SVP309P Office POE SIP 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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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
Set the interval of Subscribe.
Enable DNS SRV
Support DNS looking up with _sip.udp mode
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 recieves the patckets sent from
server
，
phone will use the source IP address, not the address in via field.
Enable GRUU
Set to support GRUU
Enable Displayname Quote
Set to make quotation mark to displayname as the phone sends out signal, in
order to be compatible with server.
