Having this outbound trunk configuration:

{
  "sipTrunkId": "ST_xxxxxxxxx",
  "name": "SIP-1",
  "address": "<http://myhost.com:5060|myhost.com:5060>",
  "transport": "SIP_TRANSPORT_UDP",
  "numbers": [
    "+11111111111",
    "+11111111112"
  ],
  "authUsername": "livekit-user",
  "authPassword": "livekit-pass"
}

please generate Asterisk pjsip.conf file entry to authenticate and authrorize livelit calls

Thanks for reaching out, Aivis. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.

what is the purpose of [identify_livekit] section?

Good question — thanks for asking. The

[identify_livekit]

section is used to map incoming requests from a remote IP/host to a specific PJSIP endpoint. In practice it does two main things: • Matches requests by source (the

match=

value) so Asterisk knows which endpoint configuration should handle traffic from that remote peer (useful when multiple trunks exist). • Ensures incoming requests are associated with the correct endpoint for routing, authentication checks, and applying endpoint settings (codecs, context, etc). When to use it: • Use an

identify

when the trunk is identified by IP/host rather than registration credentials, or when the provider sends calls to your IP and you need to bind those requests to the proper endpoint. When it’s not required: • If you register to the provider with a

registration

stanza, or the provider authenticates by credentials and realm, incoming requests may already be associated by auth/registration and an

identify

may be unnecessary. If you want, I can show an example

identify

that matches an IP address instead of hostname or add a

registration

stanza instead. Which would help more? , with interactive elements

wait, livekit is not identified by ip, is it? it has no defined range of ips

Good point — you’re correct: LiveKit does not have a fixed, published range of IP addresses, so you cannot reliably identify it by IP. • The

[identify_livekit]

stanza is only useful when you can match the provider by IP/host. • Because LiveKit has no defined IP range, do not rely on an

identify

match for LiveKit traffic. Instead use one of these approaches: ◦ Use authentication (

auth

+ `aor`/`endpoint`) and, if required, a

registration

stanza so calls are associated by credentials. ◦ Or enforce routing/authorization in your dialplan (

extensions.conf

) based on caller ID or other SIP headers. If you want, I can: • Provide a

registration

example for outbound trunks, or • Show a dialplan snippet to restrict which source numbers may place calls through the trunk. Which would you prefer?, with interactive elements

what could this message in asterisk mean?

*ERROR*[595]: *res_pjsip_outbound_authenticator_digest.c*:*450*

*digest_create_request_with_auth*: Host: 'XX.XX.XX.XX:5060': There were no auth ids available

Thanks — you’re asking what that Asterisk error means. It indicates Asterisk tried to create an outbound SIP request that required Digest authentication but could not find any configured authentication credentials (auth IDs) to use for that destination. What it typically means • Your PJSIP endpoint/AOR is not linked to an

auth

object, or the named

auth

section does not exist or is misspelled. • AOR/endpoint configuration is incomplete (endpoint missing

auth=...

or

aors=...

), so Asterisk has no credentials to add to the outbound request. • Less commonly: `outbound_auth`/transport settings are misconfigured. How to check and fix (quick steps) 1. Verify your endpoint links to the auth object: ◦ In

pjsip.conf

the endpoint must include

auth = <your-auth-name>

and

aors = <your-aor-name>

. 1. Ensure the

auth

section exists and is correct:

[livekit-auth]
   type=auth
   auth_type=userpass
   username=livekit-user
   password=livekit-pass

1. Ensure the AOR exists and is referenced by the endpoint:

[livekit-aor]
   type=aor
   contact=sip:myhost.com:5060

1. Confirm names match exactly (no typos or extra chars). 2. If your provider requires registration instead of simple auth, add a

registration

section tied to that auth. 3. After changes reload PJSIP and verify endpoints: ◦ asterisk -rx "module reload chan_pjsip.so" ◦ asterisk -rx "pjsip show endpoints" If you want, paste your current full endpoint/auth/aor sections and I’ll point out the exact mismatch. Sources: Agents telephony integration | LiveKit Documentation, with interactive elements