How to setup Asterisk13 for noobs?

Hi all,

I am struggling for a few days now trying to get asterisk13 running one LEDE 17.1
I have to say I am a total noob with regards to voip. I was so far connecting directly to my provider on a dedicated phone via wifi, bridged to WAN interface since I never manage to pass thru openwrt.
I recently understood that Asterisk was my answer, so that it would connect to my provider and serve multiple sip phone in house. As said I'm all new to voip and can only find old tutorials referring to asterisk 1.8, having no UCI config files.

In short, what is the minimum required in /etc/config/asterisk so that I can connect to my provider, and connect two phones to it behind my router?

All what I know is my provider's Domain, Proxy, SIP register server, Port and my account details...

Looking forward, I'll be glad to update the wiki with a solution

See this thread...

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Although I cannot provide an easy way to SAFELY (!!!) install and configure asterisk, I tried this myself and ended up learning a lot of things. In my opinion this is not a set-and-forget-thing, I highly recommend to step into a long and flat learning curve. Especially important is understanding what a context is and how to prevent script kiddies abusing your phone line, the basic configuration usually is prone to script attacks. Unfortunately I stopped my studies a while ago, asterisk still isn't online here, so I cannot give you a failsafe config currently. Hopefully I get time soon to complete my installation.

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The more I understand Asterisk, the more it seems over the top for what I need: connecting one or two sip phone behind my router to my sip provider. Not mentioning the increased attack surface...

Out of curiosity, I looked a bit further at available package for stun client and came upon siproxd, which seem to do exactly what I want :

Siproxd is a SIP application layer proxy for SIP-based softphones hidden behind an IP masquerading (NAT) firewall. It includes an RTP data stream proxy for incoming and outgoing audio and video data. Multiple local UAs (SIP phones) can be simultaneously masqueraded. All configuration is done via one simple ASCII text file. Its main purpose is to perform NAT traversal for SIP and the RTP traffic.

Now to testing!