Allied Telesis AT-VP504E FXS User manual

AT-VP504E FXS (SIP/MGCP)
User’s Manual
PN 990-11591-10 Rev C
VoIPTalk

Copyright © 2001 Allied Telesyn International, Corp.
960 Stewart Drive Suite B, Sunnyvale CA USA 94085
All rights reserved. No part of this publication may be reproduced without prior written permission from Allied
Telesyn International, Corp.
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Contents User’s Manual (SIP/MGCP Version)
AT-VP504E FXS iii
Contents
Chapter 1
Using the AT-VP504E FXS ...................................................................... 1
Before you Begin......................................................................................................... 1
Acronyms.................................................................................................................................2
Overview of the AT-VP504E FXS............................................................................................2
Using the AT-VP504E FXS .....................................................................................................3
Call Processes............................................................................................................. 3
Calls Involving Another Terminal.............................................................................................4
Calls Involving a Terminal and a LAN Endpoint ......................................................................4
Calls Involving an Analog Gateway.........................................................................................5
Making Calls................................................................................................................ 7
Complete Dialing Sequence....................................................................................................7
Dialing a Telephone Number or a Numerical Alias..................................................................7
Dialing an IP Address..............................................................................................................8
Using the Call Waiting Feature.................................................................................... 9
Using the Call Transfer Feature .................................................................................. 9
Call Transfer – Supervised....................................................................................................10
Call Transfer – Unsupervised................................................................................................10
Using the Call Forward Feature................................................................................. 11
Call Forward Unconditional ...................................................................................................11
Call Forward on Busy / No Answer........................................................................................12
Conferencing Calls.................................................................................................... 13
Requirements ........................................................................................................................13
Managing a Conference Call.................................................................................................13
Appendix A
Glossary.................................................................................................. 15

Contents User’s Manual (SIP/MGCP Version)
AT-VP504E FXS iv

AT-VP504E FXS 1
1Using the AT-VP504E FXS
Thank you for purchasing the AT-VP504E FXS from Allied Telesyn.
This manual illustrates some of the various call processes the
AT-VP504E FXS supports. It also describes how to make and receive
calls, as well as how to use the services the AT-VP504E FXS offers.
Before you Begin This manual assumes that:
your AT-VP504E FXS has been properly set up by your
system administrator
If you need to install and configure the AT-VP504E FXS
yourself, please refer to the Administration Manual provided
with your AT-VP504E FXS or contact your system
administrator.
theIPCommunication Serverhasbeenproperlyinstalled
and set up1
If you need to install the IP Communication Server yourself,
please refer to the IP Communication Server Administration
Manual provided with your package.
Related Documentation
In addition to this Manual, each document set of the AT-VP504E FXS
includes the following:
IP Communication Server Manual1
This manual explains how to install and configure the IP
Communication Server, which is a set of software tools that
helps manage a network of APA communication units and
other SIP devices. It is intended for a network administrator.
The manual is not printed –it is located on the Installation CD
provided with your AT-VP504E FXS.
Administration Manual
This manual explains how to install and set up the various
AT-VP504E FXS parameters. It is intended for a network
administrator. The manual is not printed – it is located on the
Installation CD provided with your AT-VP504E FXS.
1. Valid only if you have purchased the IP Communication Server.

Chapter 1 - Using the AT-VP504E FXS User’s Manual (SIP/MGCP Version)
AT-VP504E FXS 2
AT-VP504E FXS Quick Start booklet
Thisprintedbookletallows you to quickly setup andworkwith
your AT-VP504E FXS.
IP Communication Server Quick Start booklet2
Thisprintedbookletallows you to quickly setup andworkwith
your IP Communication Server.
Acronyms
Overviewofthe
AT-VP504E
FXS
The AT-VP504E FXS is an IP Telephony adaptor that connects up to
four (4) analog terminals to a LAN with access to an IP Network,
allowing high-quality, full duplex, audio/fax communication.
This version of theAT-VP504E FXS canuse either one of the following
signalling protocol:
the Session Initiation Protocol (SIP), which is a simple
2. Valid only if you have purchased the IP Communication Server.
FXS Foreign Exchange Service
IP Internet Protocol
LAN Local Area Network
LED Light Emitting Diode
MAC Media Access Control
Mb/s Megabits per Second
MGCP Media Gateway Control Protocol
PBX Private Branch Exchange
PSTN Public Switched Telephone Network
SCN Switched Circuit Network
SIP Session Initiation Protocol
SNMP Simple Network Management Protocol
TCP/IP Transmission Control Protocol/Internet Protocol
TFTP Trivial File Transfer Protocol
VoIP Voice Over Internet Protocol
WAN Wide Area Network

