[Spce-user] mr13.5.1 No PSTN gateways available

[SPCE Newbie]

1. Expected Behaviour

With allow_out_foreign_domains = true, allowed_clis populated,
and with a single SIP Peer and permissive rules, then given an
E.164 R-URI, call is routed to the SIP Peer.


2. Actual Behaviour

503 PSTN Termination Currently Unavailable


3. Configuration

https://spce:1443/v2/#/contract
has two contracts:
- one VoIP Reseller
- one SIP Peering

https://spce:1443/v2/#/billing
however says "Used (contracts): 3"

https://spce:1443/v2/#/peering
has a single SIP Peer with:
- no Inbound Rules
- a single, Default enabled (Stopper: off) with blank patterns
- two Peering Servers - same remote VoIP trunk provider FQDN
  but two different IPv4 addresses
- Preferences for both Peering Servers are defaulted apart from
  Peer Auth Authentication which is enabled and populated with
  peer_auth_pass, peer_auth_realm, and peer_auth_registrar_server


4. Log - /ngcp-data/logs/kamailio-proxy.log abbreviated:

 1: I-sl:filtered-ack: New request on proxy - M=ACK
 2: I-DEFAULT_ROUTE: New request on proxy - M=INVITE
 3: I-DEFAULT_ROUTE: Sending reply S=100 Trying M=INVITE
 4: I-ROUTE_LOAD_CALLEE_DOMAIN_PREF: Load domain prefs for 'my.domain'
 5: I-ROUTE_FIND_CALLER: Sending reply S=101 Connecting
 6: I-ROUTE_LOAD_CALLER_PREF: Load caller preferences for uuid <uuid>
 7: I-ROUTE_LOAD_CALLER_CONTRACT_PREF: Load caller contract '2' prefs
 8: I-ROUTE_LOAD_CALLER_RESELLER_PREF: Load reseller '1' preferences
 9: I-ROUTE_LOAD_CALLER_PREF: Ringback tone enabled for caller
10: I-ROUTE_DLG_MANAGE: Dialog managed, lua_dlg_callid:
11: I-ROUTE_GET_CALLER_CLI: User-Provided CLI 'Me' taken from From-User
12: I-ROUTE_CHECK_CLI_ALLOWED: User-provided CLI 'Me' rejected, using NN
13: I-ROUTE_INVITE: Setting <E.164 at my.domain> as initiating user-CLI
14: I-ROUTE_INVITE: Setting <E.164 at my.domain> as initiating net-CLI
15: I-ROUTE_FIND_CALLEE: Callee is not local R=sip:<E.164 at my.dom>:5060
16: I-ROUTE_PSTN: Call to SIP Peering R=sip:<E.164 at my.domain>:5060
17: I-ROUTE_PSTN: Load gws matching calling partsip:<E.164 at my.domain>
18: E-ROUTE_PSTN_SELECT_PEER: No PSTN gateways available R=sip:<E.164@>
19: I-ROUTE_ACC_CALLER: Setting acc source-leg for uuid <uid>
20: I-ROUTE_ACC_CALLEE: Setting acc destination-leg for uuid '0':
21: I-ROUTE_LOCAL_REPLY: Sending S=503 PSTNTerminationCurrentlyUnavailable


5. Own Research/Resolution Attempted

1 Websearch result:
https://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/2021-May/029570.html

N/A as not attempting to use a prepaid billing profile here.