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SIP trunks for PBX and contact-center platforms

Bidirectional trunks with the codec and authentication mix your platform actually uses. We test against Asterisk, FreeSWITCH, FreePBX, 3CX, Cisco UCM, MS Teams Direct Routing, and the major hosted-PBX softswitches before signing customers up.

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IP authentication

Whitelist up to 8 source IPs per trunk. The default and most secure option for B2B. No SIP digest credentials in customer config files.

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SIP registration

Username/password registration with TLS-protected SIP signaling. Suited for dynamic-IP environments and selected mobile UAs.

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Hybrid

IP-auth on a known SBC, registration on a backup. Trunks fail over between auth modes for resilience without manual intervention.

Tested against the platforms our customers actually run

Most "SIP trunk compatibility" lists are wishful thinking. We integration-test against the major softswitches, document the working dial-plan and INVITE templates, and ship them on day one of a new customer's setup.

  • Asterisk / FreePBX — pjsip and chan_sip templates, transport TCP/UDP/TLS
  • FreeSWITCH — gateway profiles for inbound + outbound, codec negotiation pre-set
  • 3CX — generic SIP trunk template with codec list and outbound rules
  • Cisco UCM — SIP trunk profile, route pattern guidance, transcoder considerations
  • Microsoft Teams Direct Routing — via SBC interop (AudioCodes, Ribbon, Oracle ACME) — TLS+SRTP only
  • Yeastar, Grandstream, Sangoma, Yealink — vendor-specific guides
  • Custom softswitch — Kamailio-fronted, OpenSIPS, kHomer-traced
# Asterisk pjsip template — outbound trunk [callivex] type = endpoint context = from-callivex disallow = all allow = ulaw,alaw,g729,opus transport = transport-udp outbound_auth = callivex_auth aors = callivex_aor direct_media = no rtp_symmetric = yes force_rport = yes rewrite_contact = yes trust_id_inbound = yes identify_by = ip,auth_username [callivex_aor] type = aor contact = sip:sip.callivex.com:5060 [callivex_auth] type = auth auth_type = userpass username = your_account_id password = REDACTED
# Trunk health — last 60 minutes Endpoint: sip.callivex.com (eu-west-1) Transport: UDP + TCP + TLS Active calls: 38 / 100 channels Inbound: 412 attempts · ASR 47% Outbound: 1,841 attempts · ASR 39% Avg PDD: 2.1s Codec mix: PCMA 71% · G729 21% · opus 8% Last 5xx: 18m ago — carrier_b 503 auto-rerouted to carrier_c

Multiple POPs, multiple carriers, automatic failover

Your trunk doesn't depend on one upstream carrier or one server. Customers point to a single FQDN that resolves to a load-balanced SBC pair, and routing decisions are made per call against current quality and cost.

  • Anycast-style FQDN with active/active SBC deployment
  • Per-call carrier selection (LCR with quality + price)
  • Auto-failover on 5xx and PDD spikes within seconds
  • Customer-facing endpoint stays stable even during carrier-side incidents
  • Maintenance windows announced 7 days ahead in writing

Bring your platform; we'll match the trunk.

Tell us what softswitch you're running and the call volumes. We'll send back the right config + a trial trunk.

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