2 Telephone Interconnect Overview Telephone Interconnect is the service included into SmartPTT Radioserver to establish interconnection between Radio and Telephone Subscribers. For communication with telephone subscribers Telephone Interconnect Service connects to VoIP Gateway having SIP trunk service support by means of SIP protocol. VoIP Gateway must be either embedded into PABX system or connected by means of digital or analog channels. For communication with radio subscribers Telephone Interconnect Service uses control stations or repeaters connected to Radioserver. To enable Telephone Interconnect functionality it is required to get telephone interconnect license per each Radioserver.
3 Telephone Interconnect Architecture SmartPTT Dispatcher Console IP Network IP Site Connect Radioserver Telephone Telephone Interconnection: - Initiated by telephone subscriber (Dial Radio ID) - Initiated from the dispatcher console - Initiated be radio subscriber by sending text message with the phone number to the radioserver PBX + VoIP Gateway
4 VoIP Gateway Requirements VoIP gateway provides the conversion interface between the public switched telephone network and an IP network for voice and fax calls. The best choice is native VoIP gateways integrated into PBX as they provide maximum flexibility and performance. In case you have PBX without integrated VoIP gateway, it is possible to use separate gateway, connected to PBX by standard interfaces: digital (e.g. ISDN PRI, ISDN BRI) or analog (FXS, FXO, E&M). Digital interfaces are preferred to analog ones, as they provide more full-functional signaling features (Caller ID, Dialed Number Identification Service (DNIS), Hang-up signals). FXS and FXO interfaces support DNIS and reliable hang-up signaling just in one way, so it s recommended to combine them for transferring calls in opposite directions (FXS/FXO gateway) or use the Direct Inward Dialing (DID).
5 Types of calls Telephone Interconnect initiated by telephone subscriber. Telephone subscriber dials Prefix + Radio ID to do the call. PABX and VoIP Gateway must be programmed correspondingly. Telephone Interconnect initiated by radio subscriber. Radio subscriber sends text message with the phone number to Radioserver to process. Phone number can have prefix to identify that text message is the phone number, but not an ordinary text. Telephone Interconnect initiated by Dispatcher Console. Radio and Telephone Subscribers can be patched from Radio Subscriber Call Window in SmartPTT Dispatcher Console.
6 Telephone call from SmartPTT Dispatcher Console SmartPTT Dispatcher has a built-in functionality to make telephone connections to telephone subscribers and vice versa. The following connection options are possible: SmartPTT dispatcher calls office PBX subscriber Office PBX subscriber calls SmartPTT dispatcher In both cases connections and addressing are implemented via SIP protocol. Voice is transmitted in full-duplex mode via RTP protocol. A valid unique SIP address for each telephone subscriber must be assigned in SmartPTT Dispatcher in accordance with the telephone subscriber's phone number.
7 Phone calls from SmartPTT Dispatcher SIP addressing system SIP determines the following addressing system for every subscriber: address is the IP address of the device (VoIP gateway for telephone subscribers, IP address of SmartPTT dispatcher PC for dispatcher). port is a UDP port that is used by device to receive and send commands, port 5060 is used by default. name is a subscriber ID which can be interpreted in various ways: for telephone subscribers names consist of an optional numerical prefix and the subscriber's phone number. for SmartPTT dispatcher the name is not important. While the VoIP gateway is being configured, it must be set to 1
8 Phone calls from SmartPTT Dispatcher Configuring Telephone Settings To configure connection parameters with telephone subscribers, expand the Settings list in the main menu of the SmartPTT Dispatcher window and click Telephone Settings. The Configuration window will be displayed for configuring connection parameters and a list of telephone subscribers (presented on next slide).
