Voip.ms also supports unlimited free SIP calling. However, the routing is different than Callcentric.
I am working on it now and just have started to figure this stuff out this week.
Incoming callsWith
Voip.ms, you can receive calls at @sip.voip.ms or at your POP server, such as @newyork1.voip.ms.
As for what goes on the left-hand side:
If you have a DID, such as 2121234567, you can put the DID without the country code on the left-hand side, such as
2121234567@sip.voip.ms or
2121234567@newyork1.voip.ms. The call will route as you specify in Edit DID Settings. For example, you can set up a Ring Group and ring all the different sub-account phones on your premises.
I am trying to see if there is a way to create a SIP-only address, without needing to buy a DID. I do not know if you can do that or not yet.
You can call a particular sub-account (i.e., SIP phone registration) with your SIP/IAX Main Username plus the two-digit customized extension number, but only at your POP server. So, if your main username is 654321 and your phone's extension is 23, you must do
65432123@newyork1.voip.ms. You cannot do @sip.voip.ms; it will not work. Call routing appears to be a feature of DIDs only; if you call sip:65432123@newyork1.voip.ms, it will ring only that device.
Outgoing callsI am still researching this.
CodecsIn all cases, it seems that
Voip.ms interposes the codecs that you check in the Advanced > Allowed Codecs section. Currently
Voip.ms supports "G.711u", "G.729a", "G.722 BETA", and "gsm". Despite how hard I have tried, there appears to be no way to let calls through
Voip.ms use Opus (or other codecs, for that matter).