[SR-Users] Unknown caller gets online user's identity
Cibin Paul
paul_cibin at me.com
Thu Jul 17 09:48:07 CEST 2014
Hello,
Try allow allowguest=no in sip.conf [general] context and create a peer for kamailio in sip.comf
Regards
Cibin
On 17-Jul-2014, at 12:52 pm, g.aloitus at gmail.com wrote:
> Hello,
>
> There is a message "Possible Security issue with Kamailio - Asterisk Realtime integration" in Asterisk users mailing list:
>
> http://lists.digium.com/pipermail/asterisk-users/2013-February/277633.html
>
> I think the problem I have is somewhat similar.
>
> Should I suppose that there is a security risk in Kamailio - Asterisk realtime integration, and if this is a case what I can do to eliminate this risk?
>
> Best,
>
> Teijo
>
> 16.7.2014 9:44, g.aloitus at gmail.com kirjoitti:
>> Hello,
>>
>> Has anybody any solution or suggestion?
>>
>> If I for example launch MicroSIP (no doubt it could be some other SIP
>> client), and simply call:
>>
>> sip:some_extension at my.public.ip.address
>>
>> call is established, if there is online user/users. Naturally this
>> incoming call should be handled by Asterisk in context where I have
>> defined unauthorized calls are handled, but in stead, the call goes
>> online user's context.
>>
>> To get this situation I don't need to define any account information in
>> MicroSIP.
>>
>> I have not set passwords for users in Asterisk to avoid double
>> authorization. May this cause the behavior? I have not set default user
>> or from user in my peer definitions. I am not registering Kamailio to
>> Asterisk - I mean I have no peer definition for Kamailio in sip.conf.
>>
>> I do not know what direction to go to. I would be happy, if I should not
>> go to the trial and error path so any help is welcome.
>>
>> Thanks in advance,
>>
>> Teijo
>>
>>
>> 14.7.2014 9:06, g.aloitus at gmail.com kirjoitti:
>>> Hello,
>>>
>>> If one places call, and tell that "my from domain is your Kamailio's
>>> IP", call is established, because Asterisk accepts requests from
>>> Kamailio. One problem is that it's unpredictable in this case what is
>>> the context where thiskind of call is handled by Asterisk.
>>>
>>> This situation requires that I change something in my setup. If I decide
>>> accept calls only from my users, I suppose that it can be quite easily
>>> done by modifying if statement referred below or at least by applying
>>> instructions found here:
>>>
>>> http://www.kamailio.org/dokuwiki/doku.php/examples:restrict-calls-to-registered-users
>>>
>>>
>>>
>>> However, I'm somewhat unsure what should I do, if I decide to accept
>>> calls from any caller - not only from my users.
>>>
>>> Best,
>>>
>>> Teijo
>>>
>>> 12.7.2014 19:36, Muhammad Shahzad kirjoitti:
>>>> Well, this
>>>>
>>>> *if (from_uri!=myself && uri!=myself)*
>>>>
>>>> Means neither source nor destination is our user. Which implies that
>>>> if our
>>>> domain is A, then call from domain "B to C" is not possible. However,
>>>> calls
>>>> from "B or C to A" and "A to B or C" are possible. That is way an
>>>> unauthorized user gets passed and reaches asterisk. Asterisk accepts it
>>>> since call is coming from kamailio and tries to route it back to
>>>> kamailio,
>>>> where kamailio finds user online and thus it goes through.
>>>>
>>>> You should really break down this,
>>>>
>>>> *if (from_uri!=myself && uri!=myself)*
>>>>
>>>> into something like this for clarity,
>>>>
>>>>
>>>> *if (from_uri!=myself) { *
>>>> * if (uri!=myself) {*
>>>> * # neither source nor destination is our user*
>>>> * } else {*
>>>> * # source is not our user but destination is our user*
>>>> * };*
>>>> *} else {*
>>>> * if (uri!=myself) {*
>>>> * # source is our user but destination is not our user*
>>>> * } else {*
>>>> * # both source and destination are our users*
>>>> * };*
>>>> *};*
>>>>
>>>> Hope this helps.
>>>>
>>>> Thank you.
>>>>
>>>>
>>>>
>>>>
>>>> On Fri, Jul 11, 2014 at 5:36 PM, <g.aloitus at gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I'm using Kamailio version 4.1.4+precise (amd64).
>>>>>
>>>>> I have followed "Kamailio 4.0.x and Asterisk 11.3.0 Realtime
>>>>> Integration
>>>>> using Asterisk Database" (http://kb.asipto.com/
>>>>> asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). One main
>>>>> difference in my setup compared to that one is that I continued use of
>>>>> Kamailio's database.
>>>>>
>>>>> The problem is as follows:
>>>>>
>>>>> I decided to put Kamailio and through it Asterisk reachable from
>>>>> internet.
>>>>> I have tried to configure Asterisk so that only calls of registered
>>>>> users
>>>>> would be possible, and they could only call to other registered
>>>>> users or
>>>>> conference rooms and echo test number.
>>>>>
>>>>> Then I took the following steps:
>>>>>
>>>>> I ensured that there was no online users with kamctl online. Then I
>>>>> launched MicroSIP (www.microsip.org), but I did not defined account, I
>>>>> simply set the protocol to tls and media encryption to mandatory,
>>>>> because
>>>>> I'm using these.
>>>>>
>>>>> I called to extension with xxx at my.public.ip.address (where xxx is
>>>>> extension) getting "unauthorized". And that was what I wanted.
>>>>>
>>>>> But if there is online users, calls go through, and incoming call is
>>>>> coming from Asterisk (in syslog I can find out that src_user=asterisk).
>>>>>
>>>>> Kamailio and Asterisk are listening the same IP address, but different
>>>>> port. I have refused connections to the Asterisk's port with iptables.
>>>>>
>>>>> I have defined my public IP address as domain in sip.conf. There is
>>>>> also
>>>>> other domain defined which corresponds to users' domain I am using in
>>>>> Kamailio's database.
>>>>>
>>>>> In kamailio.cfg there is if statement which prevents Kamailio not to be
>>>>> open relay:
>>>>>
>>>>> if (from_uri!=myself && uri!=myself)
>>>>> ...
>>>>>
>>>>> If I change this for example:
>>>>>
>>>>> if (from_uri!=myself || uri!=myself)
>>>>>
>>>>> I get what I want this time: no calls from outside, but I somewhat
>>>>> think
>>>>> that this is not a final solution.
>>>>>
>>>>> I have not found from log files such information which would have
>>>>> helped
>>>>> me. I have not yet investigated this problem so much that I could
>>>>> tell the
>>>>> logic behind the selection of online user's identity which is used.
>>>>> However, if I make a call to conference room I notice that Asterisk is
>>>>> thinking that one of online users has joined the conference.
>>>>>
>>>>> If I can recall correctly, I started with Kamailio version 3.2, and
>>>>> integrated it with Asterisk 11 (currently 11.10.2). Is there something
>>>>> which has changed in Kamailio, but what I have not changed in my setup
>>>>> which could explain this.
>>>>>
>>>>> Best,
>>>>>
>>>>> Teijo
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>>
>>>>>
>>>>> Tämä viestin rungon osa siirretään pyydettäessä.
>
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