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One100E Quick Installation Guide

1 Summary
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Summary ...................................................................................................................................................................1
Introduction ...............................................................................................................................................................2
Hardware description.................................................................................................................................................2
Making the connections.............................................................................................................................................2
Connecting to the equipment.....................................................................................................................................3
Configuring the device WAN IP address ..................................................................................................................4
Configuring the router gateway .................................................................................................................................4
Configuring the Voice routing table ..........................................................................................................................5
8.1 Outgoing (towards SIP network) Call forwarding. ...............................................................................................5
8.2 Incoming (from SIP network) calls and SIP client configuration .........................................................................5
8.3 Outgoing calls (coming from ISDN, going out towards VoIP network) configuration ........................................7
8.3.1
Adding calling number if required .........................................................................................................7
8.3.2
Forward valid outgoing calls by checking their Calling Number...........................................................7
8.3.3
Replace other Calling Numbers by a pre-defined number......................................................................8
8.4 Deleting an entry in the routing table ...................................................................................................................8
8.5 Voice routing Example .........................................................................................................................................9
9 Saving the configuration ...........................................................................................................................................10
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Debugging the network.......................................................................................................................................11
10.1
Ping the router gateway..................................................................................................................................11
10.2
Ping the SIP gateway .....................................................................................................................................11
10.3
Check the status of the ISDN ports ................................................................................................................11

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2 Introduction
This document gives the basic steps for installing the OneAccess One100E Multiservice Router, for
connection ISDN BRI subscribers to the Etisalat network. For more advanced configurations, please consult
the product user guide.
The unit is delivered with a default Etisalat configuration, as to limit the number of manipulations on-site.

3 Hardware description

The unit is equipped with following interfaces


(1) Serial Console connector (RJ45) for local configuration
(2) Ethernet uplink, for connection to ONT
(3) Ethernet swith, for connecting local user applications (optional)
(4) 2 BRI ports, for connecting local ISDN applications: telephones, PBX or ISDN video applications
(5) Power input

4 Making the connections


Following drawing shows the connections in case of a ISDN PBX. Also other types of ISDN applications may
be connected.

(1)

(4)
(5)

(2)
ONT

GPON

(1) Connect the console port to the serial port of the PC, using the console cable. As a reference, the
cable layout of this cable is given in Annex A
(2) Connect the Ethernet uplink to the ONT, using a straight Ethernet Cable
(4) Connect the ISDN application using straight RJ45 cabling. The OneAccess unit is default configured
in NT Mode.
(5) Connect the Power

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5 Connecting to the equipment


The equipment is shipped with a default configuration, adapted to the Etisalat environment. This limits the
configuration work to be executed on site.
The Multiservice Router is using the industry standard CLI for doing configuration. Other configuration
options (such as Web interface, auto-provisioning, etc, are not described in this quick installation guide).
Please start-up a terminal program on your PC (such as hyperterminal), and configure the serial port on
which the unit is connected for 9600 bps, 8N1, and no flow control (see picture)

After the device has booted, the Username prompt will be displayed in the terminal window. If you only
connected the terminal after the device has already booted, type enter to display the prompt.

Type following username and password (case-sensitive!):


Username: admin
Password: admin
The actual configuration can now be displayed by issuing the command
show run
In order to enter configuration mode, enter following command:
conf t
the prompt changes to indicate the unit is in configuration mode.

please note that in the Command-Line Interface, one can use the <tab> key to do auto-completion of the
commands.

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6 Configuring the device WAN IP address


Use following command (in configuration mode) to change the units WAN IP address:
interface FastEthernet 1/0
ip address <ip address> <ip netmask>
exit
Please use the IP address as it has been assigned for this customer.

7 Configuring the router gateway


The gateway needs to be configured in order to have the router to be able to connect to the IP core network.
The configuration of the gateway is actually done by adding a default route to the routing table, pointing to
the gateway:
ip route 0.0.0.0 0.0.0.0 <gateway address>

Please use the Router gateway address, as it is applicable for that site.
Note: Generally The router gateway address is different from the SIP gateway address. The SIP gateway
address is the address of the VoIP Softswitch and it generally will not change. It is pre-configured within the
default configuration

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8 Configuring the Voice routing table


The voice routing table in the device will execute following tasks:
Forward Outgoing calls to the VoIP trunk
Forwarding incoming SIP Voice calls to the proper ISDN port
Mapping of ISDN subscriber numbers onto SIP accounts
Verifying that outgoing calling numbers are allowed, and replace them by a general number if
required
To enter the voice routing table, issue the following command:
voice-routing

