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How to connect my livekit agent with Asterisk ser...
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f
How to connect my livekit agent with Asterisk server for inbound and outbound calls @icy-policeman-64898
t
Thanks for reaching out, Abhishek. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
To connect your LiveKit agent with an Asterisk server for inbound and outbound calls, follow these steps: Inbound Calls Setup 1. *Create an Inbound Trunk:*`client.create_sip_inbound_trunk( name: "Asterisk-Inbound", numbers: ["your-phone-numbers"], allowed_addresses: ["asterisk-server-ip"], auth_username: "sip_user", auth_password: "sip_password" )` 2. *Create a Dispatch Rule:*`client.create_sip_dispatch_rule( name: "Asterisk Rule", trunk_ids: ["your-trunk-id"], rule: "direct-rule-configuration" )` Outbound Calls Setup 1. *Create an Outbound Trunk:*`client.create_sip_outbound_trunk( name: "Asterisk-Outbound", address: "asterisk-server-address", numbers: ["your-phone-numbers"], transport: "transport-configuration", auth_username: "sip_user", auth_password: "sip_password" )` 2. *Make Outbound Calls:*`client.create_sip_participant( sip_trunk_id: "your-trunk-id", sip_call_to: "destination-number", room_name: "your-room-name" )` Ensure that the SIP service exposes a public IP address for remote SIP peers to connect to. Additionally, both LiveKit and Asterisk must be configured to communicate over Redis for session state management. Sources: lib/livekit/sip_service_client.rb | README.md
f
How do i configure the Asterisk server
t
To get your Asterisk server chatting smoothly with LiveKit, here’s a friendly step-by-step guide: 1. Set Up LiveKit's SIP Service • First, make sure your LiveKit server is up and running, and Redis is set up for session storage. • Create an inbound SIP trunk in LiveKit to link up with Asterisk. You can use the LiveKit CLI or SDK to set up the trunk and dispatch rules that route calls to your LiveKit rooms. 2. Configure Asterisk as a SIP Trunk • Edit your 
sip.conf
or 
sip_custom.conf
file to add a SIP peer for LiveKit. Here’s a quick example: • Next, set up your Asterisk dial plan in 
extensions.conf
to route calls over to LiveKit: • Don’t forget to restart your Asterisk services to make these changes take effect: 3. Enable SIP Trunking in LiveKit • Double-check that LiveKit’s SIP service is configured with a YAML file, including your API keys, Redis address, and SIP/RTP ports. • Then, start the SIP service using that configuration file. 4. Test the Integration • Give it a whirl by making some test calls from Asterisk to confirm everything routes nicely to your LiveKit rooms. 5. Security Considerations • For peace of mind, enable TLS to keep communication between Asterisk and LiveKit secure. • Also, consider using the 
hide_phone_number
option in your dispatch rules to keep caller info private. If you want me to walk you through any part of this or have questions, just let me know—I’m here to help! Sources: README.md | SIP trunk setup | LiveKit Docs | SIP Integration | livekit/livekit_composite | DeepWiki
f
can ypu give me an example sip.conf
t
I don't have the answer you're looking for. You could also try asking your question: • in one of the other Slack channels or • to https://deepwiki.com/livekit/livekit_composite which is trained on all LiveKit source code If you find the answer, please post it here to help others!
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