[SR-Users] Unknown caller gets online user's identity

Sat Jul 19 13:04:20 CEST 2014

Hello,

Well, this is still problem for me.

Best,

Teijo

17.7.2014 11:22, g.aloitus at gmail.com kirjoitti:
> Hello,
>
> I have:
>
> allowguest=no
> contactpermit=kamailio.ip.addr.ess
>
> I also have tried the approach that I have peer kamailio, but then all
> calls seems to go to to the context defined for kamailio peer. I do not
> know how I could in that case handle individual calls - for example
> determine if given phone can call to given number or not.
>
> Best,
>
> Teijo
>
> 17.7.2014 10:48, Cibin Paul kirjoitti:
>> Hello,
>>
>> Try allow* allowguest=no *in sip.conf [general] context and create a
>> peer for kamailio in sip.comf
>>
>>
>> Regards
>> Cibin
>>
>>
>>
>> 17.7.2014 10:22, g.aloitus at gmail.com kirjoitti:
 >>>
>>> 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> TÃ¤mÃ¤ viestin rungon osa siirretÃ¤Ã¤n pyydettÃ¤essÃ¤.
>