SIP Trunk Compatibility: Works with 3CX, Avaya, Asterisk & More

One of the best things about SIP trunking is that it works with the PBX system you already have. No rip-and-replace required.

Compatible Platforms

WOCOM's Flexi-SIP Trunk has been tested and certified with: 3CX, Avaya IP Office, Asterisk/FreePBX, Cisco UCM, Grandstream UCM, Panasonic KX series, Mitel MiVoice, and Yealink T-series.

Setup Is Straightforward

For most systems, configuration takes under an hour. WOCOM provides step-by-step guides for each platform, and our technical team can assist with remote configuration at no extra cost.

What You Need

A static IP address (recommended), a SIP-compatible PBX or gateway, and a stable internet connection with adequate bandwidth for your call volume.

What "compatible" actually means

Compatibility is not a single yes or no. A SIP trunk and a PBX have to agree on four things, and a mismatch in any one produces a specific, recognisable failure:

• Authentication — either the PBX registers with a username and password, or the trunk is authorised by static IP address. IP authentication is more robust where you have a fixed address; registration is necessary where you do not.
• Codecs — the compression used for the audio. Both ends must support at least one in common, or the call connects with no sound at all.
• DTMF signalling — how keypad presses travel. A mismatch here is the classic "the call works but the IVR ignores me" fault.
• NAT traversal — how the PBX advertises its address from behind a router. Get this wrong and calls connect with audio in one direction only.

Notes by platform

3CX — the most straightforward, with a provider template that populates most settings. Covered in detail on our 3CX integration page.

Asterisk / FreePBX — fully supported and highly configurable. Use PJSIP rather than the legacy chan_sip driver on current versions.

Avaya IP Office — supported; the SIP trunk is added as a line and the licensing needs to allow the channel count you are buying, which is worth confirming before the cutover date.

Cisco UCM — supported via a SIP trunk security profile; expect to work through the profile and route-pattern configuration rather than a single form.

Grandstream UCM — supported, with a simple trunk configuration page.

Panasonic KX series — supported on the IP-capable models. Older TDM-only units need a gateway, covered below.

Mitel MiVoice — supported; check the SIP trunk licence count matches your channels.

The three settings behind most failed installs

SIP ALG on the router. Almost every consumer and small-business router ships with SIP ALG enabled, and it corrupts SIP packets while believing it is helping. Turn it off. This one setting accounts for a large share of "the trunk registers but calls fail" reports.

Codec mismatch. If the PBX offers only a codec the trunk does not support, the call sets up and then sits in silence. Enable G.711 as a baseline and add G.729 if you are conserving bandwidth.

DTMF mode. Set RFC 2833 unless you have a specific reason not to. In-band DTMF breaks as soon as a compressing codec is involved.

If your PBX genuinely is too old

Some equipment predates SIP entirely — traditional analogue or digital systems with no IP capability at all. These are not stranded. A SIP-to-analogue gateway sits between the trunk and the old system, presenting familiar analogue ports to the PBX while speaking SIP to the network. You keep the handsets, the wiring and the staff training, and still retire the expensive lines.

It is a bridge rather than a destination, but it turns a forced replacement into a planned one.

Test before you cut over

Never port your main number as the first test. Provision the trunk alongside your existing lines, then work through this sequence: an outbound call to a mobile, an inbound call on a temporary number, a call held for more than five minutes to expose one-way audio and session-timer faults, keypad entry into your own IVR, a transfer between two internal extensions, and finally a check that caller ID presents your business number correctly.

Only when all six pass should you schedule the number port. Our technical team will run this with you at no extra cost — see the Flexi-SIP Trunk page, or the complete SIP trunking guide for the wider picture.