9 Phone calls from SmartPTT Dispatcher Telephone Settings Configuration dialog
10 Phone calls from SmartPTT Dispatcher Configuring Telephone Settings Dispatcher Settings section This section specifies the dispatcher's local computer settings. Allow Calls to Telephone Subscribers enables the feature for this dispatcher. Incoming Sound Port the range of ports to use for voice data transferring. Each connection with the telephone subscriber requires two ports (e.g. the first connection will use UDP port and TCP port 18601, the second will use UDP port and TCP port and so on). Thus, the specified ports must be available both for incoming and outgoing traffic. Interface IP address at the local computer, used to receive and transmit commands and voice data. If dispatcher's computer has only one IP address, the system may select it automatically; if it has several IP addresses, the IP address (interface) should be specified expressly. The specified IP must be available from the VoIP gateway end. SIP Port is a UDP port to receive and send commands to connect to telephone subscribers. The default value is Accordingly, the specified port must be available for incoming and outgoing traffic. Please note: If several network interfaces are presented on the PC you may have to configure the routing table using the command "route add...".
11 Phone calls from SmartPTT Dispatcher Configuring Telephone Settings VoIP Gateway Default Settings section If most of the telephone subscribers the SmartPTT dispatcher is connected via the same VoIP gateway, it is more convenient to specify default values in this section. These settings allow to register telephone subscribers by specifying only the internal phone number. Default VoIP Gateway Address the gateway's IP address. SIP Port a VoIP gateway port to receive and send SIP commands. By default VoIP gateways use the UDP port Dial Prefix for Outgoing/Incoming Calls default prefix number for outgoing/incoming calls in accordance with VoIP Gateway settings. Notes: 1. If the default VoIP gateway is used, the telephone subscriber's SIP address for outgoing calls is automatically composed with the following rule: SIP address = Dial Prefix for Outgoing Calls + Phone Number + Default VoIP Gateway Address + : + SIP Port For example, with Default VoIP Gateway Address = , SIP Port = 5060, Dial Prefix for Outgoing Calls = 12, system will assume SIP address for telephone subscriber with the phone number When the SmartPTT dispatcher is called by the telephone subscriber, the subscriber is identified by the incoming SIP address, which can be different from the outgoing one depending on the VoIP gateway settings. That is why a separate prefix for incoming addresses is set in the system.
12 Phone calls from SmartPTT Dispatcher Configuring Telephone Settings Telephone Subscribers list The SmartPTT dispatcher stores a list of telephone subscribers. You can manage the list using Add, Edit or Delete buttons. Telephone subscriber properties are managed in separate window. In this window you can enter the subscriber name and choose a picture to display in call window. To specify the SIP address you can select among two options: Specify just the phone number (default VoIP gateway settings will be used). Enter both outgoing and incoming SIP address manually in full form.
13 Phone calls from SmartPTT Dispatcher Telephone Subscribers access pane If the settings allow connections to telephone subscribers, a telephone subscribers access pane is available in the SmartPTT Dispatcher main window. Double click in the Telephone subscribers pane to open the Telephone Subscriber call window. To call the subscriber click Connect. To end call click Disconnect. The Telephone Subscribers pane has a speed-dial button (at top-right corner) to call the subscriber by phone number. In this case default VoIP gateway settings are used to generate the full SIP address.
14 Phone calls from SmartPTT Dispatcher Telephone Subscribers access pane To display the telephone subscriber settings window, right-click the desired telephone subscriber To set a sound for an incoming call alert, click the button under corresponding section. To display the telephone subscriber call window, click the button Call Window. To assign the selected telephone subscriber to specific categories, click the button Categories. Select desired categories in the displayed window and click ОК.
15 Telephone Interconnect Radioserver Settings SmartPTT Radioserver has integrated Telephone Interconnect service which allows establishing of communication between radio and telephone subscribers. Telephone Interconnect service uses the following protocols: SIP for signaling RTP for voice data transmission Voice data transmission is done in half-duplex mode. There are 3 ways to initiate the interconnection: Telephone subscriber initiates a call. Radio subscriber initiates a call. Call is initiated from dispatcher console. Select Telephone Interconnect item in Radioserver Configuration Window to set up Telephone Interconnect Service. Note! Telephone Interconnect Service requires corresponding license to be installed at Radioserver. Licenses are installed under Licenses section of Radioserver Configuration tool.
