1) Run DM on your PC. 2) Select System -> Link Setup.



Similar documents
SIP Trunking with Elastix. Configuration Guide for Matrix SETU VTEP

OfficeServ 7000 Series Software Version 4.30i General Availability

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Allworx 6x IP PBX

OfficeServ 7000 [V4.40 S/W] Feature Guide

SIP Trunking Quick Reference Document

SEUK. How to setup SIP Trunking?

Release Notes for NeoGate TE X

Mobile i-phonenet User Guide Android

WE VoIP User Guide For OfficeServ 7000 Series

How to Configure the Toshiba Strata CIX for use with Integra Telecom SIP Solutions

How To Use Touchtone.Com'S Digital Phone Service Web Portal User Guide (For Ip Phones) On A Pc Or Ip Phone (For A Cell Phone) On Pc Or Mac) On Your Ip Phone On A Mac Or Ip Cell Phone On

Atcom MP01 and Elastix Server

VoIPon Tel: +44 (0) Fax: +44 (0)

Configuring Positron s V114 as a VoIP gateway for a 3cx system

Configuring Avaya IP Office 500 for Spitfire SIP Trunks

Quick Installation Guide

SIP Trunk Configuration Guide. using

SIP Trunking Application Notes V1.3

InSciTek Microsystems 635 Cross Keys Park Fairport, NY Setting up Your Phones

KX-NS. Cellular Phone as Extension

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online

Contents 1. Setting up your Phone Phone Setup Phone Usage 2. User Portal 3. Softphone for your computer 4. Faxing

Quick Start Guide CREATING A NEW SITE

Benefits Of Upgrading To The Latest CIX Processor

Opera Wireless Mobilty

VoIP Intercom and Elastix Server

OfficeServ 7000 [V4.30 S/W] Feature Guide

FLX VoIP Registering with Avaya IP Office 500

How To Configure An Ip Trunk On Anconnect Sip Trunk On A Pc Or Mac Or Ip Trunk (For A Pbx) On A Ip Trunk With A Pbt (For An Ip) On An Ip Or Ip (For Pb

Vocia MS-1 Voice-over-IP Interface. Avaya System Verification. Configuring Avaya Aura Session Manager system with Biamp s Vocia MS-1

EPYGI QX IP PBXs & GATEWAYS

How to use IP-0x to connect to Skype

WorkTime UC Mobile Admin Guide

Internet Telephony PBX System. IPX-300 Series. Quick Installation Guide

3rd Party VoIP Phone Setup Guide (Panasonic b)

Application Notes for Configuring QuesCom 400 IP/GSM Gateway with Avaya IP Office using H.323 trunks Issue 1.0

Link Gate SIP. (Firmware version 1.20)

DINSTAR DAG1000-4S4O with Elastix Setup Guide

UCM61xx Configuration

Linksys SPA942 User Guide. Linksys 942 User Guide

How to Connect MyPBX to NeoGate TG via SIP Trunking

FMC VoIP. User Manual. OfficeServ 7000

ACD Manual. Version 3.1 for SV8100 R8

Cisco Unified Communications Manager SIP Trunk Configuration Guide for the VIP-821, VIP-822 and VIP-824

Cisco Unified Communications Manager SIP Trunk Configuration Guide

Quick Installation Guide

OfficeServ 7100 IP-PBX. SIP Trunking using the Optimum Business Sip Trunk Adaptor and the Samsung

Mobile i-phonenet User Guide iphone

Opera System Configuration for Apps

Hosted PBX. TelePacific Communicator User Guide. Desktop Edition

OfficeServ 7000 Series IP PBX. OfficeServ V4.60 Product Update Guide

Configuration guide for Switchvox and Cbeyond.

SIP Trunking with Allworx. Configuration Guide for Matrix SETU VoIP Gateways

SPECIAL APPLICATIONS SECTION

PPM User Guide Telephone: IP Solutions, Aldermans House, Liverpool Street, London, EC2M 3UJ

Inter-Tel IP Phone Quick Reference Guide

Allo PRI Gateway and Elastix Server

Motorola TEAM WSM - Cisco Unified Communications Manager Integration

Configuration Notes 0217

VoIP Intercom with Allworx 6x Server Setup Guide

OPENSCAPE UNIFIED COMMUNICATION (UC) WEB CLIENT USER GUIDE A B C D E F G H I

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5

Integrating a Hitachi IP5000 Wireless IP Phone

Elastix Server VoIP Intercom Setup Guide

Mediatrix Gateway 440x Series Quick Configuration Guide

Contents. Cbeyond Communicator for Mobile (ios) extends TotalCloud Phone System (TCPS) calling capabilities to an iphone.

