Hi all,<br><br>Special thanks to David and Noah for the earnest efforts...<br><br>Dan<br><br><br><br><div><span class="gmail_quote">On 02/01/07, <b class="gmail_sendername">Dave Schardin</b> <<a href="mailto:dschardin@atacomm.com">
dschardin@atacomm.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 style=""><div><br></div>Your best bet is to contact Sysmaster support at
<a href="mailto:support@sysmaster.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">support@sysmaster.com</a> or 1877-900-3993. I was talking to one of our contacts there and he said that it would be best to have you contact them. In order to get it to work for you they need to know the exact configuration you are trying to set up. We've worked with Sysmaster for some time now and they are very nice and helpful people.
<div><br></div><div>-David<div><div><div><br><div><div><br></div><div><br></div><div><br></div><div>Looking for voice over IP products? Visit our VoIP store at <a href="http://voipstore.atacomm.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
http://voipstore.atacomm.com</a></div></div><br></div><div><span class="e" id="q_10fe3e12b5b231f6_1"><div><br></div><div>On Jan 1, 2007, at 8:53 PM, Noah Miller wrote:</div><br><blockquote type="cite"><div style="margin: 0px;">
Hi Again Dan -</div><div style="margin: 0px; min-height: 14px;"><br></div> <blockquote type="cite"><div style="margin: 0px;">> Its a VOIP Switch based on SIP Proxy. Its Voice Master from SysMaster.</div><div style="margin: 0px;">
></div><div style="margin: 0px;">> VoiceMaster only authenticates IP and cant have username password based</div><div style="margin: 0px;">> authentication which asterisk can do. So i need to take some traffic from
</div><div style="margin: 0px;">> VoiceMaster to Asterisk and terminate it.</div><div style="margin: 0px; min-height: 14px;"><br></div><div style="margin: 0px;">That shouldn't be a problem.<span> </span>You can just create a sip friend/peer
</div><div style="margin: 0px;">without a username or password, and with a "host=ipaddress" statement.</div><div style="margin: 0px;"><span> </span>Like this in your sip.conf file:</div><div style="margin: 0px; min-height: 14px;">
<br></div><div style="margin: 0px;">[NoAuth-VoiceMaster]</div><div style="margin: 0px;">type=friend</div><div style="margin: 0px;">context=<your context></div><div style="margin: 0px;">host=<IP Address Of Voice Master>
</div><div style="margin: 0px;">disallow=all</div><div style="margin: 0px;">allow=<codecs you want to allow></div> </blockquote><div style="margin: 0px; min-height: 14px;"><br></div><div style="margin: 0px;">As an addendum to this, it would be a very good idea to make certain
</div><div style="margin: 0px;">that you've properly secured your asterisk server so you're not going</div><div style="margin: 0px;">to have unwanted unauthorised access.<span> </span>I would probably only do this
</div><div style="margin: 0px;">if your asterisk server is not accessible from the outside world via</div><div style="margin: 0px;">sip.</div><div style="margin: 0px; min-height: 14px;"><br></div><div style="margin: 0px; min-height: 14px;">
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