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 $rd to asterisk ip and then route
Sent from my iPhone
On 22 Jan 2014, at 12:12, "Александр Пучков"<[email protected]> wrote:
Ovidiu, thanks for your answer.
There is some misunderstanding.
I'm probably not very well formulated goal I want to achieve.
Try to describe in more detail:
UAC (1): OpenSIPs registered as a UAC on Asterisk (sip:100@asterisk).
UAC (2): UAC (2) is registered to OpenSIPs (sip:100@opensips).
Everything works well when there is a call : PSTN --> Asterisk --> UAC (1) -->
UAC (2).
I would like to learn how to make a call : UAC (2) ---> UAC (1) --> Asterisk
--> PSTN
I tried to replace roughly FROM header for OUTGOING calls with UAC (2):
if (db_is_user_in ("$ 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<[email protected]>:
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
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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
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Users mailing list
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