Auth¶

• IP: Enter the IP(s) of the customer's switch.
  FQDN can be used for Ingress-only switches.

• 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.

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

  Egress: 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.

• Dial Pattern: The default selection is the industry standard.
• CLI Prefix, Tech Prefix, Strip Digits: Do NOT Use these fields. Use the Parameter Rewrite tab to modify numbers.
• Bandwidth, Force From: Do NOT use these fields.
• 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: Set CLI Authentication for situations where Accounts are unable to use Tech Prefix to differentiate customers using the same IP.

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.

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.
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.

Example: International calls coming in with a + should be replaced with a specific country code.

• 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".
• Password: Must match with the configuration on the customer UAC.
• Channels, Flow Speed, Bandwidth; Do NOT use these fields.

• 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.

  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.

• 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.

• Smart Extension: Calls are sent to the Class5, not Class4 infrastructure. This feature is currently in Alpha and is not recommended.

  

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.
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.

See Voicemail for information on accessing Voicemail.