SIPtrix™

User Manual

v1

 

 

Introduction

 

 

This document is intended to provide information for the administrator about the SIPtrix™ IP telephone system: the features available, the network and environmental prerequisites, the default settings the system is installed with, and how those settings can be changed.

 

Due to the nature of the SIP and IAX standards, the interoperability of the SIPtrix™ platform and the plethora of IP equipment manufacturers available, it is not possible to write detailed configuration instructions for all handsets, ATAs, gateways or other VoIP equipment. However guides for a number of devices are included in the Appendices.

Neither is it within the scope of this document to provide information on the protocols and standards employed by SIP. It is, however, recommended that you read the wealth of information available on the Internet if you are new to the terms and concepts employed by VoIP technology.

 

 

Features

 

The SIPtrix™ box is a modular IP PBX with additional complimentary products and services. It can be provided in both fixed and mobile office scenarios and is based on the Asterisk platform. Asterisk is an open source IP PBX application, enriched through global development forums and therefore high on the priority list of all associated hardware manufacturers when it comes to their product development.

 

It is prerequisite that it runs on Linux-based machines, from a small PC up to an enterprise class of server. The limitation on the number of extensions and trunks is dependent on the processing power and memory of the host machine. The Asterisk management tool, “AMP”, has been tweaked with its features enriched and added to the appropriate machine to create the SIPtrix™ box.

 

IP telephony is not solely concerned with free calls between offices, but also offers wireless capability and remote configuration, especially in the mobile office environment, when set up in conjunction with the likes of a 3G router and a fixed IP SIM.

 

Features of SIPtrix™:

 

 

• Unlimited extensions
• Unlimited Voice Mail boxes
• Unlimited IVR (Integrated Voice Response). “Press 1 for sales…”
• Recordable IVR prompts
• Automatic Call Distribution
• Configurable Call Routing
• Least Cost Routing
• Multiple DID numbers per VoIP trunk
• Call routing to multiple extensions or targeted user
• Call hunting – sequential, round robin, random…
• Unified Messaging – this is voicemail delivered to different mediums, including email and a web-management portal
• Queuing – this includes position announcement, static agents, queue “sign ins”…
• Call waiting
• Speed Dialling
• Three Way Calling
• Conference Rooms
• Message Waiting Indicator
• Operator Console
• Time-based conditions
• Automated Attendant
• Phone Provisioning
• Configuration of IP phones
• Call Recording – this can be extension specific
• Call Forwarding
• Do Not Disturb
• Call Reporting
• Automatic hardware provisioning, eg with the Polycom phones. This is an example of a manufacturer building products with Asterisk in mind
• Voice prompting – developments here offer the user more choice on prompt types, eg language variations
• Date and time and Daylight Savings support
• Call duration with call time stamp stored in call logs
• Record incoming and/or outgoing calls on demand
• Syslog, Debug, Report Generation and Event Logging
• Built in web server for admin and configuration with multiple security levels
• Built in SSH server
• Remote management capability

·         Long line extension, (LAN, WAN, PAN, MAN)

 

 

 

The SIPtrix™ solution can be used in both the fixed or mobile environment. The hardware that has been chosen for each is appropriate for its surroundings.

Hardware

 

Fixed Office solution

 

For fixed office environments, standard Dell PCs/Servers are chosen. The same machine could be used for a 10 or 100 extension system. An example of such a machine is detailed below:

 

Dell PowerEdge SC1430 Server:

 

·         One Intel® Xeon™ processor 5050 at 3.0GHz with 2MB L2 cache

·         1GB FB 533MHz Memory

·         80GB 7200rpm SATA hard drive

·         2 x 512MB single rank DIMMs

·         CD-ROM 48x

 

If a rack-mount kit is required, then the SC1435 SATA would be the option:

 

Dell PowerEdge SC1435 SATA:

 

·         Opteron 2210 processor at 1.8GHz/2M 95W

·         1GB DDR2 667MHz SDRAM Memory (2 x 512MB single rank DIMMs)

·         80GB SATA 3.5inch hard drive

·         CD-ROM 24x IDE

·         4-post rack mounting kit

 

 

Mobile Office solution

 

Twister box (ruggedised mobile office solution)

 

For mobile office environments, a more ruggedised smaller device has been chosen. This is known as a Twister box and is ideal for temporary or mobile offices where the device is to be used. The purpose of the Twister case is to cool off the processor without using the fan because of the possibility of failure of the fan that will damage the processor and the system.

