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Ingress Routing

Management Customer [Customer Name] Routing

Ingress Routing is the process that allocates an incoming call (dialed by our customers) based on the assigned Customer Rate Card, which then Egresses the call to a specified provider. This makes it possible to deploy several rate cards both with and without a prefix.

First, it checks the longest prefix, then it checks the shortest prefix for a match. If no prefix gets matched, it matches the rate cards with mutually exclusive destinations. If there are several rate cards with the same prefix, you must set up a dial plan with a Tech Prefix to identify the correct card.

Routing Templates and more

Create templates for customer routing in Routing Global.

For more information on Routing, see Routing Setup in our Video Guides for a detailed walk-through.

You can find additional documentation in the Routing Overview and Routing Strategy sections.

Configure Routing

View and configure existing routes on the Routing tab in the Customer card. To create a new route, click + in Ingress Routing.

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  • Rate Card: Also known as Tariff, this allows you to select the rate card used on a customer's account. You can handle these calls in the following three ways:

    • Internal: Send a call to the ConnexCS Class5 (Voice Mail, Interactive Voice Response (IVR), etc.). If selected, the "Auto" option becomes available, which will generate dial strings from all possible internal extensions.
    • Extension: (uses SIP users in Customer Auth configured SIP Users) Send a call to a Session Initiation Protocol (SIP) Authenticated user on the account.
    • Customer IP: (uses IPs in Customer Authconfigured IPs) Send a call from an agent back to the customer's Private Branch eXchange (PBX), using either the Tech Prefix (e.g.: #9) or a Dial String (e.g.: ^[0-9](4)$).
    • To Carriers: Choose a carrier to send the call to a location outside of the ConnexCS system.
  • Tech Prefix: This lets you distinguish a route from an inbound party. When several customers share the same IP address, each customer needs an individual Tech Prefix so the switch can route calls correctly. It enables service providers to differentiate between several rate cards.

  • Dial String Prefix Set: Helpful for commonly used sets of prefixes. Rather than entering a complete list of prefixes for the UK, for example, you can create a predefined Prefix Set (defined under Setup Advanced Prefix Set) and then select it here for appropriate customers.
  • Dial String: Only allows a dialled number to pass through if it matches the defined dial string (or "dial pattern"). (If you customer enters nothing, it matches everything and attempts to send all calls).

This doesn't work if you have more than one Rate Card as the system won't know which one to use). Each prefix gets listed once per line, it allows both prefixes and regular expressions:

Prefix 441 442

Regular Expression (RegEx) ^44(1|2)

Combination (UK Landline & Canada) 441 442 ^1(204|226|236|249|250|289|306|343|403|416|418|438|450|506|514|519|579|581|587|604|613|647|705|709|778|780|807|819|867|902|905)

Using Rate Cards for multiple countries

Ingress Routing isn't independently aware of the card type you are using or,  more specifically, the appropriate dial strings it needs to send. For example, if you are using both UK and US cards, you need to enter appropriate dial strings in the routes you set up for each card type.

  • Enabled: You can enable and disable the routes here.

Price Limits

  • Capped Rate and Provider Capped Rate: Set the maximum cost of a call. Calls that exceed the set rate won't get connected. For example, for customers with flat rate accounts, which allows to dial all UK numbers but premium numbers, you would set the Provider Capped Rate at 0.01, so any call that the provider might charge over that amount wouldn't get completed.

  • Profit Assurance: When Enabled, only calls that are profitable pass-through; any call that costs more than the retail rate aren't allowed to complete. This is particularly useful for A-Z routes or NPA-NXX rate cards. Keep in mind that enabling it adds an extra Post-dial delay (PDD) to the call.

  • Block Connect Cost: Block any call that has a connection fee.

  • FTC DNC Report ANI Block (USA): When Enabled, ConnexCS will take a copy of FTC data (using the FCC's Do Not Call (DNC) Reported Calls Data API) and add it to the system. We can then block callers from known spammer CLI / ANI's.

  • DNO: Click here to know more about it.

Capacity Limits

  • Channels: Set the maximum number of channels/live calls allowed for this route.

  • Max Duration: Set the maximum amount of time (in seconds) that the call gets to exist before getting terminated, typically for the case of a missed BYE.

Call Timeouts

A VoIP call is stateful, even though its protocol is stateless. This means that both the sides should be informed on finishing of the call. They do this with a BYE message. If the BYE message goes missing, the call will continue forever.

Max Duration is a method for setting up Missing BYE Protection. Another approach is to use a SIP Ping to determine when the connection has timed out. This sends a SIP packet to the remote end of the conversation every 30 seconds. This checks to see if the other side is still aware of an ongoing conversation. If it does'nt receive a response or informed that the conversation is not active, it disconnects the call.

