Hi all,<br><br>Im quite new to SPA3000. <br><br>I have a TRIXBOX running on public address. I need my SPA3000's FXO to be used as a trunk from a dynamic address behind NAT. Is this scenario possible?<br><br>Please give me some good links if it works.. I would really appreciate any help as my TRIXBOX is in US and my SPA3000 in middle east.
<br><br>Thanks everyone<br><br>Dan.<br><br><br><br><div><span class="gmail_quote">On 02/09/06, <b class="gmail_sendername">Rich Adamson</b> <<a href="mailto:radamson@routers.com">radamson@routers.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;">That option addresses what to do with the fxs (line 1) when the<br>registration fails as opposed to what does the fxo (pstn line) does when
<br>registration fails.<br><br><br>Bob Chiodini wrote:<br>> Rich,<br>><br>> After reading a little more, how about the "Line 1 VoIP Fallback to<br>> PSTN" (section 4.9)? It looks like this is invoked when the Ethernet
<br>> link is down or registration fails. I don't have a SPA3000 up at the<br>> moment to look at what's required.<br>><br>> Bob...<br>><br>> On Fri, 2006-09-01 at 11:45 -0500, Rich Adamson wrote:<br>>> If "pstn call ring thru line 1" is enabled, all incoming pstn calls will
<br>>> ring through to the fxs port (and not to asterisk). The OP was looking<br>>> for a auto fail over function that essentially would be "pstn call ring<br>>> thru line 1 on sip failure". That doesn't exist.
<br>>><br>>><br>>> Bob Chiodini wrote:<br>>>> Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in:<br>>>><br>>>> <a href="http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22">
http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22</a><br>>>><br>>>> By default, if my asterisk went down after the SPA3000 was already<br>>>> registered, the in-bound PSTN call was lost. I probably did not wait
<br>>>> long enough and I did not have "PSTN Call Ring Thru Line 1" enabled.<br>>>><br>>>> Bob...<br>>>><br>>>> On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote:
<br>>>>> My 3000 does this natively without config.<br>>>>><br>>>>><br>>>>> Kevin Collins<br>>>>><br>>>>><br>>>>> -----Original Message-----
<br>>>>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>>>>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com
</a>] On Behalf Of Steve Kennedy<br>>>>> Sent: Friday, September 01, 2006 10:03 AM<br>>>>> To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>>>>> Subject: Re: [asterisk-users] Sipura SPA3000
<br>>>>><br>>>>> On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote:<br>>>>><br>>>>>>> I have a question on configuration of SPA3000 with asterisk.<br>>>>>>> 1. I want all incoming calls are redirected from SPA3000 to my
<br>>>>>>> asterisk server.<br>>>>>>> 2. Asterisk then should direct this call to my SIP phones (including<br>>>>>>> Sipura)<br>>>>>>> 3. In case asterisk server is down I want that call be directed
<br>>>>>>> straight to the handset connected to the Sipura Is this<br>>>>>>> configuration possible?<br>>>>>> The spa3000 does not have logic in it to support #3.<br>
>>>> I thought the SPA3K could do this, i.e. on power failure or non-ability to<br>>>>> connect to server, connect FXS to FXO.<br>>>>><br>>>>><br>>>>> Steve<br><br>
_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br>