SIP / BYO SIP
Connect Taliiq Managed SIP or bring your own carrier/PBX trunk, with termination gateway IP whitelisting and digest authentication.
Taliiq supports two ways to route phone calls to an agent over SIP:
| Path | What it is | Where it lives |
|---|---|---|
| Taliiq Managed SIP | A SIP extension hosted by Taliiq (your-id@sip.taliiq.com) — no carrier needed, good for testing or SIP-to-SIP calling | Build → Channels → SIP |
| BYO SIP | Your own carrier or PBX trunk (Twilio, Telnyx, an on-prem Asterisk/FreePBX, etc.) forwards calls into Taliiq | Manage → Integrations → SIP (trunk credentials) + Build → Channels → SIP (the channel) |
This page covers BYO SIP — bringing your own trunk. Managed SIP needs no trunk configuration; just create a SIP channel and dial the generated extension.
Need calls to stay off the open internet, with no public SIP or RTP ports on your side? Use Private SIP Link, then attach it when configuring BYO SIP or an outbound trunk.
Setting up a Custom SIP Trunk
A Custom SIP Trunk stores the connection details for your carrier or PBX. Creating one does not provision anything on its own — it becomes live once you attach it to a BYO SIP channel (see below).
Go to Manage → Integrations → SIP and click Add SIP Trunk.
Give it a Trunk Name (at least 3 characters, used for display only).
Add one or more Termination Gateways — the IP addresses or hostnames your carrier sends calls from.
If any gateway is a hostname rather than a bare IP, turn on Enable Digest Authentication and set a SIP username and password (see IP whitelist vs. digest authentication below for why).
Save. The trunk now shows up under Saved Trunks and is ready to attach to a channel.
Termination gateways
Each trunk can have any number of gateways. Every gateway gets its own:
- IP Address / Host — required. Accepts an IPv4 address, IPv6 address, or hostname.
- Port — defaults to
5060, must be 1–65535. - Transport —
UDP,TCP, orTLS, per gateway.
Multiple gateways are additive, not a fallback list — a call is accepted if it arrives from any gateway in the list. This is normal for carriers that originate SIP traffic from several media-gateway IPs (redundancy, multiple regions).
A gateway row you add but leave blank is dropped automatically when you save — it doesn't need to be removed manually, and it won't block the save.
IP whitelist vs. digest authentication
Taliiq restricts who can call in on a BYO trunk in one of two ways, and it's worth understanding which one is actually active:
- IP whitelist — every gateway that's a real IP address gets added to the trunk's allowed-address list at the network layer. Only SIP traffic arriving from one of those IPs is accepted.
- Digest authentication — if you set a SIP username and password, the carrier must authenticate each call with those credentials (set the same ones on your carrier's trunk config).
A hostname gateway (e.g. sip.example.com) can't be turned into an IP whitelist entry — SIP trunk whitelisting works on IP/CIDR, not DNS names, since the carrier's actual source IP can change or resolve differently than expected. Hostname gateways are accepted for reference but are skipped from the whitelist.
If every gateway on a trunk is a hostname and digest authentication is off, the trunk has no effective access restriction — the IP whitelist is empty (nothing to whitelist) and there's no credential check either. Taliiq rejects saving a trunk in that state; enable digest authentication so there's at least one real restriction in place.
Attaching the trunk to a channel
A Custom SIP Trunk only starts accepting calls once it's linked to a BYO SIP channel with a DID (phone number).
Go to Build → Channels → SIP and add a channel, choosing BYO SIP Trunk as the source.
Assign a voice-capable agent or a conversation squad — inbound calls on this number route to it. See Conversation squads.
Enter the Inbound Phone Number (DID) in E.164 format (e.g. +201234567890) — the number your carrier sends in the SIP To header.
Pick your trunk from the Custom SIP Trunk dropdown. Haven't created one yet? Select + Manage SIP Trunks at the bottom of that same dropdown to jump to the trunk setup page above.
Troubleshooting
| Symptom | Likely cause |
|---|---|
| "At least one gateway IP or host is required" | Every gateway row was blank, or the only gateway added was never filled in |
| "All gateways are hostnames, which can't be IP-whitelisted…" | Trunk has only hostname gateways with digest authentication off — turn on digest auth and set a username/password |
| "SIP password is required when a SIP username is set" (or vice versa) | Digest auth needs both a username and a password together — a lone one does nothing on the carrier side |
| Trunk saves fine but calls never reach the agent | Confirm the trunk is actually attached to a BYO SIP channel — saving a trunk under Integrations alone doesn't provision anything |
| Carrier reports calls rejected / no response | Double-check the gateway IP matches what your carrier actually sends from (not a hostname that resolves differently than expected), and that the port/transport match your carrier's config |
See Channels overview for web widget and outbound options.