This guide explains how to add the Talkative SIP Trunk connection in AudioCodes Live Hub. It applies regardless of what sits on the other end — Mitel, Cisco, Avaya, Webex Calling, Zoom Phone, or any other SIP-capable platform. Once this connection is in place, you can route calls to it from any other SIP connection configured in Live Hub.
For guidance on configuring a specific originating platform (e.g. Mitel), see the relevant platform-specific integration guide. This document only covers the Talkative-facing connection itself.
Overview
The Talkative SIP Trunk connection is the fixed part of any Live Hub integration with Talkative Engage — it doesn't change based on what platform you're connecting. What varies between integrations is the platform-facing connection and the Routing rule that links the two.
Your platform (any SIP-capable system)
↓
AudioCodes Live Hub — SIP connection (platform-facing, platform-specific)
↓ Live Hub Routing rule
AudioCodes Live Hub — SIP connection ("Talkative SIP Trunk", this guide)
↓
Talkative Engage / Voice AIPrerequisites
Talkative provides managed solutions using LiveHub - we can offer per minute pricing on top of the standard pricing to provide a managed LiveHub setup and integration.
- An AudioCodes Live Hub account with access to create SIP connections
- A Voice AI config already created in Talkative Engage, with your SIP trunk details available on the Voice Routing page (Engage → Settings → Voice Routing)
- Confirmation of whether your account requires encryption (TLS/SRTP) — this is set on the Talkative side and must match on this connection
Talkative uses Twilio to provide SIP trunk connectivity. Your trunk hostname takes the form {account-uuid}.sip.{region}.twilio.com, visible on the Voice Routing page in Engage. The default region is us1; if you use a different edge region, include it in the hostname and ensure the corresponding voice route in Engage reflects the same region.
Firewall/IP allowlisting: if your network path to Twilio passes through a firewall, allowlist Twilio's SIP trunking signalling IP ranges — see Twilio's IP Addresses for Elastic SIP Trunking documentation. Each region publishes multiple IP addresses that signalling traffic may use; prefer resolving the FQDN (see Step 4's Addresses table) over hardcoding a single IP, since Twilio does not guarantee a fixed address.
If your firewall is a Fortigate, be aware its SIP ALG (Session Helper) is enabled by default and is known to conflict with SIP trunking traffic, causing signalling to loop and never reach its destination. If you experience calls that never complete despite correct configuration, check whether SIP ALG is enabled and disable it.
Step 1: Create the Connection & General Settings
- In Live Hub, go to Voice channels → SIP connections → Add new SIP connection
- Name it clearly, e.g.
Talkative SIP Trunk
Field | Guidance |
Provider type | SIP Trunk |
Provider | Generic SIP Trunk |
Live Hub region | Choose the region closest to your deployment |
Encryption | Enabled if your account uses TLS/SRTP. If enabled, use port 5061 on the Outgoing tab (Step 4) — a non-secure connection will be rejected once encryption is required on the Talkative side. |
Take Calling Number from | From header, unless advised otherwise |
Registration | Typically disabled — Talkative's SIP connector can be configured to require registration, but this isn't the default. Confirm with your Talkative representative if unsure. |
Credentials | Default configurations surface credentials in the Voice Route page - they should be included here. |
Step 2: Incoming
This controls how Live Hub authenticates calls arriving from Talkative (for example, a bot transferring a call back to your platform). Live Hub requires one of:
- FQDN (Request-URI) alone, or
- Dialed Numbers (DIDs) plus one additional method (Source IP address or Credentials)
FQDN matching (the default/recommended option) is usually sufficient for this connection, since Live Hub auto-generates a unique FQDN per connection that Talkative can be configured to target.
Step 3: Outgoing
This controls how Live Hub sends calls to Talkative.
- Enter the SIP server hostname from your Voice Routing page (
{account-uuid}.sip.{region}.twilio.com)
- Add an explicit entry to the Addresses table — do not leave this empty
- Set the port and protocol to match your account's requirements (TLS on port
5061for encrypted connections; UDP on port5060otherwise)
- Set an appropriate DNS method (A-record is typical)
This is the most common misconfiguration on this connection. The SIP server hostname field only affects the content of the Request-URI and To header in outbound SIP messages — it does not tell Live Hub where to actually send the call. If the Addresses table is left empty, Live Hub has no signalling destination, and calls will appear to be accepted (you may see a 100 Trying response logged) but will never actually be sent anywhere. This produces no error and can look identical to several unrelated problems. Always populate the Addresses table explicitly, even if the hostname field is already set.
Step 4: Numbers
Set E.164 format to match the number format your platform-facing connection uses. This avoids number-matching issues later when routing rules are configured.
An MiVB system can send numbers without needing an e164 format - whereas other platforms like OpenScape require an e164 number. You can configure this accordingly here.
Step 5: Advanced
The Talkative Voice AI Trunks do not support options requests, as such they should be disabled in the Advanced Tab:
Verifying the exact endpoint address
Once this connection is live and routing calls to it, place a test call and check the Twilio Call Log page in Engage (Settings → Twilio Call Log). The To column shows the exact SIP URI Talkative received — use this to confirm the Request-URI leaving your Live Hub Routing rule matches exactly. This is the fastest way to confirm the Number customisation setting mentioned below is correct, and mirrors the same verification step used in Talkative's direct PBX integration guides.
Next steps
Once this connection is saved and tested, it can be used as the Route to destination in a Live Hub Routing rule from any platform-facing connection. See the relevant platform-specific integration guide for details on configuring that side.
When setting up the Routing rule, check the Number customisation section. Live Hub can forward a call without rewriting its Request-URI to match this connection's configured hostname, which Talkative may reject (commonly surfacing as a 403 Forbidden response). If calls reach this connection but are rejected, this is the first place to check.
Troubleshooting
Test calls are accepted (100 Trying) but nothing else happens
Check, in order:
- Is a Routing rule in place sending calls to this connection?
- Is the Addresses table on this connection's Outgoing tab populated? (See Step 4.)
- Check the Completion status field in Live Hub's Call History panel, not just the SIP ladder — a generic "No user responding" status usually means the call was accepted but never matched to a route, rather than a problem with this connection specifically.
Calls reach Talkative but are rejected (e.g. 403 Forbidden)
Check the Request-URI of the outbound INVITE (visible in the SIP ladder) — if it doesn't match this connection's configured SIP server hostname, the originating Routing rule's Number customisation settings likely need adjusting.