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FusionPBX

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Installation

Debian 11

wget -O - https://raw.githubusercontent.com/fusionpbx/fusionpbx-install.sh/master/debian/pre-install.sh | sh;

cd /usr/src/fusionpbx-install.sh/debian && ./install.sh

NAT Setting

Web Admin > Advanced > Variables > IP Addresses

  • external_rtp_ip: <server-public-ip>
  • external_sip_ip: <server-public-ip>

重啟 freeswitch

systemctl restart freeswitch

驗證

Web Admin > Status > SIP Status

  • sofia status profile internal: ext-rtp-ip, ext-sip-ip
  • sofia status profile external: ext-rtp-ip, ext-sip-ip

RTP Port

/etc/freeswitch/autoload_configs/switch.conf.xml:

<!-- RTP port range -->
<!-- If no definitation the port range would be 16384 - 32768 -->
<param name="rtp-start-port" value="16384"/>
<param name="rtp-end-port" value="17000"/>

Gateway to Asterisk

On FreePBX

  1. Added a custom context 'from-ext-sip-server' with the module Custom Contexts.
  2. FreePBX Admin > Connectivity > Custom Contexts > Add Context
    • Context: from-ext-sip-server
    • Description: Whatever
    • Outbound Routes: <allow-some-route>
  3. Add Trunk
    • Trunk Name: fusionpbx
    • PEER Details:
host=sip.osslab.tw
type=peer
context=from-ext-sip-server
nat=yes
insecure=port,invite

On FusionPBX

Web Admin > Accounts > Gateways > Add

  • Gateway: myasterisk
  • Proxy: <my-asterisk-sip>
  • Register: False
  • Profile: external
  • Enable: Checked

Web Admin > Dialplan > Outbound Routes > Add

  • Gateway: myasterisk
  • Dialplan Expression: 9 Digits
  • Prefix: <blank>
  • Enable: True

另一個方法：以 Bridge 取代 Gateway

Applications > Bridge

    Name: <自定義> Destination: sofia/Internal/$1@your-freePBX-IP:5060

    Advanced > Access Controls > Add

      Name: FreePBX Default: deny Nodes
        Type: allow CIDR: xxx.xxx.xxx.xxx/32 Domain: <CIDR 與 Domain 擇其一> Description: <自訂>

         Outbound Route 記得要選 Bridge。

         

        Voicemail to Email

        Web Admin > Accounts > Extensions > Select extension and Edit

        • Voicemail Mail to: <your-email-addr>

        Web Admin > Advanced > Default Settings > Email

        • address_type: add_address
        • method: smtp
        • smtp_auth: True
        • smtp_from: <sender-from-addr>
        • smtp_from_name: <sender-from-name>
        • smtp_host: smtp-relay.sendinblue.com
        • smtp_hostname: False
        • smtp_username: <smtp-user>
        • smtp_password: <smtp-pass>
        • smtp_port: 587
        • smtp_secure: tls
        • smtp_validate_certificate: True

        Send Test Email

        Web Admin > Status > Email Logs > TEST

        Bug Fixed:

        [ERR] switch_cpp.cpp:1465 [database] can not bind parameter: undefined parameter: email_from

        Edit /usr/share/freeswitch/scripts/resources/functions/send_mail.lua

        if (email_from == nil or email_from == "") then
        email_from = settings:get('email', 'smtp_from', 'text');
        from_name = settings:get('email', 'smtp_from_name', 'text');
end
-- added by Alang
-- fixed: [ERR] switch_cpp.cpp:1465 [database] can not bind parameter: undefined parameter: email_from
email_from = 'noreply@your.domain';

if (email_from == nil or email_from == "") then
        email_from = address;
elseif (from_name ~= nil and from_name ~= "") then
        email_from = from_name .. "<" .. email_from .. ">";
end

        Voicemail Transcription

        IBM Watson API

        IBM Cloud > Watson > STT

        • API Key: SQCKJOwC_4VoRozrhw-zm2vYFcxgztFlb2LskGr
        • API URL: https://api.au-syd.speech-to-text.watson.cloud.ibm.com/instances/{GUID}

        FusionPBX > Advanced > Default Settings > 新增以下參數

        Category Subcategory Type Value Enabled
        voicemail transcribe_provider text watson True
        voicemail watson_key text {your watson key} True
        voicemail watson_url text {watson url} True
        voicemail transcribe_language text en-US True
        voicemail transcribe_enabled boolean true True
        voicemail json_enabled boolean true True

        按下 "Reload" 套用設定。

        FusionPBX > Status > SIP Status

        按下 "Flush Cache"，"Reload XML" 與 "Rescan"。

        測試 STT API

        # Audio file: audio-file.flac
curl -X POST -u "apikey:{API_KET}" \
--header "Content-Type: audio/flac" \
--data-binary @audio-file.flac \
"https://api.au-syd.speech-to-text.watson.cloud.ibm.com/instances/{GUID}/v1/recognize"

        Auto Provisioning

        Provision (Linksys PAP2T)

        Web Admin > Advanced > Default Settings > Provision

        • enabled: True, Enabled: True
        • admin_password: <自訂密碼,硬體電話的管理存取>, Enabled: True
        • http_auth_username: <空白>, Enable: False
          NOTE: PAP2T 不支援 http 認證的 Auto Provisioning。
        • ntp_server_primary: tw.pool.ntp.org, Enable: True
        • spa_dial_plan: (*xx|*0xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|*xxxx|xxxxxxxxxxxx.), Enabled: True
        • spa_time_zone: GMT+08:00, Enabled: True
        Device

        Web Admin > Accounts > Devices > ADD

        • MAC Address: <PAP2T-MAC-Addr>
        • Domain: <選擇適合的網域>
        • Enabled: Checked
        • 其餘欄位保持空白或預設
        Extension

        Web Admin > Accounts > Extensions > Add 

        • Extension: <自訂分機號>
        • Device Provisioning:
          • Line: 1
          • MAC Address: <選擇 PAP2T 的 MAC>
          • Template: cisco/pap2t
          • Domain: <選擇合適的網域>
          • Enabled: Checked
        驗證 Provision Configuration

        瀏覽器輸入 http://<fusionpbx-ip-addr>/app/provision/?mac=<pap2t-mac-addr>

        如果有輸出 XML 格式的參數設定檔內容，表示以上的設定正確。

        Linksys PAP2T

        PAP2T Web Admin > Provisioning

        • Provision Enable: yes
        • Profile Rule: http://<fusionpbx-ip-addr>/app/provision/?mac=$MA

        TIP: 如果發生無法註冊成功，且同一個 NAT 網路環境還有其他 SIP 終端裝置。試試修改 PAP2T 的 Local SIP port 為其他 port。

        修改 PAP2T 的 Local Port (透過 FusionPBX)

        Web Admin > Accounts > Devices > Select: <PAP2T-Mac-Addr> > Lines > Line 1

        • Port: 1001 (預設是 5060) 

        重啟 PAP2T 以便重新同步新設定。

        Cisco IP Phone 8841

        Let's Encrypt

        WebRTC

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