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This is a reported post for a post in the topic <input class="cms_keep_ui_controlled" size="45" title="[post param=&quot;Jigasi setup&quot;]482[/post]" type="button" value="post Comcode tag (dbl-click to edit/delete)" />, by jacobgkau<br /><br /><comcode-quote param="2">Hello Dennis,<br /><br />The only commercial SIP provider I&#039;ve used is SIPStation, which is part of Sangoma, the company who makes FreePBX. Supporting SIPStation would be supporting free software in that regard. You can find SIPStation here: <a class="user_link" href="https://www.sipstation.com/" rel="nofollow noopener external" target="_blank" title="https://www.sipstation.com/ (this link will open in a new window)">https://www.sipstation.com/</a><br /><br />The only bad thing I&#039;ve heard about SIPStation is that its pricing (which is basically $25/month per unlimited trunk) can be less optimal if you don&#039;t use your trunks often. It may be more economical to use a pay-per-minute SIP provider. The company I set a FreePBX server up for was coming from a traditional phone system and wanted a flat rate, so this wasn&#039;t an issue for me, but it&#039;s something to consider.<br /><br />I&#039;m not sure what you mean by &quot;special interconnection layer&quot; for a &quot;regualr PSTN phone number.&quot; The SIP provider provides the connection between the telephone network and your digital network. It&#039;s called DID: <a class="user_link" href="https://en.wikipedia.org/wiki/Direct_inward_dial" rel="nofollow noopener external" target="_blank" title="https://en.wikipedia.org/wiki/Direct_inward_dial (this link will open in a new window)">https://en.wikipedia.org/wiki/Direct_inward_dial</a><br /><br />If you&#039;re thinking you&#039;re going to &quot;game the system&quot; by using a regular phone line instead of a SIP provider, you would need some sort of appliance to convert the analog line into a SIP line, basically being your own SIP provider. There are some of those devices on the market, although the ones I&#039;ve seen are more to provide an emergency fallback in case your internet connection goes down and you still need to have your SIP devices work:<br /><br /><a class="user_link" href="https://www.sangoma.com/voip-gateways/analog/" rel="nofollow noopener external" target="_blank" title="https://www.sangoma.com/voip-gateways/analog/ (this link will open in a new window)">https://www.sangoma.com/voip-gateways/analog/</a><br /><a class="user_link" href="http://www.grandstream.com/products/gateways-and-atas/voip-gateways" rel="nofollow noopener external" target="_blank" title="http://www.grandstream.com/products/gateways-and-atas/voip-gateways (this link will open in a new window)">http://www.grandstream.com/products/gateways-and-atas/voip-gateways</a><br /><a class="user_link" href="https://www.voipsupply.com/voip-gateways/analog" rel="nofollow noopener external" target="_blank" title="https://www.voipsupply.com/voip-gateways/analog (this link will open in a new window)">https://www.voipsupply.com/voip-gateways/analog</a><br /><br /><br /><br /></comcode-quote><br />//// PUT YOUR REPORT BELOW \\\\<br /><br /> Add: Add: Font Size Color [Font] Arial Courier Georgia Impact Times Trebuchet Verdana Tahoma Geneva Helvetica [Size] 0.8 1 1.5 2 2.5 3 4 [Color] Black Blue Gray Green Orange Purple Red White Yellow This is a reported post for a post in the topic [post param="Jigasi setup"]482[/post], by jacobgkau [quote="2"] Hello Dennis, The only commercial SIP provider I've used is SIPStation, which is part of Sangoma, the company who makes FreePBX. Supporting SIPStation would be supporting free software in that regard. You can find SIPStation here: [url="https://www.sipstation.com/" target="_blank"]https://www.sipstation.com/[/url] The only bad thing I've heard about SIPStation is that its pricing (which is basically $25/month per unlimited trunk) can be less optimal if you don't use your trunks often. It may be more economical to use a pay-per-minute SIP provider. The company I set a FreePBX server up for was coming from a traditional phone system and wanted a flat rate, so this wasn't an issue for me, but it's something to consider. I'm not sure what you mean by "special interconnection layer" for a "regualr PSTN phone number." The SIP provider provides the connection between the telephone network and your digital network. It's called DID: [url="https://en.wikipedia.org/wiki/Direct_inward_dial" target="_blank"]https://en.wikipedia.org/wiki/Direct_inward_dial[/url] If you're thinking you're going to "game the system" by using a regular phone line instead of a SIP provider, you would need some sort of appliance to convert the analog line into a SIP line, basically being your own SIP provider. There are some of those devices on the market, although the ones I've seen are more to provide an emergency fallback in case your internet connection goes down and you still need to have your SIP devices work: [url="https://www.sangoma.com/voip-gateways/analog/" target="_blank"]https://www.sangoma.com/voip-gateways/analog/[/url] [url="http://www.grandstream.com/products/gateways-and-atas/voip-gateways" target="_blank"]http://www.grandstream.com/products/gateways-and-atas/voip-gateways[/url] [url="https://www.voipsupply.com/voip-gateways/analog" target="_blank"]https://www.voipsupply.com/voip-gateways/analog[/url] [/quote] //// PUT YOUR REPORT BELOW \\\\ View all Use of this website implies that you agree to the website rules and privacy policy.