PBXpress Admin Guide


(c) 2004-2005 PBXpress Communications Inc. All rights Reserved. www.pbxpress.com

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Copyright notice & disclaimers

Copyright (c) 2004-2005 PBXpress Communications Inc. All rights reserved. PBXpress Admin Guide V.1.2.3 March 2005 Please address your comments and suggestions to: Sales Department, PBXpress Communications Inc., Westwood Corporate Centre, Suite 200, 2963 Glen Drive; Coquitlam, BC V3B 2P7 Canada Changes may be made periodically to the information in this publication. Such changes will be incorporated in new editions of the guide. The software described in this document is furnished under a license agreement, and may be used or copied only in accordance with the terms thereof. It is against the law to copy the software on any other medium, except as specifically provided in the license agreement. The licensee may make one copy of the software for backup purposes. No part of this publication may be reproduced, stored in a retrieval system, or transmitted in any form or by any means, electronic, mechanical, photocopied, recorded or otherwise, without a written permission from PBXpress Communications Inc. The software license and limited warranty for the accompanying product are set forth in the information package supplied with the product, and are incorporated herein by this reference. If you cannot locate the software license, please, contact your PBXpress Representative for a copy. All product names mentioned in this manual are for identification purposes only, and are either trademarks or registered trademarks of their respective owners.



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Table of contents ........................................................................

Preface ............................................................................................................................ 3 Introduction .............................................................................................4

Usage scenarios ......................................................................................6 PSTN call fail-over ....................................................................................................... 6 Network bridging ......................................................................................................... 6 Internal network only ................................................................................................ 8 PBX replacement – without outgoing VoIP........................................................ 9 Call recording box ..................................................................................................... 10

Hardware setup......................................................................................11 Networking setup ...................................................................................................... 11 Phone lines .................................................................................................................. 14 Power cable ................................................................................................................. 15 Check that PBXpress has obtained an IP address......................................... 15

System concepts ................................................................................... 16 Administrator interface............................................................................................ 16 IP PBX............................................................................................................................ 16 Call recording .............................................................................................................. 21 Voicemail ...................................................................................................................... 22 PSTN interface cards................................................................................................ 22

System configuration .......................................................................... 23 Using the admin web interface ............................................................................ 23 System configuration ............................................................................................... 27

Using the system .................................................................................. 66 Monitoring call records ............................................................................................ 67 Moving old call records to the backup storage............................................... 71 Checking voicemail messages............................................................................... 75 Using PBX features from an IP phone ............................................................... 81

Frequently Asked Questions.............................................................. 84

Appendixes ............................................................................................. 86 Appendix A - Glossary ............................................................................................. 86 Appendix B – Supported IP phones and analog phone adaptors ............ 89 Appendix C – Auto-configured IP phones and adaptors............................. 92



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Preface This document provides general information about PBXpress VoIP PBX as well as installation and configuration instructions.

Where to get the latest version of this guide

The online copy of this guide is always up-to-date, integrating the latest changes to the product. You can access the latest copy of this guide at: www.pbxpress.com/docs.

Conventions

This publication uses the following conventions: Commands and keywords are in boldface Terminal sessions, console screens, or system file names are displayed

in fixed width font Caution means ‘reader, be careful!’ You are capable of doing something that might result in program malfunction or loss of data. NOTE: Means reader take note. Notes contain helpful suggestions or references to materials not contained in this manual.

Timesaver means the described action saves time. You can save time by performing the action described in the paragraph. Tips Means the following information might help you solve a problem.

http://www.pbxpress.com/docs

Introduction


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Introduction Let’s answer these simple questions before we proceed.

What is PBX?

PBX (Private Branch Exchange) is a phone switch for your office organization. It connects phones inside of your office with the rest of the world, and allows you to reach John by just dialing 501, etc. Yes, it is that strange box which stands somewhere in the dark room, has a lot of cables connected to it, and most people in your organization have no idea how to change something in it.

What is PBXpress?