 

Specifications of the Twister:

 

 

·         1GHz Nehemiah Fanless

·         LEX CV863A Motherboard

·         Dimensions 300W x 175D x 75H mm

·         1 x PC2100 266MHz DDR slot

·         4 x 10/100 Realtek 8139 LAN Port

·         4 x rear USB ports

·         1 COM port

·         Onboard Compact Flash slot

·         2 x PCMCIA slots

·         1 PCI slot

·         Onboard DC-DC converter powered by an external AC adapter

 

 

 

In addition to the above hardware options, together with the configured Asterisk platform inside, Hugh Symons Telecom can supply the following in order to complete an IP PBX solution in or out of the office:

 

 

·         VoIP services

·         IP handsets

·         Installation

·         FCTs

·         Fixed IP SIMs

·         Pre-Site Surveys

·         Broadband and connectivity

·         Gateways

·         Support and maintenance

·         Configuration

·         Airtime

·         3G Routers

·         Telephony Services

·         SIPtrix™ hardware

 

 

Partners include:

 

 

·         Boscom

·         Digium

·         Nokia

·         Polycom

·         Dell

·         Ericsson

·         Option

·         Wyless

 

 

Prerequisites

 

When implementing a VoIP telephone system, planning is key.

 

SIPtrix™ is an IP-based PBX, this means that it operates across a TCP/IP network. Whereas data applications can tolerate a certain amount of latency and jitter, voice communications require an uninterrupted quality of service and real time guaranteed reliable delivery of information. It is therefore crucial that the underlying TCP/IP network be set up properly.

 

Before the SIPtrix™ PBX can be installed, it is vital that the site be surveyed and the networking infrastructure examined.

 

The phone system can either run on a dedicated network, or can share the same data network as PCs.

 

If a dedicated network is going to be used, is a switch present for the phones? Does it provide Power over Ethernet (PoE)? If not, are there enough power outlets for the phones? Are there sufficient network points for the phones in the desired locations?

Will users require that voicemail messages be sent to them as email attachments? If so, will there be access from the dedicated network to the email server? Does the dedicated network have Internet access?

 

If the VoIP system is going to share the data network, how many hubs and/or switches are used and what speed are they? Is the network subnetted? Is DHCP going to be used? If so, will an address range be reserved for the phones and a fixed address be assigned to the SIPtrix™ server?

 

These are all questions that need to be asked to ensure that sufficient bandwidth is available and that voice communications will not suffer due to the data activity on the network.

 

What type of phones will be used with the system?

 

How will calls be delivered to the SIPtrix™ system? Analogue BT lines? ISDN? SIP trunk? IAX trunk? How will calls be placed by the server? Will Fixed Cellular Terminals be used?

 

This is the information you need to be armed with before attempting to install SIPtrix™.

 

See the Hugh Symons Pre-Installation Site Survey document for more information.

 

Changing default settings

 

The default IP address settings of the SIPtrix™ box are as follows:

 

DHCP disabled

 

IP Address:       10.0.0.100

Subnet Mask:    255.255.255.0

 

Should these settings need to be changed, log into the SIPtrix™ server using the following credentials:

 

Username:        root

Password:        pa55word!

 

You will then be logged into the SIPtrix™ server:

 

 

Type in Netconfig and press enter, the following screen will be displayed:

 

 

Press Yes. The following screen will be displayed:

 

 

Enter the desired IP address settings.

 

Once complete, press OK.

 

Now restart the server.

 

 

 

 

The SIPtrix™ web interface

 

The SIPtrix™ box contains a built in web server providing access to the voicemail and recordings centre, the conference control tool, the administration tools as well as the flash operator panel.