16 Telephone Interconnect Radioserver Settings
17 Telephone Interconnect Radioserver Settings Active Enable the Telephone interconnect feature for radioserver. Server Settings group Interface Local IP address which will be used for communication with VoIP Gateway. Note! Radioserver IP used for communication with VoIP Gateway must be real one. NAT is not supported. SIP Port Local UDP port to receive and send SIP commands (default value: 5060). Note! In case when Radioserver and Dispatcher Console are installed on the same PC and both of them are set up for telephone interconnection, SIP ports must be different for Radioserver and Dispatcher Console. RTP Ports Range of local ports used for voice data transmission to/from VoIP-Gateway (default values: ). Private/Group Call Prefix If the name part in SIP address, received by radioserver from VoIP gateway, starts with this prefix, radioserver treats the rest of name as Radio ID/Group ID and performs private/group radio call to it. In case they are not set, all calls are treated as private.
18 Telephone Interconnect Radioserver Settings Allow Calls to Default VoIP Gateway enable the possibility to use default VoIP gateway settings to build the full SIP addresses in case just phone number was specified, when radio subscriber initiates call. Default Gateway Settings group Address network IP address of default VoIP Gateway. SIP Port SIP port which of VoIP gateway (default value: 5060). Source/Destination Prefix default prefix number, which will be added in front of radio ID/calling phone number for outgoing calls. Dialing Rules group Prefix This prefix is used to determine telephone call request. If text message sent to Radioserver starts with this prefix, Telephone Interconnect Service treats the rest of message as phone number and compose the following SIP addresses for outgoing call: From: [Default Gateway Settings-Source Prefix][Radio Settings- Interface]:[Server Settings-SIP Port] To: [Default Gateway Settings-Destination Prefix][Phone Gateway Settings-Address]: [Default Gateway Settings-SIP Port]
19 Telephone Interconnect Radioserver Settings
20 Telephone Interconnect Radioserver Settings Additional Telephone Interconnect Settings: To limit the radios authorized to use telephone interconnect service, set Allowed Subscriber List checkbox and add permitted radios to the list. Call Request Timeout Period of time, during which request from a radio (text message with a phone number sent to radioserver) will stay in queue waiting for pending calls. Telephone Ring Timeout Time to wait for telephone subscriber to answer the call. Radio Subscriber Call Timeout Time to wait for connection from telephone subscriber to radio subscriber when all available radio channels are busy. Also you can enable or disable sound notifications about events for phone and radio subscribers.
21 Telephone Interconnect General VoIP Gateway settings SmartPTT Telephone Interconnect Service connects to VoIP Gateway by means of Session Initiation Protocol (SIP) in SIP Trunk connection mode. To transfer the voice data, G.711 codec is fully supported. Also it is possible to implement G.723 and G.729 support, but additional licensing is needed. Here are some examples how to configure your VoIP gateway. Cisco, AddPac: dial-peer voice 10 voip! destination-pattern 22. // all numbers beginning with 22 session protocol sipv2 session target ipv4: codec g711alaw // Radioserver IP address
22 Telephone Interconnect General VoIP Gateway settings Using GUI of IPGear VoIP Gateway: Radioserver IP Mask number SIP
23 Voice call from telephone to radio General overview To successfully make a voice call from telephone set to radio, the following requirements must be satisfied: The number, which is typed by telephone subscriber, is routed from PBX to VoIP gateway. Then the correct SIP address is generated by VoIP gateway and routed to SmartPTT Radioserver. Radio subscriber is on-line and available. On every stage the dialed number can be modified along with settled rules: PBX can add or remove prefix in dialed by phone subscriber number. VoIP gateway can choose one of different radioservers based on prefix. Radioserver can be configured to removed unnecessary prefixes. Also on every stage it s possible to modify information about caller number (Automatic Number Identification/Caller ID (ANI/CID)). In case of using analog channels, all ANI/CID information will be lost. Radio subscribers are addressed by Radio ID, so PBX must be programmed to allocate number pool corresponding to Radio IDs.