PBXpress is a modern day replacement for the traditional PBX systems. Instead of dealing with the wires, it utilizes Internet technologies. Notably, it uses VoIP (voice over IP) technology to transmit calls.

What are the benefits?

• Reduced hardware costs. PBXpress runs on the standard PC platform, which is much cheaper than the specialized PBX equipment

• Easy deployment and scalability. With the traditional PBX you need to have a physical wire from the PBX box to the phone on your desk. What if during the building construction only one pair of cables was installed going into your room, and now you need three phone lines? What if four of your employees are sitting not in your main office building, but in the sales center several blocks away? What if you want to use your phone while on a business trip? With IP-based PBXpress all of the above is not a problem at all. You just need network connectivity to be able to use the phone service

• Huge cost savings on the long-distance and international phone calls. With PBXpress you can utilize any of the many available VoIP carriers to terminate calls to them. In just one minute you can start forwarding calls to a Moscow-based company in Russia that cost only $0.05/min, still keeping Packet8 as your carrier for most of your international destinations and Qwest as your provider for local calls.

• Easy to manage. No need to do PBX programming using console interface and cryptic language according to think manual - everything is web based.

• Ability to record all of the phone conversations. Worried about who made all these calls to the Dominican Republic? Wish you had some record of your previous verbal agreement with the

Introduction


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vendor? Would like to analyze if your new sale person does the right thing on the phone? Not a problem - PBXpress can record every conversation made through it, and store it on a CD-ROM or DVD-ROM in the format which can be played on any computer.

• Integrated voicemail and auto-attendant, music on hold • ACD – automatic call distribution queues • Solution for networks with firewall/NAT. You tried to use service

from a residential VoIP company in your office before, but cannot get any audio? The most likely reason for this is that your firewall is too strict and does not allow network traffic to pass. PBXpress can work as a bridge across the firewall for VoIP calls, eliminating such problems.

• Integration with the external billing system via RADIUS. PBXpress supports standard Cisco VSA attributes for VoIP calls, so it can send call information to an external billing solution (for example, PortaBilling100 www.portaone.com/products/billing/index.html) for further call processing/rating.

Usage scenarios


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Usage scenarios

PSTN call fail-over Internet reliability is a very important issue when using VoIP. PBXpress offers risk-free approach to it. PBXpress has an interface to your local PSTN line, so if an outgoing call via IP fails, that call will be automatically routed via your existing PSTN line. This is transparent for your end users, so the worst case scenario (no Internet connectivity is available for VoIP calls at all) when using PBXpress is exactly what you have right now, when the local telco is used for outgoing calls.

Network bridging You have an office network and a firewall, which protects it. Unfortunately, it looks like the firewall blocks VoIP UDP traffic, so when you try to call somebody, you hear one way audio, and people cannot call you at all. At the present moment you are not able to change firewall configuration (it is either not possible, e.g. firewall is built into your router, or your network administrator is unsure how to do this, or there is some other reason). PBXpress offers an easy solution to this problem: it has two network interfaces. One of them will be connected to your LAN (Intranet), so there will be no restrictions on the traffic between the PBXpress and an IP phone. The other interface will be assigned a public IP address and connected to WAN (Internet) – so again, there will be no restrictions on the traffic between PBXpress and the remote gateway or an IP Phone.

Usage scenarios


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When the call is made, PBXpress will maintain two separate calls:

• Between an IP phone on your network and PBXpress; • Between PBXpress and the remote party,

and bridge them together. The main point of concern when migrating to the VoIP technology is reliability. Yes, it is nice to save a few pennies on each call, but what if because your Internet line was down you did not make an important call and did not close a deal for $100,000? PBXpress allows you to start using VoIP without loosing reliability. PBXpress will be connected to your existing PSTN lines (or some of them). If for some reason it is not possible to establish an outgoing call via the IP – it will use one of the available PSTN lines! So the worst-case scenario involving VoIP is basically equal to what you have right now. And of course, when your Internet connection is up (and normally it is up most of the time) it means savings. Plus, coming back to the “missed important call”, it seems to be something from the past. PBXpress can take an incoming call and forward it to your IP phone anywhere: in your office, at your home, or to your notebook with the WiFi connection.

