Voice

Carrier-grade SIP trunking
for the routes that move your P&L.

Direct voice termination with in-region expertise — concurrent channels, HLR-optimised routing and daily route-quality visibility on your toughest routes.

Contact Sales →All products

Direct termination · HLR optimisation · PBX-ready

Direct routing — A to B

Direct route· 0 hopsSelected

1-HOP route· 1 hopNot selected

Reseller route· 3+ hopsNot selected

● Direct route selectedbest ASR · lowest latency

What you get

Trunks tuned to your traffic

📉

Route quality

Direct routes and HLR optimisation keep ASR up where reseller multi-hop routes quietly fail.

📡

Scales with you

From a handful of channels to 50+ concurrent, provisioned on the same platform as your numbers.

🔌

PBX-ready

Drops into any standard SIP environment — bring your own PBX or reseller stack.

Prove it first

30 days on your toughest routes

Carriers won’t admit when a route degrades. Run a 30-day pilot on your hardest 2–3 routes with daily route-quality dashboards — no commitment.

✓ Direct A–Z termination with in-region routing expertise.
✓ Daily route-quality and ASR dashboards per route.
✓ Regulated DID numbers and local CLI on the same account.
✓ 30-day route pilot with clear exit criteria.

Talk to our team →

Pilot at a glance

30d

Route pilot

50+

Concurrent channels

A–Z

Termination coverage

Daily

Route dashboards

Set up in minutes

Set up your SIP Trunk in 1 — 2 — 3

No new hardware. Point your existing PBX or softswitch at TextCall and start terminating on direct routes.

Point your PBX

Aim your PBX or softswitch — Asterisk, FreeSWITCH and any standard SIP stack — at our SIP endpoint, with IP-ACL or digest authentication.

We provision routes

We enable direct A–Z termination, your concurrent channels and local CLI — tuned to the routes that matter to you.

Go live & monitor

Start terminating and watch CDRs and daily route-quality / ASR dashboards per route — scale channels as you grow.

For developers

Built for developers — read the docs

Point your PBX or softswitch at our SIP endpoints and start terminating — IP-ACL or digest auth, standard codecs, and CDRs plus route-quality stats over the API.

✓ Register any SIP PBX — Asterisk, FreeSWITCH and any standard SIP stack.
✓ IP-ACL or digest authentication, your choice.
✓ Standard codecs: G.711, G.729, OPUS.
✓ CDRs and per-route ASR/route-quality via API.

Read the documentation →

sip.conf

# SIP trunk — point your PBX here
host     = sip.textcall.com
username = ac_10482
auth     = digest · or IP ACL
codecs   = G.711, G.729, OPUS
channels = 50 concurrent

# → registered · terminating A–Z with daily route-quality stats

Contact Sales

Test our routes against yours

Send us your toughest routes — we’ll run a 30-day pilot with daily route-quality dashboards.

Contact Sales →Explore all products

Carrier-grade SIP trunkingfor the routes that move your P&L.