Inbound Calls are Failing

A common request received by our Support team is for assistance with troubleshooting why inbound calls are not working. This article breaks down common causes of inbound call issues and what steps are needed to troubleshoot such issues.

 

Failing inbound calls can be broken into three common categories:

 

1. The SIP trunk is not registered.

 

2. The SIP trunk is registered to the wrong IP Address and/or port.

 

3. Trunk is registered correctly, but inbound routes are misconfigured.

 

Note: If you are troubleshooting a system which was previously working and no recent changes were made, try performing a simple reboot of the equipment before going in-depth into the troubleshooting process. 

 

How to Check SIP Trunk Registration

SIP.US offers the capability to view the active registrations on a SIP trunk in the customer portal. To view this, hover over the SIP Trunking menu and choose the SIP Trunks submenu option. You can view all SIP trunks on the account on this page. Click on "View Trunk Registration Status" for a SIP trunk and a pop-up will display registration details. 

 

• For more information on how to register a SIP trunk, see our General Device Configuration guide or refer to our Device Setup Guides for configuration guides with common devices. 
• For more information on how to read the registration status details in the customer portal, see our SIP Trunk Registration Status guide. 

 

 

Contact Address Doesn't Match Source Address

Note the line below “REGISTRATION STATUS.” If you have successfully registered your PBX/Phone with our service you should see a string like this (the string is edited out of the image above for privacy/security):

 

• sip:Trunk#@<YOUR CONTACT IP>:<YOUR PORT>;alias:<YOUR SOURCE IP>:<YOUR PORT>
  Example: sip:123456789@142.93.###.###:5060; alias=142.93.###.###:5060

 

The key thing to confirm is that both strings match the same IP and port combinations. This is because the first IP in the registration string reads from the CONTACT header of the SIP REGISTER packet sent to SIP.US. The second IP in the registration string displays the public IP from which the REGISTER packet originated.

 

The CONTACT header is user-defined in the SIP device, so SIP.US displays what it receives and this is how the two IPs can potentially conflict. SIP.US attempts to deliver inbound traffic to the IP defined in the CONTACT header. If your system does not receive inbound calls and there is an IP conflict, that can serve as an indicator of where to look in your configuration to apply the necessary changes. 

 

There are a number of things which can cause IP conflicts, including user error in the manual configuration, a power outage affecting a non-static IP (in which your ISP changes your public IP address but the SIP device does not detect the change), a custom-implemented NAT policy, or a firewall/security solution automatically applying NAT changes you're not aware of. 

 

Security/Firewall-related Issues

Port conflicts in the registration are also common with NAT policies and firewall port remapping. Most PBX systems use port 5060 by default. If the port is some funky number like 49784, chances are that your firewall/router is doing Port Address Translation (PAT). This doesn't always mean your calls won't work, but calls may not work intermittently (the cause of the intermittent issue is beyond the scope of this article). The best solution is to use 1:1 port translation so that your port does not change through translation.

 

SIP.US cannot help with troubleshooting a firewall, so you will need to reach out to the vendor in these scenarios. However, our Support team can assist in providing information such as what IPs and ports we see SIP responses from to assist in your troubleshooting. 

 

Note: If you have proper port forwarding in place, your system can function properly. This is all outside of the realm of support for SIP.US though. 

 

Contact Address is a Private IP Instead of Public IP

Inspect the IP address and port very carefully in the SIP register string. If the first IP address displayed falls within the following IP ranges, we will be unable to deliver your calls:

 

10.0.0.0 – 10.255.255.255

172.16.0.0 – 172.31.255.255

192.168.0.0 – 192.168.255.255

 

These are private IP addresses which are used within your Local Area Network (LAN). Private IP addresses are not routable on the internet. Your ISP gives you/your router a public IP address which is not in those ranges. That public IP address is the one that needs to be in the string. To find your public IP address, simply query a search engine “What is my IP.” Your public IP should be returned as the top result. There are also many free lookup tools available that will appear in the same search results.

 

If you cannot seem to get your public IP and port combination to show up in the string, you may want to use a STUN server. Within most phones/PBX systems there is a field to input a STUN server. You may use ours by putting in “stun.sip.us” in the field for “STUN Server” on your system and use port 3478. 

