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

[Marco Capetta]

Hi,

can you please confirm if you still have this problem?
If you have find out which was the root cause, please share it with the
community because it could be be helpful for other users.
If not, please describe better all the steps you did for the configuration,
because we are not able to replicate your problem locally.

Thank you


On Tue, Dec 16, 2025 at 9:30 AM SPCE Newbie <spce-user at tel.co.uk> wrote:

> 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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>
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