Usage scenarios


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Internal network only The fact that there is some box which has access to both internal and external network might raise concern of your security administrator. And even if PBXpress box runs Unix, supplied in the secure configuration and is pretty difficult to break into, it is better to be safe than sorry.

In this case PBXpress will be installed with the LAN connection only (of course then you have to make sure that your firewall allows VoIP communication between it and outside VoIP carriers). Everything which is said in the previous chapter remains true: you can still use PSTN lines if your Internet service is down. With this method you can still make outgoing calls to the VoIP carriers and even receive incoming calls from them.

Usage scenarios


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PBX replacement – without outgoing VoIP Even if you are not ready to start saving on outgoing VoIP calls at this moment – you still may find PBXpress extremely useful. If you looking for a PBX for a new office – why not use PBXpress instead of the traditional PBX? You do not have high-speed Internet in your office at the moment – but perhaps you should prepare for tomorrow, when you will have one?

You get solution which is:

• Cost-efficient • Extensible • Provides call recording capabilities • VoIP ready – after you get your Internet connection, you can send

outgoing calls via IP with just a few mouse clicks, which decreases your costs substantially

Usage scenarios


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Call recording box Okay, so you definitely decided to keep your nice and expensive legacy PBX box for now. Even in this case, there is a place where you can use PBXpress quite efficiently. For instance, you can extend you existing PBX with the call recording capability. Just install PBXpress between your PBX and actual telephony lines, and every incoming or outgoing call will be recorded.

Hardware setup


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Hardware setup Unpack PBXpress from the box and follow further instructions in this chapter.

Networking setup Consult the Usage scenarios chapter above and decide on the deployment scheme for the PBXpress.

Network bridging

1. Plug the USB Ethernet dongle into first available USB port using the USB cable as shown below

Hardware setup


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2. Plug the WAN cable into the USB dongle as shown below

3. Plug the WAN Ethernet cable into a USB Ethernet dongle. 4. Plug the LAN Ethernet cable into an on-board Ethernet socket.

5. Install phone connectors according to instructions in the next

chapter

Hardware setup


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Internal network only

1. Plug the LAN Ethernet cable into an on-board Ethernet socket.

2. Install phone connectors according to instructions in the next

chapter

Hardware setup


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Phone lines PBXpress F200, 2 FXO

Use two upper ports

PBXpress F400, 4 FXO

Use all available ports

Hardware setup


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Power cable Connect the power cable as shown on the picture below

Check that PBXpress has obtained an IP address

Upon successful connection to the DHCP server on your LAN PBXpress will obtain an IP address, and this address will be displayed on the LCD

System concepts


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System concepts PBXpress consists of several components:

• Administrator web interface – something you will use to manage the system

• IP PBX • Call recording engine and call records • Voicemail web interface

Administrator interface Easy to use, accessible from any platform where a modern Internet browser is available, the Administration Interface allows you to:

• Change the system’s global parameters (e.g. public IP address) • Auto-detect and provision individual IP phones • Configure connections to external carriers, where you can

terminate calls to • Browse call log history, listen to recorded conversations • Manage backup of the call records on CD-ROM or DVD-ROM

There are two modes for the administrator interface: express and advanced.

• In the express mode, web interface contains only basic configuration screens, necessary to start using the service. The express mode is used by default.

• In the advanced mode, you have access to all the features and configuration options.

