Asus ITX-220: 12 SIP IP-PBX

12 SIP IP-PBX: Asus ITX-220

Chapter 12

iPBX30 User Manual

12 SIP IP-PBX

The iPBX30 integrates the functionalities of SIP registrar server,

proxy server and voice media application, supporting up to 30 SIP

clients with all necessary call functions together with voice mail

capability.

The iPBX30 can work with any RFC3261 compliant gateway, IP

phone, ATA. iPBX30 can connect to legacy PBX by the FXS/FXO

ports of gateway, and is able to handle up to10 concurrent calls.

The following diagram shows a typical iPBX30 application scenario

in ofce.

Figure 12.1 iPBX30 application ofce scenario

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12.1 Conguration

A basic IP-PBX system should have three major components

including IP-PBX server, user clients and gateway working together

to provide necessary PBX functions. A user client can be an

embedded hardware device such as ATA, IP phone or software IP

phone running on PC, PDA. The following table shows their roles

and conguration parameters required.

Table 12.1 Conguration Parameters

IP_PBX server User clients Gateway

Function 1) Accept registration

1) Register to server

1) Register to

from user clients

server

2) Make or terminate

2) Resolve IP address

call via server

2) Accept call

of destination client for

from server to

3) Voice codec and

call invitation.

trunk port (FXO,

echo handling

FXS, or digital

3 ) P r o v i d e m e d i a

4 ) C a l l f u n c t i o n

t r u n k T 1 /E 1 /

service such as voice

handling

DSDN)

mail, IVR, DISA, etc.

5) 3-way conferencing

3) Forward call

from trunk to

server

4) Voice codec

a n d e c h o

handling

P a r a m -

1) WAN port IP

1) WAN mode/IP

1) Serve r IP,

e t e r s

signal/RTP port

2 ) G a t e w a y I P ( i f

2) Server IP

required

installed)

2) Dialing plan

3) Signaling/RTP port

for routing prex

3) Extension number,

4) User name, phone

code remove/

ID, password table

number, ID, password

add

4) Dialing plan for call

5 ) C o d e c a n d

3) Codec and

routing

parameters

parameters

5 ) P B X r e l a t e d

4 ) S i g n a l i n g

functions setting

protocol

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12.1.1 General Setting

Figure 12.2 General setting page

In the web-based conguration software, the following items can be

congured.

External IP address (Not congurable)

This IP address is the same with the WAN IP address. It shows

the same IP if the iPBX30 WAN port. The SIP server uses this

IP to distinguish the incoming VoIP call location by checking if it

is from external WAN or local LAN.

SIP codec type

Choose one codec type for SIP server from G.711u/G.711a/

G.729A. The iPBX30 SIP server uses the selected codec to

negotiate with the SIP client requesting for registration. The

iPBX30 will request every client to use the same codec for

compatibility.

Local subnet

This subnet value defines the SIP server LAN segment.

For example, if you have assigned the LAN IP segment to

192.168.10.x, the value for this eld is 192.168.10.0.

Subnet mask

SIP server uses this subnet mask to judge if the clients are

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registered in LAN environment. It can be a C-class or B-Class

mask.

Max. registration expire time

This value denes the maximum allowable expiry time for client

registration. The SIP client notifies the server its registration

expiry time when registration is in progress.

Default registration expire time

The server uses this expiry time as default value if any client

registered without expire time value attached.

Start RTP port/ End RTP port

The user can assign the starting RTP port number and End

RTP port number for VoIP service to dene the iPBX30 VoIP

RTP port usage range.

SIP port

The 5060 port is commonly used for SIP call signaling. The user

can change it if necessary.

DTMF mode

Users can select one of the three available DTMF transmission

methods: Inband, RFC2833, and Info.

Note: We recommend Infomode for normal use.

SIP client and server side should use the same

DTMF mode.

Log Level

This option allows the user to determine how detailed the log

message would be in the log le.

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12.1.2 Gateway

The user can add an SIP gateway node for the iPBX30 IP-PBX

server to provide inbound/outbound call capabilities from/to PSTN

or PBX system.

Assign the IP-PBX server IP to the SIP gateway, and the gateway

can forward PSTN incoming calls to the IP-PBX. Assign the

gateway IP address to the IP-PBX, and the outgoing calls can be

forwarded to the gateway.

