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

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