Auth¶

• IP/CIDR: Enter specific IPs or use CIDR notation to specify an entire subnet.
  FQDN can be used for Ingress-only switches. You can assign multiple IPs.

• Switch Direction: The available options are from the perspective of the customer switch (PBX, dialer, etc), and describe how that switch interacts with the ConnexCS switch. For switches that send and receive calls from ConnexCS, there should be separate entries for each direction. Two types of switches are as follows:

  This endpoint receives calls. This switch receives calls from ConnexCS. (Note: When selected, this gives the option of using the FQDN rather than the switch IP.)

  This endpoint sends calls. This switch sends calls to ConnexCS.

• Channels: Set the maximum number of concurrent calls for this switch.

• Flow Speed: Set the Calls Per Second (CPS) (0 = unlimited calls).

  

• Manufacturer and Version: These references fields allow you to enter the customer switch Manufacturer and Version, if required (these fields are not functional; they are informational only).

• Protocol: Select the protocol for SIP (call signaling) and RTP (transport / audio).

  UDP: SIP and RTP are unencrypted and transported over UDP.

  TCP: SIP is sent over TCP, RTP over UDP.

  TLS: SIP is sent over TLS (TCP), RTP over UDP.

  TLS & SRTP: SIP is sent over TLS (over TCP), RTP is encrypted with SRTP.

  SMPP: SMPP, for SMS, is not currently supported.

• Port: Default = 5060. If using TLS protocol, this should be set to 5061. ConnexCS is compatible with non-standard ports.

• Dial Pattern: The default selection is the industry standard.
• CLI Prefix, Tech Prefix, Force From: Do NOT Use these fields. Use the Parameter Rewrite tab to modify numbers.
• Username, and Password: Set when sending calls out (egress switch direction) to a remote system. Set this to allow the ConnexCS switch to operate as a client or UAC. Not typically recommended unless the customer has a very specialized system.
• Force NAT: Forces the switch to read the IP address the traffic was received from, not the IP in the SIP packet. (See Far-End NAT Traversal for more details on how ConnexCS handles NAT for SIP.)
• Intercept Reinvite: The only situation where this is recommended is when a customer's equipment doesn't support REINVITES. Enable this to correct issues with dropped calls by having ConnexCS respond to the INVITES, which can help keep calls up if they are being disconnected by the far-end switch.
• Outbound Proxy: Enter the IP address of a Proxy server for calls to route to before being sent to the carrier. This rewrites the UAC IP in the VIA field of the SIP header. This reduces management overhead as a customer only needs to authorize a single IP. Additionally, multiple addresses can be load-balanced using the AnyEdge system.

• Flags: Check CLI Authentication flag for situations where Accounts are unable to use Tech Prefix to differentiate customers using the same IP. Check Disable NAT flag to disable NAT settings.

  

  • CLI Authentication: Select this flag to distinguish between multiple customers sharing the same IP address by using CLI Authentication instead of Tech Prefix.

    • Configuration

      • CLI Routing: CLIs are used as a secondary authentication mechanism to identify which customer traffic belongs to.

      Example

      Joe and Bob share an IP address but have different CLIs; the system uses these CLIs to route calls correctly.

      • Setup Process

        • Configuring IP Authentication:

          1. Navigate to Customer Customer [Name] Auth IP Authentication click on the blue + sign.

          2. Under Advanced settings Flag Enable CLI Authentication.

          3. This step ensures that the system will use CLI Authentication to differentiate customers with the same IP address.

  

  • Setting Up CLIs or Regular Expressions:
    • Navigate to Customer Customer [Name] Routing CLI click on the blue + sign.
    • Enter the specific CLIs, or Regular expressions associated with the customer.
    • This configuration allows the system to match incoming call CLIs with the defined patterns.

  

graph LR
A[Call Arrives] --> B{Check Source IP Address}
B --> |If the IP address matches with the entry in the Authentication configuration| C{Check CLI of the incoming call}
C --> |If both CLI and IP address match the configuration for a particular customer| D{Route Call to Customer}

All Codecs are supported unless specifically set as Restricted here.

The Parameter Rewrite tab is used to manipulate data as it comes into the system, particularly useful for formatting phone numbers.

• Configuration:

  • Number Formatting: Strip leading zeros or add country codes based on regular expressions.

    Example: Changing 01782 to 441782 for international formatting.

  • Destination Number or CLI Manipulation: Apply these manipulations on either destination numbers or CLIs.

