Plain Asterisk, FreePBX and Issabel: one extension, one AMI user
Any Asterisk 13+ box can hand a call to an AI agent with the same dialplan pattern VICIdial uses. AudioSocket is supported as an alternative to per-call RTP.
- Asterisk 13 or newer (chan_rtp for UnicastRTP, or res_audiosocket)
- manager.conf user with read = call, write = call
- UDP 5001-5999 (or TCP 9192 for AudioSocket) and TCP 443 open to app.bridgeside.ai
Asterisk in 3 steps
Add the dialer
Type Asterisk, host, AMI user and secret, and the context the agent should transfer calls into (for example from-internal).
Paste the dialplan
Drop the generated `_85XXX` extension into extensions.conf (or a custom context in FreePBX / Issabel) and reload.
Route calls
Send inbound DIDs, IVR options or queue overflows to 85 + dial code. Transfers use Local channels into your context.
What you paste into Asterisk
Either media path works; RTP is the default, AudioSocket avoids opening a UDP range.
- ✓UnicastRTP or AudioSocket media
- ✓Hangup and transfer over AMI
- ✓Transfer into any dialplan context
- ✓Caller id number and name passed to the agent
- ✓Same campaign portal, scripts and voices as VICIdial
; extensions_custom.conf
[bridgeside]
exten => _85XXX,1,Answer()
same => n,Set(PORT=${CURL(https://app.bridgeside.ai/api/asterisk/rtp?token=…&agent=${EXTEN:2}&channel=${URIENCODE(${CHANNEL})}&phone=${URIENCODE(${CALLERID(num)})})})
same => n,Dial(UnicastRTP/app.bridgeside.ai:${PORT}/c(ulaw))
same => n,Hangup()
; or, with AudioSocket instead of RTP:
same => n,AudioSocket(${UNIQUEID},app.bridgeside.ai:9192)Asterisk questions
Does it work with FreePBX?+
Yes. Put the extension in extensions_custom.conf, create a Custom Destination pointing at it, and route a DID or IVR option to that destination.
No dispositions?+
Dispositions and DNC write-back are VICIdial features. On plain Asterisk the outcome, summary and transcript are kept in the Bridgeside AI portal and exportable as CSV or JSON.
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Put an agent on Asterisk this afternoon
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