RTP Time-out: This is another way to check for an active call based on whether audio is passing. If there is no audio passing for a pre-set interval, our Real-time Transport Protocol (RTP) array will notify the switch and instruct it to end the call. This won't work if RTP Mode is set to direct.

Asterisk pings

Asterisk doesn't have SIP Ping (OPTIONS) enabled by default. If your customer / carrier is using Asterisk, you may need to disable this if they don't have it enabled on their side, as calls will typically disconnect after 30 seconds.

  • Flow Speed (CPS): Limits the calls per second. You should set this for each customer card assigned to the customer account.

  • CPS Spike Buffer: Limit a spike of calls by spreading them over a longer period of time. This essentially manages a large volume of calls over a short period of time. Once the buffer limit reaches its threshold, the calls per second kicks in, distributing the spike of calls.

CPS Buffering

CPS Buffering: Used to manage large volumes of calls over a short period of time. This process maximises saturation and increases call completion within a given CPS restriction. It does this by removing spikes and borrowing capacity from future seconds.

If incoming traffic exceeds your pre-set CPS, it holds the call for one second and then tries again. You can increase the second count in the CPS Spike Buffer field.

Changing the CPS Buffering value only affects calls that exceed the CPS. The delay will show as increased PDD on the call, each second the system will emit a 100 Trying (High CPS, Buffering) response to show the status/progress of the call.

  • ASR Plus assists capacity management by helping you define how to handle connections for known failed numbers. For information on the ASR Plus options, see ASR Plus Details below.

  • Balance Disconnect this feature checks the balance every 60 seconds. It will disconnect the call when the balance plus the debit limit is below $0.


Balance Disconnect only takes into account the completed calls; it excludes any active calls.


  • ScriptForge: Set a custom JavaScript to run from within the ConnexCS platform in-line with the call. Some example operations could be checking a Do Not Call list or forcing a CLI. For more information about setup and operation, see the ScriptForge page.

  • Timeout: Set how long the script may run.

  • Timeout Action: This option lets you decide the action when the timeout occurs.
  • VARS (TOML): Select the variables you want pass into the ScriptForge script.


Used for troubleshooting, you can remove carriers from a route and run a quick test.

  • Lock (Allow): One or more rate cards from the list of available providers.
  • Exclude (Deny): Exclude access to one or more rate cards in the list of available providers.
  • DNC (Do Not Call) List: The customer won't be able to able to dial the numbers in the specified DNC list. You can add the list of numbers in the Database.

    Apart from your own DNC list you can also choose United States Federal DNC. Choosing not to accept telemarketing calls is possible because of the National Do Not Call Registry.

  • Block Destination Type: You can select and block the calls to various destinations (carriers) like Mobile, Fixed, Paging, etc.

  • Spam Scout Scoring: It blocks Spam calls based on the CLIs. You can either Block All, Allow All, Block Most Spam, or Block Least Spam.

Exclude Use Case

If a customer reports an issue with a carrier or route, you can come here and set the carrier / route to Exclude and Save, then come back and remove it, and do a Delay and Save for a later date.


  • Transcoding: Enter the number of channels allowed for transcoding. This is a limited option. The best use case is for customers in low-bandwidth areas that want to use G.729. Be aware that if you don't have enough transcoding capacity, calls will fail.
  • SIP Ping: Send regular pings to ensure both sides of a call are still up. Enabled is the recommended setting.

    Option Result
    Disabled No SIP pings will be sent
    Enabled Both Sides SIP pings sent in both directions
    Enabled (Downstream Only) SIP Pings sent to the location where the call originated
    Enabled (Upstream Only) SIP Pings sent towards where the call is TO (terminated)
  • SIP Session Timer (SST): SST is Passive by Default, but Enabled is the recommended setting. When enabled, SST ensures there is no ghost or long-duration calls get billed when one or both sides have hung up. A timer activates when the call starts and refreshes the call every X number of seconds by sending a RE-INVITE. SST has surpassed SIP Ping Timeout as the best way to prevent long-duration calls. Note that any SST shorter than sixty (60) seconds gets rejected.

    SST Option Result
    Default Passive SST, No headers gets changed and no SST gets engaged, all RE-INVITES will propagate through the system enables
    Enabled Both ConnexCS will send SIP Session Timers to both legs of the call
    Enabled (Upstream) ConnexCS will use SST with the carrier
    Enabled (Downstream) ConnexCS will use SST with the customer
    Suggest Session-Expire headers and Min-SE gets added to packets sent to the carrier encouraging the use of SST
    Disabled All timer headers are removed
  • RTP Media Proxy: This defaults to Auto, but selecting a zone (by continent) is the current recommendation. The following options allow you to set where RTP media server for this route for this customer:

Direct RTP (no proxy)- Bypass ConnexCS, so media flows directly between the customer and carrier. If the customer is using a firewall or other NAT device incorrectly, then media may not flow between the carrier and the customer. Using this setting also means that if there are audio issues, the issue can't be ConnexCS. Since it isn't likely to be the carrier, the issue would typically exist on the customer's end.

