This guide aims to help you understand the integration points for the Voice AI system. This guide assumes you are comfortable using SIP and setting up a SIP Trunk in Mitel. Each section contains a dropdown with a link indicating the minimum known support version and a setup guide. Additionally, there is information to support decisions around licensing relating to SIP channels required for operation.
Obtaining your SIP Trunk Details
To find your SIP trunk details, navigate to a Voice Route config page in engage and you will be able to see the SIP Trunk which has been configured for you:

This will be displayed at the top of each Voice Route for ease of access.
If you have an existing SIP trunk provided manually by Talkative - the credentials listed here will not match - you may continue to use the existing credentials, which will remain valid or, replace your credentials with these credentials instead.
Firewall Considerations
Talkative utilises Twilio for SIP Trunk application capabilities. Twilio’s resilient managed trunks ensure stability when routing Voice AI calls. To ensure the signalling and RTP traffic can be received, please whitelist the signalling IPs which can be found here: IP Addresses for Elastic SIP Trunking Services | Twilio - The default region is none selected is us1. Each region has 4 IP addresses which signalling traffic may originate from or be routed too. We recommend always using the FQDN and never a direct IP address.
Regional SIP Trunking
By default the SIP Trunk region is us1 - if you wish to use a different region, you may adjust your Trunk FQDN to include a regional identifier:
{uuid}.sip.{region}.twilio.comThis will force your signalling to favour a particular region for originating it’s traffic.
If you change your region in your PBX, be sure to update your route mappings in Engage to also include the region parameter.
System Proxies
The most common setups include a Mitel Border Gateway - this will contain your at edge SIP Trunk configuration. Other configuration may include a custom or bespoke SBC. This can be used in place of an MBG. Whilst an SBC is commonly found, it is not a pre-requisite, and a PBX can be integrated with directly.
Mitel - MBG
Min Version: 1.2.3
Other SBC
Any suitable SBC can be used in place of the MBG - if you have specific requirements, please contact us.
PBX/Call Handler
The current supported PBX call handler systems include MiVB, MX-One and MV-5000
MiVB
Min Version: TBC
MX-One
Min Version: TBC
MV-5000
Min Version: TBC
Contact Centre Software
Whilst not required, you can additional use contact centre platforms to allow for screen pops within platforms such as Web Ignite - this additionally allows for IVR workflows to add contextual metadata to your interactions such as the console URL and any passed interaction data.
MiCC-B
Min Version: TBC
MCX
Min Version: TBC
MiCC-E
Min Version: TBC
SIP Channel Usage
The amount of SIP channels you consume will depend on if you are utilising Voice Assist in your workflows. The details for the channels are as follows:
Voice AI
Voice AI without Voice Assist will consume a maximum of 2 SIP channels. The usage is defined as follows:
PSTN/CC ↔ PBX: 1 SIP/media leg on their side.
PBX ↔ Talkative VAI trunk: 1 SIP/media leg on their side.
Voice Assist
Voice Assist, either direct or including Voice AI triage will consume a maximum of 3 SIP channels. The usage is defined as follows:
PSTN/CC ↔ PBX: 1 leg.
PBX ↔ Talkative VAI trunk (initial call to AI): 1 leg.
Talkative VAI ↔ PBX (AI dials back / second leg to stay in): 1 leg.
The additional channel is consumed as the system makes a SIP Dial instead of a SIP Refer to remain in the call, essentially creating a 3 way call bridge.
Voice Assist requires a Contact Centre platform to run a workflow. This workflow will add required metadata to the interaction allowing the Engage console to be loaded within platforms such as WebIgnite. The amount of IVR ports required will vary depending on usage, but we estimate this value to be approximately 5-10.
Voice AI Minute Consumption
Voice AI minutes have three different consumption multipliers. They are:
- Standard Voice AI (Speech to Text) - 1x
- Premium Voice AI (Speech to Speech) 1.5x
- Voice Assist (Real time transcription) 0.4x
Minutes are consumed in bundles and then the rate of consumption depends on the tool utilising it. These minutes are then allocated within Engage according to the required breakdown.
An example of this might be a user who requires 1000 Standard Voice AI minutes per month and 5000 Voice Assist Minutes.
Tool | Required Minutes | Purchased Minutes |
Standard Voice AI | 1000 (1x multiplier) | 1000 |
Voice Assist | 5000 (0.4x multiplier) | 2000 |
Total Purchasing Requirements | 3000 |
Another example might be a customer wishing to use premium speech to speech models. If they required 1000 minutes of Premium Voice AI, and 5000 minutes of Voice Assist, the breakdown would be as follows:
Tool | Required Minutes | Purchased Minutes |
Premium Voice AI | 1000 (1.5x multiplier) | 1500 |
Voice Assist | 5000 (0.4x multiplier) | 2000 |
Total Purchasing Requirements | 3500 |
Estimated Token Usage
Token usage should be considered alongside SIP trunk consumption, as the amount of minutes available heavily determines the likelihood of total call concurrency and as such, the amount of concurrent sip legs which will be established.
The calculation for Standard and Premium voice is very similar and should be determined by working out total expected call volume and then an estimating the time a customer is likely to spend talking to the bot. The length of time they will spend with the bot will be determined by the bots abilities. For example, a simple bot which just collects a name and account number and then passed the call to a human agent may have an expected duration of 60 seconds. Conversely, a complex bot which will collect data and use tools to interact with an API or utilising KB lookups, provide customer information may last for 5-10 minutes.
If you have multiple bots covering different scenarios, ie. a triage bot which acts as an IVR and a more complex bot to handle CRM interactions, you may find benefit to calculating their expected usage separately and then combine the values when you purchase.
Common Troubleshooting Issues
Issue | Resolution |
A message plays saying endpoint not configured | This message indicates there was no matching Voice Router for the incoming dial, either SIP or PSTN - check your Voice Routes to ensure there is a mapping present. Sometimes a PBX can mutate the incoming dial parameters, so you may need to reach out to support for assistance if you are unsure why your route is not matching. |
SIP Registration Failed | Our SIP connector can be configured to require SIP endpoints to register with the SIP domain. If you have specified this in your config, but it is not enabled in our backend, you should reach out for support to enable this. |
Secure connection failed | If you opted for a secure connection, this would be enabled in the Talkative backend and any non secure connection will now fail. Ensure your configuration is using port 5061 and a secure connection. |
Transfer fails when Voice Assist is configured and there is no signalling traffic on the MBG/SBC | Voice assist uses a SIP dial - this means a new connection using a new SIP channel is established. If the MBG is not showing any signalling, it may be the PBX address is incorrectly configured or there is a firewall blocking the signalling traffic. Please contact support so we can review our logs. Please note: if you use a TLS connection, we will not have any pcap information to inform what happened. It may be neccessary to disable TLS to run packet captures to determine the cause of the fault. |
When using Voice Assist, SIP signalling packet captures indicate a 404 on the route. In systems with MBG and MiVB this 404 will originate from the PBX and be proxied upstream. | This is likely caused because the system has not got an entry for the SIP Peer Profile Assignment by Incoming DID. An entry should be made to include the IDs of the incoming dial extension and they should be pointed towards the correct SIP peer for handling. |