This guide can be used to set-up your ShoreTel Connect system with SIP.US. It assumes you have a static IP and your firewall is capable of static port translation.
Step1: Set up the codecs for use with the trunk. SIP.US supports G711ulaw (a.k.a. PCMU) and G729. Navigate to "Features" ---> "Call Control" ---> "Codec Lists" and create a new codec list as show in the image below
Step 2: Navigate to "System" ---> "Sites" then select your site and assign the codec list created in Step 1 as shown in the image below:
Step 3: Navigate to "Trunks" ---> "SIP Profiles" and create a new trunk profile. Use the default system parameters and add the two custom parameters shown in the image below:
Step 4a: Navigate to "Trunks" ---> "Trunk Groups" and create a new trunk group as shown in the image below (note that because we are going to do IP Authentication for this trunk, you DO NOT need a Username or Password entry):
Step 4b: Select the "INBOUND" tab on your Trunk Group. The major changes here are the "Number of digits from CO (this should always be 12 for SIP.US) and the Destination:
Step 4c: Select the "OUTBOUND" tab on your Trunk Group. You should use the image below as a guide, but your selections here will vary based on your needs:
Step 5: Navigate to "Trunks" ---> "Trunks" and create a new trunk according to the image below:
Step 6: Add 22.214.171.124 to your Trusted IP List. Failure to do this means no calls for you.
Navigate to your SIP.US portal, and select "Modify Trunk" underneath of your trunk. You will need to check the "IP Auth" box and input the external IP and Port we will see your traffic come from. Note, if your traffic does not come from this IP and port, we will not allow it to make calls. I don't know if we can specify strongly enough that the IP Address AS WELL AS THE PORT must match.
Once you have done this you should be able to make and receive calls on through your SIP.US trunk. If you do not have audio but can see calls in your Call Detail Records on your SIP.US portal, please open a ticket or give our support line a call, and they will enable a media proxy which should fix the audio issue.