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

[oso che bol]

Dear Bogdan,

Thanks you for your time to help me on this.

I got it solved.

Btw, what is the way we usually do to process 404 User (User not found, and
not online)?

Thanks and Regards,
Loi Ngo

On Fri, Apr 10, 2009 at 12:19 AM, oso che bol <ndlgroup1 at gmail.com> wrote:

> 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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