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
All options in this section can remain as default unless you find a specific need to change them for your setup.
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 |
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.
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