoo.com]
Enviado el: Miércoles, 20 de Julio de 2005 10:37
Para: serusers(a)lists.iptel.org
Asunto: [Serusers] Problem authorizing with radius
hi friends,
I am having problems while authorizing with the radius server. I am using
the same configuration as mentioned in the radius-howto. Authentication
works perfect as I am able to authenticate using the radius server. However
while authorizing against the radius server to make a call I get the
following error:
auth: No authenticate method (Auth-Type) configuration found for the user
request: Rejecting the user
auth: Failed to validate the user.
Delaying request 2 for 1 seconds
Finished request 2
When I authorize against the mysql database, it works fine. Any clue???
Best Regards,
Naresh
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
_____
Start <http://us.rd.yahoo.com/evt=34442/*http://www.yahoo.com/r/hs> your
day with Yahoo! - make it your home page
Are there any plans to backport this feature to 0.9.0?
Re:
sobomax 2005/04/27 16:15:21 CEST
SER CVS Repository
Modified files:
modules/nathelper nathelper.c
Added files:
modules/nathelper nathelper.h natping.c
Log:
Add support for using real SIP requests for doing natpinging instead of
UDP packets with 4 zero bytes payload. This provives workaround for brain
damaged NATs which close hole if no packets are being sent through it
from inside, even if there are some packets going from outside.
The feature can be controlled via natping_method variable. By default old
method is used, otherwise its value is taken verbatim and used as method
name for generating requests. For example setting natping_method="OPTIONS"
will instruct the module to use OPTIONS requests for natpinging.
Are there any plans to backport this feature to 0.9.0?
Re:
sobomax 2005/04/27 16:15:21 CEST
SER CVS Repository
Modified files:
modules/nathelper nathelper.c
Added files:
modules/nathelper nathelper.h natping.c
Log:
Add support for using real SIP requests for doing natpinging instead of
UDP packets with 4 zero bytes payload. This provives workaround for brain
damaged NATs which close hole if no packets are being sent through it
from inside, even if there are some packets going from outside.
The feature can be controlled via natping_method variable. By default old
method is used, otherwise its value is taken verbatim and used as method
name for generating requests. For example setting natping_method="OPTIONS"
will instruct the module to use OPTIONS requests for natpinging.
>
> Ricardo, could you please test these various solutions and
> see if they
> really work or if (where...) I go wrong :-)
> g-)
Greger, Thanks for your insightful answer, i will make the test and i will
let you know
about my results...
Thanks again!!.
Regards,
Ricardo Martinez.-
It should... but it doesn't. I have ser 0.9.0 and the latest rtpproxy
version.
WARNING: rtpp_test: can't get version of the RTP proxy
----- Original Message -----
From: "harry gaillac" <gaillacharry(a)yahoo.fr>
To: "Sebastian Kühner" <skuehner(a)veraza.com>
Sent: Wednesday, July 20, 2005 1:44 PM
Subject: Re: [Serusers] ACK
> your rtpproxy should work !
>
> --- Sebastian Kühner <skuehner(a)veraza.com> a écrit :
>
> > Hi,
> >
> > Ok, my rtpproxy doesn't work, so I try it with STUN.
> > When I look at my
> > SIP-messages I get the information, that the audio
> > stream has to go through
> > my public IP... but I don't hear anything (I have
> > the volume on maximum).
> >
> > The Invite comes with this message:
> >
> > v=0.
> > o=- 3330865830 3330865830 IN IP4 xxx.xxx.xxx.xxx.
> > <-- Public IP
> > s=SJphone.
> > c=IN IP4 xxx.xxx.xxx.xxx <--
> > Public IP
> > t=0 0.
> > a=direction:active.
> > m=audio 16482 RTP/AVP 3 8 0 101.
> > a=rtpmap:3 GSM/8000.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-11,16.
> >
> > Doesn't that mean, that the audio-stream has to go
> > through my public IP now?
