Hallo,
i'm using radius+mediaproxy and since I moved my sip proxy on a public ip I have the folowing problem. I log in regulary and make my firs call with no problem. But When I hung up, and after few minutes call again, my proxy wont authorise me. I get a message that I need to authorise with proxy. During that my FRERADIUS is recieving requests for authorisation. The second i get rejected i call again and it connects me.
I tried lowering nat_ping intervals but no succes.
my Invite snippet is a common one:
sl_send_reply("100", "Trying");
if (!radius_proxy_authorize("")) {
proxy_challenge("","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
Im still seeking my head how Vonage and others can do that? does Iptel
have any solution for that, if yes how much may that cost?
> Depends of what you want to do...
> For a real time billing for prepaid, you will need a b2bua, Ser will not
give you the possibility to do that(I think).
> For postpaid, Ser is enough.
>
> Olivier
>
> -----Message d'origine-----
> De : Waldo Rubinstein [mailto:waldo@trianet.net]
> Envoyé : mardi 7 février 2006 14:15
> À : Olivier Taylor
> Cc : lists(a)cingerr.com; serusers(a)lists.iptel.org
> Objet : Re: RE : RE : RE : [Serusers] SER + Radius + B2Bua
>
>
> Would this also be a viable solution for terminating calls to other SIP
gateways instead of just PSTN or would SER-only be sufficient for that?
>
> Thanks,
> Waldo
>
> On Feb 7, 2006, at 7:54 AM, Olivier Taylor wrote:
>
>> Well we use it for pstn calls and have canreinvites=yes, then
>> Asterisk is
>> most of the time outside of the media path.
>> The only job of this server is the billing and routing calls to pstn.
Furthermore, the number of simultaneous calls Asterisk can manage
depends on
>> the hardware u will use, transcoding and so on.
>> We had more than 50 simultaneous calls on a p4 with 2Gb ram, I
>> really don't
>> (yet) know what will be the limit.
>>
>> Olivier
>>
>> -----Message d'origine-----
>> De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la
>> part de lists(a)cingerr.com
>> Envoyé : mardi 7 février 2006 13:42
>> À : serusers(a)lists.iptel.org
>> Objet : Re: RE : RE : [Serusers] SER + Radius + B2Bua
>>
>>
>> Dear Olivier,
>>
>> Im interesting to know how asterisk behave in heavy load? how many
calls at
>> a moment is your asterisk handling?
>>
>>> Hello again,
>>>
>>> Just try asterisk b2bua, it works fine for us.
>>> http://developer.berlios.de/projects/b2bua/
>>>
>>> The developper is very kind, subscribe to his mailing list.
>>>
>>> Good luck,
>>>
>>> Olivier
>>>
>>>
>>> -----Message d'origine-----
>>> De : serusers-bounces(a)lists.iptel.org [mailto:serusers-
>>> bounces(a)iptel.org] De
>>> la part de lists(a)cingerr.com Envoyé : mardi 7 février 2006 11:54 À :
serusers(a)lists.iptel.org
>>> Objet : Re: RE : [Serusers] SER + Radius + B2Bua
>>>
>>>
>>> Hello Olivier,
>>>
>>> I have about two week spending lot of time trying to make it work but
no luck, so thank you for saving my time.
>>>
>>> Is there any known way to make prepaid/postpaid billing with SER?
>>>
>>>
>>> Best Regards
>>> Hekuran,
>>>
>>>
>>>
>>>> Just a question, do you use Vovida b2bua?
>>>>
>>>> If yes, forget it, development has been stoped 2 years ago and it's
very incomplete. For example, Vovida b2bua is unable to manage
re-invites.
>>>>
>>>> Just to avoid wasted time for you
>>>>
>>>> Olivier
>>>>
>>>> -----Message d'origine-----
>>>> De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
De la part de lists(a)cingerr.com Envoyé : mardi 7 février 2006 10:42 À
>>>> : serusers(a)lists.iptel.org Objet : [Serusers] SER + Radius + B2Bua
>>>>
>>>>
>>>> Hello,
>>>>
>>>> Im trying to make a combination of radius and b2bua to make some
prepaid/postpaid billing. Till now I have managed to make SER talk to
>>>> b2bua and b2bua talk to Radius, but Im not clear yet where does ser
or radius stores the call detail records (radacct table is always
empty).
>>>>
>>>> Also my question about prepaid: is there any database schema that
provides tables where I store user credit or so (as it says that
b2bua have bundled billing system) or should it be done using radius
attributes? If so are there any extra attributes that I have to add
in dictionary?
