<div dir="ltr">Hi Jun,<div><br></div><div>Without a doubt, I'd recommend flowroute since you're setting up your own voip server at your location. </div><div>Check out the blogs of flowroute: <a href="http://blog.flowroute.com/">http://blog.flowroute.com/</a></div>
<div><br></div><div>Reasons I like flowroute:</div><div>0. There's some kind of [in]direct support to freeswitch. I don't know the exact politics of the support, though.</div><div>1. You can easily setup failover to a DID, SIP URI, or a host address. You may think this is standard for all VOIP providers, but places like vitality charge more per-min to do failover.</div>
<div>2. Support is super responsive and very knowledgeable</div><div>3. Flowroute has blogs that are updated regularly: <a href="http://blog.flowroute.com/">http://blog.flowroute.com/</a></div><div>4. Free inums and you can forward to URI, DID, host</div>
<div>5. They just enabled cname support on incoming calls</div><div>6. Never had a service interuption</div><div>7. get Pat Fleet or Alex Kasperavicius to record your IVR menus: <a href="https://www.flowroute.com/voice/voice/">https://www.flowroute.com/voice/voice/</a></div>
<div>8. They supported calling inums before releasing inums. Some companies think of inmus as competition, but flowroute did not have that impression.</div><div>9. free calling to toll free numbers</div><div>10. Free test account with 25 cent credit. Sign up and make some calls. You won't be able to receive calls until you buy a DID, though. (that's expected)</div>
<div><br></div><div>What I would like improved at Flowroute:</div><div>0. Better DID search. As of now, you must select an area code, then a rate center to see all available numbers. There's no way to search for a string in their available list. </div>
<div><br></div><div>What I hope will be released at flowroute:</div><div>0. Public API with a very good userguide. </div><div>1. SMS support</div><div><br></div><div><br></div><div>My next favorite suggestion would be <a href="http://voip.ms">voip.ms</a></div>
<div><br></div><div><br></div><div>Hope this helps!</div><div><br></div><div>happy new year,</div><div>jungleboogie</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On 30 December 2013 10:36, Jun Sun <span dir="ltr"><<a href="mailto:jsun@junsun.net" target="_blank">jsun@junsun.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div dir="ltr">I just checked their pricing. A little higher than I thought. List for others' info.<div><br></div><div>Vitelity: 1.2c/min+$1.49/mo or $7.95/mo; $18 per port</div><div><a href="http://VOIP.MS" target="_blank">VOIP.MS</a>: 1~1.49c/min+$0.99~1.49/mo or $4.95~$6.95/mo; FREE port for US.</div>
<div>FlowRoute: 1.2c/min+$1.25/mon or $6.25/mo; porting fee not found.</div><div><br></div><div>It seems you would need to add more for E911 service as well. It looks <a href="http://VOIP.MS" target="_blank">VOIP.MS</a> is the cheapest.</div>
<div><br></div><div>Cheers.</div><div><br></div><div>Jun</div><div> </div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Sun, Dec 29, 2013 at 3:21 PM, Sean Devoy <span dir="ltr"><<a href="mailto:sdevoy@bizfocused.com" target="_blank">sdevoy@bizfocused.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">You need to choose a TISP (Telephony Internet Service Provider) also called a SIP Provider. They will be able to provide services dialing out to PSTN phones as well as your DIDs.<br>
<br>
I have personally had good success with Vitelity and <a href="http://VOIP.MS" target="_blank">VOIP.MS</a>. I know FlowRoute is a provider with pretty good FreeSwitch support.<br>
<br>
Choose one from here:<br>
<a href="https://wiki.freeswitch.org/wiki/SIP_Provider_Examples" target="_blank">https://wiki.freeswitch.org/wiki/SIP_Provider_Examples</a><br>
<div><br>
<br>
-----Original Message-----<br>
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of Jun Sun<br>
Sent: Sunday, December 29, 2013 5:57 PM<br>
To: FreeSWITCH Users Help<br>
</div><div><div>Subject: [Freeswitch-users] [OT] How do I port my home phone number to a SIP address?<br>
<br>
<br>
Hi, guys,<br>
<br>
This might be a little off topic, but I think many experts here may know this already. :)<br>
<br>
I have a VOIP server set up and running. Now I start to think how to port my home phone number to this server. Basically what I want is if someone calls my home phone number I like to see a SIP connection to my own VOIP server.<br>
<br>
Currently I'm using bbtalk as my home phone service. I'm pretty much sure they don't provide this service.<br>
<br>
Who would be able to provide this service? I live in USA.<br>
<br>
Thanks in advance.<br>
<br>
Jun<br>
<br>
</div></div><div><div><br></div></div></blockquote></div></div></blockquote></div><div><br></div>-- <br><div dir="ltr">-------<br>inum: 883510009902611<br>sip: <a href="mailto:jungleboogie@sip2sip.info" target="_blank">jungleboogie@sip2sip.info</a><br>
<div>xmpp: <a href="mailto:jungle-boogie@jit.si" target="_blank">jungle-boogie@jit.si</a></div></div>
</div></div>