> > Both sides doesn't hear anything...
> >
> > What's wrong?
> >
> > Sebastian
> >
> >
> >
> > ----- Original Message -----
> > From: "Greger V. Teigre" <greger(a)teigre.com>
> > To: "Sebastian Kühner" <skuehner(a)veraza.com>;
> > <serusers(a)lists.iptel.org>
> > Sent: Wednesday, July 20, 2005 2:24 AM
> > Subject: Re: [Serusers] ACK
> >
> >
> > > Sebastian,
> > > I know many people don't like STUN. However, I
> > have good experiences with
> > > STUN and prefer to use STUN as a "first layer
> > defence." For many NATs I
> > > then avoid the proxying. However, there are some
> > things that can go wrong:
> > > For one, you need to make sure that the STUN
> > server is running correctly
> > on
> > > two ports and two IP addresses. If you for example
> > have a firewall
> > blocking
> > > one port, STUN will give the wrong result. But the
> > biggest problem can be
> > > faulty STUN implementations in the EUCs. They
> > normally behave ok for the
> > > most standard NATs, but there are some
> > non-standard NATs and the EUC's
> > > behavior can be unpredictable. Also, some EUCs
> > try to rewrite the IP:port
> > > even if they are behind a symmetric NAT (or if the
> > STUN server is not
> > > correctly set up, the EUC will conclude with the
> > wrong result).
> > > If you know the clients you are going to use,
> > you can test and limit
> > the
> > > problems and STUN can be a great cost saver! If
> > your gateway supports
> > > active media (direction=active), then you only
> > have IP-2-IP phone calls to
> > > proxy.
> > >
> > > To your question: Sipura has a good implementation
> > of STUN, but has MANY
> > > options for NAT. Your problem is that the RTP and
> > RTCP is not traversing
> > the
> > > NAT to your Sipura. Either you don't force
> > proxying in onreply for OKs,
> > or
> > > something goes wrong. An ngrep trace of the call
> > setup will reveal what
> > the
> > > problem can be.
> > > g-)
> > >
> > > Sebastian Kühner wrote:
> > > > Thank you Nils,
> > > >
> > > > Now it's working better!
> > > >
> > > > The problem that I have now is that I don't hear
> > anything if I call
> > > > from the SIPURA to a Gateway, but the callee is
> > hearing me.
> > > >
> > > > What could be the problem of that one-way
> > conversation? Had anyone of
> > > > you the same problem using a Restricted Cone
> > NAT?
> > > >
> > > > Thanks!
> > > >
> > > > Sebastian
> > > >
> > > >
> > > > ----- Original Message -----
> > > > From: "Nils Ohlmeier" <lists(a)ohlmeier.org>
> > > > To: <serusers(a)lists.iptel.org>
> > > > Cc: "Sebastian Kühner" <skuehner(a)veraza.com>
> > > > Sent: Tuesday, July 19, 2005 3:58 PM
> > > > Subject: Re: [Serusers] ACK
> > > >
> > > >
> > > > Hi,
> > > >
> > > > On Tuesday 19 July 2005 20:53, Sebastian Kühner
> > wrote:
> > > >> I have two phones behind a Port Restricted Cone
> > NAT (both in the same
> > > >> private area) and ser is running with another
> > public IP.
> > > >>
> > > >> I want to call from one of those phone to the
> > other. The call is set
> > > >> up and I can talk, but one Softphone shows me
> > the message: "Waiting
> > > >> acknowledgement..."... and all followed SIP
> > messages don't reach the
> > > >> other phone. I'm using a STUN server.
