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.
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?
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.
[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?)
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:
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.