網路電話閘道器 1 FXS 及 1 FXO
官方連結:http://www.linksys.com/servlet/Satel...VisitorWrapper
討論區:http://forum.voxilla.com/linksys-sip...um/index2.html
設定之前必須以 Admin 登入及選擇 Advanced 模式。建議在完成設定前先不要設定密碼,以避免設定過程中需要不斷輸入密碼的麻煩,所有設定皆完成後再設定管理者密碼。
1. 檢查 RTP Packet Size
SIP->RTP Parameters
RTP Packet Size = 0.020
2. 設定 PSTN Line
PSTN Line->
SIP Settings
SIP Port = 5061
Proxy and Registration
Proxy = 你的 Asterisk box 的 IP
Make Call Without Reg = Yes
Ans Call Without Reg = Yes
Register Expires = 300
Subscriber Information
Display Name = PSTN Call
User ID = pstn-1 (這必須與 Asterisk 的 Trunk - username 相同)
Password = yourpass (同上, 必須與 Trunk 相同)
Audio Configuration
Preferred Codec = G711u
DTMF Process INFO = Yes
DTMF Process AVT = Yes
DTMF Tx Method = Auto
Dial Plans
Dial Plan 2 = (S0<:123456789>) ;取代 1234567890 為實際的 PSTN 號碼,這必須與 Asterisk 的 Inbound Route 的 DID 號碼相同.
VoIP-To-PSTN Gateway Setup
VoIP-To-PSTN Gateway Enable = yes
VoIP Caller Auth Method = None
VoIP PIN Max Retry = 3 ; I did not change this.
One Stage Dialing = Yes ; very important
Line 1 VoIP Caller DP = none
VoIP Caller Default DP = none
Line 1 Fallback DP = none
VoIP Users and Passwords(HTTP Authentication)
保留所有原本的空白及下拉選單的 1
PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable = Yes
PSTN Caller Auth Method = none
PSTN Ring Thru Line 1 = no ; I use Asterisk for my routing.
PSTN Pin Max Retry = 3
PSTN CID for VoIP CID = Yes if you subscribe to CallerID service on your PSTN line, otherwise No
PSTN CID Number Prefix = (Leave Blank)
PSTN Caller Default DP = 2 ; important - here is where it sends the calls to.
Off Hook While Calling VoIP = No
Line 1 Signal Hook Flash To PSTN = Disabled
PSTN CID Name Prefix = (Leave Blank)
《Trunks》Add SIP Trunk
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《Inbound Routes》 Add Incoming Route
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