Carrier Connectivity

C2C Connectivity — carrier routing you can actually control

We connect your contact center to multiple carriers — SIP trunks, PRI via gateways, and click-to-call APIs — behind an OpenSIPS session border controller that routes every call by caller-ID, trunk capability and load.

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  • Multi-carrier SIP & PRI
  • Caller-ID based routing
  • OpenSIPS SBC & load-balancing
  • RTP media relay

One trunk, one point of failure

Many contact centers run every call through a single carrier. When that trunk degrades — or a range of caller-IDs is only valid on a specific circuit — calls fail silently and campaigns stall.

Proper carrier routing needs an SBC layer that knows which number ranges belong to which trunk, which trunks are healthy, and how to present the right caller-ID to each carrier.

How It Works

How calls reach the right carrier

01

Call originates

A dialer, agent or IVR places a call from any Asterisk media node.

02

SBC decides the route

OpenSIPS matches the caller-ID range, campaign and trunk capabilities, and picks a healthy carrier or PRI gateway.

03

Media is anchored

RTP is relayed through rtpengine so NAT, codecs and multiple network interfaces are handled cleanly.

Features

Carrier-grade building blocks

Caller-ID range routing

Route calls to a specific SBC or PRI gateway based on the presented number range — required when DIDs belong to a particular circuit.

Load-balanced media nodes

OpenSIPS distributes calls across several Asterisk nodes and takes failed nodes out of rotation automatically.

Trunk capability whitelist

Per-trunk rules control which caller-IDs and headers (PAI/RPID) each carrier accepts, preventing rejected calls.

PRI & GSM gateways

Integrate E1/PRI lines and GSM gateways alongside SIP trunks under one routing policy.

Concurrency control

Cap simultaneous calls per trunk to match what each carrier circuit can carry reliably.

Firewall-scoped trunks

Carrier IPs are explicitly allow-listed at the firewall; SIP signalling is never exposed to the open internet.

Architecture

Signalling and media path

Every outbound and inbound call crosses a single routing policy, regardless of which carrier carries it.

  1. Dialer / Agent / IVR
    Asterisk nodes
  2. OpenSIPS SBC
    Route & LB
  3. rtpengine
    Media relay
  4. Carriers
    SIP / PRI / GSM

Compliance & Security

Telecom-compliant connectivity

Caller-ID presentation per carrier and TRAI number-series rules
Separate promotional and transactional routing where required
Source-IP allow-listing for every carrier trunk
Per-trunk concurrency and rate limits
CDRs for every call leg for reconciliation and billing
Recording on the media node independent of carrier

FAQ

Common questions

Can you work with our existing carrier?+
Yes. We integrate with any SIP trunk provider and with PRI lines through a media gateway; we can also add new carriers alongside existing ones.
What happens if a carrier goes down?+
Routes can define fallback trunks, and the SBC stops sending calls to unhealthy destinations so campaigns keep running on the remaining capacity.
Do you support inbound DIDs?+
Yes. Inbound DIDs are mapped to IVRs, queues or campaigns, with number normalisation handled at the SBC.
Is this only for outbound?+
No. The same routing layer handles inbound, outbound, click-to-call and WebRTC traffic.
Can we host it on our own servers?+
Yes. The same software runs in the cloud (your account or ours), on-premise in your data centre, or hybrid. Features are identical across deployment modes; on-premise keeps all recordings and lead data inside your network.

Related

OpenSIPS Asterisk ITSP / Carrier

Get Started

Untangle your carrier routing

Share your trunks, DID ranges and call volumes — we'll design the routing layer.

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