This guide assumes you have configured your SIP trunk in the MBG (or similar) - please review the MBG Setup Guide if you have not configured this step.
Introduction
This guide will enable you to configure your MiVB to pass calls to the Voice AI system and provide configuration to enable the calls to be returned (if applicable with a CC platform). This configuration will be done within the MiVoice platform.
Add Voice Network Element
Navigate to Voice Network → Network Elements and select Add to create a new Network Element
Complete this form using the following table as guidance. We recommend naming elements with Talkative or TK as part of the name for ease of identification.
Option | Suggested Value | Details |
Name | TalkativeVAI | Enter a name to identify this network element. |
Type | Other | |
FQDN or IP Address | {companyUuid}.sip.twilo.com | This will be the domain for your Talkative SIP trunk, it will match the domain setup in the MBG. |
Local | False (default) | |
Version | (not configurable) | |
Zone | 1 (default) | Set to your appropriate Zone |
SIP Peer | Checked | |
SIP Peer Transport | default (default) | This can be changed if you require a non default transport |
SIP Peer Port | 5060 (default) | 5061 if not using UDP |
External SIP Proxy FQDN or IP Address | blank | |
External SIP Proxy Transport | 5060 | 5061 if not using UDP |
SIP Registrar FQDN or IP Address | blank | |
SIP Registrar Transport | default (default) | This can be changed if you require a non default transport |
SIP Registrar Port | 0 (default) | |
SIP Peer Status | Auto-Detect/Normal |
Add Outbound Proxy for MBG (SBC)
Remaining in the Network Elements page select Add to create a new Network Element
Complete this form using the following table as guidance. We recommend naming elements with Talkative or TK as part of the name for ease of identification.
Option | Suggested Value | Details |
Name | TalkativeVAI | Enter a name to identify this network element. |
Type | Outbound Proxy | |
FQDN or IP Address | {mbg-ip-address} | This should be the IP address of your MBG |
Local | False (default) | |
Version | (not configurable) | |
Zone | 1 (default) | Set to your appropriate Zone |
Outbound Proxy Transport Type | default (default) | This can be changed if you require a non default transport |
Outbound Proxy Port | 5060 (default) | 5061 if not using UDP |
Add SIP Peer Profile
Navigate to Trunks → SIP → SIP Peer Profile and select Add to create a new SIP Peer Profile. This form has tabbed section. Each section will be detailed under it’s own subheading.
You will need the names of the Network elements you created in the previous steps to enter into your SIP Peer Profile
Basic
Complete this form using the following table as guidance. We recommend naming elements with Talkative or TK as part of the name for ease of identification.
Option | Suggested Value | Details |
SIP Peer Profile Label | TalkativeVAIPP | Enter a name to identify this peer profile |
Network Element | {networkElementName} | This is the network element name you created in the previous step (not the proxy) |
Registration User Name | blank (default) | |
Address Type | IP Address (default) | |
Interconnect Restriction | 1 (default) | |
Maximum Simultaneous Calls | 5 | Set to a suitable level for your licensing and requirements. |
Minimum Reserved Call Licenses | 0 (default) | Set to a suitable level for your licensing and requirements. |
Outbound Proxy Server | {proxyNetworkElementName} | This is the network element name you created in the previous step for outbound proxy |
SMDR Tag | 0 (default) | |
Trunk Service | 1 (default) | This value may vary. It should match a configuration in your Trunk Attributes. |
Zone | 1 (read only) | |
User Name | siptrunkuser | This will be provided by Talkative or via Engage. Some older accounts will have a different user name |
Password | {password} | The password provided by Talkative for your SIP Trunk |
Confirm Password | {password} | Confirm the value from the previous step. |
Authentication Option for Incoming Calls | No Authentication (default) | |
Subscription User Name | blank (default) | |
Subscription Password | blank (default) | |
Subscription Confirm Password | blank (default) | |
Digital Trunk Licenses | (read only) | |
Maximum Digital/Analog Channels | (read only) |
Call Routing
All options in this section can remain as default unless you find a specific need to change them for your setup.
Calling Line ID
All options in this section can remain as default unless you find a specific need to change them for your setup.
SDP Options
Two options may change in this section if you use TLS and SRTP connections.
Option | Suggested Value | Details |
AVP Only Peer | No | If using TLS and SRTP this should be set to No, otherwise the default value of Yes is acceptable |
NAT Keepalive | No | Some firewalls have issues with sessions persisting due to this setting, so it can be disabled. |
Avoid Signaling Hold to the Peer | Yes | This option should be enabled to ensure on-hold music functions correctly. |
Signalling and Header Manipulation
All options in this section can remain as default unless you find a specific need to change them for your setup.