In this context, the unit is already pre-configured with two logical voice ports, as they will be referenced
further in the voice routing table:
pots-group 1 corresponds to the first BRI port on the equipment (labeled 5/0 on the connector)
pots-group 2 corresponds to the second BRI port on the equipment (labeled 5/1 on the connector)

8.1 Outgoing (towards SIP network) Call forwarding.


The first voice routing entry (use the command show run to inspect current configuration) shows how all
outgoing calls are forwarded to the voice trunk:
-> Indicates a voice route in the voice routing table
route 2
->the logical voice group is the VoIP uplink
dial-peer voip 0
prefix-type outgoing called last ->this routing entry checks outgoing calls, and will check on called number
->Any called number will match and will dial out after a timer
prefix . timer
->there is no loopback of voice-calls on the VoIP trunk
no loopback-routing
exit
This voice routing entry is already part of the default configuration, and should not be altered.

8.2 Incoming (from SIP network) calls and SIP client configuration
Multiple telephone numbers, and associated multiple SIP usernames and passwords can be configured per
ISDN port.
In the default configuration, the phone numbers and associated SIP accounts for the pots-group 0 (first BRI
port) are configured in routing entry 50 and 51 of the voice routing table, those for pots-group 1 (second BRI
port) in routing entry 52 and 53
Each phone number and corresponding sip account entry has the following format:
route <route number>
-> identifies the logical voice port and identicates that sip client
dial-peer pots-group <pots group number> ua-sip
will be defined

sip-authentication <username> <password>


sip-username <identifier>

-> Sets the SIP username and password


-> Sets the SIP identifier

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prefix-type outgoing called last


prefix <MSN Number> length <MSN length>
no loopback-routing
exit

-> will filter the called number for calss outgoing on this port
-> filters on this ISDN number with associated length
-> disable forwarding to the same port where it comes from

For Etisalat network, SIP username and Sip identifier are set the same as the ISDN MSN Number. The SIP
password is set to the string value password
Example: assigning an ISDN 027711342 number to the first ISDN port (pots-group0) is done below, using
route entry 50

This process is repeated for the other phone numbers, using the next entries in the routing table (route
51,52,..)
A typical configuration could be the following:
route 50
dial-peer pots-group 0 ua-sip
sip-authentication 027711342 password
sip-username 027711342
prefix-type outgoing called last
prefix 027711342 length 9
no loopback-routing
exit
route 51
dial-peer pots-group 0 ua-sip
sip-authentication 027711343 password
sip-username 027711343
prefix-type outgoing called last
prefix 027711343 length 9
no loopback-routing
exit
route 52
dial-peer pots-group 1 ua-sip
sip-authentication 027711344 password
sip-username 027711344
prefix-type outgoing called last
prefix 027711344 length 9
no loopback-routing
exit
route 53
dial-peer pots-group 1 ua-sip
sip-authentication 027711345 password
sip-username 027711345
prefix-type outgoing called last
prefix 027711345 length 9
no loopback-routing
exit

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8.3 Outgoing calls (coming from ISDN, going out towards VoIP
network) configuration
The following configuration steps will execute following tasks
Verifying is the ISDN subscriber is sending a Calling Number, and if not add one.
Verifying the Calling Number is valid, and if not replace it by a valid number.
These tasks are executed in the default configuration from voice route entry 100 onwards.

8.3.1 Adding calling number if required


Following route syntax will check the calling number on a logical voice port, and add a number if no number
is present
route <routenumber>
->Specify voice port to check
dial-peer pots-group <number>
->check calls coming in on this port and check the calling number
prefix-type incoming calling last
->check if no number is present
prefix - length 0
insert-calling <calling number> type 2 plan 0 presentation 0 screening 0->if no number is present, insert one.
no loopback-routing
exit
Following example will do this for checking the first ISDN port (pots-group 0) and attributing the calling
number 027711342 if none would be present

8.3.2 Forward valid outgoing calls by checking their Calling Number


When ISDN subscribers are initiating outgoing calls that includes a calling number, the unit will check if the
Calling Number is a valid number. For this purpose, a routing entry will be created for checking each valid
number (as many routing entries as numbers available for this subscriber). This is done in the next voice
routing entries.
The format is the following:
route <number>
dial-peer pots-group <number>
prefix-type incoming calling last
prefix <validnumber> length <length>
no loopback-routing
exit

->Specify voice port to check


->check calls coming in on this port and check the calling number
->check if a valid calling number is present

Following example checks for 2 different calling numbers on pots-group 0 (first BRI port)

