User manual

AT-323 SIP Phone User Manual (V1.43)
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Note The free service list of Internet：www.ip-calculator.com; www.ipchicken.com;
www.ipchicken.com;www.showmyip.com;www.whatismyip.com;
www.myipaddress.com; www.whatismyipaddress.com; ip.sbbs.net;
www.whatismyipaddress.net;checkip.dyndns.org
When “nat traversal” is set to “stun”, please put the URI of the stun server
into “nat addr”, in the format as “domain name/IP address : service port”.
The default service port for stun is 3478.
● nat ttl: When IP phone sit behind a NAT device, it will send packets to
server every “nat ttl” seconds to keep the port mapping on the NAT
device alive. “nat ttl” is an integer between 10 and 65535, default value is
20.
● phone number: The local phone number or username of this phone,
usually is allocated by system.
● account: With SIP system which requires authentication, please put
the username/account into this field.
● pin: With SIP system which requires authentication, please put the
password into this field.
● register port: The local UDP port registered with server to accept
incoming handshaking messages. The default port number is 5060.
● rtp port: RTP port is the port transferring and receiving voice packets
using UDP protocol. This is an even number between 1024 and 65535,
can’t be the same as “register port”.
● tos: Set the TOS field of the IP header of the RTP packets. The bigger
this value is 0, the higher priority the packet is .
● outbound proxy : Enable/disable Outbound proxy by checking/clearing
this box. If the system has an Outbound Proxy，please set the URI of the
Outbound proxy into “sip proxy” and set the domain name of SIP proxy
server into “domain/realm”. The default service port is 5060.
● dtmf: Set DTMF signal sending way by selecting inband audio, rfc 2833
and sip info from list box.
● dtmf payload : When DTMF select rfc 2833.This parameter can be
used indicating type of RTP payload type. The value can be use integer
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