Transcript TRANSip - Neo at Home

TRANSip
Voice over IP solution for REDCOM Technologies
Corporal Neo Martinez
Communications Company
September 2013
Overview
• 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.
TRANSip
• The purpose of this document is to inform you, the user, on
fundamental TRANSip concepts and configurations to provide
VoIP capabilities with REDCOM Technologies.
Note: the audience targeted for this lesson are Marines with no
prior experience of IP telephony.
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Overview
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Media Service Circuit (MSC) Board
TRANSip Capabilities
Configuring internal VoIP phones with SIP signaling
Configuring a SIP to trunk to external nodes
TRANSip
• The areas that will be covered during this lesson are:
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• 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 HighDensity 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
Media Service Circuit (MSC) Board
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Media Service Circuit (MSC) Board
• Located in slot 15 of USMC packages.
Halt
Major
Power Good
N/A
DSP 0
RGCP Registration
DSP 1
TRANSip
N/A
DSP 2
DSP 3
DSP 4
DSP 5
TX
Link
RX
100Base TX
10Base T
RJ-45
Connector
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• 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:
http://www.neoathome.com/sip-firmware-load/
TRANSip
TRANSip
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Elements of TRANSip
• 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
• TRANSip Consist of 4 fundamental elements;
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Configuring TRANSip
• Next you will configure TRANSip to make internal calls with
voip phones.
• 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.
TRANSip
• PLEASE NOTE:
• Assuming you have basic POTS internal calling and you met
the requirements mentioned above, you may begin
configuring the REDCOM for voip purposes.
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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.
adm>slot=p enter network settings see image 1
adm/slot>ip address 192.168.1.2 assign the MSU IP address
adm/slot>net_mask 255.255.255.0
adm/slot>gateway 192.168.1.1
adm/slot>ex;ac
TRANSip
Ethernet Configuration
Assign the MSC Board an IP Address
adm>slot=15 enter MSC Board network settings see image 2
adm/slot> ip address 192.168.1.5 assign MSC IP (or call manager IP)
adm/slot>net_mask 255.255.255.0
adm/slot>gateway 192.168.1.1
adm/slot>ex;ac
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Configuring TRANSip
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
TRANSip
Device Screen | Configuring voip phones
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
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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
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
TRANSip
Setting Dynamic Station List Attributes
Secure Session Initiation Protocol (SECSIP)
adm>option
adm/opt>secsip set secure sip see image 8
adm/opt>ex;ac
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Configuring TRANSip
Increasing the DSTN table for more users
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
>gen
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Setting up 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=192.168.1.5 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
• A SIP Trunk will allow you to talk to other nodes over IP.
Configuring a SIP Trunk
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Setting up a SIP TRUNK
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
TRANSip
Configuring route screen
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
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TRANSip
TRANSip
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SIP Trunk
TRANSip
Image 9
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SIP Trunk
TRANSip
Image 10
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