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PAP2T

Compatible VoIP Analogue Adaptors

Price Ŗ34.62 Excl.Vat and shipping

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Linksys PAP2T

The PAP2T is the direct replacement for the extremely popular SPA-2002

This unit is suitable for connecting 2 plain analogue telephones to a SIP service provider, or a SIP iPBX. The two connections operate independently, with a separate SIP account. This in effect gives two telephone lines.

A fax machine can also be connected to one of the lines to provide support for sending faxes over a broadband Internet connection.

Secure Provisioning

The PAP2T supports remote provisioning using XML files transfered over HTTPS using client certificates. This is a very robust and secure mechanism that allows for easy mass deployment.

We can provide a managed provisioning service or can provide advice about setting up your own server.

The Linksys Range of VoIP adaptors used to be produced by Sipura, who were recently taken over by Linksys/Cisco. The devices are identical internally but now come housed in stylish new Linksys casings.

Product Overview

The Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports (Figure 1) enables high-quality feature-rich VoIP service through your broadband Internet connection. Just plug it into your home router or gateway and use the two standard telephone ports to connect analog phones or fax machines. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and reliable fax connections, even while using the Internet.
With Internet telephony, along with low domestic and international phone rates, impressive arrays of special telephone features are available. Choose your preferred local dialing number, regardless of where you live. Or add a virtual telephone number to have forwarded to your Internet phone. You can even add a toll-free number. The Cisco PAP2T Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephone service provider, such as caller ID, call waiting, voicemail, call forwarding, distinctive ring, and much more.

Features

Telephony

• Two voice ports (RJ-11) for analog phones or fax machines

• Impedance agnostics - eight configurable settings

• Call waiting, cancel call waiting, call waiting caller ID

• Caller ID with name/number (multinational variants)

• Caller ID blocking

• Call forwarding: no answer, busy, all

• Do not disturb

• Call transfer

• Three-way conference calling with local mixing

• Message waiting indication - visual and tone based

• Call return

• Call back on busy

• Call blocking with toll restriction

• Delayed disconnect

• Distinctive ringing - calling and called number

• Off-hook warning tone

• Selective/anonymous call rejection

• Hot line and warm line calling

• Speed dialing of eight numbers/addresses

• Music on hold

Specifications

Table 1 contains the specifications and package contents for the Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports.

Table 1. Specifications for the Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports

Specifications

*Note: Many specifications are programmable within a defined range or list of options. Please see the PAP2T Administration Guide for details. The configuration profile is uploaded to the PAP2T at the time of provisioning.

Data networking

• MAC address (IEEE 802.3)
• IPv4 (RFC 791) upgradeable to v6 (RFC 1883)
• Address Resolution Protocol (ARP)
• DNS-A record (RFC 1706), SRV record (RFC 2782)
• Dynamic Host Configuration Protocol (DHCP) client (RFC 2131)
• Internet Control Message Protocol (ICMP) (RFC 792)
• TCP (RFC 793)
• User Datagram Protocol (UDP) (RFC 768)
• Real Time Protocol (RTP) (RFC 1889) (RFC 1890)
• Real Time Control Protocol (RTCP) (RFC 1889)
• Differentiated Services (DiffServ) (RFC 2475), type of service (ToS) (RFC 791/1349)
• Simple Network Time Protocol (SNTP) (RFC 2030)

Voice gateway

• Session Initiation Protocol (SIP) v2 (RFC 3261, 3262, 3263, 3264)
• SIP proxy redundancy - dynamic via DNS SRV, A records deregistration with primary SIP proxy server
• SIP support in network address translation (NAT) networks (including Serial Tunnel [STUN])
• Highly secure (encrypted) calling via prestandard implementation of Secure RTP
• Codec name assignment

Voice algorithms

• G.711 (A-law and μ-law)
• G.726 (16/24/32/40 kbps)
• G.729 A
• G.723.1 (6.3 kbps, 5.3 kbps)
• Dynamic payload
• Adjustable audio frames per packet

Fax capability

• Fax tone detection pass-through
• Fax pass-though - using G.711
• Dual-tone multifrequency (DTMF): in-band and out-of-band (RFC 2833) (SIP info)
• Flexible dial plan support with interdigit timers and IP dialing
• Call progress tone generation
• Jitter buffer - adaptive
• Frame loss concealment
• Full duplex audio
• Echo cancellation (G.165/G.168)
• Voice activity detection (VAD) with silence suppression
• Attenuation/gain adjustments
• Flash hook timer
• Message waiting indicator (MWI) tones
• Visual messaging waiting indicator (VMWI) via frequency shift keying (FSK)
• Polarity control
• Hook flash event signaling
• Caller ID generation (name and number) - Bellcore, DTMF, European Telecommunications Standards Institute (ETSI)
• Music on hold client
• Streaming audio server - up to 10 sessions

Security

• Password-protected system reset to factory default
• Password-protected administrator and user access authority
• Provisioning/configuration/authentication:
• HTTPS with factory-installed client certificate
• HTTP digest - encrypted authentication via MD5 (RFC 1321)
• Up to 256-bit Advanced Encryption Standard (AES) encryption

Provisioning, administration, and maintenance

• Web browser administration and configuration via integral web server
• Telephone keypad configuration with interactive voice prompts
• Automated provisioning and upgrade via HTTPS, HTTP, Trivial File Transfer Protocol (TFTP)
• Asynchronous notification of upgrade availability via NOTIFY
• Nonintrusive, in-service upgrades
• Report generation and event logging
• Stats in BYE message
• Syslog and debug server records - per-line configurable web browser

Physical interfaces

• One 10BASE-T RJ-45 Ethernet port (IEEE 802.3)
• 2 RJ-11 FXS phone ports - for analog circuit telephone device (tip/ring)

Subscriber line interface circuit (SLIC)

• Ring voltage: 40-55 volts root mean square (Vrms) configurable *
• Ring frequency: 10-40 Hz *
• Ring waveform: trapezoidal and sinusoidal *
• Maximum ringer load: 3 ringer equivalence numbers (RENs)
• On-hook/off-hook characteristics:
• On-hook voltage (tip/ring): -50V nominal
• Off-hook current: 25 mA min
• Terminating impedance: 8 configurable settings, including North America 600 ohms, European CTR21

Regulatory compliance

FCC (Part 15, Class B), cUL, CE, IC-003, A-Tick

Power supply

• DC input voltage: +5V DC at 2.0A max.
• Power consumption: 5W
• Switching type (100-240V) automatic
• Power adapter: 100-240V - 50-60 Hz (26-34 VA) AC input, 1.8 m cord

Indicator lights/LED

Phone1, Phone2, Internet, Power

Documentation

• Quick-start installation and configuration guide
• User guide
• Administration guide - service providers only
• Provisioning guide - service providers only

Environmental

Dimensions

W x H x D

3.98 x 3.98 x 1.10 in.

(101 x 101 x 28 mm)

Unit weight

5.40 oz (153 g)

Operating temp.

32ē to 113ēF (0ē to 45ēC)

Storage temp.

-77ē to 158ēF (-25ē to 70ēC)

Operating humidity

10% to 90% noncondensing

Storage humidity

10% to 90% noncondensing

Package Contents

• Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports
• Power adapter
• RJ-45 Ethernet cable
• Quick installation guide

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Last modified: 14/06/2011.