Figure 12.3 Gateway page

Seq. number (Not congurable)

This field is not configurable and for sequence identification

purpose only. You may have more than one gateway in an IP-

PBX system for different call routing with pre-defined dialing

plans.

Name

This eld is for management purpose only.

IP address

Define the proper IP address assigned for this gateway that

iPBX30 can access to. The user can locate this SIP gateway in

either LAN or WAN environment.

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Port

Denes the SIP port assigned for this gateway to communicate

with.Use default 5060 port for SIP signaling. You can change

SIP signaling port value, if necessary.

Note: The server and gateway should use the same

port number.

Type

This field allows the user to configure the service type that

the gateway provides. Select the gateway to be connected to

PSTN (Trunk) or PBX (Line) extension line, or both in a single

gateway.

Location

This eld allows you to set the location for the gateway.

12.1.3 Extensions

You can add SIP user accounts in this page. The SIP extension

accounts must be added before allowing extension client devices

for registration. You may create up to 30 extensions. Some

extensions can be used for registering to ITSP or other SIP servers.

Click

Extension

item to see the current extension list, and Click

Add

to create a new extension account for clients.

Figure 12.4 Extension page

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Extension

Assign an extension number for an SIP client to register.

Extension number, the password and user ID are for

authentication requirement when registering to SIP server.

Caller ID

Assign a caller ID, either in numeric or text characters, which

will be sent to the called party when you are making a call.

Type

Select the service type for the extension. An extension can be

selected as a standard SIP extension, auto attendant virtual

extension, or ITSP registration account. To work with ITSP

service, an extension must be created for both forwarding

outbound call and accepting inbound call. The user has to do

the other conguration for ITSP operation, such as IP address,

port number assignment, ID and password setting. Refer to

“Dialing plan” section for more details.

DTMF mode

Select the DTMF mode for this extension to work with the

iPBX30 server. The available selections are: Inband, RFC2833

and info.

Note: We recommend using info mode for most

cases.

The SIP client and the server should select the same DTMF

mode to work properly.

We do not recommend using Inband' mode for

iPBX30, although it is supported.

If the SIP client enabled ‘Inband’ mode, it will send out DTMF

tone to server and server has to process the tone and decide

which DTMF code has received. Processing the DTMF tone

needs very complex calculation. This makes iPBX30 CPU very

busy and it will be incapable of processing many channels at

the same time.

Call Privilege

For different extension users, you may want to control their

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outbound call authorization level, in other words, the rights to

call local city call, long distance call, international call or only

ofce call.

Note: Check the item Recording to assign this

extension into greeting voice announcement

recording function. The user can record voice

message into the extension.

Server RTP relay

This function allows the user to choose if the IP-PBX server

should relay the RTP packets from this extension.

This function is useful when the extension client is located

behind some special NAT device and unable to make a VoIP

call successfully. The RTP packet relay will increase the

loading of the server, but this is a good solution to penetrate

NAT devices for VoIP client.

Call group

This allows the user to assign a group number for identication

when this extension makes a call to other parties. This will

enable another extension to know the group number when it

receives a call.

Pickup group

Pickup grouping allows you to group the different extensions

with the same working attribute into the same pickup group, so

the group members can answer their colleague’s phone call if

they were temporary unavailable.

NAT

You have to check this item with yes when the extension is

located behind an NAT device, and uncheck this item if the

extension is located at LAN environment.

Availability checking intervals (ms)

Assign the time duration for IP-PBX server sending out an

availability check request to the extension device. Server will

change the extension available status to “Unreachable” if the

extension has no response to this checking request, and mark

the client status as on line if the device response in time.

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Authentication

You can select the authentication algorithm for this extension

when the extension device sends registration to server.

Allowable algorithm is MD5, MD5-sess or none.

Password

Assign the authentication password here if you choose MD5,

MD5-sess algorithm for extension registration.

Email address

When this extension has a voicemail, the server will send a

email notication to this email address.

MAC

You can give the MAC address of the SIP client device (AX-112)

for this extension number. It is for auto-provision function which

AX-112 downloads the conguration le from iPBX30. iPBX30

will generate a conguration data le for each extension with

the MAC address dened here.

12.2 Dialing Plan

12.2.1 General

Figure 12.5 General Dialing Plan page

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Operator access number

Assign the code for accessing operator, for example “0” or “9”,

so the extensions can dial this access code to call the operator.

