Switching to chan_pjsip Trunk

Beginner here. So I’m attempting to add a new chan_pjsip trunk in GUI but I only have the chan_sip PEER Details script from the ISP, which looks like this:

username=username here
secret=secret here

How do I configure the chan_pjsip trunk with these details?
Do I have to contact the ISP to tell me the PJSIP configuration?

You can add a PJSIP trunk in the Web GUI by going to: Connectivity → Trunks → + Add Trunk → + Add SIP (chan_pjsip) Trunk.

The individual settings are set on the pjsip Settings tab.

You should be able to match your settings from your SIP trunk to what PJSIP is asking for. In your example I have no idea what are place holders that you blanked out for security reasons and what are actual settings used by your trunk.

If you are having trouble after that your SIP provider might have some specific instructions for you on how to get a trunk setup that works with their infrastructure.

Doesn’t do anything in this context.

Syntax error. Questionable choice of value.


insecure=port is questionable

Questionable value, although possible.

Generally trying to translate a provider chan_sip configuration is not a good idea. The obvious settings will work in most cases, although, it is possible that you do need from_user.

Better to ask them what they require in the REGISTER and INVITE requests.

I do have fromuser but I changed it for being sensitive info.

All the settings above work with Elastix chan_sip as they are.

Now I am just trying to see if I can configure chan_sip on FreePBX but I am having issues adding a second NIC.

I will keep digging.

Generally ITSP provided settings work, but often not well. The failures are often in terms of security, so they don’t affect normal operation. The problem when translating them to chan_pjsip is that you end up trying to find ways to emulate things which weren’t necessary in the first place.

username isn’t used by chan_sip in this context, so ignore.

type cannot be exactly translated, as chan_pjsip match priorities differ in FreePBX, but it should have been peer, in chan_sip, and you shouldn’t need to consider it for the translation.

fromuser should be obvious.

I don’t think the GUI supports anything other than insecure=port, although I think you can make this secure by specifying a port in the match for the type=identify section in the the conf file.

insecure=invite, means you use outgoing authentication, not bothway.

dtmfmode should be obvious, but I would question why they don’t say rfc2833 which translated to rfc4733.

canreinvite is direct media (and was renamed to that a long time ago, in Asterisk)

host is server.

I would hope everything else was obvous.

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.