SVP2000PP Business POE IP Phone User Manual
STEPHEN TECHNOLOGIES CO.,LIMITED
/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: [email protected] / Url: www.stephen-tele.com
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Enable Via rport
Enable/Disable system to support RFC3581. Via rport is special way to realize SIP NAT.
Enable PRACK
Enable or disable SIP PRACK function, suggest use the default config.
Long Contact
Set more parameters in contact field; connection with SEM server
Enable URI Convert
Convert # to %23 when send the URI.
Dial Without Register
Set call out by proxy without registration;
Ban Anonymous Call
Set to ban Anonymous Call;
Enable DNS SRV
Support DNS looking up with _sip.udp mode
Forward Type
Select call forward mode, the default is Off
Off
:
Close down calling forward
Busy
:
If the phone is busy, incoming calls will be forwarded to the appointed phone.
No answer
:
If there is no answer, incoming calls will be forwarded to the appointed
phone.
Always
:
Incoming calls will be forwarded to the appoint phone directly.
The phone will Prompt the incoming while doing forward.
Forward Phone Number
Appoint your forward phone number.
Server Type
Select the special type of server which is encrypted, or has some unique requirements or call
flows.
DTMF Mode
Select DTMF sending mode, there are three modes:
DTMF_RELAY
DTMF_RFC2833
DTMF_SIP_INFO
Different VoIP Service providers may provide different modes.
RFC Protocol Edition
Select SIP protocol version to adapt for the SIP server which uses the same version as you
select. For example, if the server is CISCO5300, you need to change to RFC2543, else phone
may not cancel call normally. System uses RFC3261 as default.
Transport Protocol
Set transport protocols, TCP or UDP;
RFC Privacy Edition
Set Anonymous call out safely; Support RFC3323and RFC3325;
Subscribe Expire Time
Overtime of resending subscribe packet. Suggest using the default config.
Enable Conference
number
Set to use sever conference.
MWI Number
Input the number of the server's voice-mail box
Click to Talk
Set click to Talk (need practical software support).
Signal Encode
Enable/Disable Signal Encrypt.
RTP Encode
Enable/Disable RTP Encrypt.
Enable Session Timer
Set Enable/Disable Session Timer, whether support RFC4028.It will refresh the SIP sessions.
Answer With Single
Codec
Enable/Disable the function when call is incoming, phone replies SIP message with just one
codec which phone supports.
Auto TCP
Set to use automatically TCP protocol to guarantee usability of transport as message is above
1300 byte
Enable Strict Proxy
Support the special SIP server-when phone receives the packets sent from server, phone will
use the source IP address, not the address in via field.
Enable GRUU
Set to support GRUU
Enable Display name
Quote
Set to make quotation mark to display name as the phone sends out signal, in order to be
compatible with server.
4.3.3.2. IAX2 Config