 

The default IP address for the SIPtrix™ box is http://10.0.0.100 (subnet mask 255.255.255.0, DHCP disabled). The web interface appears as shown below:

 

 

 

SIPtrix™ Administration

 

The SIPtrix™ administration web portal allows you to configure all aspects of the system’s functionality, monitor the status of the system and also generate reports. Click on the link to access the portal. You will be prompted to log in, the default credentials are:

 

Username:        maint

Password:        password

 

Once logged in, the interface appears as shown below:

 

 

Setup

 

The Setup section of the administration portal allows you to configure the operation of the telephone system. Areas include:

 

• Module Admin
• Time Conditions
• System Recordings
• Conferences
• Extensions
• Inbound Routes
• Outbound Routes
• Trunks
• General Settings
• Ring groups
• Digital receptionist
• Queues

 

Module Admin

 

The Module Admin section simply allows you to add or remove modules to the setup interface. It is not advised that you alter the default configuration.

 

Extensions

 

 

SIPtrix™ supports 4 types of extension:

 

• SIP
• IAX
• ZAP
• Custom

 

There is no default numbering scheme for extensions, so you can base your numbering plan:

 

• On convention: 200, 201, 202, etc
• To match the last 4 digits of a DDI range
• To match those on a previous system

 

However it is important to note that some number ranges are already reserved for used by the system:

 

• 1-99      Reserved for ring groups

• 3xx       Reserved for speed dials
• 7xx       Reserved for call parking
• 8xxx     Reserved for conferences

 

To add an extension, select the type of extension you wish to create:

 

 

Hovering the mouse over each field description will display some help on that item:

 

 

Enter a unique number for the extension. Enter the Display Name how you want that extension to appear to others. Enter a Secret of sippassword.

Now press Submit to create the extension.

NOTE – you will need to click on the red bar at the top of screen to apply any changes made.

If you wish to enable Voicemail for that extension, set Voicemail & Directory to Enabled. Further options will be displayed:

 

 

Enter a password of 1111.

If you wish voicemail messages to be emailed as attachments to the user, enter their full email address and select the option to email attachment.

 

Once you have created the extension, if you click on the entry for it on the right hand side of the interface, you will see further options available, including the option to assign the extension to one or more call groups and pick up groups. It is not advised that you alter any of the other settings.

 

NOTE – it is important to open the settings for the extension once it has been created and press the Submit button, even if you have not made any changes to the settings. This is required to create the voicemail box for the extension.

 

 

Ring Groups

 

 

Ring groups can be assigned the numbers 0-99. Three different ring strategies exist. Again, hovering the mouse over the item description will display some help:

 

 

Available ring behaviours include:

 

• Ringall:             all of the extensions will ring simultaneously
• Hunt:                each extension is rung sequentially
• Memoryhunt:     all extensions are rung, the first extension alone, followed by the first and the second

together, then the first, second and third extensions together and so on

 

Add the extensions you wish to be part of the group. Please each number on a new line. Ensure that no non-existent extensions are included.

 

Ring groups can contain both internal and external numbers. For external numbers, they should be entered exactly as they need to be dialled, followed by a ‘#’ symbol, eg 901202712700#

 

You may add a Call prefix to the group that will be presented on the display of the IP phone, allowing the recipient to know what ring group is being called.

 

You can specify how long you want the group to ring for before the call is delivered to the No answer destination, the maximum ring time is 60 seconds, however there is nothing stopping the no destination from being the same ring group.

 

If required an announcement can be played prior to the call being delivered to the ring group.

 

Specify a No answer destination.

 

Inbound Routes

 

 

How an incoming call is routed will depend upon the technology over which it is delivered. If ISDN 2e or ISDN 30e is being used, then the call can be routed based upon the digits presented by BT. If an analogue gateway is being used, then you can route the call based on the digits sent by the gateway.

 

The simplest way to route a call is to send it to an extension, or a ring group:

 

 

Calls can also be routed to any of the following destinations

 

• Time condition
• Conference
• Extension
• Voicemail box
• Ring group
• IVR
• Queue

 

Setting an inbound route to match any DID or CLI is a useful “catch-all” as it is only used if an incoming call fails to match any other routes defined elsewhere. It is the only Inbound Route required if there is no CLI or DDI being presented.

