SIP trunking

A trunk is a set of SIP credentials on one of our gateways. Your PBX, dialler or AI agent platform sends outbound calls through it, and numbers on your account can be routed to it for inbound.

Gateways

Ports: 5060 UDP/TCP, 5061 TLS. Media (RTP) uses UDP 10000–20000 from the same hosts. If a gateway is unreachable from your side, the trunk fails over to the next one in the same region automatically.

Authentication

Username + password. Your PBX registers with the vt_… username and the password from the trunk card, re-registering every 30 s. Use this behind NAT or with dynamic IPs.

IP allow-list. Enter the public IPs or CIDRs of your PBX / media servers on the trunk. No registration; we accept INVITEs from those addresses only. Use this for diallers, FreeSWITCH/Kamailio clusters and AI voice platforms with fixed egress.

Both can be changed later with Edit on the trunk card.

Codecs and transport

Offered: OPUS, G.722, G.711 A-law, G.711 µ-law, G.729. Transcoding to what the far end needs is free. Offer OPUS or G.722 first for AI agents and softphones, G.711 for legacy PBXs.

TLS on 5061 with SRTP for media is supported on every gateway and recommended for new trunks. Plain UDP on 5060 remains available.

Caller ID

Outbound caller ID must be a number you are allowed to present: any number on your VoiceTele account (verified automatically) or a number you own elsewhere, verified once by a code read out by a call. Unverified caller IDs are rejected with SIP 403 so our routes stay clean. Send the number in the From user part in E.164 (+442035550118) or set it per trunk.

Channels and limits

Pay-as-you-go trunks start at 30 concurrent channels and 10 calls per second; both rise with account level or on request. A per-trunk spend cap and destination allow-list can be set by support until these controls appear in Settings.

Route class

Premium (default) uses direct and Tier-1 routes only, caller ID guaranteed, lowest post-dial delay. Standard is least-cost routing for destinations where CLI is not critical. The class is a trunk setting and can be changed any time.

Asterisk (pjsip)

The PBX config button produces this with your values filled in:

[voicetele]
type=registration
outbound_auth=voicetele-auth
server_uri=sip:sip-ams-1.voicetele.com
client_uri=sip:[email protected]
retry_interval=30

[voicetele-auth]
type=auth
auth_type=userpass
username=vt_482113
password=YOUR_PASSWORD

[voicetele]
type=endpoint
context=from-voicetele
disallow=all
allow=opus,g722,alaw,ulaw
outbound_auth=voicetele-auth
aors=voicetele
from_user=vt_482113
from_domain=sip-ams-1.voicetele.com

[voicetele]
type=aor
contact=sip:sip-ams-1.voicetele.com
qualify_frequency=30

Dial out with Dial(PJSIP/${EXTEN}@voicetele). Inbound arrives in from-voicetele with the dialled number in ${EXTEN}.

FreePBX

Connectivity → Trunks → Add SIP (chan_pjsip) trunk. General tab: any trunk name, outbound CID = one of your numbers. pjsip Settings: Username vt_…, Secret = password, SIP Server = gateway host, Port 5060, Registration Send, Context from-trunk. Codecs: opus, g722, alaw, ulaw. Add an outbound route pointing at the trunk.

3CX

Admin → Voice & Chat → Add SIP Trunk → Generic SIP Trunk. Registrar/Server = gateway host, Auth ID = vt_…, password = trunk password, main trunk number = one of your VoiceTele numbers. Under Options keep "Register" enabled and codecs G.711 A/µ + G.722.

Diallers and agent platforms

VICIdial, Dialfire and custom Asterisk/FreeSWITCH: use IP allow-list auth and ask support to raise the CPS ceiling to what your campaign runs. ElevenLabs, Vapi, Retell, LiveKit: add VoiceTele as a "bring your own carrier" SIP trunk with the gateway host and credentials, or IP-authenticate their media IPs; route your inbound numbers to the platform's SIP URI (Numbers & routing).

Test call

Test call on the trunk card originates a short call to a number you enter and writes the result to Call records, charged at the trunk's route class. Use it to confirm rating and reporting before connecting a PBX.

Troubleshooting

• Registration fails (401/403): wrong password or username; copy both from the trunk card. Check the trunk is not paused.
• Calls rejected with 403: caller ID not verified, or destination blocked by the trunk's allow-list / high-risk prefix rules.
• 503 / no route: insufficient balance, or channel ceiling reached, see Billing and the trunk card.
• One-way audio: open UDP 10000–20000 from the gateway hosts and disable SIP ALG on your router.
• Still stuck: send the call time and destination to [email protected]; we answer with a SIP trace.