24 Voice call from telephone to radio To make a call from Telephone to Radio subscribers, it is necessary to do preliminary settings on both PBX, VoIP-Gateway and Radioserver. PBX must be programmed to allocate number pool corresponding to radio IDs (group IDs in case of group calls). For example, all radio subscribers (Radio ID= XXX) are assigned with telephone numbers = 51XXX and all radio groups (Group ID= NNN) are assigned with telephone numbers = 52NNN. PBX is programmed in the way when all calls to numbers beginning with 5 are forwarded to VoIP-Gateway. Then you can configure your systems in two different ways, which differs by the stages, where different prefixes are removed. 1. This way is suitable when your VoIP Gateway supports number modification. VoIP Gateway should be programmed to extract destination number from initial telephone numbers (remove prefix 5 in our case) and compose destination SIP-address to make the call to Radioserver: [Destination of Radioserver]:[SIP port of Radioserver] Radioserver is configured to treat prefixes in destination number to differ individual and group calls (1 and 2 respectively in our case). 2. This way is suitable when your VoIP Gateway does not support number modification. So your program VoIP Gateway to simply route all calls to Radioserver with following SIP-address: [Initial dest. of Radioserver]:[SIP port of Radioserver] Radioserver is configured to treat prefixes in SIP-name to differ individual and group calls.
25 Voice call from telephone to radio Example configuration Example 1 PBX is programmed to route to VoIP gateway all calls to numbers beginning with 5. VoIP Gateway is programmed to route all these calls from PBX to Radioserver and remove 5 in front of number. Telephone Interconnect Radioserver settings: Interface = , SIP Port = 5060, Private Call Prefix = 1, Group Call Prefix = 2. To call the radio with Radio ID = 123, telephone subscriber dials 51123, PBX route this call to VoIP gateway, VoIP gateway composes SIP address = and initiates session with Radioserver. Radioserver recognize the private call prefix (1) in front of SIP name and make private call to radio with Radio ID = 123. The same for group call. To call all radios within group with Group ID = 33, telephone subscriber dials 5233, PBX route this call to VoIP gateway, which composes SIP address = and initiates session with Radioserver. Radioserver recognize the group call prefix (2) in front of SIP name and make group call to group with Group ID = 33. Example 2 PBX is programmed to route to VoIP gateway all calls to numbers beginning with 5. VoIP Gateway is programmed to route all these calls from PBX to Radioserver. Telephone Interconnect Radioserver settings: Interface = , SIP Port = 5060, Private Call Prefix = 51, Group Call Prefix = 52. To call the radio with Radio ID = 123, telephone subscriber dials 51123, PBX route this call to VoIP gateway, VoIP gateway composes SIP address = and initiates session with Radioserver. Radioserver recognize the private call prefix (51) in front of SIP name and make private call to radio with Radio ID = 123. The same for group call. To call all radios within group with Group ID = 33, telephone subscriber dials 5233, PBX route this call to VoIP gateway, which composes SIP address = and initiates session with Radioserver. Radioserver recognize the group call prefix (52) in front of SIP name and make group call to group with Group ID = 33.
26 Voice call from radio to telephone To make a call to telephone subscriber, radio subscriber needs to send text message with the phone number to base radio station. Text message is processed by Radioserver in accordance with settings done at Radioserver Configuration window and transferred to VoIP Gateway. In Radioserver Configurator: 1. Check Allow Telephone Interconnect checkbox for every Control Station which can be used for Telephone Interconnect. 2. Check Allow Calls to Default VoIP Gateway and set Default Gateway Settings to process incoming text messages in accordance with Dialing Rules. 3. Set Dialing Rules Prefix. For VoIP Gateway: Configure gateway to translate incoming calls from SIP connection to PBX. If the digital lines are used to connect VoIP Gateway and PBX, it is possible that PBX will check the incoming Caller ID and allow calls only with CID in certain range. In this case the CID modification rules should be configured on gateway. If it is not possible, you can configure the Source Prefix in Telephone Interconnect Server Settings on Radioserver. Note! Radio subscriber can also make a call to telephone subscriber by sending it s full SIP-Address to the Base Radio. For example:
27 Voice call from radio to telephone Example configuration Radioserver settings: Server Settings: Interface = , SIP Port = 5060 Default Gateway Settings: Address = , SIP Port = 5060, Source Prefix = 1, Destination Prefix = 12 Dialing Rules: Prefix = A VoIP Gateway Settings: Gateway is programmed to transfer all calls to PBX, removing prefix 12 in front of Calling ID and add prefix 5 in front of CID. To call to Phone Number = 743, Radio Subscriber with Radio ID = 123 sends text message A743 to base radio. Telephone Interconnect service extracts phone number from the text message, add Destination Prefix and composes following SIP Addresses for calling to VoIP Gateway: From: To: VoIP Gateway processes these SIP Addresses and transfer call to PBX, in which: Caller ID: Calling ID: 743 Note! In case your VoIP gateway does not support Caller ID/Calling ID modifications, you can obtain the same result if you modify the following Default Gateway Settings: Set Source Prefix to 51. Empty Destination Prefix.
