<p>As I understand your configuration , dial-peer voice <span class="nu0">697617664</span> voip, only forward the pattern <span class="nu0">697617664(</span> destination-pattern <span class="nu0">697617664) to XXX.XXX.XXX.
<span class="nu0">115</span>:<span class="nu0">5060 ( session target ipv4:XXX.XXX.XXX.<span class="nu0">115</span>:<span class="nu0">5060) that I think is your Asterisk box.</span></span></span></p>
<div><span class="nu0"><span class="nu0"><span class="nu0">An incoming call in your E1 must much a destination pattern, your only destination pattern is <span class="nu0">697617664.</span></span></span></span></div>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0">Usually an E1 has several DID associated it in a consecutive range, 91 5344XXX for example.</span></span></span></span></div>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0"></span></span></span></span> </div>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0">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 class="nu0"><span class="nu0"><span class="nu0"><span class="nu0">You can configure asterisk , without user registration with the sip.conf insecure option</span></span></span></span></div>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0"></span></span></span></span> </div>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0"> when I copied </span></span></span></span></div>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0"><span class="gmail_quote">dial-peer voice 10 pots<br> destination-pattern 0T should be .T </span></span></span></span></span></div>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0"><span class="gmail_quote">it tells cisco 26xx router what patterns can be reached throught E1</span></span></span></span></span></div>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0"><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>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0"><span class="gmail_quote"></span></span></span></span></span> </div>
<div><span class="nu0"><span class="nu0"><span class="nu0"><span class="nu0"><span class="gmail_quote">best regards</span></span></span></span></span></div>
<div><br> </div>
<div class="de2"> </div><br><br>
<div><span class="gmail_quote">2006/12/7, FaberK <<a href="mailto:f.faberk@gmail.com">f.faberk@gmail.com</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><a onclick="return top.js.OpenExtLink(window,event,this)" href="http://pastebin.ca/270840" target="_blank">
http://pastebin.ca/270840</a><br>This is the newone with some changements.<br>Unfortunately, always the same problem.<br><br>Fran, if I add the "<span class="gmail_quote">dial-peer voice 10 pots", I receive the advise that the number does not exist.
<br>Also, I do not find the way to add "</span><span class="gmail_quote">authentication username "asterisk-uername" password XXXXXX".<br><br>The story continues...</span><span class="gmail_quote"><br><br>
Thanks<br></span><span class="sg"><br>F.<br></span></blockquote></div><br>