 

Outbound Routes

 

 

An outbound route is composed of two components:

 

• A Trunk
• A Route

 

Trunks contain the physical settings, such as IP address, username and password and can be to hardware devices, ITSPs (Internet Telephony Service Provider)s, other SIPtrix™ servers or even other VoIP PBXs.

 

Routes examine the digits dialed and match them to a specific trunk.

 

To add a route, enter a name for the Route that will identify it (eg ‘Emergency’, ‘Default’, ‘To Head Office’, etc).

 

Dial Patterns effectively act as filters: you can specify that a particular trunk be used based on the sequence of digits entered, or the number of digits entered.

 

In the Dial Patterns field, enter a dial prefix, if desired, to identify the trunk, followed by a pipe. So, for example, creating a trunk with a dial pattern of 9|. Will result in all calls dialed with a leading 9 will be placed over this trunk.

 

Again, hovering the mouse over the Dial Pattern field will display some help:

 

 

Then select the trunk to be used by the route. The resulting configuration would look as shown below:

 

 

Outbound routes are processed in order, top down. Each route can contain multiple trunks. Trunks are also processed in order, top down.

 

Trunks

 

Trunks are what connect your SIPtrix™ server to the outside world.

 

 

There are four principal types of trunk:

 

• ZAP trunk          to an internal Digium ISDN 30e or analogue card
• IAX2 trunk         to a phone service provider or to another SIPtrix™ server
• SIP trunk           to a gateway device or to an ITSP
• Custom trunk     used for ISDN 2e with CAPI

 

ZAP Trunks

 

ZAP trunks are the only trunks that communication directly with hardware. They address “channels” at hardware level, usually by bundling multiple channels together as Groups. As an example, a server with a 30 channel ISDN 30e card installed, the trunk would be known as ZP/g1 – Group 1 containing some or all of the channels.

 

To add a ZAP trunk, select the option to Add ZAP Trunk:

 

 

Enter an identifier for the trunk (g1 in this case).

Submit the changes

 

Now allocate the trunk to an outbound route.

 

If desired, the maximum number of channels can be specified to limit the maximum number of simultaneous calls.

 

IAX Trunks

 

IAX is the Inter-Asterisk eXchange protocol, and is typically used to link two Asterisk servers together. To add an IAX trunk, select the option to Add IAX Trunk:

 

 

If necessary, specify the maximum number of channels to limit the amount of bandwidth that will be used by the trunk.

In the Outgoing Settings section, enter a name to identify the trunk, for example ‘To Head Office’.

In the PEER field, enter the IP address, username and password for the target server. For example:

 

host=81.138.13.178

username=user

secret=password

type=peer

 

Click the Submit button to save the changes. Now allocate the trunk to an outbound route. Within that outbound route it is advisable to allocate a prefix in the dial pattern, followed by the extension number range used at the remote site (for example: 5|2xx)

 

To allow calls to be accepted from the other VoIP server, edit the trunk and scroll down to the Incoming Settings section:

 

 

Enter a User Context to identify it, eg ‘From Head Office’.

In the details field enter the password and type details of the other server. For example:

 

context=from-internal

secret=password

type=friend

 

Click Submit to save the changes.

 

 

SIP Trunks

 

SIP trunks are used with Internet Telephony Service Providers (ITSPs). The settings used here will need to be obtained from your service provider. At the very least you will need a server address, username and password. You may also further settings as well as a register string, all of which your ITSP will be able to assist you with.

 

Time Conditions

 

Within SIPtrix™, there are no system-wide modes of “Day1”, “Day2” or “Night”. Instead, where time conditions are required, individually named Time Conditions can be created:

 

 

A time condition can be set as the destination of an inbound route, for example.

 

To create a time condition, enter a name to identify it, then enter the time settings in the Time to match field. This information needs to be entered in a specific format. As always, the context help has details:

 

 

 

 

 

Therefore, to direct all calls to a ring group between the hours of 9.00am and56.00pm, Monday to Friday, you would enter ‘9:00-17:00|mon-fri|*|*’, then set the ‘match’ destination to the ring group, and the ‘not match’ destination to, say, a voicemail box or system recording.