• Steps for re-writing:

  1. Click +.
  2. Type: Select the parameter to modify.
  3. Current: Enter the prefix for the destination number, or the CLI. You also have the option to generate a regular expression using AI Generate Regular Expression.
  4. New: Enter what should replace the current information.
  5. Use the Testing Input field to verify if the replacement is working as expected.
  6. Click Save when done.
  7. If a parameter rewrite is already created, you will have the ability to test it from the main tab.

• SIP Profile: You can select the created SIP Profile here.
• Username: This will be the Username used for SIP authentication (must match configuration on the customer UAC). If the Customer has Internal Number Block set on the Main tab, you can only select the Username from available extensions. If a Username is already in use on the Account, they will get an error "Duplicate User Detected". Required when sending calls out to a remote system; the switch behaves in UAC (User Agent Client) mode.
• Password: Must match with the configuration on the customer UAC.
• Channels: Set the maximum number of concurrent calls for this switch.
• Flow Speed: Set the Calls Per Second (CPS) (0 = unlimited calls).

• Protocol: Select the protocol for SIP (call signaling) RTP (transport/audio).

  UDP: SIP and RTP are unencrypted and transported over UDP.

  TCP: SIP is sent over TCP, RTP over UDP.

  TLS: SIP is sent over TLS (TCP), RTP over UDP.

  TLS & SRTP: SIP is sent over TLS (over TCP), RTP is encrypted with SRTP.

  SMPP: SMPP, for SMS, is currently not supported.

• IP Allow list: Enter specific IPs or use CIDR notation to specify an entire subnet.

• NAT/SIP Ping: Set behavior of pings sent from ConnexCS back to the customer through their firewall to their UAC. This helps when there are remote agents connecting to the switch. NAT/SIP Ping is used to keep the network address translation (NAT) open, ensuring calls can be received.

  Disabled: No pings are sent

  Enabled: Send UDP pings every 60 seconds, helping to keep some longer calls (1800 or 3600 seconds) up.

  Enabled (Timeout): Send UDP pings every 60 seconds and disconnect the call (terminate registration) if the pings aren't returned.

graph TD
    A[Outbound Packet Sent] --> B[NAT Creates Temporary Port Mapping]
    B --> C[NAT Remembers Port Mapping and Routes Responses]
    C --> D[UDP Has No Built-in KeepAlive Mechanism]
    D --> E[NAT Can Close Inactive Ports After Timeout]
    E --> F[System Sends SIP Ping Every 60 Seconds to Keep Alive]
    F --> G{Check for Response from Device}
        G -- Yes --> H[Response Received, Continue Operation]
        G -- No --> I[No Response, System Times Out]
        I --> J[System Deregisters Device Due to Inactivity]

• Retain DID: When you enable this, inbound calls will retain the destination number (DID), and the call is sent into the system, rather than using the SIP Username or routing calls to a predefined destination.

  DID Setup

  • DID: +1234567890

  • Destination: Internal SIP User TestRIS1

  • Behavior:

    • Without Retain DID: Call to +1234567890 routes directly to TestRIS1.
    • With Retain DID: Call keeps +1234567890 as the destination, allowing further routing by downstream systems.

• Smart Extension: It simplifies call transfers and enables advanced features like call barging and interception by centralizing REFER message handling within the Class 5 system. Unlike traditional SIP workflows, where the UAC or softphone manages transfers, this approach offloads complexity to us, enhancing functionality, user experience, and control through seamless integration of Class 4 and Class 5 systems.

  • Smart Extension Workflow:

    graph TD
    %% Define node styles
    classDef startEnd fill:#d1e7dd,stroke:#0f5132,stroke-width:2px,font-weight:bold;
    classDef process fill:#f9f9c5,stroke:#d4a017,stroke-width:2px,font-weight:normal;
    classDef decision fill:#ffffcc,stroke:#999900,stroke-width:2px,color:#000,font-style:italic;
    
    %% Nodes
    A[UAC Initiates Call]:::startEnd
    B[Class 4 System Routes Call]:::process
    C{Upstream Carrier Determines Transfer Needed?}:::decision
    D[REFER Message Sent Back to Originating System]:::process
    E[Class 5 System Intercepts REFER Message]:::process
    F[Class 5 System Processes Transfer]:::process
    G[Class 5 System Initiates New Call]:::process
    H[New Call Flows to Class 4 System]:::process
    I[Call Routed to Updated Destination via Carrier]:::startEnd
    J[Call Continues Normally]:::process
    
    %% Links
    A --> B
    B --> C
    C -- Yes --> D
    D --> E
    E --> F
    F --> G
    G --> H
    H --> I
    C -- No --> J

    • Advantages

      • Simplified Call Transfers: The complexity of handling REFER messages is shifted from the UAC to the Class 5 system. Ensures a seamless experience for the user, as the originating device is unaware of the transfer.

