TRANSip Voice over IP solution for REDCOM Technologies Corporal Neo Martinez Communications Company September 2013.

Slides:



Advertisements
Similar presentations
© 2009 Avaya Inc. All rights reserved. IP Office control units and IP Office Expansion Modules.
Advertisements

Unified Communications
BAI613 Module 2 - Voice over IP Technology. Module Objectives 1. Describe the benefits of IP Telephony/Packet Telephony/VoIP over traditional telephone.
 WAN uses Serial ports  Ethernet Ports:  Straight through  Cross over.
IPX-300 Series PBX with FXO GW Configuration Internet Telephony PBX System Copyright © PLANET Technology Corporation. All rights reserved.
Radio over Internet Protocol
Overview of DVX 9000.
Tom Behrens Adam Muniz. Overview What is VoIP SIP Sessions H.323 Examples Problems.
Voice over IP Fundamentals
Security in VoIP Networks Juan C Pelaez Florida Atlantic University Security in VoIP Networks Juan C Pelaez Florida Atlantic University.
KX-TDE100/200 System (Version 1.0)
SG2001_VIP.ppt Page 1 PLANET Technology Corp. Product Guide 2001 VoIP Products Your Voice over Internet By Product Department.
1223 North Glenville Dr. Richardson, TX P: F: The Value-Priced Leader Now Offers Web-Control Confer III.
1 © 2006 Cisco Systems, Inc. All rights reserved. Session Number Presentation_ID Using the Cisco Technical Support & Documentation Website for Voice Issues.
Chapter 1: Computer Networks IB 300: Advanced Computer Sciences. Professor: Nabil Elmjati.
FXS Voice Gateways and Cisco Unified Call Manager Configuration Guide 3 RD Marine Division HQBN Communications Company 1LCPL MARTINEZ NEO.
Data Networking Fundamentals Unit 7 7/2/ Modified by: Brierley.
Voice over Internet Protocol (VoIP) Training and Development.
H.323/ SIP Internet Telephony Gateway
TAX-AIDE Local Area Networking July, 2013.
Integrating Voice with Data Over a Leased Line
Internet Telephony PBX System
H.323/ SIP Internet Telephony Gateway
VIP-2/4/8/16/24 port Series P2P Configuration H.323/ SIP Internet Telephony Gateway Copyright © PLANET Technology Corporation. All rights.
BASIC TELECOMMUNICATIONS
Describe the elements of a VoIP Dial Plan Design.
MiVoice Office v MiVoice Office v6.0 is mainly a service enhancement release, rather than a user feature rich enhancement release.
Internet Video Conferencing Phone
Quintum Technologies, Inc. Risk Free VoIP.
© 2007 NeoAccel, Inc. NeoAccel SGX Installation Guide Dear Customer: We are pleased to provide you with our training presentation for our SSL VPN-Plus.
Configuring Routing and Remote Access(RRAS) and Wireless Networking
Internet Addressing. When your computer is on the Internet, anything you do requires data to be transmitted and received. For example, when you visit.
Voice VLANs Lecture 7 VLANs.ppt 21/04/ Apr-17
Quintum Technologies, Inc. Risk Free VoIP.
PART 2: Product Line. Tenor Switches & Gateways Tenor AX Series Solution For Medium to Large Enterprises  Available in 8, 16, 24 and 48 port Available.
Features and Applications for Multisite Deployments
© 2008 Cisco Systems, Inc. All rights reserved.CIPT1 v6.0—4-1 Enabling Single-Site On-Net Calling Implementing MGCP Gateways in Cisco Unified Communications.
1 © 2000, Cisco Systems, Inc. ATA_overview0101 Analog Telephone Adaptor Overview Product IntroductionProduct Introduction.
CHAPTER 14 PSTN and VoIP Interworking. Cisco Packet Telephony: Connection Control Call Control Services.
Basics of IP Telephony Sam Lutgring Director of Informational Technology Services Calhoun Intermediate School District.
Internet Telephony PBX System
Telecom 101 Intro to Switch Programming Presented by: Mark Vollmer © Mark Vollmer, 2005 PreviousNext.
Cisco 3 - LAN Perrine. J Page 110/20/2015 Chapter 8 VLAN VLAN: is a logical grouping grouped by: function department application VLAN configuration is.
NETWORK COMPONENTS Assignment #3. Hub A hub is used in a wired network to connect Ethernet cables from a number of devices together. The hub allows each.
M340 Modbus Plus Proxy Link legacy systems to Ethernet
TELEPHONE NETWORK Telephone networks use circuit switching. The telephone network had its beginnings in the late 1800s. The entire network, which is referred.
ﺑﺴﻢﺍﷲﺍﻠﺭﺣﻣﻥﺍﻠﺭﺣﻳﻡ. Group Members Nadia Malik01 Malik Fawad03.
Single-Side Configuration IPX-1900 Single-Side Configuration Internet Telephony PBX System Copyright © PLANET Technology Corporation.
NATIONAL INSTITUTE OF SCIENCE & TECHNOLOGY VOICE OVER INTERNET PROTOCOL SHREETAM MOHANTY [1] VOICE OVER INTERNET PROTOCOL SHREETAM MOHANTY ROLL # EC
PTCL Training & Development
ICF-1600 P2P Configuration Internet Video Conferencing Phone Copyright © PLANET Technology Corporation. All rights reserved.
Aastra Communication Server Aastra 400
Voice Over Internet Protocol (VoIP) Copyright © 2006 Heathkit Company, Inc. All Rights Reserved Presentation 5 – VoIP and the OSI Model.
Trimble TMR1 Data Radio Operational Overview.
KX-NS PBX Remote IP Extension
Unit-4 Telephone system
© 2006 Cisco Systems, Inc. All rights reserved.Cisco PublicITE I Chapter 6 1 Cisco Routers.
HOW TO GUIDE: INEXPENSIVE INTERNET PROTOCOL TELEPHONY SOLUTION Created by: Cameron Adkisson Eastern Kentucky University
© ITT Educational Services, Inc. All rights reserved. IS3120 Network Communications Infrastructure Unit 7 Layer 3 Networking, Campus Backbones, WANs, and.
Cisco Routers Routers collectively provide the main feature of the network layer—the capability to forward packets end-to-end through a network. routers.
Rohde & Schwarz Topex VoxiPlus Advanced July 2011.
Configuring Network Devices
On-Site PBX Vs Hosted PBX.
SIX MONTHS INDUSTRIAL TRAINING REPORT
KX-HTS Step by Step Guide SIP Phone in Existing Router
Switch Setup Connectivity to Other locations Via MPLS/LL etc
Em4 Ethernet tutorial Remote connection.
© 2002, Cisco Systems, Inc. All rights reserved.
VoIP Signaling and Call Control
Voice Over Internet Protocol
Presentation transcript:

