Voice-over-IP
Voice-over-IP Features
The Vigor 2820VS has an analogue phone port and an analogue line port to provide full PSTN and VoIP integration on the same phones, via both the Internet and your regular analogue line. If you have an ISDN phone, you can also use that on the unit's 'Internal S0-Bus' to make additional VoIP phone calls.
The analogue 'Line' port connects into your regular analogue line (PSTN/POTS*). This then gives the telephones access to your analogue line to allow you to make calls as well as your VoIP facility (you can select the PSTN line instead of VoIP by dialling #0). Incoming calls are automatically switched through to your telephone(s) (either one or both) so that each phone can be used for both VoIP and POTS calls. Both telephones plugged into your router have access to VoIP and your analogue line. In addition, using the 'Digit-Map' facility you can set rules about particular call destinations using either the POTS line or your SIP/VoIP service. For example, local calls can be routed via your PSTN line (if you have a free calls package for example) whereas international calls can go via your preferred VoIP provider; there is flexibility to have several digit-map rules.
The above diagram should be used as a working schematic only, rather than an exact representation of the unit's connectivity and should be viewed with these notes: ISDN is not currently available on this model. The USB port can be used for a compatible 3G modem or a shared compatible printer connection. The secondary WAN Port ('WAN2') can be used for connection to any Ethernet feed for load balancing or Internet backup. You cannot use both WAN2(Ethernet) and USB for 3G at the same time. The LAN Ports are not shown on this diagram.
*POTS = Plain Old Telephone Service - The traditional analogue phone voice line in your home/office, e.g. B.T. That line may also be carrying your ADSL data signal.
- Vigor 2820VSn VoIP Features :
- One 'FXS' Phone Ports
- One 'FXO' Line Port for connection to analogue (POTS) line
- Automatic phone switch-over for incoming calls on either PSTN or VoIP
- Hotline (Dial preset number when handset lifted)
- Digit-Map facility for LCR selection
- Protocols: SIP, RTP / RTCP
- 12 SIP Registrar Accounts (for up to 12 VoIP providers)
- G.168 Line Echo-cancellation
- Automatic Gain Control
- Jitter Buffer ( 125ms )
- Voice Codecs:
- G.711 A / Law
- G.723.1
- G.726
- G.729 A / B
- VAD / CNG
- Tone Generation: DTMF , Dial , Busy , Ring Back , Call Progress
- DTMF Transmission: In Band / Out Band ( RFC-2833 ) / SIP info
- FAX / Modem Support G.711 Pass-through
- T.38 for FAX
- Supplemental Services (Dependent on ITSP):
- Caller ID
- Call Hold / Retrieve
- Call Waiting
- Call Waiting with Caller ID
- Call Transfer
- Call Forwarding ( Always , On Busy and On No Answer )
- DND (Do not Disturb)
- Call Barring ( Incoming / Outgoing )
- MWI ( Message Waiting Indicator ) ( RFC-3842 )