28 Patch Radio an Telephone subscribers by Dispatcher Console SmartPTT Dispatcher console has feature to patch radio and telephone subscribers. Voice call data is transferred by radioserver in this case, so license with enabled Telephone Interconnect Service is required to be installed on radioserver. 1. Open the Call Window of radio subscriber. 2. Select Call Telephone Subscriber menu item. 3. Call Telephone Subscriber window is displayed. 4. Select telephone subscriber from the stored telephone subscribers list and click Connect button (or double-click on the list item). 5. If necessary phone number is not in list, you can click Dial button to input the telephone number manually. In Dialing window, input the phone number and click or press Enter.
29 Patch Radio an Telephone subscribers by Dispatcher Console During dialing process status window will be displayed showing current state of the call. When telephone subscriber picks up the phone, connection status is changed from Ringing to Call. If the connection attempt is failed, status window shows corresponding error message. To disconnect current connection click button. To establish the same connection again, click button.
30 Tested environments There is no any additional system requirements for Telephone Interconnect Service except listed in VoIP Gateway Requirements section. But please note that the overall efficiency of the system consisting of equipment by different manufactures depends on the various factors. The main of them are the integration ability of different parts and possibility to implement unified numbering plan for PBX and radio subscribers. Equipment, listed in this list, was tested for compatibility and worked with SmartPTT telephone interconnect. VoIP Gateways: Cisco integrated services routers (recommended) AddPac VoIP gateways IPGear VoIP gateways Audiocodes Mediant & Mediapack VoIP gateways PBX: Cisco CallManager Tadiran Coral LG ipecs Asterisk (and all Asterisk-based PBX)
Valcom PagePro SIP (Session Initiation Protocol) Paging Servers, models VIP-201 and VIP-204, are compatible with Cisco Unified Communications Manager as either a Third-party SIP Device (Basic or Advanced)
AudioCodes Mediant 800 MSBG Page 1 of 66 6.40A AudioCodes Mediant 800 MSBG 1. Important Notes Check the SIP 3 rd Party Validation Website for current validation status. The SIP 3 rd party Validation Website
AudioCodes Mediant 1000 Configuration Guide 2010 FaxBack, Inc. All Rights Reserved. NET SatisFAXtion and other FaxBack products, brands and trademarks are property of FaxBack, Inc. Other products, brands
Internet Telephony Terminology Understanding the business phone system world can be a daunting task to a lay person who just wants a system that serves his or her business needs. The purpose of this paper
Application Note - IP Trunking End-to-End Configuration for IP Trunking This document gives you a detailed description of how to configure IP Trunking in a Tenor VoIP system. The following topics are included
Configuring the Synapse SB67070 SIP Gateway for Broadvox GO! SIP Trunking 2012 Advanced American Telephones. All Rights Reserved. AT&T and the AT&T logo are trademarks of AT&T Intellectual Property licensed
Valcom Network Trunk Ports, models, are compatible with Cisco Unified Communications Manager as either a Third-party SIP Device (Basic or Advanced) or as a SIP Trunk. To preserve the Caller ID information
VoIP H.323 Series VoIP Gatways: VoIP 422/404/440/800 VoIP Routers: VoIP 404R/440R/200R/110R Quick Setup Guide Important Information In this guide, you will learn how to setup our VoIP gateway and Routers
Avaya Solution & Interoperability Test Lab Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Abstract These Application Notes describe the steps to configure an Avaya
