[Help] Unstable sip calls Asterisk 13.1 on VPS

Asterisk 13.1 LTS certified without FreePBX
followed this guide: http://www.mikeslab.net/?p=330

DigitalOcean 512mb vps, firewall disabled for testing

I’m trying to configure a really basic server to make sip calls between devices.
I’ve installed Asterisk without FreePBX, because I want to manage the config files manually.

I’ve added a simple dial plan and created some extensions. I’m using two laptops with Zoiper installed to register with the server and make calls. The problem is that it works 50% of the time. Sometimes it works just fine and I get two-way audio. But I randomly get one-way audio problems.

I’ve never used Asterisk before and I don’t have any experience with VoIP servers so I’m not sure what config files to post. Please let me know if you need more info to help and I’ll post them asap.

Thanks in advance.

sip.conf and extensions.conf

Together with logging at verbosity 5 or higher.

SIP debugging may be necessary as well.