The emerging standard in enterprise voice.

One connection to thousands of calls, at a fraction of the costs.

SIP (Session Initiation Protocol) is the key to unlocking the powerful potential of next-generation IP-based communications. By creating virtual phone lines between PBXs and the PSTN, and linking your various office locations, this innovative solution allows your business to replace its costly ISDN PRIs and traditional lines with a simplified infrastructure that can reduce your costs and handle any call volume.

Capacity On Demand

Most businesses that rely on traditional voice solutions pay for up to 50% more lines than they need – just to be safe. With SIP Trunking, you can manage the same call capacity with fewer lines (or “channels”) and only subscribe to the lines that you really need because Capacity On Demand (also known as channel bursting) will always be available. There are no penalties for changing capacity up or down and no limits on the number of times you can change it.

Flex vs Fixed

With our SIP Trunking solution, you'll be able to make changes to your account instantly thanks to sophisticated online management tools. Did you know that it’s usually about 10 times faster to make changes to a SIP Trunking solution than to traditional PRI-based systems? It takes only 24 - 48 hours to add or remove SIP Trunking channels.

Aggregate channels and in-network calling

By streamlining your telephony infrastructure, SIP Trunking not only allows you to benefit form an extended in-network calling area, but also pools the available capacity across multiple locations. Instead of being locked by traditional PRIs, each with a fixed site-specific capacity, you can now dynamically allocate capacity based on demand, and enable any one site to act as a complete fail-over to ensure business continuity.

Flex vs Fixed

Thanks to our national network footprint, you can now create a local presence and eliminate long distance charges for your customers by selecting phone numbers from more than 800 Canadian cities and assigning them to a single SIP trunk.

Ensuring 100% Business Continuity

ThinkTel’s exclusive Surecall feature allows you to specify a call forwarding number for each active DID. Should a SIP trunk go down due to ISP outages, network issues, or problems with your on-site PBX, Surecall will be there to ensure your inbound calls still reach the right person

The ThinkTel Advantage

We have been delivering SIP-based voice services since their inception. In fact, our engineering team contributed to the definition of how SIP is being implemented today.

ThinkTel’s innovations in service delivery and billing typically reduce overall monthly telecommunications costs from 30-70%.

For example, by switching to a customized SIP Trunking configuration, a company with 400 employees in 10 cities (40 employees per site) will experience a massive drop in telecom costs: A savings of over $10,000 per month – a 36% reduction in the costs associated with voice.

These reductions in cost don’t come at the price of reliability, either. You can bank on our highly-redundant national voice network built on carrier-class technology. ThinkTel is also unique in that we support multiple network paths over diverse carriers and up to 8 simultaneous IP gateways.

Is SIP Trunking right for your business?

Do you have an IP-ready PBX?
Are you thinking about purchasing a new PBX?
Do you have multiple office sites?
Do you experience highly variable call volume?
Do you ever experience issues with busy signals or insufficient lines?
Are you seeking enterprise-level reliability and scalability?

If you answered YES to any of these questions, SIP Trunking is the right voice solution for your business.

Want to know more? Contact us and we'll tell you how you can try SIP Trunking... risk free!

Network specifications and capabilities

Up to 8 IP addresses per SIP Trunk for built-in redundancy
Codec support for G.711, G.726, G.721, and G.729a/b
QoS support via TOS bit and 802.1q VLAN tagging
Support for PPTP, IPSec, and L2TP VPNs
Fully redundant class 4/5 core switching powered by Metaswitches
Directory listings for all DIDs
10 or 11 digit dialling (e.164)
15 character manageable Caller ID and Caller Name
Basic and VoIP V911 with user-managed location for each DID
Private IP Cross Connection availability
Certified with Cisco CallManager and UC 520/540/560PBX’s
Certified directly with Microsoft OCS R2 and Lync

Important note regarding 911 service: VoIP 911 service has certain limitations relative to Enhanced 911 service that is available on most traditional telephone service.