[OpenSIPS-Users] Interaction between OpenSIPS as UAC and real UAC

[Александр Пучков]

11.02.2014 19:44, Bogdan-Andrei Iancu пишет:
> Hello,
>
> Please keep the list CC'ed all the time !
>
> You did no inserted the topo hiding triggering in the write place - 
> you need to do it only for initial INVITEs ; for sequential requests 
> you need the be sure to invoke the match_dialog() function. See:
> http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295144
>
> It will be helpful to provide more info on what "it is not working" - 
> like do you see any changes on the INVITE sent out by OpenSIPS, any 
> script errors, etc
I 'm trying to use your proposed function, but ran into a problem - I 
have not triggered the authorization and I can not make an outgoing call .

After making some changes :

     if (!db_is_user_in("$fu", "asterisk")) # +++ I ADDED THIS STRING
     if (!(method=="REGISTER") && from_uri==myself) /*НЕ multidomain 
версия*/

     {

         if(!check_source_address("0")){
             if (!proxy_authorize("", "subscriber")) {


                 proxy_challenge("", "0");
                 exit;
             }
                 if (!db_check_from()) {
                     sl_send_reply("403","Forbidden auth ID");
                     exit;
                 }
             consume_credentials();
         }
     }

I like making progress , but the analysis of messages using the program 
"wireshark", I saw that I had not sent a request "ACK" to the asterisk. 
The challenge does not pass.

Briefly remind you that I 'm trying to achieve :
I would like to make opensips acted as UAC to asterisk , but it was a 
server for opensips UAC. This will allow me to introduce UAC, registered 
in OpenSIPS like an extension on the asterisk.

Please, write a little piece of code for deciding my problem.

My configuration file:

http://pastebin.com/39vV4eYT

> Hello,
>
> Use the dialog based topology hiding
> http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id296001
>
> when forwarding the INVITE to Asterisk (in OpenSIPS).
>> Thank you.
>>
>> I tried it like this code fragment:
>>
>> #--8<----------------------------------------------------------------------------------
>>
>> route {
>>
>>
>>     if (!mf_process_maxfwd_header("10")) {
>>         sl_send_reply("483","Too Many Hops");
>>         exit;
>>     }
>>
>>     #force_rport();
>>
>>     if(avp_db_load("$fu","$avp(trace)")) {
>>         $avp(traceuser)=$fu;
>>         setflag(22);
>>         sip_trace();
>>         xlog("L_INFO","User $fu being traced");
>>     }
>> #...................................................................
>>     if (db_is_user_in("$fu", "asterisk"))
>>     {
>>         if(!has_totag() && is_method("INVITE")) {
>>         topology_hiding();
>>         }
>>     }
>>
>>     if (has_totag()) {
>>     ...
>>     }
>>
>>     ...
>> #--8<----------------------------------------------------------------------------------
>>
>> But it not worked :( Perhaps, this code is wrong.
>>
>> Could you indicate where the error? Could you indicate where the 
>> error? Or do I need to follow the documentation on the function 
>> topology_hiding()?
>>> On 11.02.2014 10:12, Александр Пучков wrote:
>>>>
>>>> 10.02.2014 13:24, Bogdan-Andrei Iancu пишет:
>>>>> Hello,
>>>>>
>>>>> In this scenario:
>>>>>
>>>>>     Astеrisk <-- 3 -- OpenSIPS <-- 4 -- UAC
>>>>>
>>>>> What SIP requests the UAC is sending ? REGISTER ? INVITES ?
>>>> Hello!
>>>>
>>>> UAC registered to Opensips, Opensips registered as UAC on Asterisk. 
>>>> When the INVITE request comes from the UAC, Opensips need to sent 
>>>> INVITE to Asterisk as UAC.
>>>>
>>>> Note that the UAC does not know about Asterisk and Asterisk does 
>>>> not know about UAC. Asterisk know about only Opensips.
>>>>
>>>> It is necessary to any UAC registered on Opensips could imagine how 
>>>> extension on Asterisk, without changing the configuration of UAC.
>>>>
>>>> Thank you.
>>>>
>>>>>
>>>>>
>>>>> Regards,
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developer
>>>>> http://www.opensips-solutions.com
>>>>> On 20.01.2014 13:05, Александр Пучков wrote:
>>>>>> Hi!
>>>>>>
>>>>>> OpenSIPS version 1.8.
>>>>>>
>>>>>> We have the following diagram:
>>>>>>
>>>>>> PSTN        <-- 1 --> OpenSIPS    <-- 2 --> Astеrisk    <-- 3 --> UAC
>>>>>> 192.168.1.1 <-- 1 --> 192.168.1.2 <-- 2 --> 192.168.1.3 <-- 3 --> 
>>>>>> 192.168.1.*
>>>>>>
>>>>>> I would like to make the following scheme:
>>>>>>
>>>>>> PSTN        <-- 1 --> OpenSIPS    <-- 2 --> Astеrisk    <-- 3 --> 
>>>>>> OpenSIPS <-- 4 --> UAC
>>>>>> 192.168.1.1 <-- 1 --> 192.168.1.2 <-- 2 --> 192.168.1.3 <-- 3 --> 
>>>>>> 192.168.1.2 <-- 4 --> 192.168.1.*
>>>>>>
>>>>>> Interestingly the following schema fragment:
>>>>>> Astеrisk    <-- 3 --> OpenSIPS <-- 4 --> UAC
>>>>>>
>>>>>> Here OpenSIPS need to increase control over the services for UAC. 
>>>>>> UAC should not know about the existence of Asterisk and UAC must 
>>>>>> be registered on the server OpenSIPS, and any SIP request 
>>>>>> OpenSIPS should redirect to Asterisk.
>>>>>>
>>>>>> I tried to use the module in UAC_REGISTRANT OpenSIPS, it works 
>>>>>> fine in the direction:
>>>>>>
>>>>>> Astеrisk -- 3 --> OpenSIPS -- 4 --> UAC
>>>>>>
>>>>>> But how to implement the scheme in the direction:
>>>>>>
>>>>>> Astеrisk <-- 3 -- OpenSIPS <-- 4 -- UAC
>>>>>>
>>>>>> I do not really imagine. Please tell me how it can be implemented.
>>>>>>
>>>>>> Thank!
>>>>>>
>>>>>>
>>>>>>



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Александр Пучков,
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