No , they are on different networks, and NAT has nothing to do with it
because there is no router to do so between them, only the default
gateways ( both HT and AS have public ip adresses) . As i said in the
first mail , my only problem is the config of SER to receive the voice
packages from one IP adress and send them to another IP adress and only
that ( something like : listen on : 193.226.xxx.xxx,5060 , send to:
193.230.xxx.xxx ). Actually I'm not even sure that SER cand replace the
hardware version of a Sip Server . The path I described
(phone->AS5350->SipServer->HandyTone->phone) , works on
an request/reply system and the codec negotiation is made after
the reply from the HT; if the reply message is not transmited on the
same path there is no codec negotiation , there-for no voice.<br>
Anyway , thanks for trying to help me. Unfortunately, day after day , I'm begining to think that there is no solution for my problem.<br><br><div><span class="gmail_quote">On 7/25/07, <b class="gmail_sendername">Weiter Leiter
</b> <<a href="mailto:bp4mls@googlemail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">bp4mls@googlemail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Are the AS and HT both part of the same network?<br>Otherwise, most comon diagnosis for your symtom is NAT and you might find some inspiration as instructed below, or any other NAT support resource you find on <a href="http://www.iptel.org" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
www.iptel.org</a><br><a href="ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.cfg" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.cfg
</a><br><a href="ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-mediaproxy.cfg" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-mediaproxy.cfg</a><br><br><br><br><div><div><span><span class="gmail_quote">On 7/24/07, <b class="gmail_sendername">Vlad Costea
</b> <<a href="mailto:vlad.costea@interpoint.ro" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
vlad.costea@interpoint.ro</a>> wrote:</span></span></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><span>
Hello.<br>I have just installed SER on RedHat 9 Linux with the purpose to route voice packages between a PSTN Gateway AS5350 and a Grandstream HandyTone
286.Let me tell you how it should work: I have assigned a phone number, that when it is picked-up, the IVR from the AS5350 router respods and after the press of a key it sends the packages to the HandyTone and from there to a normal phone; So far so good because it partialy works, that means that the phone rings butt there is no voice. I have managed to solve this problem using a hardware Sip Server between the two devices ( in this way both HT286 and AS5350 act as clients), but this is not possible any more because it was not my server; so i have tried with a software solution and I've installed SER. As I said before , I haven't managed to configure it to work as I wish and this is why I'am asking for your help; it would realy help me if you can provide a
ser.cfg example that would do just the routing part (no ack no authentification , no mysql,...., just routing).<br>Please excuse my bad english.<br>Thank you very much !<br>
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http://lists.iptel.org/mailman/listinfo/serusers</a><br><br></blockquote></div><br><br clear="all"><br>-- <br>"C is a language that combines all the elegance and power of assembly language with all the readability and maintainability of assembly language."
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