Vlad,<br><br><div><span class="gmail_quote">On 7/25/07, <b class="gmail_sendername">Vlad Costea</b> &lt;<a href="mailto:vlad.costea@interpoint.ro">vlad.costea@interpoint.ro</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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 ).</blockquote><div><br>SER is only handling signaling, SIP. It is unable to handle &quot;voice packages&quot; - RTP in this case - by itself. That&#39;s why two other - equivalent - helping applications exist: rtpproxy and mediaproxy.
<br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> Actually I&#39;m not even sure that SER cand replace the
hardware version of a Sip Server . The path I described
(phone-&gt;AS5350-&gt;SipServer-&gt;HandyTone-&gt;phone) ,&nbsp; works on&nbsp;
an&nbsp; request/reply&nbsp; system&nbsp; and&nbsp; the codec&nbsp; negotiation&nbsp; is&nbsp; 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.</blockquote><div><br>So, in your scenario with SER is the session established OK? In other words, is the call established and lasts on both phones for more than 30s (or more), or less than that: the callee picks up, but this is never perceived at caller?
<br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Anyway , thanks for trying to help me. Unfortunately, day after day , I&#39;m begining to think that there is no solution for my problem.</blockquote><div><br>From my experience, unless running SER, no other (hardware) box can do as much as SER&amp;friends can. You just need to clearly say what hurts. 
<br>As last resort, do a network sniffing on the SER box and send it over.<br><br>WL.<br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div><span class="e" id="q_113fcbb1ab39c859_1"><div><span class="gmail_quote">On 7/25/07, <b class="gmail_sendername">Weiter Leiter
</b> &lt;<a href="mailto:bp4mls@googlemail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">bp4mls@googlemail.com</a>&gt; 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> &lt;<a href="mailto:vlad.costea@interpoint.ro" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
vlad.costea@interpoint.ro</a>&gt; 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&#39;ve installed SER. As I said before , I haven&#39;t managed to configure&nbsp; it to work as I wish and this is why I&#39;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>&quot;C is a language that combines all the elegance and power of assembly language with all the readability and maintainability of assembly language.&quot;
</blockquote></div><br>
</span></div></blockquote></div><br><br clear="all"><br>-- <br>&quot;C is a language that combines all the elegance and power of assembly language with all the readability and maintainability of assembly language.&quot;