<br><br><div><span class="gmail_quote">On 7/25/07, <b class="gmail_sendername">Weiter Leiter</b> <<a href="mailto:bp4mls@googlemail.com">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;">
Vlad,</blockquote><div><br><br style="color: rgb(255, 204, 204);"><span style="color: rgb(255, 102, 102);">SER is only handling signaling, SIP. It is unable to handle "voice
packages" - RTP in this case - by itself. That's why two other -
equivalent - helping applications exist: rtpproxy and mediaproxy.
<br><br><span style="color: rgb(0, 0, 0);">So you are saying that I should also install these other two applications ( are they applications ? excuse me , but i'm not familiar with this and I dit not quite undestood what you ment to say)
<br><br></span>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?
</span><br style="color: rgb(255, 102, 102);"><span class="q"><br><span style="color: rgb(0, 0, 0);">No the session is not established with SER (it gives me busy tone) ; without SER (from the AS directly to the HT), the phone rings more than 30 sec , but when it is picked-up ... nothing.
<br>And this is my problem ,that with SER the phone doesn't ring and I do not know how to make this happen from the ser config file.<br>If it helps , here is the dial-peer from the AS router to send the packages to SER:
<br> dial-peer voice xxxxx voip<br> description tech support<br> huntstop<br> destination-pattern .%00111111T<br> redirect ip2ip<br> voice-class codec 11<br> session protocol sipv2<br> session target ipv4:193.226.xxx.xxx ----> the ip of the computer with SER installed
<br> dtmf-relay sip-notify<br></span></span>-I'm sure that this dial-peer is corect because , as i said before , it worked with a hardware SipServer.<br><span style="color: rgb(255, 102, 102);"><br>As last resort, do a network sniffing on the SER box and send it over.
</span><br style="color: rgb(0, 0, 0);"></div><br>This will take some time , because I will have to install Ethereal on linux.<br><div><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="q"><span class="gmail_quote">On 7/25/07, <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><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></span><div><br>SER is only handling signaling, SIP. It is unable to handle "voice packages" - RTP in this case - by itself. That's why two other - equivalent - helping applications exist: rtpproxy and mediaproxy.
<br></div><span class="q"><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'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.</blockquote></span><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><span class="q"><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'm begining to think that there is no solution for my problem.</blockquote></span><div><br>From my experience, unless running SER, no other (hardware) box can do as much as SER&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><span class="sg"><br>WL.<br></span></div><span class="q"><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><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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</span></div></blockquote></span></div><div><span class="e" id="q_113fd0753b366849_11"><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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