TRANSip Voice over IP solution for REDCOM Technologies Corporal Neo Martinez Communications Company September 2013

Overview The purpose of this document is to inform you, the user, on fundamental TRANSip concepts and configurations to provide VoIP capabilities with REDCOM Technologies. In other words (Marines): instead of setting up a Call Manager to install voip phones, you will now program TRANSip in MSC Slot 15 of your Redcom. Note: the audience targeted for this lesson are Marines with no prior experience of IP telephony. TRANSip 2

Overview The areas that will be covered during this lesson are: Media Service Circuit (MSC) Board TRANSip Capabilities Configuring internal VoIP phones with SIP signaling Configuring a SIP to trunk to external nodes TRANSip 3

Media Service Circuit (MSC) Board The MSC Board provides Dual Tone Multi Frequency (DTMF) detection, tone generation, echo cancellation and conference calling. The MSC Board is featured in the Slice2100 and the High- Density Exchange Commercial Communication Switch (CCS) inside the DTC facility. The MSC Board in compatible with the Modular Switching Units (USMC’s DEOS), however, it must run REDCOM v5.0 or higher. TRANSip 4

Media Service Circuit (MSC) Board Located in slot 15 of USMC packages. Halt Major Power Good N/A RGCP Registration DSP 0 DSP 1 DSP 2 DSP 3 DSP 4 DSP 5 100Base TX 10Base T TX Link RX RJ-45 Connector TRANSip 5

TRANSip is Redcom’s solution to IP telephony. TRANSip uses Session Initiation Protocol (SIP) protocol to communicate between devices. You will need to Configure a LAN Switch for network connectivity. Consult a Data Marine or network administrator for further guidance on setting up a LAN Switch. NOTE: TRANSip uses SIP to communicate its devices, that means you will need to modify the voip phone’s firmware. For further guidance on how to change firmware visit: TRANSip 6