IP PBX PBXpress includes a SIP server with many features. SIP stands for Session Initiation Protocol, this is a protocol used by different elements of your VoIP network to communicate between each other. SIP server provides several functions:

List of IP phone registrations

When Peter dials David’s number, David’s phone should start ringing and when David answers – they should be connected together. It is easy to do it with the analog PBX: Some time ago it was entered into PBX configuration that Peter’s phone is connected to the cable is slot #25, and David’s phone is attached to the cable in slot #46. So when Peter calls

System concepts


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David, it is just a question of connecting these wires. If David decided to move to another office or there is some other change – it requires PBX re-programming and perhaps re-installing the physical phone wires in your office. With the IP, there is no physical connection between the user’s IP phone and PBXpress. David can take his phone and connect it in another office or in the meeting room. Or tomorrow, due to re-organization, you might completely re-design your office network. All this does not matter, as long as there is a working IP network between user-user’s IP phone and the SIP server. When the David’s IP phone goes online, it connects to the SIP server and informs SIP server about its current location (IP address). SIP server remembers this information – and now it knows where the call should be routed to if somebody calls David.

Number translation

It is the SIP server’s responsibility to know that if John dialed 304, it is the abbreviated number for David, so the call should go there. So the list of number extensions (abbreviated dialing) is maintained only once, inside of the PBXpress – no need to enter it into every individual IP phone.

Call routing

So somebody tries to call Prague, the Czech Republic. Should the call go over IP to Packet8, or over IP to iConnectHere, or is it better just to use our existing PSTN line? This is going to be decided by the SIP server based on your connection settings.

List of completed/uncompleted calls

Each call attempt is stored, so later you can see who made this call, when the call was made and to which destination.

Dialing rules

In the modern telephony world, the same phone number may be dialed differently, depending on the customer’s habits or a carrier being used. For instance, when dialing a phone number within the Czech Republic, the number will be dialed as 0016041234567. The same number will be dialed as 16041234567 within the United States. PBXpress dialing rules allow the user to easily describe the numbering format which should be used by specifying important parameters such as international dialing prefix and such. After that, PBXpress will be able to automatically perform a conversion of the phone number into the correct format.

System concepts


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Connections

Connections define how the PBXpress system is connected to the other telephone (VoIP or PSTN) networks. There are two types of connections, depending on which protocol is being used to carry the call: VoIP and PSTN. For instance, if PBXpress is sending some calls via FXO ports to the local telco, and some calls over IP to the VoIP carrier – there should be two connections in the configuration, with PSTN and VoIP type respectively.

Accounts

Accounts are only applicable to VoIP connections. Since a VoIP carrier usually requires your VoIP box to authenticate itself in order to prevent service abuse – these credentials must be entered into PBXpress configuration

Outgoing caller ID

The caller identification includes two fields: • CLI (ANI) number – phone number of the caller, e.g.

16045211234 • Display name (caller info) – usually first and last names, or a

business name, identifying the caller When somebody makes a call from an IP phone, information such as phone number or identification string “John Doe” is passed along with other call information, and is visible to the called party. It works OK while you are making calls between phones in your organization, but the carrier may apply certain restrictions. In this case, it may be necessary to alter the caller info information before sending the call to the carrier. For instance, if you have a single phone line from the telco, all outgoing calls made via this line from IP phones should have the same CLI (ANI) number, the one which was assigned to you by the operator.

Dialing rules

Since different carriers may require the phone number to be sent to them in the different format (e.g. call to Seattle going via PSTN to Qwest should be dialed as 123-4567, but a call to the same number going via SuperPacket VoIP should be dialed as 206-123-4567) - dialing rules must be used to ensure that PBXpress converts the number properly.

Out routing

Defines which calls should be dialed via the particular connection. For example, calls to the USA and Canada should be terminated via PSTN connection to the local telco, and all international calls must be terminated to the low-cost VoIP carrier. This is defined by specifying set of phone prefixes for each connection. For instance, phone prefix “420*” on the connection “Nextra CZ” means that every phone number which starts

System concepts


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with 420 will go to this carrier. As a result, all calls to the Czech Republic will be terminated to this carrier. Please note that the prefixes are specified in the E.164 format – so the phone number starts with the country code, then area code and then the actual number. So the phone prefix for Washingon D.C. is 1202* and not 202*! (202* is in fact a prefix for Cairo, Egypt)

Multiple connections and fail-over

It is possible that in order to provide stable voice service, you will have several available voice carriers for the same destination. For example for the vendor SuperPacket VoIP you have specified prefix 420* (Czech Republic) and for the vendor Call4Less you have specified prefix 4202* (Czech Republic, Prague), as shown below.