1 Voice over IP (VoIP) - Basic Voice over IP - Voice over IP (VoIP) is a digital form of transport for voice transmissions, replacing analog phone systems. The benefits of VoIP are considerable: Better
SmartPTT File Transfer User's Guide Version 2.0 Introduction 2 Introduction SmartPTT File Transfer was designed to transfer files over-the-air by means of MOTOTRBO radios. It is an easy-to-use and very
Avaya Solution & Interoperability Test Lab Sample Configuration for SIP Trunking between Avaya IP Office R8.0 and Cisco Unified Communications Manager 8.6.2 Issue 1.0 Abstract These Application Notes describe
Connecting with Free IP Call Free IP Call (http://www.freeipcall.com/) offers telephone service using the VoIP standard SIP. The service allow users making/receiving VoIP calls to/from VoIP telephone numbers
Configuring a Cisco CallManager system to work with Biamp s SVC-2 card Tesira Voice-over-IP Interface Biamp s SVC-2 card allows Biamp Tesira digital signal processors to make and receive calls over any
Chapter 3 - Voice Dial Peers In order to understand the concept of dial peers, it is important to understand call legs. A voice call over a packet network is segmented into discrete call legs. A call leg
SIP Trunking using Optimum Business Sip Trunk Adaptor and the Zultys MX250 IP PBX Table of Contents Goal 3 Prerequisites 3 Zultys MX250 Configuration 4 Network Settings 4 Phone Registration and Assignment
Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670 Businesses Save Money with Toshiba s New SIP Trunking Feature Unlike gateway based solutions, Toshiba s MIPU/ GIPU8 card
Valcom Session Initiation Protocol (SIP) VIP devices are compatible with SIP-enabled versions of Avaya IP Office (5.0 and higher). The Valcom device can be added to the IP Office system as a SIP Trunk.
Peer-to-Peer SIP Mode with FXS and FXO Gateways New Rock s SIP based VoIP gateways with FXS and FXO ports support peer-to-peer mode which has many applications in deploying enterprise multi-site telephone
IPedge Feature Desc. 1/31/13 OVERVIEW Direct Inward Dialing (DID) Direct Inward Dialing is a feature offered by telephone service providers for use with their customers' PBX systems. The telephone service
How to Configure the Cisco UC500 for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the Cisco UC500 IP PBX to connect to Integra Telecom SIP Trunks.
2N VoiceBlue Next 2N VoiceBlue Next & Siemens HiPath (series 3000) connected via SIP trunk Quick guide Version 1.00 www.2n.cz 1 2N VoiceBlue Next has these parameters: IP address 192.168.1.120 Incoming
NetVanta 7100 Exercise Service Provider SIP Trunk PSTN NetVanta 7100 FXS 0/1 x2001 SIP Eth 0/0 x2004 SIP Server 172.23.102.87 Hosted by x2003 www.voxitas.com In this exercise, you will create a SIP trunk
GW400 VoIP Gateway User s Guide P/N: 956YD30001 Copyright 2006. All Rights Reserved. Document Version: 1.0 All trademarks and trade names are the properties of their respective owners. i Table of Contents
Integrating Citrix EasyCall Gateway with SwyxWare The EasyCall Gateway has been tested for interoperability with Swyx SwyxWare, versions 6.12 and 6.20. These integration tests were done by using EasyCall
Connecting with sipgate sipgate (http://www.sipgate.co.uk/) offers telephone service using the VoIP standard SIP. sipgate covers every area code within the United Kingdom and provides a local Direct Inward
Vega 100G and Vega 200G Gamma Config Guide This document aims to go through the steps necessary to configure the Vega SBC to be used with a Gamma SIP Trunk. When a SIP trunk is provisioned by Gamma a list
Avaya Solution & Interoperability Test Lab Application Notes for Configuring QuesCom 400 IP/GSM Gateway with Avaya IP Office using H.323 trunks Issue 1.0 Abstract These Application Notes describe the configuration
CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 From the Cisco CVOICE 642-432 Exam Topics Voice over IP Technologies Describe the similarities and differences between PSTN and VoIP including
Smart Web Manager for SS7 VoIP Gateways www.addpac.com AddPac Technology 2013, Sales and Marketing Contents SS7 VoIP Gateway Service Diagram SS7 VoIP Gateway Main Features SS7 VoIP Gateway Features SS7
NetVanta Unified Communications Technical Note Supporting Multiple PBXs in Hybrid Deployment Models Application Note for Hybrid Deployments The goal of this technical note is to ensure that you can leverage
SIP Trunk Configuration Guide using www.cbeyond.net 1-877-441-9783 The information contained in this document is specific to setting up SIP connections between Vertical SBX IP 320 and Cbeyond. If you require
Quick Installation Guide PRI Gateway Version 2.4 Table of Contents Hardware Setup... 1 Accessing the WEB GUI... 2 Notification LEDs (On the Front Panel of the Gateway)... 3 Creating SIP Trunks... 4 Creating
email@example.com 941.306.2200 Tech Bulletin 2012-002 Description This guide is intended to streamline the installation of AccessLine SIP trunks in the IPitomy IP PBX. In our combined testing we determined
Enabling NAT and Routing in DGW v2.0 June 6, 2012 Proprietary 2012 Media5 Corporation Table of Contents Introduction... 3 Starting Services... 4 Distinguishing your WAN and LAN interfaces... 5 Configuring
Gateways and Their Roles Understanding Gateways This topic describes the role of voice gateways and their application when connecting VoIP to traditional PSTN and telephony equipment. Analog vs. Digital
Managing Incoming Lines Incoming Lines Incoming Lines must be created to Route incoming calls to required destinations From Configuration > Telephony > Lines > Incoming Lines Click on Add a new Incoming
Avaya Solution & Interoperability Test Lab Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Abstract These Application Notes describe the procedures for configuring
ESI SIP Trunking Installation Guide 0450-1227 Rev. B Copyright 2009 ESI (Estech Systems, Inc.). Information contained herein is subject to change without notice. ESI products are protected by various U.S.
BlackBerry Mobile Voice System for SIP Gateways and the Avaya Aura Session Manager Version: 5.3 Feature and Technical Overview Published: 2013-06-19 SWD-20130619135120555 Contents 1 Overview...4 2 Features...5
Configuration Guide BUSINESS TALK IP SIP SAGEMCOM XMediusFax Server Enterprise 188.8.131.524 Version Version du 11/03/2013 1 of 21 Sommaire Configuration Guide Business Talk IP 1 Document objectives............3
H.323 / SIP VoIP Gateway VIP GW Quick Installation Guide Overview This quick installation guide describes the objectives; organization and basic installation of the PLANET VIP-281/VIP-480/VIP-880/VIP-1680/VIP-2480
SIP Trunking with Elastix Configuration Guide for Matrix SETU VFXTH Contents Setup Diagram 3 SIP User Configuration in Elastix for SETU VFXTH 4 SIP Trunk Configuration in SETU VFXTH for Elastix 7 Selecting
Configuration Guide for connecting the Eircom Advantage 4800/1500/1200 PBXs to the Eircom SIP Voice platform. 1 Contents Introduction.... 3 Installing the Applications Module... 4 Ordering a Licence for
V o I P G S M G A T E BlueGate SIP 1 Installation and setup guide V 1.0 1. General description 1.1 Technical parametres Dimensions 100 x 130 x 37 mm Weight 850 g Operating position various Operating condition
Integrating VoIP Phones and IP PBX s with VidyoGateway Updated February 2011 INDEX: I. ABSTRACT.1 II. III. IV. VIDYOGATEWAY OVERVIEW.. 1 NETWORK TOPOLOGIES AND DEFINITIONS...2 CONNECTING TO VIDYOCONFERENCES
Configuring Avaya Aura Session Manager system with Biamp s Vocia MS-1 Vocia MS-1 Voice-over-IP Interface The Biamp Vocia Message Server (MS-1) includes functionality to allow pages to be initiated over
