Asterisk (self-hosted)
SIP setup
Asterisk is the original open-source PBX. If you run it yourself (or through a VAR who gave you admin access), every extension is BYOD by definition. You add the SIP peer config by hand or through an admin UI, copy the credentials, paste into SoloDial.
Five to ten minutes
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01
Log in to your Asterisk host
SSH to the server running Asterisk. You'll edit config files directly (or use Asterisk Manager CLI if that's how you work).
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02
Pick or create an extension
In /etc/asterisk/pjsip.conf (modern) or sip.conf (legacy chan_sip), find the extension you want dedicated to the AI or create a new one. Example PJSIP config below.
pjsip.conf showing endpoint / aor / auth sections for extension 7100 -
03
Add a PJSIP endpoint + auth for the AI
Minimum viable PJSIP config block for SoloDial: [7100] type=endpoint context=from-internal disallow=all allow=ulaw,alaw,opus auth=7100-auth aors=7100 [7100-auth] type=auth auth_type=userpass username=7100 password=STRONG_PASSWORD_HERE [7100] type=aor max_contacts=1
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04
Reload PJSIP
At the Asterisk CLI: asterisk -rx "pjsip reload". Confirm the endpoint is visible with: asterisk -rx "pjsip show endpoints".
CLI output from pjsip show endpoints -
05
Copy credentials into SoloDial
Extension = 7100. SIP Username = 7100 (or whatever you set as username). SIP Password = the STRONG_PASSWORD_HERE value. Registration URL = your Asterisk server's public hostname + port, e.g. pbx.yourdomain.com:5060.
- Asterisk defaults to denying WAN registration. Open UDP 5060 (or 5061 for TLS) on your firewall AND add SoloDial's egress IP to your PJSIP transport's 'allow' list (or use match parameter on an identify section).
- If you're running chan_sip instead of PJSIP, set 'nat=yes' on the peer. PJSIP handles NAT correctly out of the box.
- For RTP, open the RTP port range your asterisk config sets (default 10000-20000 UDP). Without RTP open, registration will succeed but calls will have no audio.