What happens now if somebody dials 42021234567? PBXpress will find a connection with the best matching prefix. In our example above it would be a Call4Less vendor, since it has a 4-digit prefix match compared to 3-digit prefix match of the SuperPacket VoIP. In case if several connection have exactly same prefix, as shown on the next picture, the call be routed to the connection with the higher preference.

System concepts


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Preferences are specified as positive numbers, so preference “5” for some connection means more preferred route comparing to the other connection with preference “3”. Also, it is possible to active a certain connection as a “backup” one. It means that if the call could not be terminated via one of the connections according to prefix specification, it will go via the backup one. This allows you to easily configure situations such as “if VoIP carrier fails, send calls via our office phone like” without having to put multiple prefixes in the telco connection configuration.

In routing

In routing defines to which IP phone (or phones) the incoming calls will be connected to. So the incoming call to your office line #1 will ring on all phones of people from the sales department, and office line #2 will be forwarded to Joe’s personal phone.

Music on hold

When during the call one party places the other on hold – you may configure PBXpress to play music for the party on hold. You can upload your own music files and provide some entertainment for the waiting caller instead of the silence.

Queues

Queue is a logic group of the extensions (IP phones) for the purposes of ACD (automatic call distribution). For example, if there are five persons working in the sales department and each one of them has an IP phone – it is highly desirable to transparently route incoming calls to any available phones in the sales department.

System concepts


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In addition to specifying which phones are included in the queue, you should also specify in which fashion PBXpress should apply ringing to available extensions. It is possible to do:

• Simultaneous ringing on all the phones, which belong to the queue (ringall). The first phone that answers will be connected.

• Ring one extension at a time. If that extension does not answer within the specified time interval – ring the next extension. The order in which extensions will be dialed in this case may be controlled by:

o Extensions are tried in order in which they are defined (roundrobbin). For example, if the queue is defined as consisting of extensions 101, 102 and 103 – PBXpress will first ring the 101 extension, then 102 and then 103.

o The list of extensions is sorted by the time when the extension received a call in the ascending order (leastrecent). So extensions which received the call the longest time ago, will be on top of the list.

o The list of extensions is sorted by the number of calls previously received by each extension in the ascending order (fewestcalls). So extensions, which received the least number of calls will be on top of the list.

o List of extensions is randomly shuffled for every new incoming call (random), so it will be different every time and cannot be foreseen

Call recording Every call which passes through PBXpress is recorded into the sound file on the internal hard drive. After that, you can browse the call records and listen to the conversations online. But even if the storage capacity of PBXpress is big, still it is limited. So the solution is to periodically move the old records to the external media, such as CD-ROM or DVD-ROM and free up the space for recording new calls. Each backup set is assigned its own unique ID. So if you can still listen to the call which was made several months ago and is not available on the PBXpress internal storage anymore. You will be prompted to insert the CD or DVD with the required ID into the drive, and PBXpress will retrieve the necessary file automatically.

System concepts


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Voicemail When the certain extension is not available or not answering – instead of just dropping the call, the caller may be given an option to leave a message. After that, the called party will be informed that there is a new message available, he or she can retrieve this message and take the appropriate actions

Message waiting indicator (MWI)

User will see directly on his/her phone that there is a new message available. The actual representation depends on the phone model – it may be an envelope picture on the LCD, or a special dial-tone or something else.

Email notification

When the new voice message arrives, either a short notification or a complete voice message (as a .WAV sound file in the email attachment) can be forwarded to the user’s external email (e.g. on your company’s email server)

Retrieving voice messages

User can listen to his/her voice messages by: • Dialing a voicemail access number from his/her IP phone • Logging into the voicemail web interface from his/her PC

PSTN interface cards In case if we cannot deliver the call over the Internet, PBXpress can use your existing PSTN lines to complete the call. SIP server is an IP-based solution, so in order to interface with the PSTN network, additional hardware modules will be installed into PBXpress. The type of these modules (ports) depends on the phone lines you currently have:

• FXO (Foreign Office Exchange) – this is a port where you can connect you normal telephone line (the one where you used to have your analog phone plugged into). So if your office used to have several phone lines from your local telco operator – you can connect them to PBXpress, and PBXpress will utilize them.

