Keep your existing PBX
Asterisk, FreePBX, 3CX and other SIP-compliant systems connect without replacing handsets or extensions.
Khaltrix connects IP PBXs, cloud phone systems and contact-centre platforms to the public telephone network, with inbound numbers, outbound calling and failover routing configured and tested with your team.
How it connects
A SIP trunk replaces physical phone lines between your phone system and the public network. Phones, extensions and menus stay as they are.
Asterisk, FreePBX, 3CX and other SIP-compliant systems connect without replacing handsets or extensions.
Local, national and toll-free numbers delivered to your trunk, PBX or cloud phone system.
Concurrent channels sized to your busiest hour, separate from how many numbers you hold.
Calls in the other direction follow the same path in reverse.
Use cases
Support centres, sales teams, multi-site organisations, resellers and voice applications each need SIP configured differently.
Support numbers routed to menus and queues, with a tested failover destination.
Outbound channels sized to your agents, with approved caller ID and destination controls.
One central PBX with local numbers for each office and remote staff.
Separate trunks, numbers and limits for each of your customers.
Numbers routed over SIP to your gateway in front of a voice or AI application.
Product names are trademarks of their respective owners, listed for compatibility guidance only; no partnership or certification is implied. PBX setup guide
Inbound & outbound
The same trunk usually carries both directions, but each raises different questions. We plan each on its own terms.
Callers dial your numbers. The carrier routes each number to your trunk, and your PBX decides what happens next.
Your PBX applies its dial plan and sends calls to the trunk for delivery, presenting caller ID you are authorised to use.
Reliability & security
Audio problems, dropped calls and fraud usually trace back to a configuration nobody reviewed. These controls are planned with you before go-live.
Alternate destinations take over when your PBX is unreachable, agreed and tested before go-live.
Channel caps, destination restrictions and usage alerts suited to your traffic are agreed during setup.
TLS signalling and SRTP media where both your PBX and the route support them.
IP authentication for static addresses, and firewall guidance to accept SIP only from known sources.
International and premium-rate destinations are enabled deliberately, never by default.
Failed calls are investigated with SIP traces and response codes, and findings are shared with your team.
Technical capabilities
Each parameter is agreed with your team and recorded on your configuration sheet, so nothing is assumed.
IP-based (static public IP allow-list) or SIP registration with digest credentials.
Confirmed with you: Method per trunk and your public IPs
Capacity sized to your peak simultaneous calls, separate from how many numbers you hold.
Confirmed with you: Channel count and burst needs
SIP over UDP or TCP, and SIP over TLS for encrypted signalling where supported.
Confirmed with you: Transport and ports for your PBX
G.711 (µ-law / A-law) as the common baseline; G.729 or G.722 depend on route and platform.
Confirmed with you: Codec order on your PBX
Keypad tones sent as RTP events (RFC 4733) so menus and voicemail respond reliably.
Confirmed with you: Matching method on both sides
SRTP for call audio alongside TLS signalling, when both your PBX and the route support it.
Confirmed with you: Availability for your deployment
SIP ALG, RTP port ranges and public address settings, the usual causes of one-way audio.
Confirmed with you: Your firewall and router model
Alternate destinations for inbound calls when your PBX is unreachable; second trunks for outbound.
Confirmed with you: Failover targets and order
SIP response codes, call traces and call detail records to diagnose issues and reconcile usage.
Confirmed with you: Escalation contacts and record access
Why Khaltrix
We do not publish unverifiable uptime figures or customer logos. These are the working standards you can expect from the first conversation.
We ask about your platform, network and call patterns before proposing anything, so the quote reflects what you need.
Unsupported codecs, NAT configuration or outdated PBX firmware are identified before you commit.
Channels, numbers, minutes and one-off costs are itemised and explained line by line.
You receive the settings your trunk uses, so your team or IT partner can support it with confidence.
Inbound, outbound, DTMF, caller ID and failover are tested against a written plan before go-live.
