Unique VoIP solutions and assistance, according to your needs


VoIP Consulting

VoIP Consulting

What we can do: Well, almost everything in VoIP. You have problems configuring something in your VoIP device? You need a custom VoIP feature implemented, and nobody can help you out? You need assistance with Wholesale/ITSP soultions/calling cards/call shop/PBX/pre and post-paid billing/custom Asterisk and OpenSIPS/Kamailio configurations/fax to email, email to fax solutions? You've come to the right place!

Hosted LCR Engine

Hosted LCR Engine

Highly available and extremely lightweight, our LCR engine is able to take on 10K requests / second. Our route searching algorithm has made it the highest performing LCR engine available in the market. When a SIP invite is received, we can provide either a 300 or 302 reply message to acknowledge that request and then you can route the call accordingly.

Wholesale traffic termination

Wholesale traffic termination

  • Unlimited growing potential
  • Tier-1 Carrier Grade Equipment
  • Excellent voice quality and price
  • #1 Coverage for the World, Guaranteed
  • Guaranteed CLI on all destinations
  • T38 faxing(with fallback to G711) is supported on all landline destinations
  • Failover/Redundant networks
  • Equipment for handeling any call volume(up to 13000 new calls per second)
  • No long-term contracts
In short, we make the phones ring

Custom Frontends

Custom Frontends

Custom coded VoIP user portals, admin interfaces, user self care portals, multi tenant Asterisk and Freeswitch GUIs. Whatever your business needs, we can construct it, from the ground up. But first we need a blueprint. Calliotel works with clients to discover precisely what they are looking to achieve and how software can best meet those needs.

Building custom softswitches

Building custom softswitches

  • Custom Kamailio, OpenSIPS development, custom scripting
  • Help implementing CALEA solutions for Opensips/Kamailio, Asterisk and Freeswitch
  • CDR/Billing and Large-scale SIP deployments
  • High-availability / Resilient infrastructures
  • Multi-lateral peering - Global solutions
  • NAT traversal / Media proxies
  • MVNO solutions and assistance with SS7 interconnections

24x7 Unmatched Support

24x7 Unmatched Support

We provide "round-the-clock" 24x7 service and support, 365 days a year. When your business needs support, we're here to help. We offer critical systems and "need" support for Asterisk, Freeswitch, Kamailio and OpenSIPS. Problems cannot always be anticipated, and this is where we jumps into action. When your systems are at their worst, our staff is at their best.For customers requiring a guaranteed response time, Calliotel enters a Service Agreement (SLA) with response times guaranteed.


Give a new dimension to your VoIP service.

We know what we're doing because we've been doing it for a long time. And we're good at it, too. As a team, we bring over eighty years of industry-acquired expertise to the table. VoiP System Engineers, VoIP Software Developers, programmers, brand consultants, and technical support specialists. At Calliotel, you'll find it all, done well.

Why such a collaborative process?

Because we believe in doing it right the first time. The more we understand the requirements of the task ahead of us, the better equipped we will be to work out all of the custom-designed details in the final needs. While many companies will list for you the different platforms and vendors they develop in--and yes, we do it all--it's not technology that guarantees successful results. What matters most is listening to clients, knowing how to efficiently develop within a budget, and ultimately custom designing solutions that are both flawless and intuitive. Our experienced team of consultants and developers will design what you need, implement it, test it, and even maintain it. From building VoIP systems to modifying source codes to creating phone applications, we'll make sure that what we do works for you.
Cleanly coded website from  customer selfcare interface to the sophisticated admin portal. The following are just a few of the ways the portal helps you manage your VoIP business:
  • Manage customer features, DID numbers, credit limits, rate tables and more
  • Create SIP Trunking accounts
  • Add lines and configure free IP to IP calling
  • Assign termination credit and call restrictions
  • Order DID numbers, SIP termination and all of our other services for real-time provisioning
  • Customize your SIP Proxy
  • Create/edit rate tables
  • View CDRs
  • Customize product names, voice prompts and dial plan
  • Check international rates and ASR
  • View your accounting statement of account
  • Make manual credit card payments and set default payment methods
  • View invoices
  • Open help desk ticket
  • Set automatic recharge thresholds
  • Manage DID numbers

Custom configurations for OpenSIPS

OpenSIPS (Open SIP Server) is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server, is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing engine, OpenSIPS unifies voice, video, IM and presence services in a highly efficient way, thanks to its scalable (modular) design. What OpenSIPS has to offer, comes in a reliable and high-performance flavour - OpenSIPS is one of the fastest SIP servers, with a throughput that confirms it as a solution up to enterprise or carrier-grade class. OpenSIPS has to offer many important and interesting features. To mention some of the most important ones:
  • SIP registrar server
  • SIP router / proxy (lcr, dynamic routing)
  • SIP load-balancer or dispatcher
  • SIP front end for gateways/asterisk
  • SIP NAT traversal unit
  • SIP application server
  • SIP to SMS gateway (bidirectional)
  • SIP IM server (chat and end-2-end IM)

Easy. Fast. Anytime. Anywhere.

Internet Faxing Simplified

  • As easy as Email.
  • No setup required.
  • No hardware or software.
  • Easier than a fax machine.

Fax Faster Online

  • Confirmations delivered via email.
  • Quickly find your online fax archives.
  • Faxes delivered to your email.
  • No more paper jams.

Choose Your Own Number

  • Get the fax number you want.
  • Choose by area code or by city.
  • Largest selection of worldwide numbers.
  • A real fax number for your business.

Private and Secure

  • No more shared fax machine hassles.
  • Internet faxes delivered straight to your email.
  • Easily control confidentiality.
  • Private faxes remain private.

Save Money

  • No more fax machine.
  • No toner needed.
  • No extra phone line needed.
  • No paper needed.

Save Time by Going Digital

  • Freedom to fax from anywhere.
  • Save, forward, and organize as easily as email.
  • Online fax archive.
  • No more lost faxes.

Green Technology

  • No need to print before faxing.
  • Preserve the environment.
  • Save on electricity without a dedicated fax machine.
  • Reduce waste of paper, energy and consumables.

Start Faxing in Minutes

  • No installation required.
  • All you need is email and an selfcare portal account.
  • Instant activation of your fax number.
  • Start faxing immediately.

Do it all with using WebRTC

Imagine a world where your phone, TV and computer could all communicate on a common platform. Imagine it was easy to add video chat to your web application. That's the vision of WebRTC. Want to try it out? WebRTC is available now in the customer selfcare portal, as a module. One of the last major challenges for the web is to enable human communication via voice and video: Real Time Communication, RTC for short. RTC should be as natural in a web application as entering text in a text input. Without it, we're limited in our ability to innovate and develop new ways for people to interact. Historically, RTC has been corporate and complex, requiring expensive audio and video technologies to be licensed or developed in house. Integrating RTC technology with existing content, data and services has been difficult and time consuming, particularly on the web. Gmail video chat became popular in 2008, and in 2011 Google introduced Hangouts, which use the Google Talk service (as does Gmail). Google bought GIPS, a company which had developed many components required for RTC, such as codecs and echo cancellation techniques. Google open sourced the technologies developed by GIPS and engaged with relevant standards bodies at the IETF and W3C to ensure industry consensus. In May 2011, Ericsson built the first implementation of WebRTC. WebRTC has now implemented open standards for real-time, plugin-free video, audio and data communication.  

Calliotel LLC
Voice over IP solutions for Opensips, Kamailio, Asterisk and Freeswitch
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