How to Use Voice-over-IP (VoIP)

Many modern phone systems use voice-over-IP (VoIP) protocols. The basic idea behind these systems is to carry voice traffic across packet (Internet style) networks. VoIP telephones are usually network appliances that are plugged into Ethernet jacks. These systems can be used with Active Call Center, but they will require either an analog telephone interface between the network and Active Call Center OR the VoIP system should have TAPI / wave 1.4-2.0 and drivers.

Voice-over-IP Gateways and Analog Telephones

The following are required in order to use Active Call Center with a VoIP gateway and analog telephone interface:

• Either the VoIP phone system must have analog jacks that offer a standard analog (POTS) interface (i.e., the analog lines must have dial tones, busy signals, etc.).
• The VoIP phone system must translate touch tone digits encoded in VoIP formats (out-of-band DTMF) to audible analog DTMF. Conversely, the phone system should take audible analog DTMF and convert to VoIP out-of-band DTMF.
• The VoIP phone system must offer consistent audio quality.

Several network gear makers have analog telephone adapter products for VoIP networks. One that we tested successfuly with Active Call Center is the Cisco ATA 186 (Analog Telephone Adapter). The ATA 186 offers up to two analog ports, but there are other products that are similar that offer 4 or more lines. For more details on the Cisco ATA 186 device visit Cisco's ATA 186 information page.

Voice-over-IP systems with TAPI / Wave Drivers

Many newer IP telephony systems include TAPI and TAPI / wave drivers (for example, Avaya IP Office). Many systems that have TAPI 1.4-2.0 TAPI and TAPI / wave drivers can be used successfully with Active Call Center just as if they were analog telephony cards.