SIP.US New User Troubleshooting Guide

The following information will help in troubleshooting the initial setup and configuration for SIP.US trunks.

For further assistance, please open a support ticket by clicking on the Support link in the upper right corner of the SIP.US website.

Outbound calls not working?

Check to see that your trunks are Registered:

In the SIP.US control panel, when you view your SIP trunk, you will see a Registration Status.  Make sure that you see that your trunks are registered.

Note: Technically, trunk registration is not required to make an outbound call as it is primarily used to store the contact information used by SIP.US to send you inbound calls, however it is a great indicator to see that things are setup properly.

Make sure you are sending calls with a ‘1’ in front of the number.  SIP.US routes outbound calls with Country Code + Area Code (or city code) + Number, so you must send us calls with the ‘1’ for North America. 

NOTE: CHECK YOUR CDR on SIP.US to see what you are sending!

Inbound calls not working? 

Check to see that your trunks are Registered:

In the SIP.US control panel, when you view your SIP trunk, you will see a Registration Status.  Make sure that you see that your trunks are registered.  You must have a valid registration in order to receive an inbound call!

Make sure your registration status shows the PUBLIC IP Address of your PBX or 

Device.  This is a common issue where we see a private IP address in the registration string of the registration status.  Our systems will attempt to deliver the call 

If you are using an Asterisk PBX on a machine with a private IP address, make sure that you have the Public IP address set properly in your Asterisk SIP Settings.

Make sure you have an inbound route setup for the number.  When we deliver the call to the IP address specified in the Registration status, we are sending you the call with the ‘1’ in front of the number.  Make sure that you can answer the call and route to an IVR or extension making sure you are using the format 1NXXNXXXXXX for the DID.

Make sure you have port 5060 UDP open on your router/firewall and port forwarded to your pbx.  SIP.US trunks communicate SIP signaling information over port 5060.  Your PBX or device must be able to communicate on this port and respond to requests from SIP.US servers.  If not, calls will fail.

No-Audio or One-Way Audio?

Typically no-audio or one-way-audio problems are related to NAT or Firewall issues.  If your PBX or Device is on a private IP address behind NAT, you need to make sure you have the following ports open on your firewall to properly pass Audio:

5060 UDP  (SIP Signaling Port used for Messaging (call set-up, tear-down, etc.)

10000-20000 UDP (RTP Media port range used for call audio by most PBXs)

(sometimes you can set the RTP range in your device.  If you can, and you restrict the RTP ports, make sure that they are port-forwarded in your router for that specific range that you set)

If your device supports STUN, commonly used in ATAs and Soft Phones, you can use our STUN server at: stun.sip.us