SIP trunking on a dedicated voice line

Hi Guys

Our ISP has installed a dedicated voice line and supplied us only the following information.

IP address of IPPBX ( this will be our asterisk server)
SiP proxy : enabled

and the necessary codecs that are required.

I am trying to find out where I would register to these this trunk.


The first three in the OS. The codecs in sip.conf, or the equivalent pjsip .conf file.

You will also normally need to know the IP address of their end. There is quite a lot else that you should probably know, although some of it you will be able to guess and some you can deduce from error messages.

(It is not strictly necessary to know which codecs they support, as the default is to offer all ones included in your Asterisk build.)

So what sip settings do I use to create the trunk.I have been researching and I gather that this is a setup
without registration so I need to know which settings should I use when creating the trunk.


The minimum requirements are:


You will probably need more, but you haven’t provided enough information to provide that.

You will need something in extensions.conf, but you haven’t provided enough to allow that to be provided.

Look at sip.conf.sample, in the source distribution (or the equivalent pjsip file, for other possible options.