Need help about configuration users for ari

Hello,
In fact I was setting up a secure voip network using tls and srtp with blink soft phones
By configuring tls I had no problem,
However, when I downloaded and installed srtp I only managed to have one blink subscriber.
And I have as error: error ari/config.c no configured users for ari.

Unless you are using ARI, then ignore the ARI error. It is not the cause of any problems.

Ok, why i can’t have any blink subscribers?

You haven’t given enough information. What does “blink subscriber” mean? What is the configuration? What are you actually trying to do?

In fact I am setting up a secure voip network using tls and srtp.
Well with tls the calls go through without problem but as soon as I downloaded and installed the srtp library I can’t even register a client.
I use blink as a softphone.

chan_sip is no longer supported, and will be removed in less than a year.

You have ignored other deprecation warnings.

You have ignored a warning about having something other than a plain file in your certificates directory. I’d suggest fixing that before worrying about anything else encryption related.

You’ve ignored a warning about an unresolved reference in your dialplan.

You haven’t provided your account page. The online documentation is too incomplete.

Note that Asterisk is not a proxy, although it might tolerate configuring it as a proxy (but not to the extent of then pretending to a system other than that set in the account section). Actually it looks like they make statements that SIP Proxy is the same as the server and registrar address, which is not actually true in general, but is probably true enough here to get away with. It looks like it would really prefer to work with DNS SVR records.

The only media encryption mentioned in the Blink web site is zrtp, which Asterisk does not support.

We generally prefer plain text, not images, of configuration files and logs, so we can search them.

Looking at VoIP: SIP-over-TLS and sRTP: Blink it looks like it does support SRTP. Have you gone through the procedures listed in this article, except possibly for NAPTR records?

That also suggests it uses an obsolete PJSIP library. Whilst the chan_pjsip and chan_sip libraries should interwork properly, that’s a good sort of reason for using the, supported, chan_pjsip, on Asterisk.