Elements of TRANSip TRANSip Consist of 4 fundamental elements; SIP Call Manager: provides functions such as call setup, control, termination, and directory services for IP users. Media Gateway: supports various compression algorithms/CODECS such as FoIP, G.711 and G.729. Media Gateway Controller: TRANSip acts as a gateway to allow different media/protocols to communicate. (I.e. VoIP, BRI, PRI, DTMF, ect. ) Legacy Support: this feature makes TRANSip compatible with external T!, E1, E&M and LSRD. TRANSip 7

Configuring TRANSip Next you will configure TRANSip to make internal calls with voip phones. PLEASE NOTE: Before you begin you need to set a LAN Switch for network connectivity. You MUST change the voip phone’s firmware to SIP rather than SCCP. Click on the hyperlink and follow this firmware tutorial If you need guidance on how to do this protocol change for your voip phones.firmware tutorial Assuming you have basic POTS internal calling and you met the requirements mentioned above, you may begin configuring the REDCOM for voip purposes. TRANSip 8

Configuring TRANSip Follow the commands and understand what you’re typing to the console. Refer to the snap shots in the following slides for a visual reference/guidance. Ethernet Configuration adm>slot=p enter network settings see image 1 adm/slot>ip address assign the MSU IP address adm/slot>net_mask adm/slot>gateway adm/slot>ex;ac Assign the MSC Board an IP Address adm>slot=15 enter MSC Board network settings see image 2 adm/slot> ip address assign MSC IP (or call manager IP) adm/slot>net_mask adm/slot>gateway adm/slot>ex;ac TRANSip 9

Configuring TRANSip Device Screen | Configuring voip phones adm>device enter device settings see image 3 adm/device>entry=22 adm/device>state=on turn sip service on adm/device>link=23 adm/device>entry=23 adm/device>state=on adm/device>port=5060 layer 4 port number for sip adm/device>ex;ac Configuring Line Group adm>group create a line group for internal voip phones see image 4 adm/group>new=lin adm/group>group=1 note:your first group adm/group>type=voip set protocol to voip adm/group>name=“IP-hone” adm/group>members see image 5 TRANSip 10

Configuring TRANSip adm/group>add=15/0/0 add MSC circuit board to your voip line group adm/group>qty=15 number of devices adm/group>ex;ac Setting Dynamic Station List Attributes adm>dtsnlist>station=100 create a new station see image 6 adm/dtsnlist>name=Major adm/dtsnlist>dtsn=100 access the voip station you just made adm/dtsnlist>signaling=sip set parameters see image 7 adm/dtsnlist>voip_group=1 should match you voip line group adm/dtsnlist>sip_user=100 adm/dtsnlist>ex;ac Secure Session Initiation Protocol (SECSIP) adm>option adm/opt>secsip set secure sip see image 8 adm/opt>ex;ac TRANSip 11

Configuring TRANSip Increasing the DSTN table for more users >gen gen>data this allows more space for voip users, default is only 2 gen/data>table=dstint see image 9 gen/data>current=20 set user value to the number you wish gen/data>ex;ac TRANSip 12

Setting up a SIP TRUNK A SIP Trunk will allow you to talk to other nodes over IP. Configuring a SIP Trunk adm>group adm/grp>new=trk create a second group for your sip trunk adm/grp>group=2 adm/grp>name=“SIPtrunk” adm/grp>dialing=sip set protocol to sip adm/grp>dct=11 route it out dial code table 11 adm/grp>d_host_sip= your own MSC IP is host adm/grp>d_name_sip=“sip” adm/grp>members adm/grp>add=15/0/0 add the MSC circuit board adm/grp>qty=24 adm/grp>ex;ac TRANSip 13

Setting up a SIP TRUNK Configuring route screen adm>route=1 create a route for your sip trunk see image 10 adm/rte>name=“sip to xxx” adm/rte>group=2 assign it to match the sip trunk adm/rte>del=0 adm/rte>out=7 adm/rte>ex;ac Configuring Dial Code Tables adm>dct adm/dct>dct=7 adm/dct>entry=9;pattern=xxxx adm/dct>type=rte;val=2 adm/dct>dct=8 adm/dct>entry=9;pattern=nnxx adm/dct>ex;ac TRANSip 14

TRANSip Image 1 TRANSip 15

TRANSip Image 2 TRANSip 16

TRANSip Image 3 TRANSip 17

TRANSip Image 4 TRANSip 18

TRANSip Image 5 TRANSip 19

TRANSip Image 6 TRANSip 20

TRANSip Image 7 TRANSip 21

TRANSip Image 8 TRANSip 22

SIP Trunk Image 9 TRANSip 23

SIP Trunk Image 10 TRANSip 24