PBX and IP telephony compatibility: connecting 3CX, Cisco, Yeastar or Asterisk to ODOIP

You already own a phone system — a 3CX, a Cisco, a Yeastar or an Asterisk — and you would like to benefit from ODOIP's IP telephony without replacing everything? Good news: that is exactly the point of an infrastructure built on the SIP protocol. Your existing PBX can connect to our services, keep your numbers and your habits, while gaining in reliability and controlled costs. Here is how to proceed, and what to watch for a smooth interconnection.

A universal foundation: the SIP protocol

Compatibility does not depend on the brand of your equipment, but on respecting a common standard: the SIP protocol (Session Initiation Protocol). As soon as a PBX, an IPBX or a phone speaks SIP correctly, it can talk to the ODOIP platform.

In practice, this covers the vast majority of the installed base: 3CX, Cisco (CUCM/CME), Yeastar, Asterisk/FreePBX, Vodia, Grandstream, Snom, Yealink, Fanvil… You are therefore not forced to change hardware: you plug in your existing system and preserve your investment.

Two ways to connect your PBX

There are two interconnection methods, to be chosen according to the size of your site.

Registration mode: the PBX seen as an extension

The PBX registers with ODOIP as a simple extension, using a SIP login and password. This is the simplest solution, ideal for a small site with a single number. Setup is fast and requires no particular network configuration.

SIP trunk: recommended in production

This time the PBX connects as a trunk, that is by IP address and without registration, with the routing of several numbers or DIDs. This is the recommended mode for a real multi-extension system: it is more robust, clearer and perfectly suited to intensive professional use.

The settings that make the difference

A successful interconnection comes down to a few well-chosen parameters on the PBX side.

Server and transport

• Server / Registrar: sip.odoip.fr, port 5060.
• Transport: TCP is recommended (UDP remains possible). TCP notably makes call teardown more reliable behind a NAT router.

Codecs and DTMF signalling

• Codecs: favour G.711 (PCMA/PCMU), then G.729 as a fallback.
• DTMF: use RFC 2833 for key transmission (voice servers, codes, etc.).

Security: SIP-TLS and SRTP

To encrypt signalling and voice, ODOIP offers SIP-TLS on port 5061, combined with SRTP. This is particularly useful for Cisco or 3CX environments in secure mode.

Session timers

Session timers (RFC 4028) are enabled on our platform. We recommend enabling them on the PBX side as well, especially on Cisco and 3CX systems, to avoid "ghost" calls that would remain open after a conversation ends.

Interoperability points to watch

Three behaviours deserve particular attention during the connection.

• Ringing and early media: by default, ODOIP plays the called party's ringback tone and ignores the PBX's early media. If your PBX or voice server must play its own announcements, the "let the PBX media through" option is available on the extension. And if your PBX answers immediately (200 OK), configure it to ring without answering.
• Call teardown (BYE message): some software (for example MicroSIP over UDP) defers the BYE. TCP is preferable here. In all cases, a safety net automatically clears a call whose audio stream stops.
• Fax: T.38 can be enabled case by case for sending and receiving faxes.

Quick reference by model

Each system has its own recommended settings:

• 3CX: "Generic" SIP trunk, TCP or TLS transport, DTMF RFC 2833, session timers enabled.
• Yeastar (S and P series): VoIP Trunk in "Register" or "Peer" mode, G.711a/G.711u/G.729 codecs, symmetric RTP.
• Asterisk / FreePBX: pjsip trunk with direct_media=no, rtp_symmetric=yes, rewrite_contact=yes and rfc4733 DTMF.
• Cisco (CUCM / CME): SIP trunk, session timers, often TLS and SRTP on port 5061.
• Vodia: SIP trunk in "register" or "IP" mode, G.711, DTMF RFC 2833.

How a connection with ODOIP works

Commissioning follows a simple, proven path:

1. Choose the mode: registration for a small site, trunk for production.
2. Create the extension or trunk on the ODOIP side and assign the number(s).
3. Authorise the public IP address of the PBX (in trunk mode).
4. Configure the PBX according to the settings above, preferably over TCP.
5. Test everything: inbound call, outbound call, ringing, teardown on both sides and DTMF.

In short

Migrating to IP telephony does not mean discarding your current system. Thanks to SIP, your 3CX, Cisco, Yeastar or Asterisk connects to ODOIP in a few steps, with proven settings and support at every stage. You keep your numbers and your working comfort, while enjoying a modern, secure and cost-effective platform.

Need help connecting your PBX to IP telephony? Our team supports you end to end: contact us.