2A STRA

SIP-trunk tie-line used for Private networks, unlike to Public networks it could provide some extended telephone services, e.g. Callback (RFC 6910), Call Intrusion (RFC 3911, 5850) and other services over SIP-trunk. SIP tie-line could be used between two MX-ONEs or MX-ONE and another PBX with support of appropriate RFCs. Additionally MX-ONE supports some proprietary services.  Create a route towards another MX-ONE using "MX-ONE tieline" profile. Local and remote IP addresses and domain names should match each other in Route setting for MX-ONE (A) and MX-ONE (B). Route association ID should be the same on both sides. mxone_admin@mx73:~> sip_route -print -route 2 -short Route data for SIP destination route : 2   protocol       = udp   profile        = MXONE-tieline   service        = PRIVATE_SERVICES   uristring0     = sip:?@10.130.10.20;tgrp=2;trunk-context=mx73b.mitel.lab   fromur...

MiVo 400 PBX has build-in conference bridge. You could dial-in to a bridge number and enter PIN code to join a conference. SIP-subscribers should occupy DSP resources for VoIP calls. MiVo 415 could handle six parties conferences and have on board DSP for 3 VoIP/RTP streams. Additional DSP modules available for expansion resources.  Before start to enter a conference please check that voice prompts are uploaded to the PBX from vendor server. It could be done from MiVo 400 Web-admin interface. In order to create a conference room, you've need to activate Self Service Portal (SSP) for particular user in their permissions set and mark possibility to manage conferences. Then login to SSP with the user name and password (or PIN) and create new conference room. Default dial-in conference bridge number is #896. You will hear voice prompt to enter access code. Enter PIN code using DTMF over RTP mode on SIP-phones to join the conference. Please find pdf slides with detailed instructions -...

Voice over WLAN (or VoWiFi) handset model: 5634 (WH2) could connect over wireless network to IP infrastructure and register on SIP proxy for making/receive calls. The best way to configure the phone is to connect it to PC over USB cradle (programmer) and configure using WinPDM java application. After installing WinPDM and connecting the phone, you will need to: - create new "Site" in the application, - import Parameter definitions file from SW bundle, - create new Template and configure Network and VoIP settings, - create new Number using the template above, - associate the number with device. The handset will syncing configuration with WinPDM, connect to WiFi network and register SIP extension. Please find pdf slides with detailed instructions -> pdf . The phone is successfully registered on Asterisk PBX.