<br><br><div class="gmail_quote">On 1 July 2010 22:41, Dmitri Korotkov <span dir="ltr"><<a href="mailto:dmitri.korotkov@festart.ee">dmitri.korotkov@festart.ee</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#ffffff" text="#000000">
Hi,<br>
<br>
voice:/# ps auxf |grep rtpproxy |grep -v grep<br>
rtpproxy 1291 0.0 0.0 26800 876 ? Ssl Jun18 0:10
/usr/sbin/rtpproxy -u rtpproxy rtpproxy -l my.public.ip.here -s
udp:localhost 7722<br>
voice:/#<br>
<br>
<br>
kamailio.cfg:<br>
#!define WITH_MYSQL<br>
#!define WITH_AUTH<br>
#!define WITH_ACCDB<br>
#!define WITH_NAT<br>
#!define WITH_PSTN<br>
<br>
#!ifdef WITH_NAT<br>
loadmodule "nathelper.so"<br>
#!endif<br>
<br>
# ----- nathelper -----<br>
#!ifdef WITH_NAT<br>
modparam("nathelper", "rtpproxy_sock", "udp:<a href="http://127.0.0.1:7722" target="_blank">127.0.0.1:7722</a>")<br>
modparam("nathelper", "natping_interval", 30)<br>
modparam("nathelper", "ping_nated_only", 1)<br>
modparam("nathelper", "sipping_bflag", 7)<br>
modparam("nathelper", "sipping_from", <a href="mailto:sip:pinger@kamailio.org" target="_blank">"sip:pinger@kamailio.org"</a>)<br>
modparam("registrar|nathelper", "received_avp", "$avp(i:80)")<br>
modparam("usrloc", "nat_bflag", 6)<br>
#!endif<br>
<br>
<br>
<br>
02.07.2010 0:32, dotnetdub пишет:
<div><div></div><div class="h5"><blockquote type="cite"><br>
<br></blockquote></div></div></div></blockquote><div>I'm not overly familiar with rtpproxy as we use mediaproxy but you will need to engage it somewhere in your script, are you doing that?</div><div><br></div><div>
Take a look at <a href="http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy">http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy</a></div><div><br></div><div>Can you see any rtpproxy messages in syslog?</div>
<div><br></div><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div bgcolor="#ffffff" text="#000000"><div><div class="h5"><blockquote type="cite">
<div class="gmail_quote">On 1 July 2010 21:53, Dmitri Korotkov <span dir="ltr"><<a href="mailto:dmitri.korotkov@festart.ee" target="_blank">dmitri.korotkov@festart.ee</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">Hello,<br>
<br>
I have kamailio installation WITH_PSTN, WITH_NAT and rtpproxy.<br>
Using following scenario:<br>
[kamailio]<-sip trunk ->[asterisk gw] <->sip trunk
<-> [PSTN provider]<br>
<br>
All kamailio sip subscribers are behind nat in different networks.<br>
<br>
1. OK. Local kamailio users can call one to other even they are on
different networks behind nat.<br>
2. OK. Outgoing calls from kamailio users to PSTN work also very well.<br>
3. Not OK. Incoming from PSTN side calls have only one way audio.<br>
<br>
I tcpdump'ed kamailio box and found, that pstn provider sends RTP
packets to kamailio IP in case of answered call.<br>
<br>
I guess that rtpproxy is not active in case of pstn call. Is it true ?<br>
<br>
I am using more less default kamailio config<br>
<br>
Could you please suggest solution ?<br>
<br>
BR,<br>
Dmitri<br>
<br>
</blockquote>
<div><br>
</div>
<div><br>
</div>
<div>Hi Dmitri,</div>
<div><br>
</div>
<div>Check out the nathelper module.</div>
<div><br>
</div>
<div>Regards,</div>
<div>Brian </div>
</div>
<br>
</blockquote>
<br>
</div></div></div>
</blockquote></div><br>