Paytia

Connect Your Business Telephony
to Paytia

In order for Paytia to capture sensitive business and identity data it must be active on a telephone call. Two proven telephony connection modes for Digital SIP and Analogue PSTN call forwarding. Dual Mode for complete audio control between call legs, Single Mode for simple inbound call handling.

Instant Processing
Bank-Grade Security
Cost Effective
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Telephony Call Flow Configuration

Dual Mode • Single Mode • SIP • PSTN

Dual Mode Telephony

Paytia sits between Call Leg 1 and Call Leg 2, controlling audio flow on both sides of the conversation. Perfect for Channel Separation and DTMF Suppression services where Paytia needs full control over what each party hears.

Single Mode Telephony

Paytia answers a single inbound call leg and reacts to incoming tones. Used for Conference Pay, API-driven payments with SIP headers, and self-service IVR where only the customer is on the call.

Connectivity Options

Choose the connectivity method that best fits your infrastructure

Digital SIP

Session Initiation Protocol

Paytia's proprietary SIP technology stack can be adapted to meet any telephony connection scenario with advanced call control integration.

SIP Call Back URI

Pass the SIP endpoint Paytia should route received calls back to. Enables flexible call routing and return paths for dual mode operation.

X-Paytia-Pbx1 SIP Header

Pass additional routing information or call control settings to Paytia. Can be extracted and used on call leg 1, or relayed back to sender on call back call leg 2 route.

Advanced Integration

SIP headers enable much more integrated call control - perfect for modern cloud platforms, unified communications, and digital contact centers.

PSTN Call Forward

Traditional Analogue

Direct dial telephone numbers for simple, reliable connectivity over traditional analogue infrastructure.

Business Account Level

Direct dial numbers assigned at business account level for centralized call routing. All calls from your business forward through your dedicated Paytia number.

Agent/User Level

Individual numbers assigned per agent or user for granular call routing. Each user forwards and receives calls from their own Paytia number.

Traditional Infrastructure

Works with legacy PBX systems, traditional phone lines, and mixed technology environments. No SIP infrastructure required.

Telephony Call Flow Diagrams

SIP sequence diagrams showing how Paytia connects to your telephony infrastructure

Dual Mode Call Flow

Paytia controls audio between two call legs (Agent ↔ Paytia ↔ Customer)

AgentCall Leg 1PaytiaAudio ControlCustomerCall Leg 21INVITE (SIP/PSTN)INVITE (SIP/PSTN)200 OK200 OKBoth call legs established - Paytia sits in the middleAgent and Customer connected via Paytia2RTP AudioRTP Audio (forwarded)RTP AudioRTP Audio (forwarded)Paytia passes audio through normallyBoth parties can hear each other3Start Payment (DTMF/API)Paytia Takes Control of AudioChannel Separation: Disconnects audio between legsDTMF Suppression: Masks keypad tones4Hold Music + ProgressPayment PromptsDTMF (Card Details)Agent and Customer hear different audio - Full control

Single Mode Call Flow

Paytia answers single inbound call leg (Conference Pay, API+SIP Headers, Self-Service IVR)

AgentCustomerPBXConferencePaytiaInbound1Call ConnectedAgent and Customer on normal call2Place on Hold3Conference Paytia NumberINVITE200 OKPaytia joins as third party - Single inbound call leg4DTMF (Agent ID)Agent authenticates via keypad5Resume CallAll three parties now on conference call6Start Capture (UI Action)Payment Prompts (All Hear)DTMF (Card Details)Paytia processes on single inbound leg - Agent can hear tones

Single Mode: API + SIP Headers

Agent ID passed via SIP headers, payment triggered via API

CRM/ContactCenter SystemPaytia APIPaytiaInbound Call1POST /payment/start(agentId, customerId, amount)paymentId, phoneNumberAPI creates payment session with Agent ID2INVITE + X-Agent-ID HeaderSIP Header: X-Agent-ID: (agentId)200 OKPaytia matches SIP header Agent ID to API sessionSingle inbound leg with context from API3Payment PromptsDTMF (Card Details)4Payment ResultWebhook: payment.completedAgent ID from SIP header links call to API session - No conference needed