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AT-VP504E FXS 3
signalling protocol for Internet conferencing and
telephony.
the Media Gateway Control Protocol (MGCP) version
draft-huitema-megaco-mgcp-v0r1-05. MGCP is a
protocol for controlling Voice over IP (VoIP) Gateways
from intelligent external call control elements.
Using the
AT-VP504E
FXS
Now that your administrator has properly set up the AT-VP504E FXS
and IP Communication Server configuration settings, you can dial any
number on your phone (which is connected to the AT-VP504E FXS)
and place the call.
However, you should be aware that the administrator has probably set
permissions and restrictions regarding local and long distance calls.
Should you encounter any calling problem, please discuss it with your
administrator to remedy the problem and grant the necessary
permissions.
Call Processes Thefollowingexamplesillustratesomeofthe variouscallingprocesses
theAT-VP504EFXSsupports.Theseprocessescanbeadaptedatwill
to suit your needs and requirements.
The AT-VP504E FXS can communicate with the following devices:
Another terminal on the IP network such as the AT-
VP504E FXS.
Any LAN Endpoint on the IP network such as:
• a Soft Phone
• an IP phone directly connected to the IP network
A SCN phone or fax. However, the AT-VP504E FXS
would need to contact an analog gateway such as the
AT-VP504E FXO.
For instance, the Switched Circuit Network (SCN) could be a
Public Switched Telephone Network (PSTN) or some Private
Branch eXchange (PBX).

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AT-VP504E FXS 4
Calls Involving
Another Termi-
nal
The following example illustrates how to reach a phone or fax on
another AT-VP504E FXS terminal.
XPhone/Fax -> AT-VP504E FXS A -> AT-VP504E FXS B -> Phone/Fax
A user makes a call with the phone/fax connected to an AT-VP504E
FXS, which in turn contacts another AT-VP504E FXS, then reaches
the corresponding phone/fax.
Calls Involving
a Terminal and
a LAN Endpoint
The following examples illustrate how a phone/fax connected to an
AT-VP504E FXS terminal can communicate with a LAN Endpoint on
the IP network.
XPhone/Fax -> AT-VP504E FXS -> LAN Endpoint
A user makes a call with the phone/fax connected to an AT-VP504E
FXS, which reaches the corresponding LAN Endpoint on the IP
network.
IP / LAN
Telephone Telephone
AT-VP504E FXS AT-VP504E FXS
Fax Fax
IP Communication Server
yIP phone
ySoft Phone
SIP User Agent
IP / LAN
Telephone Telephone
AT-VP504E FXS AT-VP504E FXS
Fax Fax
IP Communication Server
yIP phone
ySoft Phone
SIP User Agent

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AT-VP504E FXS 5
XLAN Endpoint -> AT-VP504E FXS -> Phone/Fax
A LAN Endpoint contacts the AT-VP504E FXS, which reaches the
corresponding phone/fax connected to the AT-VP504E FXS terminal.
Calls Involving
an Analog
Gateway
The following example illustrates how a telephone/fax connected to an
AT-VP504E FXS terminal and a SCN phone can communicate via an
analog gateway.
XPhone/Fax -> AT-VP504E FXS -> AT-VP504E FXO (Gateway) ->
SCN
A user makes a call with the phone/fax connected to an AT-VP504E
FXS, which in turn contacts an AT-VP504E FXO gateway, then
reaches the corresponding SCN Phone.
IP / LAN
Telephone Telephone
AT-VP504E FXS AT-VP504E FXS
Fax Fax
IP Communication Server
yIP phone
ySoft Phone
SIP User Agent

