How to modify the SIP login name

Hi Mates,

currently I have connected all of my family members by using It is an asterisk provider running ast + a modified freepbx version. If i set up an extension then I can login with my SIP phones by using the loginname bungee-1000 for the extension 1000. The prefix “bungee-” is used to distinguish between the different tenants on the same pbx System.

Now I have running an own AST+freePBX on a vServer. I want to switch all family members to my own productive PBX now, by changing the DNS A record pointing to the new productive system with the possibility to switch back immediately during maintenance. Currently I have to login to my own PBX by using the extensionnumber ONLY.

How can I change the configuration of my vServer to accept a loginname like bungee- ?

I found a the following thread:

Should that function be the key for a resolution of my problem or does anybody know a better way.

I would be very appreciated if someone could help me in this issue.

Thanks and kind regards,

Well this way the PBXes is the main box passing calls to your other box.

You get full use of your box.

OK login at
create a exten now go back into the exten do down to the box
dial and change it to sip/[email protected]

save it

Your box must allow for the inbound sip calls

If I create an IVR I create an inbound route for it.

say my exten on PBXes is 7000 and I want all calls which hit it to hit my IVR I create a rule for DID 7000 to the IRV I want

If you call the 1.50 a month DID I have at for testing it get forwarded to server in new orleans LA via my sip address ([email protected]) so no outbound charges for that “leg” of the call
That box then takes the inbound call and sends it out via SIP “FREE” trunk to my Offices / cell phone “paid”

OK I have a few folks who have @home running at home behind
a cable or DSL router. here is what I have them do.

Sign up at PBXES.COM (so the box is always up)

And point the PBXES exten to the @home box (that way they get the UPTIME of the remote hosted box VM if the @home is down)
Using Vitelity and the subaccount feature they can alway have the outbound calling on both boxes.

use sip/[email protected] in the dial rule and set your box to accept the call
handle it as you would any call. You can setup a inbound route based on the DID (the pbxes exten)

They get the best of both worlds FREE / CHEAP remote pbx and the @home box use for FAXes / IVR / CONFerence calling

Hi Bubba,

your advice sounds promising :wink: Unfortunately I dont understand exactly your instructions. At this time all 6 existing sip phones are registered with I cannot see any option in the interface to ‘point’ my existing ext. to my own PBX.

Additionally i cannot configure to use sip/[email protected] as destination, I can only choose one of my existing trunks.

I think that I am totally wrong and dont understand what you mean. I want to use only as backup system to my own productive system with root access.

Please, could you be so kind to describe more detailed what I shall configure on my own pbx and on pbxes? :wink:

Thanks and kind regards,