Send2SIP is completely free to use, and we don’t even require a credit card.

Nope, use Send2SIP as much as you like. We allow multiple incoming calls from your line (though your carrier may impose limits), and no limits on the number of calls or minutes used. Use Send2SIP as much as you like, completely free.

Send2SIP will work with telephone numbers in the US provided by many telephone companies including landlines, cable lines, cell phones, VoIP providers. We provide setup instructions for many of the largest companies that we, or our customers have verified as working in our Wiki.

Don’t see your provider listed? Test your number, completely for free. If you find a carrier that isn’t already listed in our Wiki, please let us know by Contacting Us and we’ll add it!

The numbers we’ll provide are located in the United States, specifically in New York State.

You can use Send2SIP for anything you like, but the most common uses would typically be:

• Permanently forwarding an existing telephone number via VoIP to a SIP URI, without needing a dedicated VoIP enabled number to forward it to.
• Temporarily forwarding a number via VoIP to a SIP URI, such as when traveling internationally.
• If your provider supports Simulring (simultaneous ringing of multiple numbers), having your local number ring as well as a SIP URI, at the same time.

Sure! Just login to your account at anytime, make the change, and it will instantly be updated.

Absolutely. Just disable the Call Forward or Simulring at your original carrier at anytime, and calls will cease forwarding / simulringing. You don’t even need to tell us about it, or change anything on your account.

Nope! You don’t need to port (LNP) your telephone number to Send2SIP to use our service. You keep control of your number, and can enable or disable Send2SIP at anytime.

If you’re on a flat-rate plan with all calls in the United States included, most providers do not charge anything to forward a call or simulring to another US number (all Send2SIP numbers are in the USA). Your provider will indicate if there is a charge for forwarding.

Imagine John calls Jane, and Jane is using Send2SIP on her telephone line to forward to the SIP URI: [email protected].

When John calls Jane’s number the call is forwarded (or simultaneously rung) to a Send2SIP number, Send2SIP identifies this call is for Jane, and sends the call over VoIP to the SIP URI Jane has chosen, in this case: [email protected].

When Echo Test Mode is enabled, Send2SIP will immediately answer all incoming calls that are forwarded to us, and provide prompts allowing you to record a message which will be played back to you. This allows you to place a test call and see that calls from your telephone number are reaching Send2SIP properly.

If Echo Test Mode isn’t answering your call, see this FAQ.

If Echo Test Mode works, but forwards to your SIP URI do not, see this FAQ.

If you’re calling your telephone number, and not reaching the Send2SIP Echo Test, you should double-check the following. Have you?:

• Added your specific telephone number to your account, and received a forwarding number here?
• Is Echo Test Mode enabled on the specific number you’re testing?
• Have you already setup forwarding on your existing telephone number, to the specific forwarding telephone number you were provided by us? NOTE: you may be provided different forwarding numbers for each of your telephone numbers if you add more than one (1) telephone number on your account.
• Is your telephone carrier listed as supported in our Wiki? If not, it’s possible that your carrier is not forwarding / simulringing calls in a way that Send2SIP can detect and support.
• If you’ve double-checked all of the above, you can Contact Us providing further details and we’ll do our best to help.

If you’re calling your telephone number, and calls are not being forwarded to the SIP URI you configured, you should double-check the following. Have you?:

• Turned on Echo Test Mode and verified the Echo Test is working?
• If Echo Test Mode is working, but forwarding to the SIP URI you configured isn’t, we recommend you try forwarding to a known working SIP URI that you can reach via other means, such as a public test SIP URI. A few examples that appear to be reliable are listed below:
  • [email protected]
  • [email protected]
  • [email protected]
  • [email protected]
• If one or more of the above examples work, but your SIP URI doesn’t, you should:
• Double-check you’ve entered a correct SIP URI including the user part (before the @ sign), the domain (after the @ sign), and if a port number is necessary, be sure to include it (IE “:5090” at the end of the URI).
• Check traces / logs at the destination SIP URI, or contact the support for the SIP URI you’re trying to forward to.
• If you’re still having problems, you can Contact Us, however we can provide limited support if Echo Test Mode for your number and publicly accessible SIP URI’s are working, but your specific SIP URI is not.

If you’re hearing an error message “No matching results”, you should verify the following:

• You’re NOT directly calling one of the “Forward To” numbers we’ve assigned on your account. These numbers cannot be directly called, rather you must forward a call to them from an existing telephone line.
• Have you added your specific telephone number to your account, received a forwarding number here, and are forwarding your existing number to the specific number you were assigned?
• Double-check that you added your existing telephone number correctly. If there is a typo in the number you entered, you will hear the “No matching results” message when forwarding / simulringing calls from your number.
• Is your telephone carrier listed as supported in our Wiki? If not, it’s possible that your carrier is not forwarding / simulringing calls in a way that Send2SIP can detect and support.

Yes, some carriers will not work properly with Send2SIP. Please see this Wiki article.

If you are forwarding to a SIP URI that has a firewall in front of it, and need to open access from specific IPs, the range of IPs that Send2SIP will originate calls from is:

• CIDR format: 199.87.144.0/21
• Or, as a range: 199.87.144.0 - 199.87.151.255