[OpenSIPS-Users] Problem NAT RTPproxy

[Bogdan-Andrei Iancu]

Hi Magnus,

attaching cfg files is useless, as no one will debug the script, but we 
will help you to debug your script.

So, for the non-working case (PSTN to SIP) does your script force 
RTPproxy in INVITE and 200 OK ?

Regards,
Bogdan

On 03/29/2012 01:52 AM, magnusadilsom at gmail.com wrote:
> I have phones (some behind NAT) connecting to Opensips server an 
> Asterisk and an rtpproxy as seen below:
>
> rtpproxy started with
> ps -aux | grep rtpproxy
> root     15666  0.0  0.0  14472   920 ?        Ssl  Mar23   0:05 
> ./rtpproxy -F -l 189.254.2.19 -s udp:* 7890 -d DBUG LOG_LOCAL3
>
>
>
> UAC1 username = 
> 100------------Firewall/router--------------------Opensips 
> 1.7---------- RTP PROXY------------Asterisk 1.6
> 192.168.1.10                    192.168.1.1                    
> 65.254.63.212          189.254.2.19           190.61.201.89
>                       external ip dinamic 169.254.2.2
>
>
> - Calls between UAC are OK (both SIP and RTP).
> - Calls UAC for PSTN is OK.
> - Did numbers is received in Asterisk, and destination for UAC 
> registered in opensips, but no work audio .
> (EX User call cellphone for DID 54115368566, call is received in 
> asterisk, and destination for user 100, registered in opensips)
>
>
>
>
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-- 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

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