Solutions · For developers

A SIP URI for every voice agent

The agent answers the calls that reach its SIP URI on the platform's switchboard. Your PBX or trunk sends the call there like any other SIP destination; the part before the @ chooses the agent.

How it works

Each agent has an address like sip:ag_7f3a2b91@pbx.swissai.dev, shown in the agents list. Calls are accepted only from PBXs you have declared in Telephony › External PBXs with their IP address: there is no username or password, the PBX is recognised by its address and can call every agent of that customer.

Audio is G.711 (alaw, ulaw). The agent picks up, talks, calls its tools and hangs up or transfers. Calls from an undeclared address are rejected, and an address that keeps trying is blocked for a while.

Example: Asterisk

The guide has the same example for chan_sip, FreeSWITCH and Kamailio or OpenSIPS.

; extensions.conf: extension 200 rings the agent
exten => 200,1,Dial(PJSIP/ag_7f3a2b91@swissai,60)
 same => n,Hangup()

; pjsip.conf
[swissai]
type=endpoint
context=from-swissai
disallow=all
allow=alaw,ulaw
aors=swissai

[swissai]
type=aor
contact=sip:pbx.swissai.dev:5060

Three ways to bring a call to the agent

Testing

From the dashboard you can call any agent from the browser before touching the PBX; test calls count as minutes like any other call. The transcript and every tool call appear live.

Frequently asked questions

Do I need to register my PBX with a username and password?

No. You declare the public IP address of the PBX and the platform accepts calls from it. If the address changes, update it in the dashboard before sending calls.

Which codecs are supported?

G.711 alaw and ulaw.

Can the agent transfer the call back to my PBX?

Yes. The built-in transfer tool sends the call to the fallback number you configure on the agent.

Related pages

Try it with your own calls

Create an account, set up the agent in the dashboard and call it from the browser. 30-day free trial with 100 minutes, no card required.

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