Scheduled maintenance until Sunday 20 September: new sign-ups and number orders are paused. Existing customers are not affected.

SIP forwarding to your own server

Deliver calls to your SendMyCall number straight to your own SIP server or PBX: Asterisk, FreePBX, 3CX, FreeSWITCH, or any SIP endpoint with a public address.

How it works

When a number is set to Your own SIP server, every call to it is sent as a SIP call from our network to the host you name, with your number as the called party and the caller's own number as caller id. There is no registration, no username and no password: we simply send the call to your address, so your server has to accept calls from ours. Nothing is charged per call; the number's price covers it.

  1. Sign in, open the number under My account and choose Your own SIP server under "Where should calls to this number go?".
  2. Enter your server's public host name or address, the port if it is not 5060, UDP or TCP, and (optionally) the extension or user the call should be addressed to.
  3. Allow our gateway addresses on your firewall and in your PBX's trunk or ACL settings (below).
  4. Route calls for your number inside your PBX to a user, a ring group or an IVR. The change applies from the next call.

Our SIP gateway addresses

Calls to your server come from these addresses. Allow every one of them: which one a call uses depends on the route it takes.

Addresses
45.77.78.79207.246.101.92
Host name
sip.sendmycall.comresolves to the addresses above, for systems that allow-list by name
SIP signalling
UDP or TCP, port 5060
Media (RTP)
UDP, ports 16384 to 32769
Codecs
G.711 µ-law (PCMU) · G.711 A-law (PCMA) · DTMF as RFC 2833 (telephone-event)
Called number
your number in international format without the plus (or the user you set)
Caller id
the caller's number in international format

Firewall checklist

  • Allow SIP from 45.77.78.79 and 207.246.101.92 to your server on port 5060, UDP and TCP.
  • Allow RTP (UDP) from the same addresses, ports 16384 to 32769, to your server's media port range, and let your server's replies out.
  • Your server needs a public address or a port forward: we cannot reach private addresses (10.x, 172.16-31.x, 192.168.x).
  • If your PBX sits behind NAT, set its external address so the media it offers is reachable; otherwise you get one-way audio.
  • Keep the same settings on every address: a call may come from any of them.

Notes for common phone systems

Asterisk / FreePBX (chan_pjsip)
A trunk without registration and without outbound authentication. In its identify section (FreePBX: the trunk's "Match (Permit)" list) enter all our addresses, one per line or comma separated. Add an inbound route for your number.
FreeSWITCH / FusionPBX
Add our addresses to an access control list (Advanced → Access Controls) used by the profile that listens on your public address, then a public inbound route for your number.
3CX
A generic SIP trunk with registration off; under source identification list our addresses so 3CX accepts the calls, and route the DID to a destination.
Anything else
Accept SIP calls from our addresses without authentication, and route calls addressed to your number (or the user you set on the number page) where you want them to ring.

Moving from our previous platform?

If your numbers were delivered to your SIP server before, calls came from 208.89.106.93. During the move they come from the addresses above as well, so allow both for now. You can remove the old address once your numbers have moved and calls arrive from the new ones. We mail every affected customer a notice with a confirmation link before moving their numbers, and we move them only after that confirmation.

If calls do not arrive

  • Ringing but never answered, or a time-out: our INVITE is dropped by a firewall, or your server listens on another port or transport than the one you entered.
  • Rejected with 403 or 401: your server expects a registered peer or a password. Set the trunk to accept calls from our addresses without authentication.
  • Connected, but no or one-way audio: RTP is blocked or your PBX advertises a private address. Open the media ports and set the external address.
  • Wrong destination inside the PBX: the call is addressed to your number in international format without the plus; set a user on the number page if your PBX wants an extension instead.

Still stuck? Write to [email protected] with the make of your phone system and the number.