Setup guides

Connect SIPTrunkHub to your platform

12 step-by-step guides with real config: what you need, IP vs register auth, codec order, NAT notes, how to test, and what each SIP error means on that platform.

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IP PBX

Office and contact-centre phone systems.

PJSIPAsteriskThis guide connects SIPTrunkHub to Asterisk 18 or newer using chan_pjsip. You will add an endpoint, an AOR and either an identify section (IP auth) or a registration (digest auth), then route outbound calls from the dialplan.Open guide
GUI / PJSIPFreePBXFreePBX writes the PJSIP config for you. You create a chan_pjsip trunk in the GUI, add an outbound route that uses it, and an inbound route for each DID. Field names below match FreePBX 16 and 17.Open guide
Gateway (GUI)FusionPBXFusionPBX is a multi-tenant web interface on top of FreeSWITCH. A trunk is a gateway on the external profile. You add the gateway, allow our IP in the providers ACL, create an outbound route, and add a destination for each DID. Menu names below match FusionPBX 5.x.Open guide
Generic SIP trunk3CXSIPTrunkHub is not a pre-listed 3CX provider, so you add it as a Generic SIP Trunk. You fill in the server, pick the auth mode, set caller ID and codecs, then create outbound and inbound rules. Screens below match 3CX v20.Open guide
P-Series / S-SeriesYeastarYeastar calls an IP-authenticated trunk a Peer Trunk and a credential trunk a Register Trunk. Field names below are from the P-Series; S-Series differences are noted.Open guide
UCM seriesGrandstreamOn a Grandstream UCM, a SIP trunk is either a Peer SIP Trunk (IP auth) or a Register SIP Trunk. You create the trunk, set codecs and caller ID, then add outbound and inbound routes.Open guide
Via CUBECisco CUCMCUCM should not talk to a carrier directly. CUBE (an IOS-XE router) sits at the edge: CUCM sends calls to CUBE over a SIP trunk, and CUBE sends them to SIPTrunkHub. This guide covers the CUBE side and the CUCM pieces that point at it.Open guide
Trunk configIssabelIssabel uses the FreePBX 2.x style trunk screen with free-text peer details. You paste a peer definition, optionally add a register string, then build outbound and inbound routes. The firewall module needs a rule for our IPs.Open guide

Softswitch

Programmable switching platforms.

Gateway XMLFreeSWITCHIn FreeSWITCH a SIP trunk is a gateway on the external Sofia profile. You add one XML file, a dialplan extension that bridges to the gateway, and an ACL entry so inbound calls from our IP are accepted.Open guide

Dialer

Outbound and blended contact-centre dialers.

Carrier entryVicidialVicidial adds trunks as carriers. One carrier entry holds the SIP peer, an optional registration, a global variable and a dialplan. Campaigns then use a dial prefix to send calls through it.Open guide

Billing switch

Platforms you use to resell minutes.

Provider trunkMagnusBillingMagnusBilling is an Asterisk-based billing softswitch. You add SIPTrunkHub as a provider, create a trunk for it, put the trunk in a trunk group, and point your plan’s rates at that group. Your customers then dial through MagnusBilling and the calls leave via us.Open guide

SIP proxy

Load balancing and routing at the edge.

DispatcherKamailioKamailio is a SIP proxy, not a PBX. Here it sits in front of your media servers and sends outbound traffic to SIPTrunkHub with the dispatcher module, failing over to a second destination on timeouts and 5xx responses.Open guide

Platform not listed?

Any SIP-compliant system works. Point it at the SIP server, use IP or digest auth, send E.164 digits and offer G.711 first. The Asterisk guide is the closest generic reference, or ask the NOC.

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