Associating a call queue with a trunk / DID

Hi all,

I have my first asterisk + freepbx set up configured and running. Using Xlite as the client softphone. sipstation sip trunk is configured and is working fine, except for a few minor call breaks. Need to attend this.

Can someone please guide me on how to route a DID to a queue. Thanks.

You do that from Inbound Routes section in FreePBX. Enter the DID and select your queue as destination.

Thanks Mikael, I could do it good. Created a queue and added some static members and selected the queue in the inbound route. Only later I realized that the order is important i.e the queue first and then the route :-). Thanks a lot for the help, queue is working now.

Can I ask one last question, this is with the digital receptionist. From what you said I could realize that for setting up an IVR I can go and set up a DID to use a particular IVR from the inbound section.

Now on the IVR/Digital Receptionist page, I can see that there are three repeating sections after the basic settings. Do you know which dtmf tones do they correspond to, that info is not avail on the page. Thanks again for the help.

On a voice call to an extension or pstn number, every 30 seconds or 1 minute there is a slight pause and the voice is inaudible for a while then it comes back. Happens for just a sec or two. Any clue what this could be ?

You can assign which ever digit or digits you want to the IVR options and you can create more than just 3 (though good IVR design is usually limited to 3).

As far as the pause, it must be something going on in your PBX, or possibly related to your Internet Connection??? There is no feature in the PBX nor any such behavior in SIPSTATION that would create this incident.

Dear Philippe,

It could be with the freepbx/asterisk set up that is causing the problems. Just now I finished an outbound call to a PSTN number and the call got disconnected repeatedly at the 30-32 seconds mark. It happened yesterday also and after that I updated the freepbx to the latest stable version.

The intermittent break is still there but that happens at a longer duration than the above, may be every 3-5 mins and hence couldnt hear it this time. I can see that there is a similar problem like this specified in the forum at,

The server load, memory everything looks good and freepbx is running on a public ip with no nat. Extensions are at remote locations behind nat-ed routers. The computer from which I was making all the calls were on wi-fi. Hence I made a test call directly connected to the lan switch bypassing wifi. Still the problem persists.

To enhance performance today I disabled innodb and bd in my.cnf and memory consumption went down from 60% to 30% in the freepbx admin console. Hope the info is of some help. Should I paste any specific settings here ?


Could you please tell me how to check the debugging on the server ? Is it by logging into asterisk console with -vvvvv option. On the client side I figured out that I could enable debugging in X-Lite. And how do the hang up messages look like ? I’ll post the entire debug output here for reference :-). Thanks for the help. Looks like I am closer to solving this.

Btw, there is a small development, after I rebooted the system and reloaded sip call disconnection is not happening but the intermittent blackouts still exist. Again as you said a debug output might help. Will try that. Anything else should I check ?

For the firewalls at the server side, will it be fine by checking it via the button provided in SIPStation ?

I dont know whether its you Philippe/Mikael who responded from support, the update is there are no call drops at the trunk side and asked me to check the drops in registrations. Asked to do a sip reload too. While making outgoing calls the calls hung up at around 30-32 secs the first time and second time it went through.

your hangups sound like they are either related to something on the firewall/router (even though you say it has a public ip, I’ve seen plenty of issues still related to a firewall/router and remember you have remote extensions that also have their own that may also be having issues). Alternatively, it may be the phone itself initiating the hangups.

You’ll need to turn on SIP debugging and see who is initiating them and why.

Just competed the firewall test and it shows green in the pbx console, under SIPStation module. I will set up port forwards at my home router. My connection is ADSL router --> Wireless N router --> Desktop -> Xlite. On desktop the ports are open as I enabled them during install. Will set up port forwarding in both wireless and adsl routers.