> > > >>
> > > >> Call from 14@xxx.xxx.xxx.xxx:5060 to
> > 13@xxx.xxx.xxx.xxx:1024:
> > > >>
> > > >> 14 -> ser:
> > > >> ----------
> > > >> IVITE 13@ip.of.ser.xxx@5060 (Contact:
> > 14@192.168.1.101:5060)
> > > >>
> > > >> ser -> 13:
> > > >> ----------
> > > >> INVITE 13@xxx.xxx.xxx.xxx:1024 (Contact:
> > 14@xxx.xxx.xxx.xxx:5060)
> > > >
> > > > sorry but what do you use STUN for if the UAs
> > still use their private
> > > > IPs and
> > > > your SER is re-writting the Contact? If you
> > allready fixing the IP it
> > > > should be easy to fix the port as well.
> > > >
> > > > Conclusion: throw away STUN. In case of
> > symmetric NATs you have to
> > > > find another solution anyway. And you really do
> > not want to try to
> > > > determine the NAT type with STUN.
> > > >
> > > > Nils
> > > >
> > > >> 13 -> ser
> > > >> ---------
> > > >> Trying and ringing (Contact:
> > 13@xxx.xxx.xxx.xxx:5060)
> > > >> (!!!!!!!!!!! <-- wrong port !!!!!!!)
> > > >>
> > > >> 13 -> ser
> > > >> ---------
> > > >> OK (Contact: 13@xxx.xxx.xxx.xxx:5060)
> > > >> (!!!!!!!!!!! <-- wrong port !!!!!!!)
> > > >>
> > > >> ser -> 14
> > > >> ----------
> > > >> OK (Contact: 13@xxx.xxx.xxx.xxx:5060)
> > > >> (!!!!!!!!!!! <-- wrong port !!!!!!!)
> > > >>
> > > >> 14 -> ser
> > > >> ----------
> > > >> ACK 13@xxx.xxx.xxx.xxx:5060
> > > >>
> > > >> ser -> 14
> > > >> ---------
> > > >> ACK 13@xxx.xxx.xxx.xxx:5060
> > > >>
> > > >> 14 -> ser
> > > >> ----------
> > > >> ACK 13@xxx.xxx.xxx.xxx:5060
> > > >>
> > > >> ... and so on... until timeout.
> > > >>
> > > >> Does anybody know what is the problem... or
> > better: the solution?
> > > >>
> > > >> Thanks!
> > > >>
> > > >> Sebastian
> >
> === message truncated ===
>
>
>
>
>
>
>
>
___________________________________________________________________________
> Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
> Téléchargez cette version sur http://fr.messenger.yahoo.com
>
>
Hello,
I have two phones behind a Port Restricted Cone NAT (both in the same
private area) and ser is running with another public IP.
I want to call from one of those phone to the other. The call is set up and
I can talk, but one Softphone shows me the message: "Waiting
acknowledgement..."... and all followed SIP messages don't reach the other
phone. I'm using a STUN server.
Call from 14@xxx.xxx.xxx.xxx:5060 to 13@xxx.xxx.xxx.xxx:1024:
14 -> ser:
----------
IVITE 13@ip.of.ser.xxx@5060 (Contact: 14@192.168.1.101:5060)
ser -> 13:
----------
INVITE 13@xxx.xxx.xxx.xxx:1024 (Contact: 14@xxx.xxx.xxx.xxx:5060)
13 -> ser
---------
Trying and ringing (Contact: 13@xxx.xxx.xxx.xxx:5060)
(!!!!!!!!!!! <-- wrong port !!!!!!!)
13 -> ser
---------
OK (Contact: 13@xxx.xxx.xxx.xxx:5060)
(!!!!!!!!!!! <-- wrong port !!!!!!!)
ser -> 14
----------
OK (Contact: 13@xxx.xxx.xxx.xxx:5060)
(!!!!!!!!!!! <-- wrong port !!!!!!!)
14 -> ser
----------
ACK 13@xxx.xxx.xxx.xxx:5060
ser -> 14
---------
ACK 13@xxx.xxx.xxx.xxx:5060
14 -> ser
----------
ACK 13@xxx.xxx.xxx.xxx:5060
... and so on... until timeout.
