Strange issue with setup of Inbound SIP from IPCOMMS

I am a new user of Freepbx and have followed a number of demo and examples on line to setup and test my installation.

However, I have spent the better part of the last 3 nights and cant figure out how to make inbound calling work without have the “Allow Anonymous Inbound SIP Calls?” on and no trunks created.

Here is the setup. I did a base install of PBX in a Flash, ran all the updates, fixes. Followed the instructions to edit sip_nat.conf.

I created one extension that rings to a softphone on my computer and can dial voicemail/other extension I created.

I even installed the Demo test from Nerd Vittles and am able dial to the test site successfully.

I would like to setup an IVR (whole point of this project), so I signed up for a test account with IPCOMMS.

Following instructions from the IPCOMMS <a href='> here ', I created the trunk, then added an inbound route with the DID Number set to the number provided and directed to the extension running on my softphone.

When I called the number, I will I got was a busy signal and no record of the call in the reports section. I tried to watch/diagnose by reading the sip debug information, but I have not been able to determine anything.

Next, comes the part I really cant figure out. If I remove the trunk entry and dial. I will get a “this number isnt in service message” and when I look at the reports section it is going to ‘s’.

I found this article on the Freepbx site here . However, when I use this I have to have a trunk created and it casuses the message to go to a busy signal.

Finally, if I remove the trunk and set "Allow Anonymous Inbound SIP Calls? to yes, then I receive the call no problem. However, from everthing I have read I this is a security risk and the provider (IPCOMMS) is actually providing the correct information.

Here is a piece of information pulled from the debug file that shows they appear to be providing the DID.

To: sip:[email protected]:5060;tag=as2996ec9c
From: sip:[email protected]:5060;tag=SD6u34001-gK0c764b2d

Any advice or suggestions? I imagine there is one setting somewhere that I have wrong and I am not able to piece together what it is from the forums/postings.

Thanks for any help that can be provided.

If it works with anonymous SIP turned on, why not leave it on. As long as you do not have an Any/Any inbound route, it is pretty safe, I think. I have had anonymous sip on for a couple of years and have had no problems.

Anonymous Sip works like a regular phone line. If the caller knows one of your DID’s, then your server will allow the call in, otherwise, it is refused. Same applies with a POTS line. If the caller knows your phone number, the call is allowed in. The key here is to not have an Any/Any route. Route all inbound calls by DID.


Thanks for the information. I wasn’t sure if I was going to open up some large security hole in my system and was having a difficult time finding information either way.

I will allow anonymous and monitor to make sure nothing strange happens.

Thanks for the help.