 

Conferences

 

SIPtrix™ has the ability to host multiple conferences simultaneously.

 

 

Callers can be added to a conference either by dialling it’s internal number (say, 8000), or by dialling into directly via an Inbound Route (which would have an external DDI and the conference set as the Destination).

 

Queues

 

 

 

A call queue is an alternative location to which you can send an incoming call.

 

Set the various parameters as required.

 

 

General Settings

 

 

These are system-wide settings affecting the behaviour of SIPtrix™.

 

Nothing on this page should be changed without the direction of a support engineer – in particular the fax support. This is included here for future releases and fax-to-email is not currently supported by SIPtrix™. The parameter should be left as disabled.

 

System Recordings

 

 

This is simply a method of recording and naming sound files for use elsewhere in the system. For instance, you can record a message here to be played to all incoming calls to let them know their calls are being recorded for training purposes.

 

To create a system record, enter your extension number into the box designated and click Go:

 

 

Now dial *77 from your extension and record your message. Hang up when done. To listen to the recording, dial *99. To re-record the message, dial *77 again.

 

Once you are happy with the recording, enter a name to identify it, then click Save.

Digital Receptionist

 

This is a very powerful feature normally only found on much larger and more expensive systems. This is one of the functions available within SIPtrix™ which sets it apart from other VoIP PBXs.

 

 

To create an IVR, you should first use the System Recordings tool to record all of the message(s) you will need to have played within the IVR, such as, for example, “Press 1 for Sales, or 2 for Accounts”.

 

Once you have recorded your message(s), click on the option to Add IVR:

 

 

Enter a name for the IVR to identify it.

 

Set the Announcement to the system recording you recorded earlier, it will be listed by name.

 

To follow our example, set the first option to 1, and set a destination of either an extension or a ring group for ‘Sales’.

Set the second option to 2, and set the destination to either an extension or a ring group for ‘Accounts’.

Click Save.

Now set the IVR as the destination of an inbound route.

 

To test the IVR, dial 7777 from an extension (to simulate in incoming call), then follow your prompts.

 

This is an example of a simple IVR, however the digital receptionist within SIPtrix™ is extremely powerful: menu options on one IVR can deliver to another IVR sequence and so on. Any given option anywhere in a sequence can deliver to an extension, a voicemail box, ring group, time condition, queue or even the same IVR!

 

If you are planning a complicated IVR sequence, it is recommended that it be planned on paper first!

 

 

On-hold Music

 

It is possible to upload your own hold music to the system.

 

The Music On Hold module will need to be enabled within the Module Admin view before the option will appear in the left-hand navigation pane:

 

 

However, this is not supported by Hugh Symons due to the potential issues of copyright.

 

SIPtrix™ Tools

 

 

The following tools are available within the SIPtrix™ administration interface:

 

• Module Admin
• System Status
• Sys Info
• Backup & Restore
• Asterisk Info
• Config Edit
• Asterisk Logs

 

The Module Admin tool allows you turn the different modules of Asterisk’s functionality on and off in accordance with your requirements.

 

The System Status tool provides simple information on what services are running and how long the server has been up for.

 

The Sys Info tool provides detailed information on hardware, memory, network and file system performance.

 

The Asterisk Info tool provides information on registered extensions, peers, trunk and channel usage.

 

The Config Edit tool allows you to edit the individual system configuration files that together make up SIPtrix™. It is not advised that you edit any of these files without the guidance of a support engineer.

 

The Asterisk Logs tool provides access to the last 2000 lines of the log file generated by SIPtrix™ on a continual basis.

 

 

Backup & Restore

 

The Backup & Restore tool allows you to take a snapshot of the configuration of the system. Backups can be set to run at regular intervals, however it should be remembered that these backups are stored on the hard disk of the SIPtrix™ server. It is advised that this backup file be copied to an alternate storage location for safe keeping.

 

To create a backup schedule, select Backup & Restore from within the Tools interface:

 

 

Select the option to Add Backup Schedule:

 

 

Enter a name for the schedule to identify it.

 

Select the system components you wish to include in the backup.

