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

[SPCE Newbie]

On Tue, 16 Dec 2025 08:50:10 +0100
Dario Busso <dbusso at sipwise.com> wrote:

> Make sure you populate the "outbound peering rules" for the peer
> group, otherwise the system cannot route any calls to that peer group.
> Check the on-line HB
> https://www.sipwise.com/doc/mr13.5.1/spce/ce/mr13.5.1/basicconfiguration/basicconfiguration.html#outbound_peering_rules
> Dario

It says "add a default rule to route all calls via this group".  I did.

> 
> On Mon, 15 Dec 2025 at 22:07, SPCE Newbie wrote:
> 
> > 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.
> >
> > --
> > Spce-user mailing list
> > Spce-user at lists.sipwise.com
> > http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com
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