Does anybody know what is the problem... or better: the solution?
Thanks!
Sebastian
Hello Naresh.
I guess there is an error in the way you call the authorization for the
INVITE. As far as i know for the REGISTER message (authentication) you need
the statement :
radius_www_authorize
But for the INVITE you need to call "radius_proxy_authorize". This is
what i have in my ser.cfg
if (method=="INVITE") {
if (!radius_proxy_authorize("")) {
proxy_challenge("","1");
break;
};
};
maybe you can try this and tell me how it works.
Good luck
Ricardo Martinez.-
-----Mensaje original-----
De: Naresh Parmar [mailto:naresh_parmar14@yahoo.com]
Enviado el: Miércoles, 20 de Julio de 2005 12:10
Para: Ricardo Martinez; serusers(a)lists.iptel.org
Asunto: RE: [Serusers] Problem authorizing with radius
Hi Ricardo,
We are using freeradius server 0.9.1 and SER 0.9.3. The version of radius
client is radiusclient-ng-0.5.1. The users file in the radius server looks
like as below:
test(a)sip2.zone <mailto:test@sip2.zone> Auth-Type := Digest, User-Password
== "cisco1234"
Reply-Message = "Authenticated",
Sip-Rpid = "1970"
test(a)sip2.zone <mailto:test@sip2.zone> Auth-Type := Accept
Reply-Message = "Authorized",
Sip-Group == "ld"
The radius authentication and authorization parts in the ser.cfg file are
given below:
if (uri=~"^sip:9[0-9]*@") {
if (method=="INVITE"){
if (!radius_www_authorize("")) {
www_challenge("", "1");
break;
}else{
if (radius_is_user_in("Credentials",
"ld")){
forward(192.168.2.101,5060);
break;
}else{
break;
};
};
};
};
And finally the error is as below:
Invalid operator for item Suffix: reverting to '=='
modcall[authorize]: module "preprocess" returns ok
modcall[authorize]: module "chap" returns noop
rlm_eap: No EAP-Message, not doing EAP
modcall[authorize]: module "eap" returns noop
rlm_digest: Converting Digest-Attributes to something sane...
Digest-User-Name = "test"
Digest-Realm = "sip2.zone"
Digest-Nonce = "42de75b2e9e39194a286e8ccd284646ffa14bcc2"
Digest-URI = "sip:94161000@sip2.zone"
Digest-Method = "INVITE"
Digest-QOP = "auth"
Digest-Nonce-Count = "0000000a"
Digest-CNonce = "753F926DB8F5415D8D56EE7816410E33"
rlm_digest: Adding Auth-Type = DIGEST
modcall[authorize]: module "digest" returns ok
rlm_realm: Looking up realm "sip2.zone" for User-Name = " test(a)sip2.zone
<mailto:test@sip2.zone> "
rlm_realm: No such realm "sip2.zone"
modcall[authorize]: module "suffix" returns noop
users: Matched entry test(a)sip2.zone <mailto:test@sip2.zone> at line 226
modcall[authorize]: module "files" returns ok
modcall[authorize]: module "mschap" returns noop
modcall: group authorize returns ok
rad_check_password: Found Auth-Type Digest
auth: type "digest"
modcall: entering group authenticate
A1 = test:sip2.zone:cisco1234
A2 = INVITE:sip:94161000@sip2.zone
KD =
53d3b82970bada131a062103f553b8b8:42de75b2e9e39194a286e8ccd284646ffa14bcc2:00
00000a:753F926DB8F5415D8D56EE7816410E33:auth:18227b358ffe96049a3745eeb449fae
2
modcall[authenticate]: module "digest" returns ok
modcall: group authenticate returns ok
radius_xlat: 'Authenticated'
Login OK: [test(a)sip2.zone/<no User-Password attribute>] (from client proxy
port 5060)
Sending Access-Accept of id 203 to 192.168.2.1:32831
Reply-Message = "Authenticated"
Sip-Rpid = "1970"
Finished request 6
Going to the next request
--- Walking the entire request list ---
Waking up in 6 seconds...