Operator type

Assign the operator type as a single extension or a group of

extensions. If the operator type is “Group”, all the extensions

defined in operator group rings when the operator code is

dialed.

Operator extension

Assign a single extension number here as the operator.

Operator extension group

Select all the operator extension numbers if operator type is set

as “Extension group”.

External prex digits

Dene the outbound call (call to PSTN) access prex code. It

is associated with “localprivilege in Extensions conguration

page. Do duplicate with other access code that is already used.

IDD prex

Dene the “International Direct Dial” prex code for call privilege

control checking. It associated with “International” privilege in

Extensions configuration page. When an extension is limited

and not able to make international long distance call, the server

will check the dialing number from extension with this IDD

prex, and the call will be denied if the prex matched.

DDD prex

Similar with above, this eld allows user to dene the “Domestic

Direct Dial prefix code for call privilege control checking.

It is associated with National Long Distance privilege in

Extensions conguration page.

External trunk gateway IP

Enter the gateway IP for IDD, DDD and outbound call access.

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External trunk gateway port

Assign the port for the gateway.

12.2.2 ITSP Server

The iPBX30 allows you to link with Internet Telephony Service

Provider (ITSP) providing SIP service. You must have a valid ITSP

user account and password for iPBX30. To link the iPBX30 to ITSP

account, register the iPBX30 to ITSP and assign an extension

number with Auto attendant or ITSP Operator type as the ITSP

inbound call operator.

Figure 12.6 ITSP Server page

Seq. no: (Not congurable)

The iPBX30 allows you to dene more than one ITSP service

account, and this eld is used for sequence identication.

ITSP Address

Enter the ITSP server IP address or domain name here.

ITSP Port

Enter the ITSP server port number here. Port 5060 is usually

for SIP.

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ITSP Operator

Select one extension number as the ITSP inbound call reception

operator. The extension must be predened in the “Extension”

page. All incoming calls from ITSP SIP will be forwarded to this

extension.

User Name/Password

The ITSP will give you an account name and password for

device registration authentication. Enter the account name

here, followed by password and authentication method.

Authentication

The iPBX30 supports MD5 authentication method while

registering to ITSP. Normally the SIP server registration is

protected for preventing unauthorized user login.

12.2.3 PrexRouting

“Prex routing” enables user to dene a prex code mapping for

routing calls to a specied destination.

The destination could be a gateway or ITSP service server. Prex

routing must work with “Gateway” or “ITSP” setting.

Figure 12.7 Prex Routing page

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Prex

Enter the prex code here to be matched with each user call

attempt. If the digits from a call fail to be matched with any

prex digits dened in prex routing, it will be treated as a local

extension call.

Action

Select the action of a call if the prex digits were matched. You

can select to forward the call or block the call.

Destination type

Select the destination type, either a gateway or ITSP service.

Destination protocol

The iPBX30 supports SIP protocol .

Destination

Select the available destination gateway or ITSP from the pull

down menu. The available gateways must be predened in

Gateway conguration page.

Digits to remove

The user can dene the length of digits to be removed from a

call before forwarding to a destination gateway or ITSP server.

Prex code removing is necessary because the ITSP will not

recognize these codes and may cause call failure.

Digits to prex

The user can dene the length of digits to be added to a call

before forwarding to a destination gateway or ITSP server.

Prex code removing is only necessary when ITSP or gateway

devices needed to parse your dialing rule.

12.3 Status

12.3.1 Extensions status

You can check all the extension client registration status in this

page. This page automatically refreshes every 30 seconds. You

can make call to the extension if it is indicated in “OK” status. .

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You can also check the extension’s IP address, port number and

NAT setting here

Figure 12.8 Extensions Status page

12.3.2 Channel status

You can check the extensions call status in this page, and this page

is empty when there is no any extension making calls or ITSP/

gateway activities in progressing.

12.3.3 ITSP registry

If you have ITSP SIP account and it is properly setup, you can

check the ITSP registration status here. The iPBX30 keeps trying

to register to ITSP account until it is successful. The iPBX30 allows

you to register to multiple ITSP accounts at the same time.

Figure 12.9 ITSP Status page

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12.4 Features

12.4.1 Voicemail

The iPBX30 supports voice mail feature, the caller party can leave

a message to the called party when the call is not answered, and

IPBX30 can send the user an e-mail notification when the voice

message recording is done.