 

Configure the schedule details.

 

Once finished, click Submit to save the changes.

 

 

To restore from a backup, select the option to Restore From Backup. Any backups that have been taken will be listed.

 

To manually browse the file system on the server and copy the backup file to a second location, open a web browser and connect to:

 

http://10.0.0.100:10000              (where 10.0.0.100 is the address of the SIPtrix™ server)

 

Login in with:

 

Username:        admin

Password:        pa55word!

 

The webmin interface will be displayed:

 

 

Click on the Others tab and select File Manager. NOTE – you will need to have Java installed on your PC.

 

In the right hand pane, navigate to /var/lib/asterisk/backups/Immediate (where Immediate is the name of the backup schedule). Highlight the files you wish to copy. Click the Save button from the toolbar. Save the file to a suitable location on your computer.

 

 

Reports

 

 

The reporting tool built into SIPtrix™ offers the ability to display call history data. The interface is straightforward and easy to use. Detailed instructions are beyond the scope of this document.

 

 

Flash Operator Panel

 

The Flash Operator is a web-based application that displays information about your PBX in real time. It is so-named as it requires Adobe Flash Player to be installed.

The Panel can be used to display information on which extensions are busy, available or ringing, trunk availability, queue status as well as messages waiting.

 

 

Recordings

 

 

The Asterisk Recording Interface (ARI) allows you to view and manage the contents of your voicemail box via a web browser.

 

Log in with your extension number and password (1111 by default):

 

 

The Call Monitor provides a quick-glance history of the calls made and received by that extension:

 

 

The Settings page allows you to configure call forwarding, voicemail behaviour as well as call recording settings:

 

 

 

Web Meet Me Control Panel

 

The Web Meet Me Control Panel displays information on any conferences in progress, and allows the administrator to control each party’s involvement in that conference:

 

 

Connecting to SIPtrix™ via Putty

 

The SIPtrix™ box also features a built-in SSH server. This allows the administrator to connect to the server using a PC-based SSH client application, such as Putty, which can be downloaded free of charge from:

 

http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html

 

By default, the SSH service on the SIPtrix™ box has been configured to accept connections on port 9016.

 

To initiate a remote connection to the SIPtrix™ box, launch the Putty executable, the following window will be displayed:

 

 

Enter the IP address of the server (10.0.0.100 by default), and set the port to 9016. ensure that SSH is the protocol selected. Click Open. You will now connect to the SIPtrix™ and be prompted to log in. The default credentials are:

 

Username:        root

Password:        pa55word!

 

 

Configuring a Polycom handset for use with SIPtrix™

 

Polycom phones are the recommended devices for use with SIPtrix™, for 4 reasons:

 

• They can be provisioned from the SIPtrix™ server
• All phone functions are supported
• The buttons do what they are supposed to do
• They are quality devices

 

The Polycom phones are configured by two files stored in the /tftpboot directory on the SIPtrix™ server:

 

• <mac_address>.cfg
• phone<mac_address>.cfg

 

The MAC address is the Media Access Control address, a 12-digit code unique to that device, found on the rear of the unit.

 

Each Polycom phone therefore needs 2 files to be created and saved to the SIPtrix™ server in the /tftpboot directory.

 

If we have a phone with a MAC address of 0004f2044893, the files required are: ‘0004f2044893.cfg’ and ‘phone0004f2044893.cfg’

 

The format of the <mac_address>.cfg file is as follows:

 

<?xml version="1.0" standalone="yes"?>

<!-- Default Master SIP Configuration File-->

<!-- Edit and rename this file to <Ethernet-address>.cfg for each phone.-->

<!-- $Revision: 1.13 $  $Date: 2004/11/26 23:30:44 $ -->

<APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone0004f2058ab3.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY=""/>

 

This file defines the location of the sip.ld file (this is a global configuration file defining the SIP variables employed by the system) as well as the phone<mac_address>.cfg file.