 

SIP Device Routing Issues

If you are registered correctly, but calls are still failing to reach you, it's time to consider your inbound routes. Each SIP device uses a different configuration for inbound routes. Please refer to the SIP.US Device Setup Guides to check for your device and ensure the inbound route configuration is correct. 

 

SIP.US Inbound call INVITE headers are delivered in the following formats by default:

• FROM: 10-digit format (NPANXXXXXX)
• TO: 11-digit format (1NPANXXXXXX)

 

If your number is 333-444-5555, we will deliver it as “13334445555” to your SIP device. Many SIP systems have a “wildcard” inbound route that allows any number delivered to the system to ring and reach a destination. If you are unsure how to program your device, we recommend using wildcard routes for troubleshooting and then locking the system down later.

 

Note: SIP.US applies an ANI Strip value of 2 to the FROM header for most numbers to deliver a 10-digit format. If you need to modify the FROM header, go to the DID settings of the number you would like to modify and set the ANI Strip to 0. You can find this setting in the customer portal by navigating to the expandable SIP Trunking menu option and choosing Telephone Numbers. This will load a page with all DIDs on your account. Click the pencil edit icon next to the relevant DID and an option table will load with the ANI Strip setting.

 

Finally, be sure that you have actually routed calls somewhere. You need to specify a destination for calls. You could direct your calls to an extension, voicemail, an automated attendant, etc. We recommend getting an extension set up and directing calls there to start out. However, if you are having trouble, you may want to switch to voicemail or an automated attendant to eliminate the extension as the issue.

 

Network Security Restrictions

Inbound calls may show as Trying in the SIP.US customer portal and you may notice that calls are not reaching your phone equipment at all. If you have network security in place, such as a firewall, you will need to ensure your network is configured to accept all SIP traffic from SIP.US gateways. Further ensure there is not any packet inspection or other security restrictions within the SIP device. 

 

• Please see the article on Interconnecting with SIP.US for the complete list of all IPs we send traffic from. 
• If you are experiencing missing audio from your calls, please refer to the article on Direct Media Delivery and Missing Audio. 

 

Other Causes for Failing Inbound Calls

If nothing else stands out as a reason to why you're experiencing issues with inbound calls, a review of your account may reveal a cause. Please consider the following:

 

Is the DID assigned to the correct SIP trunk with an active billing plan?

Ensure there is an active SIP trunk to pass the calls to the equipment and that the DID is assigned to the correct SIP trunk (check Primary and Secondary). If you are using an unlimited channel-based plan, ensure you have available channels to process the call, otherwise, SIP.US will automatically fail the call and post an alert on your account. Learn more in this article about ALERT: CHANNEL LIMIT EXCEEDED. 

 

Does the account have a positive fund balance?

If the account balance falls below the negative threshold, all call traffic on the account will automatically cease. SIP.US provides billing tools in the Payment Center of the customer portal to set up Auto Replenish for when funds get low and to receive low balance notifications. We recommend setting up both of these tools, especially if calls are critical to your operations and you wish to avoid service disruptions. 

 

Note: CDRs will not populate when the balance is below the negative threshold. 

 

Are there any call forwards in place?

If there is a call forward on a number, it can bypass your phone equipment entirely. You can check this in the customer portal by hovering over the "SIP Trunking" menu option and then clicking on the "Telephone Numbers" submenu option. Find the relevant DID and check that the PSTN Forward field is not forwarding calls out to an external number. Next check the PSTN Backup. If your calls are forwarding to that number, it means your issue is related to one of the topics in the previous sections of this article. 

 

 

Do the inbound calls appear in the Call Detail Records (CDRs)?

If you do not see any of your inbound call attempts to a particular number and there is no ringing, there are a few possibilities to consider.

• Check that the account balance is not below the negative threshold (see above section). 
• Does the DID you're testing show in the account?
• Was the number recently ported into your account? - If yes, open a ticket with our Support team.
• Is it possible the number ported out? - It is possible your client or the previous MSP ported the number out and it was not documented. If you are unsure or determine there was an unauthorized port out, please contact our Support team immediately for assistance. 

 

If you have gone through this document and still cannot receive calls, please open a support ticket with SIP.US at support@sip.us and we will assist you further with troubleshooting.

 

We also recommend reading Call Failover Options for adding redundancy to your SIP configuration in the event that inbound calls fail to your primary setup.