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8.3.3 Replace other Calling Numbers by a pre-defined number


In case the outgoing call (initiated by the ISDN subscriber) would contain a non-valid number, the following
voice routing entry will replace the number by a default number.
This route will only replace non-valid calling numbers, as a match on the valid calling numbers was already
done in the previous entries.
route <number>
->Specify voice port to check
dial-peer pots-group <number>
->check calls coming in on this port and check the calling number
prefix-type incoming calling last
->This will check on any calling number that was not yet matched in another routing entry
prefix . length 0
translate . <mainnumber> calling
->replace the number by a main number
no loopback-routing
exit
following example will replace for calls originated from pots-group 0 (First BRI port, the calling number by
number 027711342

8.4 Deleting an entry in the routing table


When editing the routing table, it is possible to delete a specific voice route by issuing following command
no route <number>

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8.5 Voice routing Example


Following shows a complete voice routing table, as based on the description in the previous paragraphs. It
shows a situation where each of the 2 BRI ports has 2 assigned ISDN numbers. This is also included as
default configuration of the device.
voice-routing
route 2
dial-peer voip 0
prefix-type outgoing called last
prefix . timer
no loopback-routing
exit
route 50
dial-peer pots-group 0 ua-sip
sip-authentication 027711342 password
sip-username 027711342
prefix-type outgoing called last
prefix 027711342 length 9
no loopback-routing
exit
route 51
dial-peer pots-group 0 ua-sip
sip-authentication 027711343 password
sip-username 027711343
prefix-type outgoing called last
prefix 027711343 length 9
no loopback-routing
exit
route 52
dial-peer pots-group 1 ua-sip
sip-authentication 027711344 password
sip-username 027711344
prefix-type outgoing called last
prefix 027711344 length 9
no loopback-routing
exit
route 53
dial-peer pots-group 1 ua-sip
sip-authentication 027711345 password
sip-username 027711345
prefix-type outgoing called last
prefix 027711345 length 9
no loopback-routing
exit
route 100
dial-peer pots-group 0
prefix-type incoming calling last
prefix - length 0
insert-calling 027711342 type 2 plan 0 presentation 0 screening 0
no loopback-routing
exit
route 101
dial-peer pots-group 0
prefix-type incoming calling last
prefix 027711342 length 9
no loopback-routing
exit
route 102
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dial-peer pots-group 0
prefix-type incoming calling last
prefix 027711343 length 9
no loopback-routing
exit
route 103
dial-peer pots-group 0
prefix-type incoming calling last
prefix . length 0
translate . 027711342 calling
no loopback-routing
exit
route 104
dial-peer pots-group 1
prefix-type incoming calling last
prefix - length 0
insert-calling 027711344 type 2 plan 0 presentation 0 screening 0
no loopback-routing
exit
route 105
dial-peer pots-group 1
prefix-type incoming calling last
prefix 027711344 length 9
no loopback-routing
exit
route 106
dial-peer pots-group 1
prefix-type incoming calling last
prefix 027711345 length 9
no loopback-routing
exit
route 107
dial-peer pots-group 1
prefix-type incoming calling last
prefix . length 0
translate . 027711344 calling
no loopback-routing
exit
exit

9 Saving the configuration


After making changes in the configuration, it is possible to get out of the configuration by issuing the
command exit.
Typing multiple times exit may be required, depending on the level within the configuration you were editing.
After exiting configuration mode, the command save run will save the changes you applied to the
configuration
Note: failure to issue the command save run will cause all applied changes to be lost when the equipment is
rebooted.

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10 Debugging the network


When after making the changes, the unit fails to connect, following methods are recommended to make a
first basic analysis:

10.1 Ping the router gateway.


Check if the router gateway is reachable.
Issue the command ping <gatewayipaddress> at the command prompt

If the router gateway is not reachable, please check the cabling to the ONT, or check if the correct Router
gateway has been configured

10.2 Ping the SIP gateway


To make sure that the SIP gateway (softswitch) is reachable, make ping to the SIP Gateway.
The IP address for the Eitisalat SIP gateway is pre-configured for 194.170.147.196.
The proper configuration of the IP address can be verified in the configuration (show run)

10.3 Check the status of the ISDN ports


Make sure the ISDN ports are properly connected by checking the port status
For checking the status of the first port, issue the command
show voice voice-port bri index 0
for checking the status of the secondf bRI port, issue the command
show voice voice-port bri index 0
if layer1 shows deactivated in the status, please check the cabling towards the connected ISDN device.

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Annex A:
Console Cable

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The following figure shows the console cable assembly: