<div>Sorry for my delay in answer you.</div>
<div>S0 is how pstn line is identified into spa3000. It means that an incoming call from S0 will be forwarded to "extension@asteriskbox"<br>yor must configure in sip.conf an account for the sip pstn line and a context , in
extensions.conf the same context with a pattern</div>
<div>or number for dialing.</div>
<div> </div>
<div>example</div>
<div>sip.conf</div>
<div>[100]<br>host=dynamic<br>type=friend<br>context=GW-PSTN-ZEL<br>secret=xxxxx<br>qualify=150<br>authuser=100<br>username=GW-PSTN-ZEL<br>accountcode=ZEL-100<br>port=5061<br>disallow=all<br>allow=ulaw</div>
<div> </div>
<div>extensions.conf</div>
<div>[GW-PSTN-ZEL]<br> exten=>s,1,Answer<br> exten=>s,2,NoOp(${CALLERID})<br> exten=>s,3,GotoIfTime(10:00-18:00,mon-fri,*,*?HLaboral,s,1)<br> exten=>s,4,GotoIfTime(09:00-10:00,mon-fri,*,*?DesvioFax,s,1)
<br> exten=>s,5,Goto(Cerrado,s,1)<br> </div>
<div>I Have configured the spa 3000 with <font color="#550055">S0<:s@Asterisk_ip_address></font></div>
<div><font color="#000000">when spa 3000 receive a call, is forwarded to s extension in asterisk, but before, it is very important that sip user line be registered with asterisk for making calls.</font></div>
<div> </div>
<div>I hope it will help you<br> <br> </div>
<div><span class="gmail_quote">2006/11/29, Larry Alkoff <<a href="mailto:labradley@mindspring.com">labradley@mindspring.com</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Fran when you say "specify the next hop" do you mean the S0 line be an<br>extension in sip.conf or a context in
extensions.conf?<br><br>Or should the line simply be tacked on to my [default] context?<br><br>Larry<br><br>Fran Oliveira wrote:<br>> I think it is wrong. You should specify the next hop with some like this<br>> S0<:
66610@Asterisk_ip_address><br>><br>><br>><br>> 2006/11/23, Larry Alkoff <<a href="mailto:labradley@mindspring.com">labradley@mindspring.com</a>>:<br>>><br>>> Problem: SPA3000 phone does not ring for incoming PSTN call although I
<br>>> can dial out.<br>>><br>>> I set up my Sipura with the Voxilla Wizard which is pretty good but<br>>> leaves out some important details.<br>>><br>>> The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab ->
<br>>> Dial Plans -><br>>> Dial Plan 8 (<S0:66610>)<br>>><br>>> Should I put extension [66610] in sip.conf with a context in<br>>> extensions.conf that will contain dialing instructions?
<br>>><br>>> Can someone please tell me what the entries under [66610] and the<br>>> associated context would look like?<br>>><br>>> Or just tell me how to handle this - I'm been stuck for some time with
<br>>> this.<br>>><br>>> The Wizard was nice enough to give detailed settings for sip.conf and<br>>> extensions.conf but nothing about to handle Dial Plan 8 except "You'll<br>>> need to enter the extension you wish to forward all incoming PSTN calls
<br>>> to on your Asterisk server". I don't understand how to do that.<br>>><br>>> Larry<br>>><br>>> --<br>>> Larry Alkoff N2LA - Austin TX<br><br><br>--<br>Larry Alkoff N2LA - Austin TX
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