Outbound calls failing — permission_denied then SIP 408 timeout on custom SIP trunk

Support Help
1

Subject: Outbound calls failing (permission_denied, then telephony_provider_unavailable / SIP 408) on custom SIP trunk

Hi Retell team,

I’m on the Pay-As-You-Go plan with a custom telephony (SIP trunk) setup for outbound calling, and my calls are failing to connect. I’ve done thorough troubleshooting on my end and would appreciate help identifying what’s happening on Retell’s side.

Setup:

  • Self-hosted FreePBX/Asterisk server on a UK-based VPS (Vultr, London datacenter, Debian 12)
  • SIP Trunk Username, Password, and Termination URI configured on the phone number page, and confirmed matching on the Asterisk server directly via pjsip show auth
  • Outbound Call Agent is assigned, Allowed Outbound Countries includes the destination country
  • Custom Telephony CPS shows a limit of 1 under Settings > Limits, so the trunk appears recognized by the system

What I’m seeing:

  1. Earlier batch calls and dashboard test calls failed instantly (0s duration, $0 cost) with disconnection reason telephony_provider_permission_denied.
  2. After re-verifying and re-saving the SIP trunk password, a newer test call instead failed after a delay with:
    SIP status code: 408 (Request Timeout)
    Disconnection reason: telephony_provider_unavailable
    (from my call log: “Error dialing to user, SIP status code: 408 … Error: dial failed: SIP 408 Request Timeout”)
  3. I checked my own server logs (Asterisk pjsip logger) during this test call and saw no incoming INVITE from Retell’s servers at all during that call attempt — the server appears to never receive the request.
  4. I’ve ruled out firewall blocking on my end: Vultr’s firewall has no firewall groups configured (fully open), and the FreePBX firewall is also currently disabled. So nothing on my infrastructure is blocking inbound traffic.

Given the above, it looks like the request may not be reaching my server, or the response path is failing somewhere outside my infrastructure — possibly on Retell’s side reaching my SIP trunk, or a routing/permission issue tied to my account’s custom telephony setup.

Could you please check:

  • Whether outbound calling / telephony access needs manual activation or additional approval on my account
  • Whether there’s a known issue with custom SIP trunk connectivity from Retell’s infrastructure to external FreePBX/Asterisk servers

Happy to provide the phone number ID, account email, call IDs, or full Asterisk logs if useful.

Thanks,
Eman Maryam (emancreates)
Account email: emancreates01@… (Pay-As-You-Go)

2

Hello @emancreates01

telephony_provider_unavailable + SIP 408 = the request either never reached your server or got no response in time. Since your Asterisk logs show no incoming INVITE at all, the request likely isn’t reaching your VPS — this could be Retell’s traffic not reaching your termination URI (firewall/network path issue), or a misconfigured/incorrect termination URI on the number import.

  1. Double-check your termination URI and credentials are exactly what you provided to Retell (delete and re-import the number if anything was wrong — configuration cannot be edited, only re-imported).
  2. Check the call’s PCAP file in call logs (only available if agent data retention is “Everything” and transport is UDP/TCP, not TLS) to see whether Retell actually sent an INVITE and where it went.
  3. If PCAP/logs show Retell attempted the INVITE but it didn’t land, that points to a network path issue outside Retell (routing, provider-side ACL, etc.) — worth confirming with your VPS/Asterisk provider’s edge too.

Additionally Could you please share a recent failed Call ID?

We’ll review the call details and SIP logs on our side to determine whether the request is reaching your configured termination URI and investigate this further with the relevant team.

Thank you!