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AT-VP504E FXS 6
A SCN user can also contact the AT-VP504E FXO gateway, which in
turn contacts the AT-VP504E FXS, then reaches the corresponding
phone/fax.
Calls Without a SIP Server
If the AT-VP504E FXS isnot configured to register with a SIP Server3,
you can make SCN to IP network calls by dialing an IP address.
XIP Address Call
You can dial anothercommunication unit (gateway or terminal) without
the help of a SIP Server by entering its IP address. If you are dialing
the IP address of an AT-VP504E FXS (which has four ports), by
default, you will reach the telephone/fax connected to Port 1 of this 4
ports terminal.
3. Only valid for units that run the SIP signalling protocol.
IP / LAN
Telephone
AT-VP504E FXS
Fax
IP Communication Server
PSTN Phone
PSTN
4 telephone lines
connected to the PSTN
AT-VP504E FXO
(Gateway)
IP / LAN
Telephone Telephone
AT-VP504E FXS AT-VP504E FXS
Fax Fax
yIP phone
ySoft Phone
SIP User Agent
Note: This type of dialing can only be possible if the AT-VP504E FXS
is not configured to register with a SIP Server.

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AT-VP504E FXS 7
Making Calls Users that have telephones or faxes connected to an AT-VP504E FXS
will dial as if they were on a standard telephony system.
Complete Dial-
ing Sequence There are three ways to indicate the dialed number sequence is complete and the AT-VP504E FXS can dial the number:
your administrator has set up the dialing process so that
you must end your telephone number with a particular
character to indicate it is complete, e.g. a “#”.
your administrator has set up the dialing process with a
timer.Thistimerchecksthe dialing processand,whenno
further digits have been dialed for 4 seconds, it assumes
the number is complete and dials it.
your administrator has set up the AT-VP504E FXS so it
knows exactly how many digits it must collect before it
places the call. It finds the number of digits to collect by
looking at the first few numbers dialed. For example: a
telephone number beginning by 1 should be followed by
10 more digits in North America.
Consult your administrator to determine which dialing process is
defined.
Dialing a Tele-
phone Number
or a Numerical
Alias
This section assumes that the AT-VP504E FXS is configured to do
SCN emulation. The AT-VP504E FXS could be configured to do any
other kind of emulation, thus its users would simply have to dial as if
they were using their old system.
XTo dial a Standard Call:
1. Dial the telephone number as if you were using a standard
telephone, with country code and Area Code when required.
Examples:
829-8749
1-514-570-1234
A Standard Call uses the server to contact the remote dialed
user. The server takes the decision as to redirect the call on
the SCN or to keep it on the network. Keeping the call on the
network takes precedence over redirecting it on the SCN. If

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AT-VP504E FXS 8
the call needs to go on the SCN, the server will redirect it to
a proper analog gateway (such as the AT-VP504E FXO) that
will place the call to the SCN network.
XTo dial a Forced SCN call:
1. Dial “**”.
2. Dial the telephone number as if you were using a standard
telephone, with country code and Area Code when required.
Examples:
**829-8749
**1-514-570-1234
A Forced SCN Call allows you to specify that the user you
want to reach is located on the SCN network. This leaves no
decisions to the server; it must find a proper gateway and
place the call on the SCN. This option can be useful only
when a SCN number isshadowedby anetwork number. This
type of call should not be used often.
Dialing an IP
Address You can dial another AT-VP504E FXS without the help of the IP Communication Server by entering its IP address. Youwillonly be able
to reach the phone or fax connected to Port 1 of this AT-VP504E FXS
This procedure will only work if your AT-VP504E FXS is running the
SIP signalling protocol. See your network administrator for more
details.
XTo dial an IP address:
1. Dial “***”.
2. Dial the numerical digits of the IP address and use “*” for the
“.” of the IP address.
3. Dial “#” to terminate the IP address.
Forexample,to dial192.168.0.23,theuserdialsthefollowing
digits:
Note: You can dial one star numbers *xx (such as *69). These
numbers will be automatically inserted in the Request-URL of the SIP
INVITE request.
Note: A forced SCN call will only be possible if an analog gateway
such as the AT-VP504E FXO is available on the IP network.