Page 1 of 16 1 Introduction The MX systems support traditional fax machines to send and receive faxes over a standard analog FXS interface. This method of sending and receiving faxes has several drawbacks
Link Gate SIP (Firmware version 1.20) User guide v1.0 1 Content 2 1. Technical parameters - Dimensions 133 x 233 x 60 mm - Weight 850 g - Operating position various - Operating condition temperature: +5
Network Load Balancing Step by Step installation of Network Load Balancing in Windows Server 2008 R2. Prerequisite for NLB Cluster 1. Log on to NODE1 Windows Server 2008 R2 system with a domain account
Configuring the Synapse SB67070 SIP Gateway from AT&T for Clearfly SIP Trunking January 2013 1 Introduction This guide was created to assist Synapse partners with configuring the Synapse SB67070 SIP Gateway
Tech Bulletin 2011-004 Description This guide is intended to streamline the installation of Paetec SIP trunks in the IPitomy IP PBX. Procedure Add Provider 1. Navigate to the IPitomy IP PBX web interface
MAX T1/E1 TM VoIP Gateway Quick Start Guide Version 1.0 Contents INTRODUCTION 1 Hardware Needed Software Needed 1 1 NET2PHONE MAX SET UP Hardware Set Up Software Set Up Set Up Internet Protocol (IP) Address
Table of Contents Device SIP Trunking Administrator Manual Version 20090401 Table of Contents... 1 Your SIP Trunking Service... 2 Terminology and Definitions... 2 PBX, IP-PBX or Key System... 2 Multi-port
Evolution PBX User Guide for SIP Generic Devices Table of contents Introduction... 1 Voicemail... Using Voicemail... Voicemail Menu... Voicemail to Email... 3 Voicemail Web Interface... 4 Find Me Rules...
Quick Installation Guide MegaPBX Version 2.1 Quick Installation Guide v2.1 www.allo.com 2 Table of Contents Initial Setup of MegaPBX... 4 Notification LEDs (On the Front Panel of the Gateway)... 5 Create
CHAPTER 1 This section describes the hardware and software features of the Cisco Analog Telephone Adaptor (Cisco ATA) and includes a brief overview of the Skinny Client Control Protocol (SCCP). The Cisco
OAISYS SIP Trunk Integration 05/11/2011 Americas Headquarters OAISYS 7965 South Priest Drive, Suite 105 Tempe, AZ 85284 USA www.oaisys.com (480) 496-9040 OVERVIEW OAISYS introduces the ability to record
SIP Trunking with Allworx Configuration Guide for Matrix SETU VoIP Gateways Disclaimer Matrix Comsec Pvt. Ltd. reserves the right to change, at any time, without prior notice, the product design, specifications,
Building a Scalable Numbering Plan Scalable Numbering Plan This topic describes the need for a scalable numbering plan in a VoIP network. Dial Plans Dial plans contain specific dialing patterns for a user
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between Metaswitch MetaSphere CFS and Avaya IP Office Issue 1.0 Abstract These Application Notes describe the steps
NetVanta 7060/7100 Configuration Checklist AOS Versions Supported: AOS A1.01.00 and above. AOS Versions Supporting SIP Trunking and Networking: AOS A2.02.00 and above. This document is designed to provide
1 SIP Carriers 1.1 Priority Telecom 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can
OfficeMaster Gate (Virtual) Enterprise Session Border Controller for Microsoft Lync Server Quick Start Guide October 2013 Copyright and Legal Notice. All rights reserved. No part of this document may be
61200796L1-29.1G March 2013 Configuration Guide Configuring Shared Line Appearances over Analog Trunks This configuration guide explains how to configure shared line appearances (SLAs) on AOS voice products
Quick Start Guide v1.0 Table of contents : 01. Quick Start Guide...03 O2. Configuring your VoIPOffice appliance...14 03. Adding a VoIPtalk trunk...21 04. Configuring UADs for use with VoIPOffice...25 05.