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Enabling legacy PBX for SIP Trunks

VoIP is more efficient in providing the required number of channels in respect of demand than traditional POTS (Plain Old Telephone Service) and PRI (Primary Rate Interface). Whenever a company required an extra channel even if only used for a few minutes during peak hours, it had to purchase an additional line that would not be utilized most of the time. Since channels are virtual with SIP trunks, the same company can now use the precise number of channels needed commensurate with demand. This feature of dynamic allocation adds flexibility over traditional systems. Correspondingly, the main reason to implement SIP trunks is the savings it can bring to a business. It can render expensive voice E1 circuits or a number of pots lines obsolete. SIP trunk providers offer much cheaper rates than conventional providers mainly because the Internet is used to deliver the calls bypassing any long distance carrier.

How does it Work?

Most new PBXs are already capable of handling SIP trunks, however in this case we are dealing with a legacy PBX that continues to provide service to the business.
The only requirement is to configure the traditional Trunk on the PBX; the rest is all transparent to it. At the legacy PBX, the call is converted to SIP and sent over the IP network to the SIP trunk provider which is in charge of terminating the call on the PSTN.
When a user needs to make an outbound call, it will initially be handled by the legacy PBX that routes the call to the trunks connected to the Voip gateway also located at the company premises. This VoIP gateway will convert the analog or TDM signal to an IP signal. This signal will then flow across the IP network, usually the public Internet, to the SIP trunk provider’s network which can route and terminate calls directed to any PSTN number. Inbound calls follow in exactly the opposite direction.


As shown in the diagram, a VoIP gateway is inserted between the Internet link and the PBX. Implementing this solution is fairly simple; however, the following fundamental requirements need to be taken into consideration:
The number of trunk channels available in the PBX and the type of bandwidth available will define the supported number of concurrent calls that can be sent through the SIP trunk. The Voip gateway can be analogue or digital, again, depending on the kind of trunk ports available in the legacy PBX. If it is analogue an FXS Voip gateway will be required.
The Internet link will carry the voice calls and the rest of the data from the company’s network therefore the required bandwidth needs to be carefully calculated and, in many cases, QOS/Voip prioritization policies need to be implemented in the company’s Internet facing router. We currently believe that for many businesses who stand to gain from such an infrastructure, the initial outlay can be recuperated in 6 months thereby reducing ongoing costs to near zero notwithstanding routine maintenance. Please contact Rustyice Solutions for more details on the required equipment and help in configuring it.


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