200ms agents in your dial plan.
Calls your Asterisk or FreePBX dial plan sends to Bland reach agents that respond in 200 milliseconds, on a voice stack Bland hosts end to end.
Bland + Asterisk and FreePBX
Connect Asterisk or FreePBX to Bland over SIP and put 200ms voice agents on your DIDs and extensions, with transfers straight back into your dial plan.
What Bland does inside Asterisk and FreePBX.
Built for your PBX.
Asterisk and FreePBX are a guided provider in the SIP setup wizard. Enter your public IP or hostname to start.
Discovery finds Asterisk.
Discovery resolves A, AAAA, and SRV records, probes SIP ports, and detects Asterisk or FreeSWITCH along with supported codecs.
Passwords that survive your config.
The password generator avoids ambiguous characters and anything that needs escaping in a PBX config or SIP URI, so it pastes safely into Asterisk.
Dial-plan extensions as targets.
Attach extensions like 2001 to the trunk, and point Transfer Call nodes at an extension or a SIP URI such as sip:2001@pbx.example.com.
Custom headers on transfers.
Map call variables to SIP headers on the trunk's outbound mapping, and they ride along on transfers to your Asterisk box.
How it deploys.
Start a trunk.
At All agents, open Dispatch, then Phone numbers, then SIP, click Add trunk, and choose Asterisk / FreePBX.
Enter your PBX address.
Use the public IP or hostname of your Asterisk or FreePBX server and let discovery recommend port and transport.
Authenticate.
Trust Bland's signaling and media IPs as SIP peers, or set up registration with a generated password, and save the trunk.
Attach extensions and test.
Attach the DIDs and extensions to route, press Test connection, and call an extension routed to Bland.
Our FDE team scopes the integration, maps Bland to your stack, and stays on through go-live. Enterprise deployments go live in 30 days.
Drop Bland into your editor.
MCP server runs in Claude Code, Cursor, and any MCP-aware editor. Or hit the REST API directly.
curl -X POST https://api.bland.ai/v1/calls \
-H "Authorization: $BLAND_API_KEY" \
-H "Content-Type: application/json" \
-d '{
"phone_number": "+14155552120",
"task": "Qualify refi lead",
"voice": "june"
}'
Frequently asked questions.
Can Asterisk or FreePBX send calls to an AI voice agent?
Yes. Add a SIP trunk in the Bland setup wizard, choose Asterisk / FreePBX, and enter your public IP or hostname. Attach the DIDs or extensions to route, and calls your dial plan sends to Bland are answered by a Bland agent.
Does Bland work with FreeSWITCH or Kamailio?
Bland works with any standard SIP endpoint. Discovery detects Asterisk, FreeSWITCH, Kamailio, 3CX, and Cisco, among others, and recommends the port and transport. Asterisk and FreePBX have a guided entry in the wizard; others use Other.
How do I set up Asterisk to accept transfers from Bland?
Trust all of Bland's media IPs as SIP peers, since transfers connect from them, and enable SRTP on the trunk toward Bland. A 403 means an IP is missing from the trust list; a 488 means SRTP is off.
Can the agent transfer to an extension and keep context?
Yes. Target an extension or SIP URI from a Transfer Call node, and map call variables to SIP headers on the trunk's outbound mapping. With SIP REFER, have your PBX copy those headers onto the new INVITE, or switch REFER off and Bland sends them directly.
Can I keep my numbers on my current carrier?
Yes. A SIP trunk carries the numbers on your own carrier, so they stay where they are. Bland also supports porting numbers in if you would rather move them, which typically takes 7 to 14 business days.
Related integrations.
See every integration on the integrations hub.
With our team
Our FDE team scopes the integration with you.
They map Bland to your stack, design the rollout, and stay on through go-live. Enterprise deployments go live in 30 days.
On your own
Or drop Bland into your editor.
The Bland MCP server runs in Claude Code, Cursor, and any MCP-aware editor. Or hit the REST API directly. Free sandbox, no credit card.