FreePBX, Asterisk, Issabel

How Voice Logica connects

The PBX registers a SIP trunk to Voice Logica (way 1). Voice Logica creates the trunk and gives you a server, a username and a password; the PBX sends a REGISTER out, so no port forwarding is needed.

• Calls routed to this trunk are answered by the AI agent.
• The agent transfers callers back over the same trunk; they land in the trunk’s context (from-pstn) and the inbound route rings the extension.

1. Create the connection in Voice Logica

Telephony - Trunks - Connect your PBX, pick FreePBX / Asterisk and the agent. Keep the dialog open until it says Connected.

2. Add the trunk

FreePBX 16 / 17

1. Connectivity → Trunks → Add SIP (chan_pjsip) Trunk. Trunk Name voicelogica.
2. pjsip Settings → General: Username and Auth username = the username, Secret = the password, Authentication = Outbound, Registration = Send, SIP Server = the server, SIP Server Port = 5060, Transport = 0.0.0.0-udp, Context = from-pstn.
3. pjsip Settings → Advanced: From Domain = the server, DTMF Mode = RFC 4733, Qualify Frequency 60.
4. Codecs: alaw first, then ulaw.
5. Submit and Apply Config.

Issabel / Elastix

1. PBX → PBX Configuration → Trunks → Add SIP Trunk.

2. PEER Details:

   host=<server>
   username=<username>
   fromuser=<username>
   secret=<password>
   fromdomain=<server>
   type=peer
   insecure=port,invite
   dtmfmode=rfc2833
   disallow=all
   allow=alaw&ulaw
   qualify=yes
   context=from-pstn

3. Register String: <username>:<password>@<server>/<username>

4. Submit and apply.

Plain Asterisk (pjsip.conf)

[voicelogica]
type=registration
outbound_auth=voicelogica-auth
server_uri=sip:<server>:5060
client_uri=sip:<username>@<server>
contact_user=<username>

[voicelogica-auth]
type=auth
auth_type=userpass
username=<username>
password=<password>

[voicelogica]
type=aor
contact=sip:<server>:5060

[voicelogica]
type=endpoint
context=from-voicelogica
disallow=all
allow=alaw,ulaw
dtmf_mode=rfc4733
outbound_auth=voicelogica-auth
aors=voicelogica
from_domain=<server>

[voicelogica]
type=identify
endpoint=voicelogica
match=<server>

3. Send calls to the agent

• A short code: Connectivity → Outbound Routes, a dial pattern (for example 9|. or a dedicated code) using the Voice Logica trunk.
• A number, an IVR option, a ring group or queue failover, after-hours calls: send it to a Misc Destination that dials over this trunk.

4. Let the agent transfer back to extensions

Connectivity → Inbound Routes for calls from this trunk, with the extension as destination (or a route whose DID matches the extension numbers). Then, in Voice Logica, add the extensions as transfer destinations: an External destination with the extension number (for example 201), or a User with that extension. Use Warm transfer.

5. Check it works

• The dialog says Connected. On the PBX, asterisk -rx "pjsip show registrations" shows Registered.
• Dial the route from an extension: the call appears in Calls.
• Ask the agent for a colleague: the extension rings.

Good to know

• Behind NAT: set External Address and Local Networks in Settings → Asterisk SIP Settings (Issabel: PBX → Tools → Asterisk SIP Settings).
• Turn off SIP ALG on the office router.