On Avaya IP Office, a carrier trunk is a SIP Line: in Manager, select Line › New › SIP Line, point the Transport tab’s ITSP Proxy Address at your carrier, add SIP Credentials if the account registers, and license enough SIP Trunk Channels for your simultaneous calls. Generic ITSP trunks are fully supported — the platform gates capacity with licensing, not which carrier you may use.
IP Office’s trunk model is the most formal of the major PBXes: six configuration tabs, explicit license entitlements, and one operational trap that drops live calls. This guide — from SIPNEX, an FCC-licensed carrier — walks all of it.
Prerequisites: system settings and licenses
Before the line exists, two things must be true:
- System › LAN1 (or LAN2) › VoIP: check SIP Trunks Enable, set the SIP Domain Name, and confirm the Layer 4 protocol/ports and RTP port range for your network.
- Licensing: SIP trunking needs SIP Trunk Channels licenses (cumulative — sized to your maximum simultaneous SIP calls) or an IP Office Subscription on R11.1+. You can build the line without them; calls need them.
Building the SIP Line
In Manager (File › Open Configuration, then Line › right-click › New › SIP Line) — or the equivalent path in Web Manager under System Settings › Line — the work spreads across tabs:
SIP Line tab: set the ITSP Domain Name — the host part IP Office writes into From/SIP URIs. Enable In Service and Check OOS (periodic SIP OPTIONS so a dead trunk is detected instead of silently eating calls). Session Timers live here too.
Transport tab: the ITSP Proxy Address — where signaling is actually sent. This accepts up to four comma- or space-separated addresses; leave it blank and IP Office DNS-resolves the ITSP Domain Name instead. Set Layer 4 Protocol (UDP/TCP/TLS) and Send Port (5060, or 5061 for TLS), and pick Use Network Topology Info to bind the line to the right LAN’s NAT settings.
The domain-vs-proxy distinction is the classic IP Office trip-up: ITSP Domain Name shapes the SIP URIs; ITSP Proxy Address is the wire destination. Carriers like SIPNEX give you both (often the same value) at provisioning.
SIP Credentials tab: for registration-mode accounts — User name, Authentication Name, Password, Expiry, and the Registration required checkbox. IP-authenticated accounts skip this tab; the carrier whitelists your public IP instead.
Call Details / SIP URI tab: add a URI and set its Incoming Group ID and Outgoing Group ID — the hooks the rest of routing hangs on — plus Max Sessions, which should match your licensed channel count. Local URI, Contact, and Display Name typically stay on “Use Internal Data.”
VoIP tab: Codec Selection — System Default or a Custom ordered list from G.711 ULAW/ALAW, G.722, and G.729a. For US trunks, G.711 ULAW first. DTMF Support = RFC2833/RFC4733 (the payload type, default 101, is system-wide under System › Codecs). Fax runs as G.711 or T.38.
Routing: inbound needs an Incoming Call Route whose Line Group ID matches the URI’s Incoming Group ID; outbound needs an ARS entry or short code (Feature: Dial) targeting the Outgoing Group ID.
NAT: the Network Topology tab
IP Office documents NAT handling per LAN under System › LAN1/LAN2 › Network Topology: STUN server address and port (default 3478) with Run STUN discovery, a Firewall/NAT Type selector, Binding Refresh Time to keep NAT pinholes alive, and manual Public IP Address/Port overrides when you’d rather declare than discover. The SIP Line applies whichever LAN’s topology data its Transport tab selects.
The edit trap: SIP Line changes drop calls
Most SIP Line settings are not mergeable — pushing a config with trunk edits restarts the SIP Line and drops every call on it. Only the ITSP Proxy Address and Calls Route via Registrar merge cleanly. Treat trunk changes as service-affecting maintenance: schedule them, don’t hotfix them mid-shift.
Frequently asked questions
Does Avaya IP Office support generic SIP trunk providers?
Yes — any standards-based SIP carrier can be configured as a SIP Line. Avaya’s licensing controls how many simultaneous SIP trunk calls you may run (SIP Trunk Channels licenses, or Subscription on R11.1+), not which ITSP you connect to.
What’s the difference between ITSP Domain Name and ITSP Proxy Address?
ITSP Domain Name (SIP Line tab) is the host part IP Office uses when building SIP URIs and From headers. ITSP Proxy Address (Transport tab) is where the signaling packets are actually sent — up to four addresses, with DNS resolution of the domain name as the fallback when it’s blank. Mixing them up produces trunks that register but can’t call, or vice versa.
How many SIP trunk licenses do I need?
One SIP Trunk Channel per simultaneous trunk call at peak — the licenses are cumulative, and the SIP URI’s Max Sessions setting should match the licensed count. Size peak concurrency with the same math as any trunk: our sizing guide and calculator cover it.
Why did editing my SIP Line drop active calls?
Because most SIP Line settings are non-mergeable: sending a configuration containing trunk edits restarts the SIP Line, dropping calls in progress. Only the ITSP Proxy Address and Calls Route via Registrar merge without a restart. Make trunk changes in a maintenance window.
SIPNEX is an FCC-licensed carrier providing SIP trunks for any PBX — registration or IP auth, unlimited carrier-side channels, A-level STIR/SHAKEN attestation under our own certificate, and engineers who have stared at Monitor traces before. Connect your IP Office or see rates.
Keep reading.
The carrier built by operators, for operators.
FCC-licensed carrier with its own STIR/SHAKEN SP certificate. Operator-owned. SIP trunks built for operators who dial at volume.