      • Advanced Call Manipulation: By routing calls through the Class 5 system, advanced features such as call barging, whispering, and interception are enabled.

      Examples

      1. Call Barging: Useful in customer service scenarios where a supervisor might need to join an ongoing call to assist or guide the agent.
      2. Whispering: Ideal in training situations, allowing a mentor to provide instructions to the agent without the customer hearing.
      3. Interception: Crucial in security or compliance settings, enabling a manager to take over a call in progress if necessary.
      • Centralized Complexity Management: Keeps call logic and transfer handling within the provider's infrastructure, reducing the burden on end-user devices.

      Use Case Example

      graph TD
      %% Define node styles
      classDef startEnd fill:#d1e7dd,stroke:#0f5132,stroke-width:2px,font-weight:bold;
      classDef process fill:#f9f9c5,stroke:#d4a017,stroke-width:2px,font-weight:normal;
      classDef decision fill:#ffffcc,stroke:#999900,stroke-width:2px,color:#000,font-style:italic;
      classDef normal fill:#f2f2f2,stroke:#666666,stroke-width:1px,color:#000;
      
      %% Nodes for With Smart Extension
      A1[UAC Calls 12345]:::startEnd
      B1[Class 4 System Routes Call]:::process
      C1{Upstream Carrier: Number Unavailable?}:::decision
      D1[Upstream Carrier Sends REFER Message]:::process
      E1[Class 5 System Intercepts REFER]:::process
      F1[Class 5 System Processes Transfer]:::process
      G1[Class 5 System Initiates New Call to 12346]:::process
      H1[New Call Flows Through Class 4 System]:::process
      I1[Call Routed to Updated Destination via Carrier]:::startEnd
      
      %% Links for With Smart Extension
      A1 --> B1
      B1 --> C1
      C1 -- Yes --> D1
      D1 --> E1
      E1 --> F1
      F1 --> G1
      G1 --> H1
      H1 --> I1
      
      %% Nodes for Without Smart Extension
      A2[UAC Calls 12345]:::startEnd
      B2[Class 4 System Routes Call]:::process
      C2{Upstream Carrier: Number Unavailable?}:::decision
      D2[Upstream Carrier Sends REFER Message]:::process
      E2[UAC Receives REFER Message]:::process
      F2[UAC Handles Transfer:Potential Failure]:::normal
      G2[UAC Initiates New Call to 12346]:::normal
      H2[New Call Flows Through Class 4 System]:::normal
      I2[Call Routed to Updated Destination via Carrier]:::startEnd
      
      %% Links for Without Smart Extension
      A2 --> B2
      B2 --> C2
      C2 -- Yes --> D2
      D2 --> E2
      E2 --> F2
      F2 --> G2
      G2 --> H2
      H2 --> I2
      
      %% Subgraphs (optional for grouping)
      subgraph With Smart Extension
          direction LR
          A1 --> B1 --> C1 --> D1 --> E1 --> F1 --> G1 --> H1 --> I1
      end
      
      subgraph Without Smart Extension
          direction LR
          A2 --> B2 --> C2 --> D2 --> E2 --> F2 --> G2 --> H2 --> I2
      end
      
      %% Notes outside of subgraphs
      I1 --> |Note:| N1["Seamless Transfer"]:::normal
      I2 --> |Note:| N2["Complexity and Potential Failures"]:::normal

      Important Note

      An additional charge of $0.003 per call applies to use the Smart Extension capability.

All Codecs are supported for the SIP user unless specifically set as "Restricted" here.

The Parameter Rewrite tab is used to manipulate data as it comes into the system. It is most useful when you need to create automatic replacements for destination numbers or CLI, so a number is formatted in the appropriate E164 format.

1. Click +.

2. Type: Select the parameter to modify.

3. Current: Enter the prefix for the destination number, or the CLI. You also have the option to generate a regular expression using AI Generate Regular Expression.
4. New: Enter what should replace the current information.
5. Use Testing Input field to verify if the replacement is working as expected.
6. Click Save when done.
7. If a parameter rewrite is already created, you will have the ability to test it from the main tab.

If you enable Voice Mail, you can set which email address receives messages, reset the Voicemail Password, and view and delete current messages.

Key Features:

• Enable voicemail for specific extensions.

• Configure email notifications for new voicemails.

• Set up a voicemail password for retrieval.

• To access voicemail, dial *1 (subject to routing settings).

See Voicemail for information on accessing Voicemail.