• T1/E1 – this is a connection when a single set of physical wires can carry many simultaneous calls. Normally, it is available for larger companies and big phone switches.

PBXpress is supplied in several configurations, with different PSTN interface cards to accommodate your needs.

System configuration


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System configuration Using the admin web interface

Navigation on the admin web interface

All PBXpress admin screens look similar. On the top, there is a list of tabs (e.g. System, Extensions, etc.), they define different parts of the system configuration. To switch between the tabs, simply click on the name of the required tab. When the certain tab is selected, you will see in the left column a list of different configuration screens (Status, WAN, LAN etc., see the picture below). To switch between configuration screens, just click on the button with its name.

NOTE: If you have changed something on the screen make sure you press Apply before going to another configuration screen.

Adding new records

Normally, when you open some form, it is in the mode for creating a new record. Enter required values in the available fields and then press Apply button to save changes.



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Editing the existing record

As described earlier, when you enter some form, it is in the “add new mode”. If there are some existing records, you will see them in the table at the bottom of the screen. To edit a certain record, click on the Edit icon next to it.

Press Apply button to save your changes.



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Deleting the existing record

When you see a list of available records, there is a Delete icon in the leftmost column. This icon allows you to delete a specific record. If the record cannot be deleted because it is referenced by another configuration element – the Delete icon will be absent.



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System configuration Connect internal network interface PBXpress to your LAN, make sure that there is a DHCP server available. Turn the PBXpress on, wait until it will finish booting up. The PBXpress IP address will be visible on the LCD display, as shown below: Now, on the PC connected to the same LAN, open your web browser and type http://<IP>, where <IP> is the IP address you saw on the LCD display (see more about this in the Check that PBXpress has obtained an IP address section on the Hardware setup part). Admin interface is protected by a password. The default User name is “admin” and the Password is “admin” as well.



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After that you will see PBXpress admin main menu.

The following chapter gives you a detailed description of all the options, available in the Advanced mode, please, switch to the advanced mode by clicking on the Advanced button, so your PBXpress screen will be similar to the one below. Consult the PBXpress Quick Start Guide for description of the Express mode.



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General configuration

Status

This is a read-only page which provides information about the current status of the network parameters.

WAN configuration

Press WAN button on the left side to access the configuration menu for the public (Internet) network interface.



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The values for IP, Net mask, Default Gateway and DNS servers should be provided to you by your ISP or network administrator. Press Apply button after you have entered all of the values.

LAN configuration

Usually, PBXpress will get an IP address for the internal network interface using your existing DHCP server, so the default settings are fine. Press the LAN button to review the current configuration or change configuration parameters if necessary.

Time synchronization configuration

In order to maintain correct call connect/disconnect times, PBXpress server must synchronize its internal clock with the other servers on the Internet. The default settings are probably suitable for most of the users.



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Administrator access options

This section controls access to the PBXpress administrator interface.

• User section. Here you can change the User Name and Password

used to access the admin interface. NOTE: Make sure you remember the new User Name and Password after you change them. • Remote management section. If you would like to access

PBXpress admin interface not only from your internal network (LAN), but also from other computers on the Internet (e.g., from you home DSL line) you may enable remote management (it is turned off by default, so admin interface is only accessible from the internal network). It is highly recommended to list IP addresses of all hosts allowed to access admin interface in the IP Access List field. Then if an intruder (even knowing the username and password) will try to access the system from his/her computer, such attempt will be rejected.

• LCD management section. Controls if you are allowed to do simple administration tasks via the LCD display and command buttons located directly on the PBXpress box.

• Factory Reset section. Enter Factory Reset in the Confirm field and press the Reset button to clear the current PBXpress configuration and start the system configuration from the very beginning.