Timers
One option changes in this section, it is listed below
Option | Suggested Value | Details |
Session Timer | 1800 | Twilio returns a 422 session interval too small error - whilst this does not break the setup, it reduces signalling log noise. |
Key Press Event
All options in this section can remain as default unless you find a specific need to change them for your setup.
Profile Information
All options in this section can remain as default unless you find a specific need to change them for your setup.
Add ARS Route
Navigate to Call Routing → Automatic Route Selection (ARS) → ARS Routes and find an ARS Route which is free to use. Choose from the list and select Change.
Take note of the route number chosen as it will be required in the next step
Option | Suggested Value | Details |
Route Number | read only | |
Routing Medium | SIP Trunk | |
Trunk Group Number | read only | |
SIP Peer Profile | {sipPeerProfile} | Choose the SIP peer profile you created in the previous step |
PBX Number / Cluster Element ID | N/A | |
COR Group number | 1 (default) | Change if appropriate |
Digit Modification Number | 1 (default) | Change if appropriate |
Digits Before Outpulsing | 4 (default) | Change if appropriate |
Route Type | blank (default) | |
Compression | read only |
Add ARS Digits Dialled
The ARS digits dialled generates the route to the Voice AI bot. This number will be dialled and call into the SIP Trunk for VAI. This configuration will allow 1 or more routes which will be mapped to 1 or more bots in Engage. For example, in the below scenario, you may call 1000 - and this route would be mapped to a Bot in engage, then you may dial 1001 and this may be mapped to another bot in engage. Additionally they can be mapped to a Voice Assist configuration.
Navigate to Call Routing → Automatic Route Selection (ARS) → ARS Digits Dialled and select Add to create a new ARS Digits Dialled entry.
Option | Suggested Value | Details |
Enter the number of records to add | 1 (default) | If you require mulitple records, enter a larger number |
Digits Dialed | 100 | Substitute this with any number you require |
Number of Digits to Follow | 1 | Allowing 1 number gives you 10 routes to the bot. |
Termination Type | route | Choose the SIP peer profile you created in the previous step |
Termination Number | {arsRouteNumber} | Enter the ARS route number from the previous step |
Map Route to a Voice AI Bot in Talkative
Once your ARS Digits Dialled entries are configured in MiVB, each route needs to be mapped to a Voice AI Bot in Talkative. Navigate to Settings → Voice Routing and click Create Voice Route (https://eu.engage.app/voice-routes).
Make a test call through your MiVB ARS route, then go to Settings → Twilio Call Log (https://eu.engage.app/twilio-call-log). The To column shows the exact full SIP URI (including any headers) that your MiVB dialled into Talkative. Copy this value directly into the Endpoint field — no reformatting needed.
Option | Suggested Value | Details |
Label | e.g. MiVB Route 1000 | A friendly name for internal identification |
Endpoint | Full SIP URI from the Twilio Call Log | Copy the To value verbatim |
Type | Voice AI Config | |
Voice AI Bot | {botName} | The bot to handle calls on this route |
Click Save. Repeat for each ARS Digits Dialled entry.
You can have multiple routes mapping to the same bot - we recommend keeping the PSTN number available to make testing your bot easier as this can dial your bot at any time.
Add Direct Inward Dialling Service (optional)
If you wish to have an external PSTN number automatically dial a Voice AI route, you can do so with the Direct Inward Dialling Service.
Navigate to Call Routing → Call Handling → Direct Inward Dialling Service and select Add to create a new Direct Inward Dialling Service entry.
Option | Suggested Value | Details |
Enter the number of records to add | 1 (default) | If you require mulitple records, enter a larger number |
DID Number | {pstnNumber} | Substitute this with any number you require |
Primary Node Id (PNI) | {primaryNodeId} | Choose the correct primary node ID for your setup. |
Destination Number | {arsDigitsDialledEntry} | Choose a digit dialled entry from the previous step |
DID Type | Standard DID (default) |
Once this is configured, you will be able to dial the PSTN call and have it route directly towards the Voice AI route.
Routing calls back to MiVB
Within the Voice AI configuration you can call a SIP address by using a fully formed SIP address. By default, no contextual information about the call will be included, so it will just look like a normal inbound call.
Simple Handoff via SIP Refer
The simplest way to return a call to your MiVB is using a SIP Refer. This suits deployments where Talkative's console is not surfaced in your contact centre platform, or for testing basic transfer connectivity. No additional MiVB configuration is required beyond what has already been set up in this guide.
