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Any phone system that can register a SIP trunk to a provider connects the same way (way 1): create the connection with Telephony - Trunks - Connect your PBX (pick Other SIP PBX), and enter on the PBX:

On the PBXValue
Server / registrar / proxy / domainthe server from the dialog
Port, transport5060, UDP
Username and authentication IDthe username
Passwordthe password
Registrationon
CodecsG.711 A-law, then u-law
DTMFRFC 2833 / RFC 4733

Then three things on the PBX, whatever the make:

  1. Calls to the agent: an outbound route, short code, IVR option or overflow that uses the Voice Logica trunk.
  2. Transfers back: calls arriving on the trunk reach extensions by number (DID/DDI routing).
  3. In Voice Logica: add the extensions as transfer destinations (External with the extension number, or a User with that extension) and use Warm transfer.

A system that cannot register a trunk but takes SIP extensions connects through an edge device. A system with no SIP: Analog and ISDN systems.

MiVoice Office 250: System → Device and Feature Codes → SIP Peer → SIP Trunk Groups → right-click Create SIP Trunk Group. In its configuration: Registration → Enable Registration = Yes, Authentication username and password, address and port, Operating State = In-Service. System → IP-Related Information → Call Configurations: DTMF Encoding RFC2833, G.711 A-law, 20 ms. Back to extensions: Call Routing Tables (DID patterns) as the trunk group’s Ring-In Type. Needs SIP trunk licences.

MiVoice Office 400 (Mitel / Aastra 400): WebAdmin Configuration Assistant → Incoming call routing → Add under Network interfaces → SIP provider (registrar = the server, port 5060), then a SIP account: SIP ID and User name = the username, Password, Registration required ticked. Back to extensions: the DDI plan. Needs one SIP Access Channels licence per call.

MiVoice Business: SIP Peer Profile (UDP, the server, port 5060) with SIP Authentication, a SIP trunk group, ARS to send calls to it and DID routing back.