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Prompts

Here you can manage a set of voice prompts to be played to the calling (or called) party. For instance, you can play the announcement that the call may be recorded.



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There are several pre-configured voice prompts supplied with the system and you may upload your own ones. When uploading a custom prompt, you can give it a short name and specify the path to the original sound file on your workstation. It is highly recommended to type the full prompt’s text into the Text field, so it will be stored in the database.



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AAA - Interface to the external RADIUS server

PBXpress may connect to the external RADIUS server for the purposes of the call authorization and accounting.

Field Description RADIUS section Server Host Hostname of the RADIUS server. If you are using a

hostname, make sure it is correctly registered in the DNS. Otherwise, use the IP address.

Authentication Port

Port on which the RADIUS server is listening to the authentication requests. 1812 is the default port according to the new RFC, some older servers may use port 1645.

Accounting Port

Port on which the RADIUS server is listening to the accounting requests. 1813 is the default port according to the new RFC, some older servers may use port 1646.

Secret RADIUS secret key, make sure the value you enter there matches the one specified in the RADIUS server configuration.

Number of retransmits

How many times PBXpress will try to re-send every RADIUS request in case if no response is received from the RADIUS server.

Retransmit Interval, sec.

How long PBXpress will wait for the response from the RADIUS server before the next retransmit attempt.

RADIUS source interface

Which network interface (LAN or WAN) shall be used to send RADIUS requests



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Field Description Accounting section Accounting Controls the way accounting packages are sent to the

RADIUS server • None – disable call accounting • Start-Stop – send one accounting immediately

when the call is connected (start accounting) and another one when the call is disconnected (stop accounting)

• Stop-Only – send one accounting record, when the call is completed (disconnected).

Alive Enable While the call is in progress, periodically send keep-alive RADIUS requests, informing the RADIUS server about the current call status.

Alive Interval, sec.

How often keep-alive packages should be sent.

NOTE: Call authentication and authorization are not currently supported.

PBX

Here you may configure parameters for the PBX features, such as voicemail system or call parking.



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Field Description

Access number

Which number should be dialed from an IP phone in order to listen to the user’s voice messages

Ringing timeout, .sec

Default timeout parameter. If the call is not answered within this interval, the call is forwarded to the voicemail. Please, note that this parameter is only used for calls between extensions. For the incoming calls timeout parameter is defined in the connection’s In routing settings.

SMTP Relay Host

If your users would like to receive notifications about new voice messages, PBXpress must be able to deliver notification emails to external mail servers (e.g., gmail.com). The best way to do this is to forward these emails through your company’s mail server or via your ISP’s mail server. Put name of this server here.

PBXpress Hostname

When sending such notification emails, PBXpress must include information about itself in the email. Please, make sure the name you assign to it (e.g., pbxpress.mydomain.com) is properly registered in the DNS.

Extension Call park extension (700 by default). If the call is transferred to this extension, it will be parked. The first parked call will be assigned this extension number increased by one (701 in the default configuration), the second will get an extension number increased by two (702), and so forth. You will hear the number of the actual parked extension after you make a transfer. 20 simultaneous parked calls are supported, so you cannot overlap the call parking extension range with the normal extensions, i.e. in the example above you cannot have extension 710.

Timeout, sec. How long the call may remain parked, before it will be returned back to the party, who parked the call.

Extensions - phone extensions configuration

Extensions

Extensions are abbreviated numbers for your IP phones. So when somebody needs to call John, they can just dial 121 instead of his full number. Here you can manage the list of the available extensions – add new entries, edit or delete existing ones.



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You need to create an extension for every person in your organization who will use an IP phone.



