Linksys PAP2/SPA3000/SPA3102

    版本為 07:53, 24 Nov 2024

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    SPA3000_med.jpg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

     

    如何設定與 Elastix(Asterisk)  連接
    ◆ 在 SPA3000 的設定

    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)

     



     

     

     

     

    ◆ 在 Elastix 的設定

    《Trunks》Add SIP Trunk

    Outbound Caller ID : "PSTN Caller"<PhoneNumber>
    Maximum Channels : 1
    Trunk Name : pstn
    PEER Details : 

    disallow=all
    allow=ulaw
    canreinvite=no
    context=from-trunk
    dtfmode=rfc2833
    host=<SPA3000 IP>
    incominglimit=1
    nat=never
    port=5061
    qualify=yes
    secret=<pass>
    type=friend
    username=<username>
     

    Incoming Settings : 保持空白 

     

    《Inbound Routes》 Add Incoming Route

    Description : PSTNOut
    DID Number : <與 SPA3000 的一樣>
    Set Destination :  

    免電腦的設定技巧

     

     

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