The format of the phone<mac_address>.cfg file is as follows:

 

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>

<!-- Example Per-phone Configuration File -->

<!-- $Revision: 1.67 $  $Date: 2005/03/11 17:05:46 $ -->

<phone1>

   <reg reg.1.displayName="Joe Bloggs" reg.1.address="201" reg.1.label="201" reg.1.type="private" reg.1.thirdPartyName="" reg.1.auth.userId="201" reg.1.auth.password="password" reg.1.server.1.address="10.0.0.100"

 

……

 

</phone1>

 

The actual file is much larger, a complete sample file can be found on the web site.

 

Don’t panic about the size of the file: it is only the information highlighted in red that needs to be configured, the rest of the information is generic and common to all Polycom handsets.

Therefore it is a relatively straightforward process to make 2 files for each extension and simply alter the display name and extension details.

 

Now connect a Polycom handset to the network and power it on. When you see START  SETUP  ABOUT displayed on the screen, press SETUP. Enter a password of 456 (this is the default). Press the down arrow 3 times to highlight the Server Menu. Press SELECT.

Verify that the Server Type is set to Trivial FTP. If not, press EDIT. Press the Up and Down arrows until Trivial FTP is displayed. Press OK, then EXIT, followed by EXIT again and then REBOOT.

 

The phone will now reboot and then register itself to the SIPtrix™ server as extension 201 (in this example).

 

Configuring a Linksys PAP2 ATA for use with SIPtrix™

 

There are occasions when it is necessary to use an analogue telephone or fax machine with a VoIP PBX. The Linksys PAP2 ATA is the device which enables this to be done.

An ATA (Analogue Telephone Adapter) is configured with the IP address and the registration settings of SIPtrix™ and defined as an extension. Then, once registered, any analogue device connected to the ATA can make and receive calls through the IP phone system.

 

Configuration

 

Any pre-existing settings need to be cleared prior to using the PAP2 with the SIPtrix™ PBX.

 

1. Plug an analogue phone into “Phone1” using the adapter provided
2. Dial ****73738#
3. Press 1 to confirm
4. Replace the phone handset. The PAP2 will reset to factory settings
5. Connect the PAP2 to the LAN with a suitable cable

 

The PAP2 must now be given a suitable IP address for the network on which it is to be connected. If your network provides DHCP, the PAP2 will obtain a suitable address automatically. If you network does not provide DHCP, the PAP2 will need to be manually configured with a suitable address.

 

Networks with DHCP – find the IP address of the PAP2

 

1. From the analogue phone plugged into “Phone1” dial ****
2. Dial 110#
3. The IP address assigned to the PAP2 will be read out to you
4. Note down the IP address and replace the handset

 

Networks without DHCP – assign and address to the PAP2

 

1. From the analogue phone plugged into “Phone1” dial ****
2. Dial 101#0#1 and replace the handset. DHCP now turned off
3. From the analogue phone plugged into “Phone1” dial ****
4. Dial 111# then the IP address you require, using a “*” instead of “.”
5. Dial # to complete the address, then 1 to confirm
6. Dial 121# then the subnet mask you require, using a “*” instead of “.”
7. Dial # to complete the subnet mask, then 1 to confirm
8. Replace the handset

 

Use your browser to navigate to the IP address of the PAP2; either the DHCP address noted above or the address you set manually above.

 

 

1. Click “Admin Login”
2. Click “Switch to advanced view”
3. Click on “System”

 

Verify the “Static IP” address and the “Netmask” settings. If the address was assigned manually, as above, ensure that “DHCP” is set to “No”

 

1. Click on “Save Settings”

 

The unit will save the settings and restart; you will be directed back to the same page.

 

1. Click on “Line 1”
2. Scroll down and set “Proxy” to be the address of the SIPtrix™ server (10.0.0.100 by default)
3. Scroll down and set “Display Name” to the extension number the phone will have; eg 203
4. Set “User ID” to match “Display Name”; eg 203
5. Set “Password” to match “Display Name”; eg 203
6. Scroll down to “Dial Plan”. Remove all the existing entries and replace with the following:

 

(2xx|3xx|4xx|*8|*97|*411|*982xx|80xx|9[2-9]xxxxx|9[0-1][1-9]xxxxxxxxx|900xxxxxxxxxxxxx)

 

1. Click on “Save Settings”

 

The unit will save the settings and restart; you will be automatically directed back to the web page.