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AT-VP504E FXS 9
***192*168*0*23#
Using the Call
Waiting Feature The call waiting4feature allows you to put a call on hold. For instance,
if you are already on the phone and a second call happens, a “beep”
will be heard and repeated every ten (10) seconds to indicate there is
a second incoming call.
The caller identity (the friendly name and phone number of the calling
user) is displayed on telephones/faxes properly equipped with a LCD
display.
XTo put the current call on hold:
1. Perform a Flash-Hook.
This will put the call on hold and the second line is
automatically connected to your line.
XTo switch from one line to the other:
1. Perform a Flash-Hook eachtime you want to switch between
lines.
XTo terminate the first call before answering the second call:
1. Hang up the phone.
2. Wait for the phone to ring.
3. Answer the phone.
The second call is on the line.
Using the Call
Transfer Feature The call transfer5features allows you to transfer a current call to any
other extension or phone number. There are two (2) types of call
transfer features available:
by consultation
unsupervised
4. Only valid for units that run the SIP signalling protocol.
Note: It is important to answer a call on hold, or your second call will
not have any answer.
5. Only valid for units that run the SIP signalling protocol.

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AT-VP504E FXS 10
Call Transfer –
Supervised The call transfer – supervised allows you to transfer a current call to any other extension or phone number. However, the individual at the
other extension or phone number must answer to complete the
transfer.
XTo transfer a current call supervised:
1. Perform a Flash-Hook.
This will put the call on hold.
2. Wait for the transfer tone (three “beeps”).
3. Dial the number to which you want to transfer the call.
The third party will answer.
4. Hang up your phone.
The call will be transferred.
5. If you want to get back to the first call (the call on hold), you
must perform two Flash Hooks.
You are back with the first call and the third party is released.
Call Transfer –
Unsupervised The call transfer – unsupervised allows you to transfer a current call to any other extension or phone number. The individual at the other
extension or phone number does not need to answer to complete the
transfer.
XTo transfer a current call unsupervised:
1. Perform a Flash-Hook.
This will put the call on hold.
2. Wait for the transfer tone (three “beeps”).
3. Dial the number to which you want to transfer the call.
4. Wait for the ringback tone, then hang up your phone.
The call will be transferred. You can also wait for the third
party to answer if you want. In this case, the call transfer
becomes supervised.
Note: If the number to which you want to transfer the call is busy or
does not answer, quickly depress and release the plunger in or the
actual handset-cradle twice. The busy tone or ring tone is cancelled
and you are back with the first call.

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AT-VP504E FXS 11
5. If you want to get back to the first call (the call on hold), you
must perform two Flash Hooks.
You are back with the first call and the third party is released.
Using the Call For-
ward Feature There are three types of Call Forward.
Call Forward
Unconditional The Call Forward – Unconditional6feature allows you to forward your
calls to another extension or line.
When forwarding your calls outside the system, a brief ring will be
heard on your phone to remind you that a call forward has been
established. You can still make calls from your phone.
XTo forward calls:
1. Take the receiver off-hook.
2. Wait for the dial tone.
3. Dial *25.
4. Wait for the transfer tone (three “beeps”) followed by the dial
tone.
5. Dial the number to which you want to forward your calls.
Dial any access code if required.
6. Wait for three “beeps” followed by a silent pause.
The call forward is established.
7. Hang up your phone.
The calls are checked against the dial maps set up by your
system administrator. See your system administrator for
more information.
6. Only valid for units that run the SIP signalling protocol.
Note: The “*25” sequence can be user-configured. Check with your
system administrator for the exact sequence actually implemented.

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AT-VP504E FXS 12
XTo check if the call forward has been properly established:
1. Take the receiver off-hook.
2. Wait for the dial tone.
3. Dial your extension or phone number.
The call is forwarded to the desired phone number.
4. Hang up your phone.
XTo cancel the call forward:
1. Take the receiver off-hook.
2. Wait for the dial tone.
3. Dial *26.
4. Wait for the transfer tone (three “beeps”) followed by the dial
tone.
The call forward is cancelled.
5. Hang up your phone.
Call Forward
on Busy / No
Answer
You can automatically forward incoming calls7to a pre-determined
extension within your system if you do not answer before a specific
number of rings or if you are already on the line.
Note: The “*26” sequence can be user-configured. Check with your
system administrator for the exact sequence actually implemented.
7. Only valid for units that run the SIP signalling protocol.
Note: The call forward on busy / noanswer can only be set up by your
system administrator. See your system administrator for more
information.

