[OpenSIPS-Users] [OPENSIPS] How to route calls out Openser to Voicemail GW with Right RURI

[oso che bol]

Dear all,

Do we need to involve RTPPROXY when Opensips sent out INVITE to Asterisk
Voicemail? in this case - involve RTPPROXY, Asterisk and UA are both UA
Client but Asterisk have Public IP and UA have private IP?

Please help to point me out.

Thanks,
-LN

On Thu, Apr 9, 2009 at 7:24 PM, oso che bol <ndlgroup1 at gmail.com> wrote:

> Dear Bogdan,
>
> Could you please help how to set RTPPROXY for 200OK reply?
>
> Why onreply_route[1] -->
> onreply_route[1] {
>     append_hf("P-hint: HTK: On_reply_route[1] processing\r\n");
>     if (status=~"(180)|(183)|2[0-9][
> 0-9]") {
>         if (nat_uac_test("1")) {
>             fix_nated_contact();
>         }
>         force_rtp_proxy();
>     }
> }
>
> --> do not apply for 200OK from ASTERISK_IP to OPENSIPS_IP?
>
> Thanks and Regards,
> -LN
>
>
> On Thu, Apr 9, 2009 at 6:17 PM, Bogdan-Andrei Iancu <
> bogdan at voice-system.ro> wrote:
>
>> Hi,
>>
>> yes, Asterisk will send media to RTPproxy IP and not to the UAC. Looking
>> at the first trace you send, I see that rtpporxy was not set for the 200 OK
>> reply (only for INVITE request) - because of this, the media is broken.
>>
>> Regards,
>> Bogdan
>>
>>
>>
>
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