 

Once the unit has restarted, it will register with the SIPtrix™ server. Verify that this has happened by dialling the extension assigned to the PAP2 (ie the “User ID” set above) from another phone.

 

Using both ports of the PAP2

 

The PAP2 device has two analogue ports, and they can be registered independently to an IP PBX to provide two separate extensions. However, it is not possible to use both extensions simultaneously under all conditions. The specific condition under which you can do this is when both extensions are configured to use the G.711 codec. If your PBX is using anything other than this codec for either extension, the second extension will ring engaged whilst the first is in use. Similarly, it would not be possible to place a call on the second extension whilst the first is in use.

 

Ringing

 

The PAP2 is essentially an American device. American telephone systems rely on the telephones themselves to contain the circuitry to generate the ringing tone. BT, on the other hand provides this circuitry (usually in the “master” BT socket at the customer premises) so UK phones generally do not have this circuit inbuilt. In many cases a UK phone plugged into the PAP2 will not ring to alert an incoming call.

 

If this is the case, the RJ11-BT converter supplied with the PAP2 needs to be replaced with a converter containing the ringing circuit.

 

To change the ring cadence from “ring” to “ring-ring” as per a standard BT phone, locate the “Ring 1 Cadence” section under “Distinctive Ring Patterns” on the “Regional” page of the PAP2 interface.

 

1. Delete the existing values
2. Set the value to the following:

 

60(.4/.2,.4/2)

 

1. Click on “Save Settings”

 

Setting up a remote extension

 

In order to set up a remote extension which registers to the SIPtrix™ server over the Internet, the SIPtrix™ server must have a connection to the Internet and also have a fixed, public, ‘real-world’ IP address.

 

The remote extension must also have a connection to the Internet.

 

Create a new SIP extension within the SIPtrix™ admin portal. When creating the extension, set the nat field to yes (as opposed to never, which is the default value):

 

 

When configuring the device, enter the public IP address of the SIPtrix™ server rather than it’s internal address.

 

There are some further modifications to be made to the SIPtrix™ server. Browse to the admin portal and select Tools à Config Edit.

 

Locate the sip_nat.conf file. Enter the following lines:

 

nat=yes

externip=81.138.13.178

localnet=10.0.0.100/255.255.255.0

 

Subsitute the IP address details for your configuration.

 

If the SIPtrix™ server sits behind a firewall, then UDP port 5060 must be allowed through to the SIPtrix™ server, as well as UDP ports 10000 -20000.

 

If the remote extension does not have a fixed public IP address (which is very unlikely), then the address of a public STUN server may need to be entered as a proxy server, like stun.fwdnet.net (note – if you use the name of the proxy rather than its IP address then access to a DNS server is also required).

Setting up a Nokia E-series handset as a wireless (WiFi) extension with SIPtrix™

 

The Nokia E-series range of handsets all have a SIP client built into the operating system that can be used with SIPtrix™ in conjunction with a wireless (802.11b) network. This means that your mobile phone can be used an extension registered with SIPtrix™ when in the office, and as a normal mobile phone when you roam outside the coverage area of your wireless network.

To configure the phone as a wireless extension, do the following:

 

 

On the SIPtrix™ server

 

Add an extension in the correct number range (for example, 1xx) for the handset to identify it as a wireless extension.

Set the Outbound CID number to match of the person’s desk extension.

Set the secret to password.

Disable voicemail.

 

<Optional>

Create a ring group and add the person’s desk extension and the E series handset as members. Set the Inbound route for the person’s DDI to point to the ring group.

Set the ring strategy to ring all.

 

 

On the Nokia handset

 

Ascertain the MAC address of the wireless adapter in the handset by typing *#62209526#

Make a note of this address.

 

 

On the Nokia handset

 

Press the menu button. Select Tools and then Settings.

Select Connection

Select Access Points

Select Options, and then New Access Point à Use default settings

Enter a name for the connection to identify it (the SSID of the wireless network is recommended)

Set the Data Bearer to Wireless LAN

Set the WLAN net. Name to <SSID>

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