To enable mail notification function, the iPBX30 needs a mail

sender account to send mail. You can set up the mail account and

tag message here.

The iPBX30 voice mail message data are kept in on-

board ash memory.

Voicemail is a centralized system of managing telephone mes-

sages for a large group of people. In its simplest form it mimics the

functions of an answering machine. Voicemail systems are much

more sophisticated than answering machines in that they can:

Answer many phones at the same time

Store incoming voice messages in personalized mailboxes

associated with the user’s phone number

Enable users to forward received messages to another voice

mailbox

Store voice messages for future delivery

Send email to notify the uset a message has arrived in the

mailbox

Transfer callers to another phone number for personal

assistance

Voicemail messages are stored on hard disk drives or on board

flash memory, media generally used by computers to store other

forms of data. Messages are recorded in digitized natural human

voice similar to how music is stored on a CD. You can call the sys-

tem from any phone, logs on using DTMF codes (clearing security)

to retrieve messages. Multiple users can retrieve or store messages

at the same time on the same voicemail system.

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There is 8Mbytes space for voice message as default and the

recording time depends on the voice codec you selected. G.711

(uLaw, aLaw) is 8K bytes per second, and G.729 is 1K bytes per

second. If you have attached the USB device onto iPBX30 USB

port and setup the USB storage device for CDR and voice mail,

then the voice message recording time is limited by the USB stor-

age size.

Figure 12.10 Voice mail page

Access Number

Defines the voicemail box function access number for

extensions to dial. When the SIP extension receives the

notification email, dial this number to enter the voicemail

system to listen to the voice message.

System E-Mail

The iPBX30 email address to send out mail.

Sender

The data in this eld is shown on the email “Sender” eld.

SMTP Servers/SMTP Port

The iPBX30 uses this SMTP Server to send the email

notication out.

SMTP Authentication Type

The iPBX30 supports two authentication types to mail server:

LOGIN and PLAIN

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SMTP Authentication User/Password

The user name and password for SMTP server’s authentica-

tion.

In the above conguration example, follow the instructions below to

access your voice message:

1. Dial “8500” from your IP phone to enter iPBX30 voicemail box

function main menu.

2. Enter the extension number and password to access your

message after entering voicemail box.

3. The password is identical to the “registration password” which

you set up in the Extension page. Refer to section 12.1.3

Extension.

4. Follow the voice prompt for more operation.

Voice mail function main menu owchart:

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Press 1 to enter play back message menu,and refer to the

following owchart.

To enter message option menu, press “3in the message option

menu after message played. You can select to reply message to

caller or repeat the message again.

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To enter message forward menu, press “8” in the message option

menu after message played. You can select to forward message

directly to another extension or record a tag message and forward.

12.4.2 Auto-provision (for AX-112)

The iPBX30 supports auto provisioning function for ASUS SIP ATA

whose model name is AX-112. Since there is no standard algorithm

for provisioning, so it’s nature that iPBX30 only supports the device

now. Autoprovision function allows system maintainer to dene the

configuration data for SIP client devices on iPBX30 server GUI.

This function can minimize the deployment efforts of ATA, IP phone

and increase the consistence and flexibility when replacing client

devices.

The iPBX30 must have a copy of conguration data for client device

so the client can download the conguration data when provisioning

function is activated at client side. For this reason, the

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provision conguration of iPBX30 is basically the same with the GUI

of AX-112.

The conguration data here will be saved into a le together with

the MAC address dened in ‘Extensions’ conguration page.

Figure 12.11 Auto Provision page

12.5 Report

CDR (Call Detail Record) Report

You can check the call log in this page which includes

information of caller party, called party and call duration. But

these call log are default recorded in SDRAM memory and will

be lost when the system powers off. The log can be recorded to

an external USB storage device if user has attached the USB

device onto iPBX30 USB port. If an USB storage device has

been congured for CDR and voice mail, then the records data

will not be lost when iPBX30 is powered off.

12.6 Utilities

12.6.1 Hot reload

Every time you make changes on the IP-PBX settings, you have

to tell the IP-PBX server to “reload” the new conguration and

activate. Click the confirm button and the reload process will

begin, taking about 10 seconds to load the new conguration.

