<div dir="ltr">Make sure you populate the "outbound peering rules" for the peer group, otherwise the system cannot route any calls to that peer group.<div>Check the on-line HB <a href="https://www.sipwise.com/doc/mr13.5.1/spce/ce/mr13.5.1/basicconfiguration/basicconfiguration.html#outbound_peering_rules">https://www.sipwise.com/doc/mr13.5.1/spce/ce/mr13.5.1/basicconfiguration/basicconfiguration.html#outbound_peering_rules</a></div><div>Dario</div></div><br><div class="gmail_quote gmail_quote_container"><div dir="ltr" class="gmail_attr">On Mon, 15 Dec 2025 at 22:07, SPCE Newbie <<a href="mailto:spce-user@tel.co.uk">spce-user@tel.co.uk</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">1. Expected Behaviour<br>
<br>
With allow_out_foreign_domains = true, allowed_clis populated,<br>
and with a single SIP Peer and permissive rules, then given an<br>
E.164 R-URI, call is routed to the SIP Peer.<br>
<br>
<br>
2. Actual Behaviour<br>
<br>
503 PSTN Termination Currently Unavailable<br>
<br>
<br>
3. Configuration<br>
<br>
<a href="https://spce:1443/v2/#/contract" rel="noreferrer" target="_blank">https://spce:1443/v2/#/contract</a><br>
has two contracts:<br>
- one VoIP Reseller<br>
- one SIP Peering<br>
<br>
<a href="https://spce:1443/v2/#/billing" rel="noreferrer" target="_blank">https://spce:1443/v2/#/billing</a><br>
however says "Used (contracts): 3"<br>
<br>
<a href="https://spce:1443/v2/#/peering" rel="noreferrer" target="_blank">https://spce:1443/v2/#/peering</a><br>
has a single SIP Peer with:<br>
- no Inbound Rules<br>
- a single, Default enabled (Stopper: off) with blank patterns<br>
- two Peering Servers - same remote VoIP trunk provider FQDN<br>
but two different IPv4 addresses<br>
- Preferences for both Peering Servers are defaulted apart from<br>
Peer Auth Authentication which is enabled and populated with<br>
peer_auth_pass, peer_auth_realm, and peer_auth_registrar_server<br>
<br>
<br>
4. Log - /ngcp-data/logs/kamailio-proxy.log abbreviated:<br>
<br>
1: I-sl:filtered-ack: New request on proxy - M=ACK<br>
2: I-DEFAULT_ROUTE: New request on proxy - M=INVITE<br>
3: I-DEFAULT_ROUTE: Sending reply S=100 Trying M=INVITE<br>
4: I-ROUTE_LOAD_CALLEE_DOMAIN_PREF: Load domain prefs for 'my.domain'<br>
5: I-ROUTE_FIND_CALLER: Sending reply S=101 Connecting<br>
6: I-ROUTE_LOAD_CALLER_PREF: Load caller preferences for uuid <uuid><br>
7: I-ROUTE_LOAD_CALLER_CONTRACT_PREF: Load caller contract '2' prefs<br>
8: I-ROUTE_LOAD_CALLER_RESELLER_PREF: Load reseller '1' preferences<br>
9: I-ROUTE_LOAD_CALLER_PREF: Ringback tone enabled for caller<br>
10: I-ROUTE_DLG_MANAGE: Dialog managed, lua_dlg_callid:<br>
11: I-ROUTE_GET_CALLER_CLI: User-Provided CLI 'Me' taken from From-User<br>
12: I-ROUTE_CHECK_CLI_ALLOWED: User-provided CLI 'Me' rejected, using NN<br>
13: I-ROUTE_INVITE: Setting <E.164@my.domain> as initiating user-CLI<br>
14: I-ROUTE_INVITE: Setting <E.164@my.domain> as initiating net-CLI<br>
15: I-ROUTE_FIND_CALLEE: Callee is not local R=sip:<E.164@my.dom>:5060<br>
16: I-ROUTE_PSTN: Call to SIP Peering R=sip:<E.164@my.domain>:5060<br>
17: I-ROUTE_PSTN: Load gws matching calling partsip:<E.164@my.domain><br>
18: E-ROUTE_PSTN_SELECT_PEER: No PSTN gateways available R=sip:<E.164@><br>
19: I-ROUTE_ACC_CALLER: Setting acc source-leg for uuid <uid><br>
20: I-ROUTE_ACC_CALLEE: Setting acc destination-leg for uuid '0':<br>
21: I-ROUTE_LOCAL_REPLY: Sending S=503 PSTNTerminationCurrentlyUnavailable<br>
<br>
<br>
5. Own Research/Resolution Attempted<br>
<br>
1 Websearch result:<br>
<a href="https://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/2021-May/029570.html" rel="noreferrer" target="_blank">https://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/2021-May/029570.html</a><br>
<br>
N/A as not attempting to use a prepaid billing profile here.<br>
<br>
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</blockquote></div>