Plug in Ethernet cable to router and wait for several seconds
Listen to IP Address which is read back to you
Enter IP Address into your web browser (For example. 192.168.1.10)
You should now see the Linksys PAP2T Web Interface as below
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.
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
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)
Adjustable audio frames per packet
Dual-tone multifrequency (DTMF): in-band and out-of-band (RFC 2833) (SIP information)
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
Echo cancellation (G.165 and G.168)
Voice activity detection (VAD)
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)
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