IP Address Range for a Voip Trunk


I have a 6102 with admittedly old firmware ( which works great for us until now. My inbound VOIP provider (Flowroute) now sends incoming calls from any of 4 IP address ranges of 16 ip addresses. The UCM rejects the calls unless there is a trunk with each of the 64 ip addresses! This is untenable to specify. I tried to enter both the range (e.g. and the specification in the “Host Name” field of the trunk. The former does not work (though it is accepted), and the latter is not accepted as an input in that field. Is there a way to specify an IP address range for a trunk in this version of the firmware? If I upgrade the firmware are IP Address ranges accepted?


None that I know of.

If your provider is sending from a range of trunks you need a trunk and inbound route for each IP.


How was your original trunk with FR set?


My original trunk had a single url: sip.flowroute.com. Now there are 64 iP addresses (4 sets of 16) from which they can send calls.


The notion seems odd that 64 IP addresses would be needed when a FQDN is present. Is it possible that some of the IPs being seen are blacklisted?


Asterisk is definitely capable of CIDR notation for endpoint matching, I’m surprised the UCM doesn’t allow for this.


They are not blacklisted in my UCM. Flowroute claims that their FQDN refers to a range of 16 ip addresses (which it may for sending), but if I just enter that FQDN in a trunk, incoming calls can come from only thre first IP addresss in the range. When I added another trunk for other IP addresses in the range, I can receive calls from them. The UCM does not accept CIDR notation. The error message says it is not a proper host name.


Well, I wondered as I saw that FR removed the UCM setup link, but saw other systems using the same FQDN as you.

I can only suggest contacting FR and/or GS support.

Maybe someone else is using FR and can advise.


Cheat: Buy a Raspberry Pi and install Kamailio on it. Configure it as a SIP proxy with brain dead stupid routing from Internet to your UCM. Then set it up so that it’s your destination for your incoming SIP provider. Then your UCM only needs to know the IP of your Kamailio server (aka, the Raspberry Pi).

Or, in other words: set up a VERY small SIP proxy that you can more easily control than the UCM, send all your incoming calls to that SIP proxy, and then send calls from the SIP proxy to your UCM. You manage the provider’s IP list on the SIP proxy and your UCM only has to allow calls from your SIP proxy.

IP Bound to the registered SIP trunk on incoming calls

I have had similar issues and would like to see this fixed as a feature of new firmware.


Closing this topic to prevent chronojacking.