SPA3000 設定筆記
NAT穿透
Line 1-> NAT Keep Alive Enable: yes
台灣來電顯示及電話規格
Regional
Distinctive Ring Patterns
Ring1 Cadence: 60(1/2)
Ring and Call Waiting Tone Spec
Ring Voltage: 70
Ring Waveform: Sinusoid
Miscellaneous
Caller ID Method: DTMF(Denmark)
PSTN Line
FXO Timer Values (sec)
PSTN Answer Delay: 2 (2秒延遲, 來電顯示解析完成後才呼叫sip)
PSTN Disconnect Detection
Disconnect Tone: 480@-30,620@-30;1(.5/.5/1+2)
International Control
FXO Port Impedance: 320+1050||230nF
SPA To PSTN Gain: 2
PSTN To SPA Gain: 6
PSTN-to-VoIP
PSTN Line
Network Settings
Network Jitter Level: low
Jitter Buffer Adjustment: disable
Proxy and Registration
Register: no
Make Call Without Reg: yes
Ans Call Without Reg: yes
Dial Plans
Dial Plan 2: (S0<:userid your.sip.server="">)
PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: yes
PSTN CID Number Prefix: <空白>
PSTN CID For VoIP CID: yes
Off Hook While Calling VoIP: no
PSTN Caller Default DP: 2 (Dial Plan 2)
Line 1 Signal Hook Flash To PSTN: Disabled
Line 1
Network Settings
Network Jitter Level: low
Jitter Buffer Adjustment: disable
Proxy and Registration (註1)
Proxy: sip.linphone.org (或其他已有註冊帳號之sip server)
Register: yes
Make Call Without Reg: no
Ans Call Without Reg: no
Subscriber Information
Display Name: user name
User ID: userid (sip server註冊帳號)
Use Auth ID: no
VoIP Fallback To PSTN
Auto PSTN Fallback: yes (註1)
註1: 向sip server註冊失敗時 (例如斷電或網路不通), Line1可以直接PSTN撥號
因為我手機使用同一sip server的userid,所以連不上sip時一定也叫不到, 故 Make Call Without Reg設為no
熱線與外線
Line 1
Dial Plan
DialPlan: (<888:userid sip.server="">|<#0:>S0<: gw0="">|xxxxxxxx.)
話筒拿起撥888可直接呼叫 sip user
話筒拿起先撥 #0, 聽到 Tone 後用 PSTN 撥外線
888:userid>
Line 1-> NAT Keep Alive Enable: yes
台灣來電顯示及電話規格
Regional
Distinctive Ring Patterns
Ring1 Cadence: 60(1/2)
Ring and Call Waiting Tone Spec
Ring Voltage: 70
Ring Waveform: Sinusoid
Miscellaneous
Caller ID Method: DTMF(Denmark)
PSTN Line
FXO Timer Values (sec)
PSTN Answer Delay: 2 (2秒延遲, 來電顯示解析完成後才呼叫sip)
PSTN Disconnect Detection
Disconnect Tone: 480@-30,620@-30;1(.5/.5/1+2)
International Control
FXO Port Impedance: 320+1050||230nF
SPA To PSTN Gain: 2
PSTN To SPA Gain: 6
PSTN-to-VoIP
PSTN Line
Network Settings
Network Jitter Level: low
Jitter Buffer Adjustment: disable
Proxy and Registration
Register: no
Make Call Without Reg: yes
Ans Call Without Reg: yes
Dial Plans
Dial Plan 2: (S0<:userid your.sip.server="">)
PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: yes
PSTN CID Number Prefix: <空白>
PSTN CID For VoIP CID: yes
Off Hook While Calling VoIP: no
PSTN Caller Default DP: 2 (Dial Plan 2)
Line 1 Signal Hook Flash To PSTN: Disabled
Line 1
Network Settings
Network Jitter Level: low
Jitter Buffer Adjustment: disable
Proxy and Registration (註1)
Proxy: sip.linphone.org (或其他已有註冊帳號之sip server)
Register: yes
Make Call Without Reg: no
Ans Call Without Reg: no
Subscriber Information
Display Name: user name
User ID: userid (sip server註冊帳號)
Use Auth ID: no
VoIP Fallback To PSTN
Auto PSTN Fallback: yes (註1)
註1: 向sip server註冊失敗時 (例如斷電或網路不通), Line1可以直接PSTN撥號
因為我手機使用同一sip server的userid,所以連不上sip時一定也叫不到, 故 Make Call Without Reg設為no
熱線與外線
Line 1
Dial Plan
DialPlan: (<888:userid sip.server="">|<#0:>S0<: gw0="">|xxxxxxxx.)
話筒拿起撥888可直接呼叫 sip user
話筒拿起先撥 #0, 聽到 Tone 後用 PSTN 撥外線
888:userid>
1 個意見:
您好:
看了您的筆記.想問一下
Dial Plans
Dial Plan 2: (S0<:userid your.sip.server="">)
Line 1
Dial Plan
DialPlan: (<888:userid sip.server="">|<#0:>S0<: gw0="">|xxxxxxxx.)
以上二個要如何填寫,可有範例嗎?
thanks
由 山可斯 提供, 於 7:24 上午
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