Dominic,<br><br>I am not familiar with Trixbox, however a similar service from Sipgate in the UK is configured in Asterisk quite easily. Check out this link.<br><br><a href="http://www.sipgate.co.uk/faq/index.php?aktion=artikel&rubrik=715&id=540&lang=de&highlight=asterisk">
http://www.sipgate.co.uk/faq/index.php?aktion=artikel&rubrik=715&id=540&lang=de&highlight=asterisk</a><br><br>From what I understand, these type of 'trunks' are really just SIP accounts. Having Asterisk or Trixbox register the account gives you control of how to use the line, as opposed to configuring a SIP account on a telephone.
<br><br>Hope this helps,<br>Colin<br><br><div><span class="gmail_quote">On 8/14/06, <b class="gmail_sendername">Dominic Son</b> <<a href="mailto:dominicson@gmail.com">dominicson@gmail.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;">
<div>Thank you Mark. I've went from "The number you are dialing is not in service, please check the number and dial again" to a fast busy tone...I think I'm getting closer..</div><div><span class="q"><br>-- <br>
Anything else, let me know.<br><br></span></div><div><span class="sg">
-Dominic Son<br><a href="http://www.DominicSon.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">www.DominicSon.com</a></span></div><div><span class="e" id="q_10d0e2c510f5a018_3"><br><br><div><span class="gmail_quote">
On 8/12/06, <b class="gmail_sendername">Mark Phillips</b> <<a href="mailto:g7ltt@g7ltt.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">g7ltt@g7ltt.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;">Sounds to me like you don't have a proper connection with Stanaphone.
<br>
The only time you'll get these problems is when they cannot contact you<br>to forward the call to your system.<br><br>Double check you firewall settings. They need to be able to reach your<br>system on port 5060UDP (assuming SIP) as well as ports 10000-20000UDP
<br>(Asterisk default media ports).<br><br>They'll contact yo when a call comes in. You'll accept the call and at<br>the same time tell them which port to send the incoming audio to.<br>They'll also tell you where to send your outgoing audio.
<br><br>Hope that helps.<br><br>Mark<br><br>On Fri, 2006-08-11 at 15:45 -0700, Dominic Son wrote:<br>> Hi. Can someone explain to a right brained person what is going on<br>> with In/out bound trunks, how it connects to my Trixbox..
<br>><br>> 1. i get issued a free NY phone number from a voip service like<br>> stanaphone .<br>> 2. i then call this number, it connects to the stanaphone voicemail<br>> 3. i turn off the voicemail because i want it to connect to my
<br>> Askterisk, I've set up all the trunks in the PBX setup,<br>> ( sip.stanaphone, etc)<br>> 4. now i call my NY number, and it says 'this phone is not in service,<br>> please check the number and dial again'
<br>><br>> my Q: how does this work, more specifically, if i turned off the VM,<br>> how does stanaphone then know to look for my asterisk server to use<br>> the trixbox?<br>><br>> --<br>> Anything else, let me know.
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