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hi, yes, rtpproxy is active in invite 200<br>
<br>
onreply_route[3] {<br>
if ((isflagset(5) || isbflagset(0)) && status =~
"(183)|(2[0-9][0-9])" && has_body("application/sdp")) {<br>
if (rtpproxy_answer()) {<br>
log("L_INFO: rtpproxy_answer NAT");<br>
}<br>
}<br>
if (!subst_uri('/(<a class="moz-txt-link-freetext" href="sip:.*">sip:.*</a>);nat=yes/\1/'))
{<br>
search_append('Contact:.*sip:[^>[:cntrl:]]*',
';nat=yes');<br>
}<br>
exit;<br>
}<br>
<br>
<br>
But i'm implemented this in invite route<br>
<br>
if (is_method("INVITE") {<br>
if ($si == "IP ASTERISK" && is_method("INVITE")) {<br>
fix_nated_contact();<br>
fix_nated_sdp("1");<br>
xlog("L_INFO", "NAT detected3 PSTN for SIP");<br>
setflag(5);<br>
return;<br>
}<br>
}<br>
<br>
<span id="result_box" class="short_text" lang="en"><span class="hps">and</span>
<span class="hps">worked, </span></span><span id="result_box"
class="short_text" lang="en"><span class="hps">but I think it</span>
<span class="hps">is not</span> <span class="hps">correct</span></span><br>
<br>
tansk<br>
<br>
<br>
Bogdan-Andrei Iancu wrote:
<blockquote cite="mid:4F79D798.1080707@opensips.org" type="cite">
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Hi Magnus,<br>
<br>
attaching cfg files is useless, as no one will debug the script,
but we will help you to debug your script.<br>
<br>
So, for the non-working case (PSTN to SIP) does your script force
RTPproxy in INVITE and 200 OK ?<br>
<br>
Regards,<br>
Bogdan<br>
<br>
On 03/29/2012 01:52 AM, <a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:magnusadilsom@gmail.com">magnusadilsom@gmail.com</a>
wrote:
<blockquote cite="mid:4F739644.30100@gmail.com" type="cite">
<meta http-equiv="content-type" content="text/html;
charset=ISO-8859-1">
I have phones (some behind NAT) connecting to Opensips server an
Asterisk and an rtpproxy as seen below:<br>
<br>
rtpproxy started with<br>
ps -aux | grep rtpproxy<br>
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
<br>
<br>
<br>
<br>
UAC1 username =
100------------Firewall/router--------------------Opensips
1.7---------- RTP PROXY------------Asterisk 1.6<br>
192.168.1.10 192.168.1.1
65.254.63.212 189.254.2.19 190.61.201.89<br>
external ip dinamic 169.254.2.2<br>
<br>
<br>
- Calls between UAC are OK (both SIP and RTP).<br>
- Calls UAC for PSTN is OK.<br>
- Did numbers is received in Asterisk, and destination for UAC
registered in opensips, but no work audio .<br>
(EX User call cellphone for DID 54115368566, call is received in
asterisk, and destination for user 100, registered in opensips)<br>
<br>
<br>
<br>
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</pre>
</blockquote>
<br>
<br>
<pre class="moz-signature" cols="72">--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
</blockquote>
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