<a href="http://pastebin.ca/271763">http://pastebin.ca/271763</a><br><br>Hi to all,<br><br>To Fran:<br><div><span class="gmail_quote"></span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<p>As I understand your configuration , dial-peer voice <span>697617664</span> voip, only forward the pattern <span>697617664(</span> destination-pattern <span>697617664) to XXX.XXX.XXX.
<span>115</span>:<span>5060 ( session target ipv4:XXX.XXX.XXX.<span>115</span>:<span>5060) that I think is your Asterisk box.</span></span></span></p></blockquote><div><br>you are right, <span><span>XXX.XXX.XXX.<span>115
</span>:<span>5060 is my * box where I've created a "friend" called </span></span></span><span>697617664</span></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><span><span>An incoming call in your E1 must much a destination pattern, your only destination pattern is <span>697617664.</span></span></span></span></div>
<div><span><span><span><span>Usually an E1 has several DID associated it in a consecutive range, 91 5344XXX for example.</span></span></span></span></div></blockquote><div><br>here too, you are right, but I'm trying to receive at leat 1 call to
<span>697617664, then for all the others will be not a problem. But first i need to let it works...!!!</span> </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><span><span><span>otherwise, for outgoing calls you must configure a pots dial peer ,you can put a randon name to the dial peer.</span></span></span></span></div>
<div><span><span><span><span>You can configure asterisk , without user registration with the sip.conf insecure option</span></span></span></span></div>
<div><span><span><span><span></span></span></span></span> </div>
<div><span><span><span><span> when I copied </span></span></span></span></div>
<div><span><span><span><span><span class="gmail_quote"><span class="q">dial-peer voice 10 pots<br></span> destination-pattern 0T should be .T </span></span></span></span></span></div>
<div><span><span><span><span><span class="gmail_quote">it tells cisco 26xx router what patterns can be reached throught E1</span></span></span></span></span></div>
<div><span><span><span><span><span class="gmail_quote">IŽll take a look into the cisco web site for sip user authentication, I have a configuration done, but with FXS interfaces and worsk fine.
</span></span></span></span></span></div></blockquote><div><br>For outgoing calls, at this moment I'm not interested.<br><br>On the new configuration, I've also changed the codecs, leaving the g711 only.<br>Unfortunately always the same: calling my number, the call reach the 2600(infact I hear the tone), but is not forwarded to the sip-server.
<br></div><br>To Pavel:<br>thanks for your suggestion regarding MGCP, but the fact is that I got all sip, and never worked with mgcp.<br><br>Thanks to all<br>Best Regards<br><br>F.<br></div>