CISCO IP PHONE 6945 Series User Guide

Avaya IP Office Platform Web Self Administration

Integrating VoIP Phones and IP PBX s with VidyoGateway

3COM VCX PBX Server VoIP Intercom Setup Guide

User Guide. Updated

Application Notes for Configuring SIP Trunking between Metaswitch MetaSphere CFS and Avaya IP Office Issue 1.0

Your Small Business Phone System

Personalizing Your Individual Phone Line Setup For assistance, please call ext. 102.

Integrating Citrix EasyCall Gateway with SwyxWare

Quick Start Guide v1.0

IP Office Technical Tip

3CX PBX SIP server 3CX softphone X-Lite softphone HW IP phone - Alphatech IP video door entry phone station

Standard Telephone User Guide

SIP Trunking using the EdgeMarc Network Services Gateway and the Mitel 3300 ICP IP-PBX

FortiVoice. Version 7.00 User Guide

3rd Party VoIP Phone Setup Guide (Panasonic UT )

SIP Trunking using the Optimum Business SIP Trunk adaptor and the AltiGen Max1000 IP PBX version 6.7

MDS Amiba Cloud PBX. Getting Started

Tech Bulletin IPitomy AccessLine SIP Provider Configuration

My Account Quick Start

Android Softphone App for the Opera IP System. Installation and user guide

Softphone User Manual

3CX PBX v12.5. SIP Trunking using the Optimum Business Sip Trunk Adaptor and the 3CX PBX v12.5

How to Configure the Avaya IP Office 6.1 for use with Integra Telecom SIP Solutions

Evolution PBX User Guide for SIP Generic Devices

Optimum Business SIP Trunk Set-up Guide

Table of Contents. Mitel 3000 Getting Started Guide

How to Configure the Allworx 6x, 24x and 48x for use with Integra Telecom SIP Solutions

SIP Proxy Server. Administrator Installation and Configuration Guide. V2.31b. 09SIPXM.SY2.31b.EN3

Digium Switchvox AA65 PBX Configuration

Multiline Telephone User Guide

Nokia Call Connect v1.1 for Cisco User s Guide. Part Number: N Rev 003 Issue 1

Avaya IP Office 8.1 Configuration Guide

Transcription:

ED 2.0 (2013.03.18)

1) Run DM on your PC. 2) Select System -> Link Setup. How to connect OfficeServ system through OfficeServ DM(Device Manger). 1) Standalone mode - Execute osdm.exe file. 2) Web browser mode MP20 MP40 MP20s OS7100 OS7070 OS7030 V4.60 Private network http(s)://system IP/dm/ http(s)://system IP Public network http(s)://system IP http(s)://system IP/dmp/ http(s)://system IP/dm_public/ - osdm files need to be uploaded to the DM directory in a SD card. But in case of OS7070, all files are already included. MP40/MP20 : osdm.jar, osdmhelp.jar, osdm.jnlp, osdm_public.jnlp MP20s/MP10a/MP11/7030 : osdm.jar, osdmhelp.jar - Java6 is required for a normal operation of DM(V4.60 )

3) You can make the connection list as below. - Enter the Site name and Destination IP address. After then click Apply button. 4) Choose the site in the list for the DM connection.

5) After entering a password, OfficeServ DM screen will be shown as below.

For the Public zone service, IP type of OfficeServ system and MGI/OAS cards should be set to Private with Public Programming 1) In DM 2.1.0, enter the system IP address. 2) In DM 2.1.2, set the system s IP Type to Private with Public. 3) In DM 2.2.2, enter the MGI card s IP address and set its IP Type to Private with Public.

1) Trunk type - To use the various PBX features of WE VoIP client, SIP & ISDN trunk are recommended. Because main features of WE VoIP are operating on the MOBEX s basic functional structure. With other trunk types, it will work as normal SIP client having a basic call function. 2) Group Conf. Cabinet assignment - For Call move and Smart routing service, one or more Group Conf cabinet should be assigned in DM 6.3.2 Virtual Card Change option. 3) WE VoIP Client Expire Time - In DM 5.2.12 SIP Extension Configuration, set SIP Expire Time to below 600 seconds. If it is set to over 600 seconds, the WE VoIP client will not try to register to the system any more. 4) G.729 codec is not supported yet - To use WE VoIP client with the OfficeServ system, do not use G.729 codec for MGI/OAS/Handset. 5) Remote dial service - This service is not commercially released. If you want the trial test, please contact your partners in HQ first.

