SIP registration or IP authentication
Register with a username and password, or authenticate by a fixed source IP — whichever your PBX, dialer or softswitch supports.
SIP trunk provider for PBX, dialers and softswitches
SIP trunking for PBX, dialers, softswitches and resellers
A SIP trunk is the connection between your platform and CloudyVoice; SIP termination is what happens on the other end of that connection — the call being completed to its destination. This page covers the trunk itself: how to connect, authenticate and size it.
Trunk, not just termination
SIP trunk vs. SIP termination — the distinction that matters
A SIP trunk is the connectivity itself: the registered account or IP-authenticated link between your PBX, dialer or softswitch and CloudyVoice, carrying SIP signalling and RTP media. SIP termination is the service delivered over that trunk — completing your outbound calls to the destination network. You need both: the trunk is how you connect, termination is what the connection is for. CloudyVoice's trunks are provisioned for outbound termination; number porting, DIDs and inbound origination are not asserted here unless confirmed for your account during onboarding.
Setting up a trunk
Authentication, dialing and capacity
E.164 dialing
Numbers are dialled as country code plus national number, with no local trunk prefix, spaces or brackets.
Channels and CPS
Concurrent channels and calls per second are agreed per account against your expected peak — a predictive dialer generating high CPS can breach carrier limits long before it reaches its channel ceiling.
Platform compatibility
Works with any SIP-capable endpoint: softphones, IP-PBX platforms including Asterisk, FreeSWITCH and 3CX, call-centre dialers such as ViciDial, and VOS3000 softswitch deployments.
Who trunks into CloudyVoice
Typical trunk use cases
PBX operators trunking a business phone system out for outbound calling; call centres and BPOs trunking a dialer such as ViciDial for outbound campaigns; and resellers or ITSPs trunking a softswitch to sell termination onward under their own brand. Caller ID behaviour on any trunk is subject to the route and to the law in the destination — CloudyVoice does not support or promote caller ID spoofing intended to mislead, and traffic review applies before high-volume trunking is enabled, per the acceptable use policy.
Buyer reference
Technical specifications
- Authentication
- SIP registration (username/password) or static IP authentication
- Dialing format
- E.164 — country code plus national number, no trunk prefix
- Billing increments
- Published per route (1/1, 60/1 or 60/60) — see each destination’s rate page
- Balance model
- Prepaid USD wallet; balance does not expire while the account is active
- Payment model
- Cryptocurrency prepaid — BTC, ETH, USDT (ERC-20 and TRC-20); no monthly commitment
- Rate effective dates
- Published on every destination page
- Route testing
- Recommended before volume commitment — see the route-quality guide
- Supported platforms
- Any SIP-capable softphone, IP-PBX, gateway, dialer or softswitch — including Zoiper, MicroSIP, Linphone, Asterisk, FreeSWITCH, ViciDial, VOS3000 and 3CX
- Codecs
- Confirmed during onboarding
- SIP transport
- Confirmed during onboarding
- DTMF
- Confirmed during onboarding
- CDR availability
- Full CDR access on a dedicated VOS/softswitch deployment; hosted-account CDR access is confirmed during onboarding
- Rate deck format
- Confirmed during onboarding
- Channels
- Confirmed during onboarding
- CPS
- Confirmed during onboarding
- Failover
- Available on wholesale and VOS routes; confirmed per destination during onboarding
Related services
What the trunk carries and who resells it
SIP termination
What happens after the trunk connects — the call being routed and completed to the destination network.
Wholesale VoIP
Volume A–Z capacity for ITSPs and resellers sending traffic over their own trunk.
Call center VoIP
Trunking sized for dialer and call-centre traffic specifically — ViciDial, predictive dialers and high-CPS bursts.
Questions
Frequently asked questions
What is a SIP trunk?
A SIP trunk is the SIP connection between your PBX, dialer or softswitch and a termination provider — it carries call signalling and media. It is the connectivity layer; the service delivered over it (completing calls to a destination) is SIP termination.
Do you provide inbound calling or DIDs on this trunk?
This page describes outbound trunking for termination. Inbound origination and DID availability are not asserted here — confirmed during onboarding if that is part of your requirement.
Can I authenticate by IP instead of registering?
Yes. Both SIP registration (username/password) and static IP authentication are supported.
How many channels or how much CPS can I get on a trunk?
Capacity is agreed per account against your expected peak traffic — confirmed during onboarding once you share your platform and expected volume.
Which platforms can trunk into CloudyVoice?
Any SIP-capable PBX, dialer, gateway or softswitch — including Asterisk, FreeSWITCH, 3CX, ViciDial and VOS3000 — plus SIP softphones and desk phones for smaller setups.
Is caller ID guaranteed on every route?
No. CLI presentation depends on the specific route and on what the destination network supports and permits; non-CLI routes do not present a number at all. CloudyVoice does not support caller ID spoofing intended to mislead.
Tell us what you need to carry
Send your destinations, expected daily minutes, traffic type and platform. We will come back with a route recommendation and a test plan.