In your Voice Bot, add a Transfer Tool and configure a destination as follows:
Option | Value | Details |
Transfer Type | SIP Refer | Issues a SIP REFER to transfer the call |
Destination | sip:{extension}@__ORIGINATOR__ | The username is the MiVB extension or hunt group to transfer to. Use __ORIGINATOR__ as the domain — Talkative automatically substitutes this with the originating call domain, ensuring resiliency. |
Prompt | e.g. "Use this to transfer to the agent queue" | Tells the AI when to invoke this transfer |
Voice Assist Config | Leave blank | Adding a Voice Assist Config switches the transfer type to SIP Dial — covered separately in the Interaction Parking guides |
Note: Using __ORIGINATOR__ as the domain is recommended — Talkative substitutes it with the originating call domain automatically, which avoids hardcoding a specific domain and improves resiliency.
If Voice Assist is a requirement you will also need to configure SIP Peer Profile Assignment by Incoming DID for your Hunt Group IVR extensions used by Interaction Parking. A range of the extensions will need to be configured in this section to prevent the SIP Dial from returning a 404.
If you have a Contact Centre linked to your MiVB like MCX or MiCC-E you may configure workflows to utilise Interaction Parking to retrieve the meta data from Engage to apply to the interaction.
Have a Contact Centre application setup. Click here to view our next guide will provide information on setting up the required workflows, including Interaction Parking. This guide will additionally cover licensing requirements for IVR Ports
Glossary of terms
Plain-English meanings of the abbreviations and technical terms used in this guide.
Term | Stands for | What it means |
AI | Artificial Intelligence | Software that can understand and respond to people. In this guide, it is the Voice AI bot that answers your calls. |
ARS | Automatic Route Selection | The MiVB feature that decides which route a dialled number takes, by matching its digits. |
AVP | Audio Video Profile | The standard way of carrying call audio without encryption. An AVP Only Peer accepts only unencrypted audio. |
CC | Contact Centre | A team and system that handles customer calls, with agents, queues and reporting. |
COR | Class of Restriction | A MiVB setting that controls which calls a route or extension is allowed to make. |
DID | Direct Inward Dialling | A service that sends a call to an external phone number straight to an extension or route, without an operator. |
FQDN | Fully Qualified Domain Name | A complete host name, for example voiceai.example.com. |
Hunt group | (a phone system term) | A group of extensions that share incoming calls. |
Interaction Parking | (a workflow technique) | A way for a contact centre workflow to retrieve the details of a call (its metadata) from Engage and apply them to the interaction. |
IP | Internet Protocol | The addressing system networks use. An IP address is the number that identifies a device on the network, for example 192.168.10.25. |
IVR | Interactive Voice Response | An automated menu or script that answers calls before they reach a person. Interaction Parking workflows use IVR ports. |
MBG | MiVoice Border Gateway | The Mitel device at the edge of your network that sits between MiVB and the Voice AI platform. It acts as the SBC. |
MiVB | MiVoice Business | The Mitel phone system (also called a PBX) that this guide configures. |
NAT | Network Address Translation | A device or service that maps a public IP address to a private one. |
Network element | (a MiVB term) | A MiVB entry that describes a remote system, with its address and type. |
Outbound proxy | (a SIP term) | A server that outgoing SIP calls are sent through first. Here it is the MBG. |
PBX | Private Branch Exchange | Your phone system. |
PNI | Primary Node Identifier | The identifier of the MiVB node that an external number belongs to. |
PSTN | Public Switched Telephone Network | The ordinary public phone network that you call landlines and mobiles over. |
SBC | Session Border Controller | A device that sits between your phone system and an outside network (here, the Voice AI platform). It protects your system and makes sure the two sides can understand each other. |
SDP | Session Description Protocol | The part of a SIP message that describes the audio of a call, such as the codecs and encryption. |
Session timer | (a SIP setting) | A timer that regularly checks that a call is still active. |
SIP | Session Initiation Protocol | The standard way of starting, managing and ending calls over an IP network. A SIP trunk is a connection that carries many calls using SIP. |
SIP Dial | (a transfer type) | A transfer that places a new call back into your phone system and keeps the Voice AI platform in the call. Voice Assist uses it. |
SIP peer | (a MiVB term) | A remote SIP system that MiVB sends calls to and accepts calls from, such as the MBG. |
SIP Refer | (a transfer type) | A SIP message that asks the other side to start a new call, used to transfer a call. The Voice AI platform uses it to return a call to your phone system. |
SMDR | Station Message Detail Recording | The MiVB call records. The SMDR tag marks which records belong to a trunk. |
SRTP | Secure Real-time Transport Protocol | The encrypted version of the protocol that carries call audio, so nobody listening on the network can hear the call. |
TLS | Transport Layer Security | Encryption that protects a connection, the same technology that protects secure websites. |
UDP | User Datagram Protocol | A faster way of sending data that does not set up a connection or check delivery. |
VAI | Voice AI | The Voice AI service that answers calls. VAI appears in names such as the SIP trunk for VAI. |
Voice Assist | (a Voice AI feature) | Real-time transcription of a call that stays with a human agent. It uses a SIP Dial back into your phone system instead of a SIP Refer. |