>>>>
>>>> Question about ser: When a user registers with radius it
>>>> registers as
>>>> username@domain (100(a)some-domain.com) but when I try to call, SER
send the call to b2bua using only username ([100]) so the user can
not be found. is there any way to register the user only by
>>>> username?
>>>>
>>>> Best Regards
>>>> Hekuran
>>>>
>>>> _______________________________________________
>>>> Serusers mailing list
>>>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Serusers mailing list
>>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>>>
>>>
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>
>
Hello,
Im trying to make a combination of radius and b2bua to make some
prepaid/postpaid billing. Till now I have managed to make SER talk to
b2bua and b2bua talk to Radius, but Im not clear yet where does ser or
radius stores the call detail records (radacct table is always empty).
Also my question about prepaid: is there any database schema that provides
tables where I store user credit or so (as it says that b2bua have bundled
billing system) or should it be done using radius attributes? If so are
there any extra attributes that I have to add in dictionary?
Question about ser: When a user registers with radius it registers as
username@domain (100(a)some-domain.com) but when I try to call, SER send the
call to b2bua using only username ([100]) so the user can not be found. is
there any way to register the user only by username?
Best Regards
Hekuran
Hi,
Just a question regarding the received-Parameter in usrloc:
Is it ALWAYS in the format "sip:<ip>:<port>" or may there also be
variations like "sips:<ip>:<port>"? What about TCP-connections? Could
there be a transport identifier?
Thanks,
Andy
Dear all,
wen a sip client(minisip) tries to connect with OpenSER with TLS
enabled on both sides, I get the following error
ipMessageTransport: sendMessage: creating new socket
Creating new SSL_CTX
SSL connect: Error in system call.
Could not get server certificate
SipMessageTransport: sendMessage: exception thrown!
SipMessageTransport: sendMessage: exception thrown!
SipMessageTransport: sendMessage: exception thrown!
SipMessageTransport: sendMessage: exception thrown!
SipMessageTransport: sendMessage: exception thrown!
SipMessageTransport: sendMessage: exception thrown!
SipMessageTransport: sendMessage: exception thrown!
regarding the certificates, using the scripts given in OpenSER, I
created a rootCA and with the same rootCA I created the following for server
and client
Server:
server-cert
server-privkey
server-calist
Client:
client-cert
client-privkey
client-calist
and loaded the sever certs in the server config file and the client
certs in the client config file.....
wat am I missing here, why is it not able to get the server
certificate ? I am posting the openser.cfg here. kindly guide me
thanks and regards,
Pjothi
____________________________________________________________
openser.cfg
# $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/openser_fifo"
#
# uncomment the following lines for TLS support
disable_tls = 0
listen = tls: 192.168.0.4:5061
tls_verify = 1
tls_require_certificate = 0
tls_method = SSLv23
tls_certificate = "/usr/local/etc/openser/user/user- cert.pem"
tls_private_key = "/usr/local/etc/openser/user/user- privkey.pem"
tls_ca_list = "/usr/local/etc/openser/user/user- calist.pem"
--------------------------------------------------------------------------------------------------------------
hi all!
can anyone help me with my little problem? i can't make my budgetone register to my ser box? i've used xlite softphones and they're able to call to/from each other. and they're quite fast loggin in. but when i used my budgetone, to test to my ser box, it won't register though i used the same configs on my softphones with the grandstream phone. is there any issue with ser and grandstream phones?
ryan
__________________________________________________
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Hi,
As many of you may know, we are undertaking several tests in order to test
the interoperability between several PBX IP from different vendors. Until
now, we were trusting that the VoIP IP PBX were good enough to be
interconnected directly, however, one of the vendors have presented the
"SBC"
concept.
The "SBC" (Session Border Controller) is not a new concept since we were
using it anyway when we setup a (Asterisk+SER+SIP Proxy) Box to handle the
"on-net dialout" calls.
I'm now overwhelmed with the amount of SBCs that are suggested by the
vendors
to implement a solution.
(http://www.juniper.net/solutions/literature/solutionbriefs/351085.pdf)
Can anyone drop me some lines about this? I urgently need some feedback on
this.
Thanks!
Joao Pereira
www.fccn.pt
Hi all,
I have found a number of calling card applications for Asterisk which have
been modified to support a prepaid sip service for origination/termination.
Does something like this exist for SER?
TIA,
Max
--
Max Clark
http://www.clarksys.com
hello all,
I am able to route calls from asterisk to ser and vice versa. is it
possible to make simultaneous calls from asterisk clients to ser
clients without getting a 486 busy.
-AA