Single Mode: Self-Service IVR

Customer only - no agent on the call

CustomerPaytia IVR1INVITE200 OKSingle inbound call - No agent2Welcome + Menu OptionsDTMF (Menu Selection)3Payment PromptsDTMF (Card Details)4Success MessageBYEFully automated - No human involvement

Key Differences

Single Mode

  • Single inbound call leg to Paytia
  • Paytia reacts to incoming tones
  • Three use cases:
  • • Conference Pay (agent + customer)
  • • API + SIP Headers (agent ID in header)
  • • Self-Service IVR (customer only)
  • No call routing changes needed

Dual Mode

  • Two separate call legs through Paytia
  • Full audio control over both sides
  • Can disconnect audio between parties
  • Can play different audio to each party
  • Requires call routing through Paytia
  • Enables Channel Separation & DTMF Suppression

Telephony Mode Controls

Paytia provides configurable controls to manage call completion behavior

Answer & Ring Tone

Answer call leg 1 and play ringing tone to the caller while establishing connection. Provides familiar audio feedback during call setup.

Configurable Delay

Set delay (in seconds) between receiving call leg 1 and starting call leg 2. Ensures phone systems have time to clear agent's call before return call.

Auto-Disconnect

Drop call legs when transaction completes. Paytia auto-disconnects when request finishes, leaving host phone system back in control of active call legs.

Multi-Endpoint Routing

Try multiple endpoint trunk routes in linear order for disaster recovery. Automatic failover ensures call completion even if primary routes fail.

Payment Services Activation

After your telephony is connected to Paytia using either Dual Mode or Single Mode, you can activate secure payment capture services on the call:

Channel Separation

Requires Dual Mode connectivity

DTMF Suppression

Requires Dual Mode connectivity

Conference Pay

Uses Single Mode connectivity

How Telephony Connection Works

Dual Mode • Single Mode • SIP • PSTN

1

Choose Connection Mode

Select Dual Mode for full audio control (Channel Separation/DTMF Suppression) or Single Mode for conference-based integration (Conference Pay).

2

Configure Connectivity

Connect via SIP for modern digital systems or PSTN call forwarding for traditional infrastructure.

3

Activate Payment Services

Once telephony is connected, activate Channel Separation, DTMF Suppression, or Conference Pay on the call.

Frequently Asked Questions

Dual Mode means Paytia sits between two call legs (Agent and Customer) with full control over audio on both sides - used for Channel Separation and DTMF Suppression. Single Mode means Paytia answers a single inbound call leg as a third party - used for Conference Pay.
Choose SIP for modern cloud-based telephony platforms, unified communications, and digital contact centers. Choose PSTN call forwarding for traditional PBX systems, legacy infrastructure, or mixed technology environments. Both support Dual and Single mode configurations.
Yes, your telephony infrastructure can support both modes simultaneously. The mode used depends on which payment capture service you activate on each call - Channel Separation/DTMF Suppression use Dual Mode, while Conference Pay uses Single Mode.
Yes, Dual Mode requires calls to be routed through Paytia so it can sit between the two call legs. This is typically done through SIP configuration or PSTN call forwarding rules. Single Mode requires no routing changes - agents simply conference Paytia in when needed.
SIP connectivity typically takes 1-3 days for configuration and testing. PSTN call forwarding can usually be set up within 1 day. Our technical team will guide you through the process and handle most of the configuration.
If you have existing SIP trunks, we can integrate Paytia into your existing configuration for Dual Mode operation, or provide a separate inbound number for Single Mode (Conference Pay). Our team will work with your SIP provider to ensure seamless integration.

Ready to Configure Your Telephony Connection?

Our technical team will help you choose the right connection mode and connectivity option for your infrastructure and business requirements.

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