How Are You Facilitating Your Inbound and Outbound DID’s?
Direct Inward Dialing and how it works
Many of us may have questions about how DIDs work and how to provision them. DID stands for “Direct Inward Dialing”. DIDs are typically used in conjunction with an IP PBX, to route incoming and outgoing calls to their correct source or destination. Almost every IP PBX has a method of facilitating DID’s, whether that be internal or external to the server. However, products used to facilitate them are often in question.
There are three ways that most businesses are “bringing in” their DID’s. The first method is Analog Trunking or “POTS” (Plain Old Telephone Service). Analog Trunks may be comprised of physical copper PSTN lines paid for and supplied by your local telephone company. These are pure RJ-11 analog lines, no different from your “landline” wall jack at home. Most SOHO applications utilize analog POT’s lines since they are more cost effective. The typical number of physical PSTN lines is usually around four to eight, and will vary depending upon the number of inbound/outbound calls the business needs to support. Each physical POTS line is equal to one channel, and represents a 1:1 ratio. Each physical PSTN line also has a single DID number associated with it. If you have four analog POTS lines, you have four DID’s or channels available to make inbound and outbound calls.
Removing Voip from the picture for a second.
Take VOIP out of the picture for a second…. in a true analog environment, each PSTN line would be connected to an analog telephone (user), and that telephone would be associated with a specific DID number. Let’s bring back VOIP now….analog telephone lines are NOT connected to the phones themselves, since in most cases, you will be using VOIP phones, but rather into the central location… the IP phone system. Analog lines are facilitated within the IP PBX via FXO PCI cards. If you have eight incoming RJ-11 PSTN lines, you will essentially need an eight-port RJ-11 analog PCI card much like the Sangoma A20004D or Digium AEX808E. Simply connect each RJ-11 connection into the ports on these cards, most IP PBX’s will auto-detect their presence, and you are now permitted to configure your analog trunks or channels within the IP PBX. Since the analog DID’s are now facilitated at a central level, they are not on a pure 1:1 basis because when a VOIP phone accesses this trunk group to make an outbound call, it is not associated to that one DID always, it will simply grab the next available DID within the channel group and use that. This allows users to add more VOIP phones to the scenario without physically increasing their number of POTS lines. Essentially, if you have 16 VOIP users, but only experience around 8 concurrent calls at a time, you would only need 8 POTS lines, rather than 16. Please note, you are not limited to four or eight-port analog PCI cards. These numbers were offered as a very basic example. Please check out Sangoma and Digium on voipsupply.com for further clarification on these cards.
Digital connections are becoming very popular amongst larger organizations because of ease of use and cost savings over Analog POTS Trunking. In most large applications, there is a need for 24, 48, or even 96 + voice channels. The easiest method to facilitate this number of voice channels is via digital T1 lines using a T1 provider. Most IP PBXs have the ability to integrate digital T1 connections, either through T1 digital PCI cards or external T1 gateways. A T1 is an essential 24 individual lines (equivalent to 24 analog POTS lines) delivered over a single pipe. A T1 is configured at the IP PBX level through digital trunk groups or channel groups.
2 Types of T1 Provisioning
T1’s can be provisioned in two flavors. The first is through PRI signaling (Primary Rate Interface). PRI’s contain 24 channels in total but only allow for 23 configurable channels. The remaining channel is a “work-horse” channel so to speak, performing all of the overhead signaling work for each of the 23 available channels. The second method of T1 signaling is a method called CAS (Channel Associated Signaling). CAS signaling with the “T” allows for 24 channels to be configured as voice, data, or both. Each channel performs its own work to allow for proper signaling to take place. Please check with your T1 provider to ensure which method they are using, and opt for the best method to fit your needs. Digital T1 lines are integrated with an IP PBX very easily….simply connect the T1 to a Sangoma A101 card or Digium TE-122P card, which are both Single T1 cards. However, you are not limited to a single T1; a standard digital PCI card can have up to four T1 ports incorporating 96 channels within the system, but that doesn’t mean you can’t add a second quad T1 card. For those larger T1 applications I just spoke about, please refer to the Sangoma A104D card or the Digium TE-420B.
SIP Trunking
The final method to bring in your DID connections is through a method called SIP Trunking. SIP Trunking is done completely over a data connection (Typically T1, Fiber, ADSL or Cable), and is provisioned to your PBX through a connection to the WAN and your SIP Trunk provider. SIP Trunking is becoming one of the most cost-effective methods of acquiring inbound and outbound channels because there is no need for a physical connection on the premises. The only down-side? Certain providers will only provide local SIP Trunks to specific geographic locations. Check with your prospective VoIP provider for availability of SIP Trunking services in your area. SIP Trunking is quickly gaining in popularity with businesses of all sizes. I think you will see more providers start to offer this solution, if they don’t already, and we should see SIP Trunking to almost every local DID as time and demand progress. Everything nowadays seems to be “up in the clouds.” SIP Trunking is no different, and is conforming to the future specifications of what consumers, both large and small, are expecting.