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12.6.2 Service restart

If you need to restart the IP-PBX server software, click this item

and conrm. The iPBX30 will kill the old IP-PBX program task

to restart it again. Your request for IP-PBX service restart won't

affect the NAT function of iPBX30.

12.7 ConguringExamples

The diagram below shows a typical iPBX application scenario, and

the iPBX30 plays the role of router and SIP server at the same

time. Following sections describe how to setup 2 ATA in LAN, 1

ATA in internet and 1 SIP gateway in LAN. We have to give some

assumption for these scenarios:

• iPBX30 WAN public IP: 210.80.66.110

DHCP server is enabled for iPBX30 LAN, and LAN IP segment

is 192.192.1.x

• Two AX-112 ATA in LAN with extension number 1001/1002, and

the AX-112 WAN is in DHCP client mode, this means AX-112

will get IP from iPBX30 built-in DHCP server.

One AX-112 ATA in Internet with extension number 1003, and

its WAN IP is 192.168.10.10 which is obtained from the home

router.

One SIP gateway connected to PSTN lines with LAN

IP:192.192.1.10.

Figure 12.12 Typical iPBX30 application set up

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Create extensions

You have to create extension accounts in iPBX30 for the registration

from three SIP ATA devices.

Click

IP-PBX -> Conguration -> Extensions

to open the page as

shown below.

Figure 12.13 Extension List page

Click the

Add

button to open the extension conguration page.

Figure 12.14 Extension page

We recommend you set up parameters for extension 1001 as

following:

1. Set “Extension”, “Caller ID”, “Password” to “1001”.

2. Check “Type” as “SIP”, “DTMF mode” as “Info”

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3. Check server RTP relay as “No”, “NAT:” as “No”

4. Click “Add” to complete this conguration.

5. Follow the instructions above to set up extension “1002”.

6. For AX-112 user under NAT device over internet (extension

1003), all settings are the same except that you should check

the “NAT:” eld as “Yes”.

7. After all setting are completed, click “Utility” item from the left

side menu , and click “Hot reload” to make iPBX30 reload all

the settings and take effective.

ConguretheSIPclientdevices

After you have created these AX-112 accounts, you have to setup

proper parameters for each AX-112 account.

Figure 12.15 SIP page

To congure the SIP client devices:

1. Connect AX-112 to iPBX30 using RJ45 cable, connect an

analog phone set to AX-112 using RJ11 cable, and power on

AX-112.

2. Pick up the phone and dial “****” to hear AX-112 IVR (Interactive

Voice Response) menu.

3. Dial “100#” and AX-112 reports you the device status. Listen

carefully for WAN IP reporting and open a browser with this IP.

4. Enter AX-112 GUI and click “SIP” on the above menu. Enter

iPBX30 LAN IP (192.168.1.1 as assumed) in “*IP” eld. Enter

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the Phone number, Caller ID, User Name and Password, and

make sure they are identical to the settings in iPBX30. Click

Apply

after completing all settings.

5. Click “Advanced” on the menu to congure advanced setting:

a. Choose “Silence Suppression” as “Off” for G.729 and G.711.

b. Choose “INFO” mode for “DTMF method”

c. Click “Apply” to update setting and then reboot AX-112.

6. Now your AX-112 with extension number 1001 can login

iPBX30 and make a call.

7. Follow the above procedures to congure extension 1002

8. For AX-112 under NAT device over Internet, only the “*IP:” (SIP

server IP address) setting is different. Assign the public IP of

iPBX30 to this eld (it is 210.80.66.110 in this example).

Enable the ITSP service

There are 3 steps to enable ITSP service.

Create an extension for ITSP

It is necessary for iPBX30 to use an extension as an UA (User

Agent) to register to ITSP SIP server, and also to accept the

incoming call from ITSP. Select an extension number for ITSP

registration and click the type as “ITSP operator”.

Set up ITSP account

Enter the ITSP server public IP address or domain name, and

the proper user name and password for authentication. You can

have multiple gateways or ITSP service accounts at the same

time.

Add a routing rule for ITSP service

You have to create a routing rule for ITSP call, just like the

gateway prex routing setup.

After you have nished all the setup, go to

IP-PBX -> Utilities

and click the “Hot Reloadto make all the settings effective.

Go to

IP-PBX -> Status

to check if the ITSP registration is

successful, and make a call with proper prex number to check

if the call can be routed to gateway or ITSP server accordingly.

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