Update Anveo Direct dial-prefix warning with live-confirmed findings

The dialing-prefix ambiguity is resolved: Anveo's own trunk edit screen has
no required-field marker on Dialing Prefix, confirming it's optional. Also
documents the separate Outbound Service Trunk vs. inbound SIP Trunk
distinction discovered while walking through a real account setup.
This commit is contained in:
Claude
2026-07-23 16:05:52 +00:00
parent 3661c8fbca
commit 4c0326ca5f
+9 -6
View File
@@ -1315,12 +1315,15 @@ install_pstn-trunk() {
log_warning "requires a verified/Anveo-owned number as Caller-ID — any DID you order"
log_warning "through them satisfies that automatically (see this service's personal-"
log_warning "number feature for assigning a specific one per extension)."
log_warning "UNVERIFIED: Anveo's own outbound-trunk page documents dialing as"
log_warning "[PREFIX]PHONENUMBER@sbc.anveo.com (a per-trunk prefix) — this contradicts"
log_warning "an earlier no-prefix-needed finding from their FAQ. This dialplan dials"
log_warning "the bare number, no prefix. Check whether your configured trunk's Prefix"
log_warning "field can be left blank before relying on this — if it can't, outbound"
log_warning "calls through that trunk won't match and will fail."
log_info "Confirmed live (2026-07-23): the trunk's own Dialing Prefix field has no"
log_info "required-field marker on Anveo's Edit Call Termination Trunk screen — it's"
log_info "optional. Leave it blank; this dialplan dials the bare number, no prefix."
log_warning "Anveo Direct also separates two different 'trunk' concepts — don't confuse them:"
log_warning "an Outbound Service (Call Termination) Trunk handles calls OUT (IP-authenticated,"
log_warning "what this installer configures), while inbound routing for a DID is a SEPARATE"
log_warning "'SIP Trunk' object (Account -> the DID's Call Options tab -> Destination SIP"
log_warning "Trunk) that forwards to a SIP URI — e.g. \$[E164]\$@<this box's public IP>:5060."
log_warning "Create both in Anveo's portal; this installer only handles the Asterisk side."
;;
2)
_default_provider_name="VoIP.ms"