3 Recenze SKU: l12301

Voip internet adaptér SPA122 ATA Voip Telefon Hlasové Adaptér SIP Voip ATA s Routerem Telefonní Adaptér retail box doprava zdarma


Plug in Ethernet cable to router and wait for several seconds

Dial ****110#

Listen to IP Address which is read back to you

Enter IP Address into your web browser (For example.

You should now see the Linksys PAP2T Web Interface as below

IP :

Username: admin

Password: admin

Click on SIP Click on the ‘SIP’ tab Change the following SIP Timer Values: Reg Max Expires = 600 Reg Retry Intvl = 10 Reg Retry Long Intvl = 20

(Changing the above will allow your PAP2T to recover from registration failures more quickly.) Change the following SDP Payload Types: RTP-Start-Loopback-Codec: G711a Change the following NAT Support Parameters (optional): STUN Enable: yes STUN Test Enable: yes STUN Server: stun.kiwilink.co.nz

(The STUN settings should be optional. Your PAP2T should work fine without STUN enabled, so you can enable/disable these settings if you are having issues registering.)

Features and Benefits

The SPA122 provides an easy-to-use VoIP solution that offers:

• Toll-quality voice and carrier-class feature support: The C isco SPA122 delivers clear, high-quality voice communication under a variety of network conditions. Excellent voice quality in challenging, changeable IP network environments is made possible through the advanced implementation of standard voice coding algorithms. The C isco SPA122 is interoperable with common telephony equipment such as fax, voicemail, private branch exchanges (PBXs) and key telephone systems (KTSs), and interactive voice response systems.

• Large-scale deployment and management: The C isco SPA122 enables service providers to provide customized services to their subscribers. It can be remotely provisioned and supports dynamic, in-service software upgrades. A highly secure profile upload saves providers the time and expense of managing and preconfiguring or reconfiguring customer premises equipment (CPE) for deployment.

• Outstanding security: The C isco SPA122 supports highly secure, encryption-based methods for communication, provisioning, and servicing.

• Compact size: Designed for small spaces, the C isco SPA122 can be installed as a desktop unit or mounted on a wall.

• Comprehensive feature set: The standards-based C

isco SPA122 is compatible with Internet VoIP provider features such as caller ID, call waiting, voicemail, call forwarding, distinctive ring, and much more to provide a complete, affordable, and highly reliable VoIP solution.

• Easy installation and changes: The web-based configuration utility enables quick deployment and easy changes.

• Investment protection: Businesses that are growing rapidly can use the solution with other C isco Unified Communications solutions, providing industry-leading investment protection.

• Peace of mind: C

isco solutions deliver the solid reliability you expect from C isco. All solution components have been rigorously tested to help ensure easy setup, interoperability, and performance. Table 1 lists the specifications for the C isco SPA122 ATA with Router.

Table 1. Product Specifications



*Note: Many specifications are programmable within a defined range or list of options. Please see the C isco SPA100 Series Administration Guide for details. The configuration profile is uploaded to the C isco SPA122 at the time of provisioning.

Data networking

M A C address (IEEE 802.3)

IPv4 (RFC 791) upgradeable to IPv6 (RFC 1883)

Address Resolution Protocol (ARP)

Domain Name System (DNS) A record (RFC 1706) and SRV record (RFC 2782)

Dynamic Host Configuration Protocol (DHCP) server and client (RFC 2131)

DHCP client reservation

DHCP Options 159 and 160

Point-to-Point Protocol over Ethernet (PPoE) client (RFC 2516)

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) and type of service (ToS) (RFC 791/1349)

VLAN tagging (IEEE 802.1p)

Simple Network Time Protocol (SNTP) (RFC 2030)

Upload data rate limiting: static and automatic

QoS voice packet prioritization over other packet types

M A C address cloning

Port forwarding

SIP channels support both UDP and TCP transport

VPN pass-through with IP Security encapsulating security payload (IPsec ESP), Point-to-Point Tunneling Protocol (PPTP), and Layer 2 Tunneling Protocol (L2TP)

Voice gate way

SIPv2 (RFC 3261, 3262, 3263, and 3264)

SIP proxy redundancy: dynamic through use of DNS SRV A records

Reregistration with primary SIP proxy server

SIP support in Network Address Translation (NAT) networks (including Serial Tunnel [STUN])

Highly secure (encrypted) calling using Secure RTP (SRTP)

Codec name assignment

G.711 (A-law and μ-law)

G.726 (32 kbps)

G.729 A

Dynamic payload

Adjustable audio frames per packet

Dual-tone multifrequency (DTMF): in-band and out-of-band (RFC 2833) (SIP information)

Voice features

QoS (Ethernet port upstream bandwidth control)

Independent configurable dial plans with interdigit timers and IP dialing (1 per port)

Call progress tone generation

Jitter buffer: Adaptive

Frame loss concealment

Full-duplex audio

Echo cancellation (G.165 and G.168)

Voice activity detection (VAD)

Silence suppression

Comfort noise generation (CNG)

Attenuation and gain adjustments

Flash hook timer

Message waiting indicator (MWI) tones

Visual messaging waiting indicator (VMWI) using frequency shift keying (FSK)

Polarity control

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