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AT-VP504E FXS 13
Conferencing
Calls A conference call8between three parties can be set up.
Requirements For the conference call to occur succesfully, all parties must meet the
following requirements:
Support at least one of the PCM codecs (G.711 µ-law
and G.711 A-law) enabled on the port that is having the
conference.
Ability to dynamically change codec during a call.
Managing a
Conference
Call
If you are on the phone with one person and want to conference with a
third one, you can do so. In the following examples, let’s assume that:
A is the conference initiator.
B is the person called on the first line.
C is the person called on the second line.
XTo initiate a three-way conference (A and B already connected):
1. A performs a Flash-Hook.
This will put B on hold and the second line is automatically
connected. A hears a dial tone.
2. A dials C’s number.
A and C are now connected.
3. A performs another Flash-Hook.
Thecall on hold (B)is reactivated. A is now conferencing with
B and C.
XA wants to transfer B to C during the conference:
1. A hangs up.
The conference is terminated. B and C are now connected.
8. Only valid for units that run the SIP signalling protocol.

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AT-VP504E FXS 14
XA wants to terminate the call with C and get back to the call with B
during the conference:
1. A performs a Flash-Hook.
The conference is terminated and the call with C is
disconnected. A and B are still connected and can go on with
their conversation.
XB (or C) hangs up during the conference:
1. B (or C) hangs up during the conference.
The conference is terminated, but the call between A and C
(or B) is not affected and they are still connected.

AT-VP504E FXS 15
AGlossary
Area Code The preliminary digits that a user must dial to be connected to a
particular outgoingtrunk group or line. In NorthAmerica, an Area Code
has three (3) digits and is used with a NXX (office code) number. For
example, in the North American telephone number 561-955-1212, the
numbers are defined as follows:
Outside North America, the Area Code may have any number of digits,
depending on the national telecommunication regulation of the
country. In France, for instance, the numbering terminology is defined
as xZABPQ 12 34, where:
In this context, the Area Code corresponds to the Zportion of the
numbering plan. Since virtually every country has a different dialing
plan nomenclature, it is recommended to identify the equivalent of an
Area Code for the location of your device.
Table 1: North American Numbering Plan
No. Description
561 Area Code, corresponding to a geographical zone in a non-LNP
(Local Number Portability) network.
955 NXX (office code), which corresponds to a specific area such as
a city region.
1212 Unique number to reach a specific destination.
Table 2: France Numbering Plan
No. Description
x Operator forwarding the call. This prefix can be made of 4 digits.
Z (regional) geographical zone of the number (in France, there are
5 zones). It has two (2) digits.
ABPQ First 4 digits corresponding to a local zone defined by central
offices.
12 34 Unique number to reach a specific destination.

Appendix A - Glossary User’s Manual (SIP/MGCP Version)
AT-VP504E FXS 16
CC Acronym for Country Code.
1. In international direct telephone dialing, a code that consists
of 1-, 2-, or 3-digit numbers in which the first digit designates
the region and succeeding digits, if any, designate the
country.
2. In international record carrier transmissions, a code
consisting of 2- or 3-letter abbreviations of the country
names, or 2- or 3-digit numbers that represent the country
names, that follow the geographical place names.
Dual-Tone Multi-
Frequency (DTMF) In telephone systems, multi-frequency signaling in which a standard set combinations of two specific voice band frequencies, one from a
group of fourlow frequencies and the other from a group of four higher
frequencies, are used. Although some military telephones have 16
keys, telephones using DTMF usually have 12 keys. Each key
corresponds to a different pair of frequencies. Each pair of frequencies
corresponds to one of the ten decimal digits, or to the symbol “#” or “*”,
the “*” being reserved for special purposes.
Dynamic Host
Configuration
Protocol (DHCP)
TCP/IP protocol that enables PCs and workstations to get temporary
or permanent IP addresses (out of a pool) from centrally-administered
servers.
Flash-Hook Quickly depressing and releasing the plunger in or the actual handset-
cradle to create a signal to a PBX or Centrex that special instructions
will follow such as transferring the call to another extension.
FXS Line Foreign Exchange Service/Station. A network-provided service in
which a telephone in a given local exchange area is connected, via a
private line, to a central office in another, i.e., “foreign”, exchange,
rather than the local exchange area’s central office. A FXS line is
normally connected to a standard telephone, fax or modem.
Gateway A device that links two different types of networks that use different
protocols (for example, between the packet network and the Public
Switched Telephone Network).
IP Acronym for Internet Protocol. The IP protocol is a standard describing
software that keeps track of the Internet’s addresses for different
nodes, routes outgoing messages, and recognises incoming
messages.