Cant get second DID to ring on specific lines of my aastra phone

Hello all,

Im running PBX in a flash version 1.4, and i have 2 DID’s from vitelity. My problem is that i want my first DID to ring lines 1 and 2 on the aastra phones and the second DID to ring lines 3 and 4. There has to be a way to do this right?? I mean i shouldnt need to get a second sip phone at each desk with a seperate extension for my second DID.

If i could get the server to indicate which DID was receiving the call (and pass that to the phones display) then it wouldnt really matter what lines it rang on as the user would be informed as to how he should answer the phone (with company a’s greeting or with company b’s greeting).

Also when i look at my trunks my secondary trunk INbound page it gives me a warning that its not in use and it cant be used for outbound calls… Isnt this my INBOUND trunk?? im confused.

my config for that trunk is:




USERNAME:[email protected]

Does that look about right?

Also, my system status reads ip trunks online are 4 and ip trunk registrations are 2. that would indicate that at the very least im communicating with vitelity properly… i think.

Any help or suggestions would be great!



I didnt have the route selected for the trunk sequence in outbound routes, BUT i still have the other issue with identifying which DID is calling inbound…

sorry guys…


In freepbx if you have a any DID any CID that will catch all inbound numbers and over write any other inbound routes.

If you want to send calls to different places for different numbers you need to have a did in your in bound routes that matches to that number.

so create a inbound route for each number then send that inbound route to where every you want it to go.

command “asterisk -rvvvvvvvv” which is a great debugging tool will show you which number is coming in so if for some reason you are not routing correctly you can use that to see whats going on.

if you have an incoming call and do not see anything in the debugging you might not have your inbound route setup correctly.

you can use “sip set debug on” in the CLI(command line interface) which is the accessed through asterisk -rvvvvvvvvvv

That helps me troubleshoot sip trunks.

You can do the same for iax2 “pronounced EEKS”

That debug method is great now i can see soo much more about whats going on in the background, thanks Nemus. Where would i find more info on how to use these cli tools.

When i did asterisk -rvvvvvvvv, i got a different cursor than if i was logged in normally. is this because its in verbose level 8 rather than level 3? How do i get back to the normal cursor? I tried putting it back to level 3 by issuing a “asterisk -rvvv” command but im obviously doin something wrong.

Thanks for helpin the noob.


I fixed the trunk issue by adding the trunk to a route. and i fixed the other issue of knowing which DID is calling in by filling in the CID name prefix area under the inbound routes page. the only problem now is that the caller id on the phones read like so when an incoming call is received:


so it inserts the prefix, like i want it to, but trys cramming the number in on the top line, then writes the full number below it. Any way to get it to not write the partial # after the prefix, and just have the prefix on the first line the number on the second??

Thanks again,