[OpenSIPS-Users] Problem NAT RTPproxy

[Bogdan-Andrei Iancu]

Well, you know, one is what we want to do , another we actually get.

I was rather asking if, making a sip capture (with ngrep) you see in 
your call the RTPproxy insertion - check it in traffic, not in script.

Regards,
Bogdan

On 04/02/2012 10:05 PM, magnusadilsom at gmail.com wrote:
> hi, yes, rtpproxy is active in invite 200
>
> onreply_route[3] {
>     if ((isflagset(5) || isbflagset(0)) && status =~ 
> "(183)|(2[0-9][0-9])" && has_body("application/sdp")) {
>         if (rtpproxy_answer()) {
>             log("L_INFO: rtpproxy_answer NAT");
>         }
>     }
>     if (!subst_uri('/(sip:.*);nat=yes/\1/')) {
>         search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
>     }
>     exit;
> }
>
>
> But i'm implemented this in invite route
>
> if (is_method("INVITE") {
>      if ($si == "IP ASTERISK" && is_method("INVITE")) {
>             fix_nated_contact();
>             fix_nated_sdp("1");
>             xlog("L_INFO", "NAT detected3 PSTN for SIP");
>             setflag(5);
>             return;
>         }
>   }
>
> and worked, but I think it is not correct
>
> tansk
>
>
> Bogdan-Andrei Iancu wrote:
>> 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)
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> -- 
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>
>
> _______________________________________________
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-- 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

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