How to modify the sip login name

Hi Mates,

currently I have 7 extensions connected to a multi tenant asterisk provider connected. They are using not only the sip extension number as login name for the sip clients (pbxes.org). They modified it to bungee-8000 and password in my case to distinguish between the different tenants, even if the tenants are using the same extension numbers.

I created an own dns record, pointing to pbxes.org and put it into my sip phone configs.

Now I have my own astersik server running and I want to switch all sip phones by changing the dns record to my own box. Unfortunately my box accepts only the extensionnumber and password as the login for a specific extension.

What should i do that the prefix “bungee-” is accepted by my own box (root access)?

During maintenance of my box I want to switch all phones by simply changing the dns record?

I would be very happy if someone have an idea how to do that.

Kind regards,
Enno

First question is: who created your own Asterisk “box”? It’s almost impossible for a fresh install of Asterisk to be picky about user name. How is your box configured? sip.conf/users.conf, extensions.conf?

Hi,

my box accepts sip phone logins with the extensionnumber, thats defaut and normal after a usual installation. But my pbx provider modified his box, so any sip phone has to log in as -.

username is my tenant name on the box. (pbxes.org)

How can i change my configuration that a phone can aqlso connect with bungee-

Would you like to see my configs? I only configured one extension (8000) and want to connect as bungee-8000 and the extension password.

Kind regards,
Enno

[quote=“bungee”]my box accepts sip phone logins with the extensionnumber, thats defaut and normal after a usual installation.
[/quote]

Maybe I’m missing something. Which Asterisk version you have installed? I don’t know of a version that has default installation with extension number. Could it be AsteriskNow instead? TrixBox? These could be very different from Asterisk. Or maybe you installed a binary Asterisk package in CentOS, Debian? (Which Linux distro, any way?)

Can you post your sip.conf and extensions.conf files

Thanks,
Suresh

Hi Mates,

sorry for the incomplete details. I am using FreePBX as well. Its generating my config files. Now I understand that the loginname for a sip device depends on the section within my sip_additional.conf:

Example! SIP Phone Username 8000:

[8000]
type=friend
secret=xxxxxxxx
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=8000@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/8000
context=8000
canreinvite=no
callerid=device <8000>

Example SIP Phone user name bungee-8000:

[bungee-8000]
type=friend
secret=xxxxxxxx
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=8000@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/8000
context=8000
canreinvite=no
callerid=device <8000>

If I change the Start of the section from [8000] to [bungee-8000] then it works fine with the sip phone login. Unfortunately the contexts are then broken, incoming routes are wrong and I dont know about all other side effects. I think thats not the right way. Additionally if I change something within freepbx the file will be overwritten.

Do you think that that there is an easy way to circumvent this behavior without hacking FreePBX? I yes i will post all necessary config files as well.

Thanks for your help!

Enno

Don’t know FreePBX, but the easiest way to do such custom configuration is plain Asterisk. What features do you need from FreePBX?