You work with people who understand SIP and can discuss your configuration with your engineers.
Deployment
Each step produces something you can review: notes, a configuration sheet and test results.
Tell us about your phone system, users, numbers, destinations and expected call volumes.
We check your PBX and network, and confirm number and route availability.
Authentication, codecs, channels, numbers, failover and commercial terms are documented and agreed.
Your trunk is provisioned and your PBX configured with the agreed settings.
Inbound to each number, outbound to key destinations, DTMF, caller ID and failover.
Numbers are switched over or ported, live traffic is checked and early issues are reviewed.
Coverage & numbers
Number availability and calling routes vary by country and change with regulation. We confirm coverage for your list rather than publish a map that may be out of date.
Pricing
Instead of a single price list, you receive an itemised quote built from these factors, with every line explained.
Simultaneous calls you need, and any burst capacity.
Rates differ by country and between mobile and landline.
How many numbers, in which countries and of which type.
Toll-free and some inbound numbers carry per-minute costs.
Expected minutes shape metered, bundled or committed pricing.
Second trunks, alternate sites or geographic failover.
Work beyond a standard PBX connection, such as a voice application.
Number documentation and regional rules that need extra handling.
Who we work with
Guides
FAQ
Short answers here, with longer explanations in our guides.
A SIP trunk is a voice connection delivered over IP. It links your phone system (an IP PBX or a cloud phone system) to the public telephone network, so calls to and from ordinary phone numbers travel over your internet or private data connection instead of copper or ISDN lines.
Read the full explanation in What is SIP trunking?
Usually, yes. Most modern IP PBXs support standard SIP trunks. Before anything is ordered we review your platform, software version, network and call-routing needs, so any incompatibility (for example an old PBX without SIP support, or a missing codec) is found up front. Older PBXs can often be connected through a SIP gateway.
A channel is capacity for one simultaneous call, in either direction. Ten channels let ten calls happen at the same moment, whether inbound, outbound or a mix. Channels are separate from phone numbers: a business with 50 numbers may need only 8 channels if no more than 8 people are ever on external calls at once. See how to estimate channel requirements.
Not because of the SIP trunk. The trunk connects to your PBX, not to individual handsets, so the phones registered to your PBX keep working. You only need new devices if you are replacing the PBX itself, for example when moving to a cloud PBX that does not support your current handsets.
Yes. A SIP trunk normally carries both directions. Inbound calls arrive on the numbers (DIDs) assigned to the trunk and are delivered to your PBX; outbound calls are sent from your PBX to the trunk for delivery to the destination. Whether specific outbound destinations are enabled, and which caller IDs can be presented, is agreed during setup.
Send us the countries, cities or number types you need (local, national or toll-free) and the quantity. We confirm availability, any documentation the regulator requires (often proof of a local address or business registration), lead time and cost before anything is ordered. Start with a quote request or read about how coverage is confirmed.
Quotes are built from your actual requirements: the number of concurrent channels, the destinations you call, how many numbers you rent and where, the mix of inbound and outbound traffic, expected monthly volume, redundancy needs and any custom integration or compliance work. We explain each line of the quote.
It depends on what is involved. Connecting an already-compatible PBX for outbound calling can be quick once terms are agreed. New numbers, regulated countries and number porting add time that is set by number suppliers, regulators and carriers. We give you a realistic timeline for your specific project before you commit, and we test inbound and outbound calling before go-live.
Rough numbers are fine. The quote form walks you through them.
The biggest risk with any SIP connection is toll fraud: someone gaining access to your PBX or credentials and placing expensive calls. Good practice includes IP allow-listing, strong unique SIP credentials, keeping the PBX patched and off the open internet, restricting international and premium destinations you do not need, setting channel and spend limits, and encrypting signalling and media (TLS and SRTP) where supported. See SIP security best practices.
Request a quote
Share your phone system, the countries you need and roughly how many calls take place at once. For a detailed request, use the full quote form.