[OpenSIPS-Users] How to use UAC_REGISTRANT module?

[Александр Пучков]

Thanks, Jordan!

I tried the following structure:
---8<-----------------------------------------------------------------------------------------------------------------------

if (db_is_user_in("$fu", "asterisk"))
     {
         $rd=$var(asterisk_ip);
     }

     if(!dp_translate("0","$rU/$rU")){
         send_reply("420", "Invalid Destination");
         exit;
     }

     if (db_is_user_in("$fu", "asterisk"))
         {
         $avp(dest)="media";
         }

     xlog("$avp(dest)");

     if ($avp(dest)=="usrloc") {
         route(m4_USRLOC);
     }

     if ($avp(dest)=="pstn") {
             route(m4_PSTN);
     }

     # route to Asterisk
     if ($avp(dest)=="media") {
         route(m4_MEDIA);
     }

---8<-----------------------------------------------------------------------------------------------------------------------
Unfortunately, it did not help.

In the picture chart calls for this approach:

http://tinypic.com/r/27zb0qb/5

*.*.3.1 - UAC (2)
*.*.1.4 - OpenSIPS [as UAC(1)] - extension on Asterisk
*.*.1.5 - Asterisk

What else can I do?


22.01.2014 14:16, Justin Zondagh - MWEB пишет:
> Set $rd to asterisk ip and then route
>
> Sent from my iPhone
>
>> On 22 Jan 2014, at 12:12, "Александр Пучков"<mailex at poig.ru>  wrote:
>>
>> Ovidiu, thanks for your answer.
>>
>> There is some misunderstanding.
>> I'm probably not very well formulated goal I want to achieve.
>> Try to describe in more detail:
>>
>> UAC (1): OpenSIPs registered as a UAC on Asterisk (sip:100 at asterisk).
>> UAC (2): UAC (2) is registered to OpenSIPs (sip:100 at opensips).
>>
>> Everything works well when there is a call : PSTN --> Asterisk --> UAC (1) --> UAC (2).
>>
>> I would like to learn how to make a call : UAC (2) ---> UAC (1) --> Asterisk --> PSTN
>>
>> I tried to replace roughly FROM header for OUTGOING calls with UAC (2):
>>
>> if (db_is_user_in ("$ fu", "asterisk"))
>>     {
>>     $var(masking_number) = "sip:" + $fU + "@" + $var(asterisk_ip);
>>     uac_replace_from ("", "$var(masking_number)");
>>     }
>> but it did not help.
>>
>> Why it need: the client (UAC) registered to OpenSIPS, could provide services Asterisk, without reconfiguring the client equipment.
>>
>> It would be great if a piece of code, as I'm no guru in Opensips.
>>
>> Thanks!
>>
>> 21.01.2014 20:41, Ovidiu Sas пишет:
>>> When the call is coming in for UAC (1) you need to perform a lookup for UAC (2).
>>> See:http://www.opensips.org/html/docs/modules/devel/registrar#id294037
>>> You need to set the AOR to UAC (2).
>>>
>>> Regards,
>>> Ovidiu Sas
>>>
>>> 2014/1/21 Александр Пучков<mailex at poig.ru>:
>>>> Hi!
>>>>
>>>> I use the module UAC_REGISTRANT. OpenSIPS registered on Asterisk as UAC (1).
>>>> There are UAC (2) that is registered on OpenSIPS. When calling with Asterisk
>>>> to UAC (1) there is a call UAC (2)
>>>> as UAC (2) is registered in the "binding_URI", table "registrant".
>>>>
>>>> How to make an outgoing call UAC (2) a call came from the UAC (1)?
>>>>
>>>> Thank!
>>>>
>>>>
>>>> Александр Пучков,
>>>> Системный администратор,
>>>> Тел.: +7(496) 569-24-24 доб.тел. 255;
>>>> ООО "ПОИГ" (Интернет-провайдер г. Щелково)
>>>> Факс: +7(496) 569-24-24 доб. 103;
>>>> Адрес: 141108, М.О., г.Щелково, Пролетарский пр., д.11;
>>>> WEB:http://www.schelkovo-net.ru
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> --
>> С уважением,
>> Александр Пучков,
>> Системный администратор,
>> Тел.:      +7(496) 569-24-24 доб.тел. 255;
>> ООО "ПОИГ" (Интернет-провайдер г. Щелково)
>> Факс:      +7(496) 569-24-24 доб. 103;
>> Адрес:     141108, М.О., г.Щелково, Пролетарский пр., д.11;
>> WEB:http://www.schelkovo-net.ru
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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