rad_recv: Access-Request packet from host 192.168.2.1:32831, id=204,
length=53
User-Name = "test"
Sip-Group = "ld"
Service-Type = Group-Check
NAS-IP-Address = 192.168.2.1
NAS-Port = 0
modcall: entering group authorize
Invalid operator for item Suffix: reverting to '=='
Invalid operator for item Suffix: reverting to '=='
Invalid operator for item Suffix: reverting to '=='
modcall[authorize]: module "preprocess" returns ok
modcall[authorize]: module "chap" returns noop
rlm_eap: No EAP-Message, not doing EAP
modcall[authorize]: module "eap" returns noop
modcall[authorize]: module "digest" returns noop
rlm_realm: No '@' <mailto:'@'> in User-Name = "test", looking up realm
NULL
rlm_realm: No such realm "NULL"
modcall[authorize]: module "suffix" returns noop
modcall[authorize]: module "files" returns notfound
modcall[authorize]: module "mschap" returns noop
modcall: group authorize returns ok
auth: No authenticate method (Auth-Type) configuration found for the
request: Rejecting the user
auth: Failed to validate the user.
Login incorrect: [test/<no User-Password attribute>] (from client proxy port
0)
Delaying request 7 for 1 seconds
Finished request 7
Going to the next request
Waking up in 6 seconds...
As you can see from the above configuration, the authentication works
perfect, its only in the authorization where it fails. Also can you please
let me know about the accounting configuration??
Thanks a lot..
Naresh
Ricardo Martinez <rmartinez(a)redvoiss.net> wrote:
Hello Naresh
I have authentication, authorization and accounting (AAA) through radius
working fine. What radius server are you using?, can you send us more
information about the configuration?
Cheers,
Ricardo.-
-----Mensaje original-----
De: Naresh Parmar [mailto:naresh_parmar14@yahoo.com]
Enviado el: Miércoles, 20 de Julio de 2005 10:37
Para: serusers(a)lists.iptel.org
Asunto: [Serusers] Problem authorizing with radius
hi friends,
I am having problems while authorizing with the radius server. I am using
the same configuration as mentioned in the radius-howto. Authentication
works perfect as I am able to authenticate using the radius server. However
while authorizing against the radius server to make a call I get the
following error:
auth: No authenticate method (Auth-Type) configuration found for the user
request: Rejecting the user
auth: Failed to validate the user.
Delaying request 2 for 1 seconds
Finished request 2
When I authorize against the mysql database, it works fine. Any clue???
Best Regards,
Naresh
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
On Jul 20, 2005 at 18:37, Dan Pascu <dan(a)ag-projects.com> wrote:
> On Wednesday 20 July 2005 18:15, Ricardo Martinez wrote:
> > 2.- I'm downloading SER from CVS with the command :
> >
> > cvs -z3 -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co
> > -rrel_0_9_0 sip_router
> >
> > To download the changes you introduce in mediaproxy.c i need to run
> > the same command ????
>
> You can do this, but you don't have to download the whole ser cvs tree again.
>
> Instead, you can go inside the previously checked-out source tree, and type
>
> cd sip_router/
> cvs update -Pd
>
> watch for the output and there you should see a line with:
>
> P modules/mediaproxy/mediaproxy.c
>
> (the P in front of the file means that your copy was updated with the changes
> retrieved from the cvs server)
>
> >
> > 3.- How do i compile SER without optimization?
>
> One simple way to do this is to compile ser with:
>
> make CC_EXTRA_OPTS="-g -O0"
>
> There may be other ways to do the same, someone more intimate with ser build
> internals may be able to point you to them.
make mode=debug
will do the same thing + define EXTRA_DEBUG (=> very verbose).
Andrei