How to modify the sip login name

Asterisk Asterisk Support
#1

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

#2

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?

#3

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

#4

[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?)

#5

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

Thanks,
Suresh

#6

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

#7

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