To use the WE VoIP client service, enter a license key to DM 2.1.4 (MMC860) as below. - No SIP phone license is required.

1) Enter the User ID and Password in DM 2.7.2 (MMC842) 2) Login profile in DM 5.2.24 - Enter the mobile number This number will be used for making a unique name for both a mobile and login profile as below. <sec_mobile_01095303304.xml> <sec_login_01095303304.xml> - Check the public zone service option To use WE VoIP in public area(not in your office WIFI zone), this option should set to Enable. - Check the Scan 5G Only option For reducing a handover delay, a proper channel scanning method should be chosen considering your network environment. -

3) Mobile profile in DM 5.2.23 - AP SSID Basically, this should be identical to your local access point. If it differs from the SSID of local AP, WE VoIP client will not recognize that it is in a office where the OfficeServ system is located in and then will try to register with a public IP address of the system. - Public zone service Using an unpopular SSID is strongly recommended. When it is unfortunately same to other one in a public area, the public zone service will not work at that area.

Multi-ring service In DM 4.2.1 Station Pair option, make WE VoIP client paired with a desk phone. - When user receives an incoming call on a paired desk phone(or WE VoIP), a missed call will not happen on the WE VoIP(or a desk phone). - Setting or clearing CFWD or DND options will be applied to both paired devices. Single CID number service In DM 4.2.1, enable the Single CID Number option. - When WE VoIP is paired with a desk phone having a primary number by Single CID number option, the primary number & name will be delivered instead of the CID number of WE VoIP. - This feature is supported on an internal call(including station group) only. - For an external call, you can use Send CLI Number option in DM 2.3.4 instead of Single CID number. - This CID number will not be shown on the SMDR log.

Seamless call move between a desk phone and WE VoIP client. Desk phone WE VoIP WE VoIP Desk phone When a user selects MOVE key on a desk phone, WE VoIP client will be ringing. After then a user answers the call on WE VoIP client, the call in progress on the desk phone will be automatically disconnected. When a user selects MOVE key on a desk phone, WE VoIP client will be disconnected from the call in progress and it will be moved to the desk phone with no ringing. Restrictions Although other type of handsets can be paired with the WE VoIP client, there are two restrictions. 1) In case of SLI/WIP phones, Call move is only supported by pressing a pre-assigned number for MOVE feature code. 2) SIP phone/client except for WE VoIP client doesn t support the Call Move feature. Programming 1) In DM 4.2.1 Station Pair option, make WE VoIP client paired with a desk phone(ipp/dgp). 2) In DM 4.9.2 Station Key option, make MOVE feature key on a desk phone. 3) In DM 5.14.2, set the Move Wait Time (Default : 20secs)

Handover from Wi-Fi to mobile network When a user presses the button To Mobile on WE VoIP client, the VoIP call in progress will be redirected from its Wi-Fi to mobile network. - Mobile phone will be ringing for establishing a mobile call. - The VoIP call will be hold during the manual handover. Programming 1) In DM 2.7.5 Mobile Extension, assign a trunk number and input a mobile number in the outgoing digit option. 2) In DM 2.7.2 SIP Phone Information, enter the Mobex station number to the destination of Call forward unreachable option. 3) In DM 2.8.0 Numbering Plan, assign any non-duplicated number to MOVE key.

Case 1 : Plug-out When a WE VoIP client is disconnected normally, an incoming call to the client will be transferred to a pre-assigned call number in the Call forward unreachable option. Case 2 : No response In case that a WE VoIP client does not respond(180/183 ringing) to an INVITE message from the OfficeServ system within a timer set in the Call forward unreachable Time option, the call will \ be transferred to a pre-assigned call number in the Call forward unreachable option. Programming 1) In DM 2.7.5 Mobile Extension, assign a trunk number and input a mobile number in the outgoing digit option. 2) In DM 2.7.2 SIP Phone Information, enter the Mobex station number to the destination of Call forward unreachable option. 3) In DM 5.14.2, set the Call forward unreachable Time. (Default : 5secs)

Cost reduction by the smart routing service When a user makes an external call via Mobile button on WE VoIP client, the OfficeServ system checks if the dialed number has a corresponding extension number and the station s status is idle. If exists, the call will be redirected in the system to pass through its own trunk line. Dialing to A-mobile number Smart Routing to A-ext. number 3G/4G AP A Programming Office 1) In DM 2.7.2 SIP Phone Information, enter the Mobex station number to the destination of Call forward unreachable option. 2) In DM 5.2.24 Login Profile, enter a mobile number that WE VoIP client is having.