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, [email protected] 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, [email protected] 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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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

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