PJSIP config for Anveo Direct?

I looked on the Anveo website for a config, but they only support regular SIP, not PJSIP. Does anyone have a working pjsip.conf setup and dialplan for incoming/outgoing?

I’m using Asterisk 18.2.0 with no GUI.

Thanks in advance!

Why do they need a separate incoming configuration? In most cases you don’t.

Have you tried slotting their parameters into the “A SIP trunk to your service provider, including outbound registration” example in res_pjsip Configuration Examples - Asterisk Project - Asterisk Project Wiki ?

It seems they do IP authentication, so you don’t need any authentication section.

Looking at their FAQ, Anveo Direct: FAQ , the chan_sip configuration is rubbish, anyway, as it has insecure=invite, on a nominally outbound trunk, with no secret, which doesn’t make sense, as the insecure=invite will do nothing, and doesn’t apply to outgoing calls. It also has insecure=invite inbound, but no secret, and I doubt you can configure it with a password for your system. Also it has type=friend, but the section name is very unlikely to be what appears in the SIP user field (which I suspect is the original dialled number). Like most ITSP provide configurations, you should ignore it.

It looks to me as though the only unusual feature in an otherwise rather simple configuration, is the identify section and match lines for the four possible IP addresses.

There is no reason why a provider should support chan_sip and not PJSIP at the end it should be transparent for the provider as long as you send the correct parameters to establish a session

What I mean is that Anveo Direct provides an example configuration for sip.conf (chan_sip), but not for pjsip.conf.

Anveo Direct doesn’t use registration, apparently. It relies on static IP addesses. No user or pass.

Thanks for your reply. I’ll see what I can do, but it’s pure voodoo to me despite decades of experience in IT. Frustrating.



send_diversion=yes     ; Send the Diversion header conveying the diversion
                        ; information to the called user agent (default: "yes")
send_pai=yes    ; Send the P Asserted Identity header (default: "no")
trust_id_inbound=yes   ; Accept identification information received from this
                        ; endpoint (default: "no")
rewrite_contact=yes  ; necessary if endpoint does not know/register public ip:port


Use this configuration if your carrier authenticate calls based on your source IP

As I pointed out, their example configuration is bad, as is the case with most such provider supplied configurations. Even with chan_sip you should really start from first principles, even if the one they provide sort of works.

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