Zone- Choose any of the regional servers, but it's recommended that you select a location close to a provider or your customer. Temporarily selecting a different region to route media traffic can be helpful in diagnosing call problems.

The recommended RTP Media Proxy servers are the Closest (To ConnexCS) Server or the Closest (Elastic) Server.


SIP Ping and SIP Session Timers can't be enabled at the same time.

  • RTP Proxy Mode: If a connection via our service fails and you have selected relaxed, it will automatically fail over to the backup.

Strict- This will enforce the proxy engagement. If the proxy can't engage with the call, the call won't get established.

Free accounts are limited to how many RTP Proxy channels get enabled, this may prevent calls from connecting if you have more channels than our free accounts allow you to have.

Relaxed- This will make the best efforts to engage the RTP Proxy; if it can't get engaged because of either network errors, or because you don't have enough RTP capacity, the calls will connect directly.

When should I use RTP Proxy?

Use an RTP Proxy if you don't want your customers to know your providers.

When should I avoid using an RTP Proxy?

You have other equipment in your SIP set-up that will act as a Media Relay or you want to run a test to see if audio problems are related to the ConnexCS switch.

RTP Proxy distinctions

When a call gets established between the customer and the provider, audio can be setup in one of two ways:

With RTP Proxy Without RTP Proxy
Audio Path Indirect Direct
Audio Quality Excellent Unbeatable
Latency Low Lowest
Information Leakage No Yes*

*While it's doubtful that any information will get logged in the customer / providers switch when the audio gets engaged, it's possible for an engineer to learn this information from a SIP trace, PCAP, or by looking at transit locations. DTMF Detection ONLY works when RTP Proxy mode gets enabled.

  • Call Recording: This allows you to record and store calls, which are then found in:

  • Logging

  • Management Customer [Customer Name] CDR
  • Management File Recording

An extra charge per recorded call of $0.003 gets added to existing fees or charges, so choose carefully how many calls to record:

Disabled- no calls get recorded.

Sampling- Choose from a 1%, 5%, 25%, or 50% sample of your calls (e.g: 1% will record 1 of every 100 calls, 25% will record 25 of every 100 calls, etc.).

Enabled (Always On)- Record all calls.

The Call Recording setting is disabled

You need to enable the feature first on the account in Setup Settings Packages before it gets enabled here for individual customers.

  • Block DTMF: This option allows you to either pass or block DTMF through your calls.

Make sure your carrier supports the DTMF feature.


For advanced routing, click to select a Prefix Set and assign a Routing Strategy. This gives you greater control over how routes get selected for a given customer.


  • Public Notes: Notes entered here get displayed on the Customer Portal when logged in.
  • Private Notes: These will get displayed to the customer in the Control Panel.


  • Fraud Profile: Apply one of the Fraud Profiles configured under Setup Advanced Fraud Profile.
  • Fraud Mode: Specify how the profile will gets into application, this is dependent on the Fraud Mode Thresholds configured in the Profile.


    Low - Alert or Block Calls

    High - Block Calls or Account

Disabled Routes

Routes highlighted in red on the customer Routing page gets disabled. Open the route, click Enabled, and then Save to enable them

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Use Case for the Tech Prefix

Using Tech Prefix with SIP User "Parameter Rewrites" allows for significant granularity to manage permissions for how to connect a user's calls.

  1. Use Parameter Rewrite on the SIP User (found in Customer Auth SIP User Parameter Rewrite) to add a number for calls from this SIP User:

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  2. Add a Tech Prefix for that user in Routing. In this example, it would be 1234.

  3. Set how you want those calls routed: Internal to Class5, out to a provider, etc.

Answer Seizure Ratio Plus Details

ASR (Answer Seizure Ratio) is the number of connected calls divided by the total number of calls (represented as a %).

ASR Plus is a proprietary ConnexCS technology that filters known failed, non-existent / working numbers between the customer and the terminating, or destination, carrier. This is useful with larger call volumes. Unless it's turned off or customized otherwise, ASR+ is active for 90% of calls, which grants the opportunity for the database replenishment.

Value Description
Off ASR+ Disabled
ASR+ (Low) Active on 30% of calls
ASR+ Active
ASR? When ASR+ gets enabled on the provider card
ASR+? When ASR+ gets enabled on the provider card, only known connected calls pass-through specific providers
ASR++ Only known connected calls pass-through (not used frequently because it's typically overly strict)

Advantages of ASR

  • Quick failure of known bad numbers.
  • Reduces response time for your customers.
  • Improves the ASR of the traffic that your upstream carrier sees.
  • Highly effective for call centre traffic.

Disadvantages of ASR

  • Marginal impact on your NER due to false positive matches. This is usually kept within tolerances of < 0.1%.
  • Doesn't offer improvements for all destinations.