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Column Description Status • Empty – this extension is not assigned to any

phone, or the phone with this extension is currently off-line

• - this extension is blocked. • - this extension is provisioned on the phone,

this phone is currently online, so this extension can send and receive calls

VM • Empty – voice mail is disabled for this extension

• - voice mail is activated (by default it is activated for every new extension)

Extension # Abbreviated number (extension) for this user Phone # Complete phone number for this user Display Name Name of the user IP If the phone is currently online – IP address of the

phone User Agent Type of IP phone currently used



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Browser configuration for pop-ups (Internet Explorer) NOTE: In order to make sure that auto configuration works properly, please, ensure that pop-ups are allowed for this web site in your browser. Should you see the message similar to the one below:

Right click on the warning text and choose “Allow Pop-ups from this side”:

Browser configuration for pop-ups (Mozilla)

Mozilla/Netscape users may see a message similar to the one below. Click No, so the pop-ups will be allowed.



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Phones

Usually, when putting IP phones on your network, you need to configure each one of them individually before they can be used. PBXpress allows you to substantially simplify this process. You can now simply connect IP phones to your LAN in the default factory configuration. PBXpress will be able to detect and configure all of them automatically.

Specify the starting and ending IP (in most cases the provided default values are ok) and press Search.



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You will see a list of IP phones which were detected on your internal network. Notice, that since these phones are not yet configured, they are marked as unregistered. Now, you can assign one of the phone extensions to the IP phone (or to the specific port on it). PBXpress will update the phone configuration, so it will be registered to your server.



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Press Close button in order to close the popup window and return to the phone configuration screen. PBXpress will re-scan the network and display the current status of phones. Note that during the IP phone configuration another popup window will be opened. This window will contain additional information about this IP phone model. You will see an information screen similar to the one below.



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In case the configuration of the IP phone is password protected, you will be prompted to enter it (you can also store it in the PBXpress for future usage).



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After you configure several IP phones, you can go back to the list of extensions (click on the List button on the left). Extensions which are registered to the server at the moment will have the green status light :

IP phone firmware version

If during the IP phone detection PBXpress will detect that one of the phones is currently using too old firmware (which may have known bugs or may not have all the required features) you will see a warning icon, as shown below.

Please consult the IP phone information page (popup window which appear during the phone configuration) to find out which firmware version is recommended



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Dialing rules

Around the world, people dial phone numbers differently. For instance, if Jan lives in Prague, the Czech Republic, and his home phone is 23456789, his friend John from Canada will have to dial his number as 01142023456789, his partner Hans from Germany will dial it as 0042023456789 and his girlfriend Zuzana will just dial 23456789. If migrating to the IP telephony technology will bring the necessity to train every user to dial the phone numbers in a new fashion, this will overcomplicate the migration process. PBXpress allows any company around the world to use their preferred dialing format, so nothing is going to change for the end user. On the dialing rules configuration screen, you can specify parameters, so the system will know how to interpret dialed phone numbers.



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Most of parameters here are self-explanatory, but if confused, you may load and learn the example settings provided. Also note the Check Yourself area – verify that the system “guessed” correctly the way you dial phone numbers.

Field Description Your country code

Country E.164 prefix, e.g. 1 for US & Canada or 420 for the Czech Republic

Your area code(s)

Code of the area or city where your phone belongs to. If there is more than one area code which is considered to be local (as for instance area codes 604 and 778 in Vancouver, BC) enter all of them, separated by comma

Include area code

If checked, it means that you always dial the number including the area code, even when dialing within your “local” area. For instance, even being within 604 area code and dialing a local number, you still must dial 604 NNN-NNNN

Prefix for accessing the outside phone network

Dial (if applicable) to get an outside line (e.g., when using some office PBX systems you first must dial 0 and then the actual number)

Prefix for domestic calls

Dial before the phone number when dialing domestic long-distance. (E.g., 1 in the USA & Canada, 0 in many European countries)

International dialing prefix

Dial before the phone number when making an international call. (E.g. 011 in the USA & Canada, 00 in many European countries)

Exceptions Comma-separated list of the special numbers. These numbers will not be treated according to the dialing rules, but rather processed “as-is”. For example if you setup that you are in the US (country code 1) and 206 area code and the user dials 911 on his phone – the system should not convert this number into something like 1206911 – the number should stay 911.

NOTE: You must configure proper routing in the connections for the exceptional numbers in order for your users to actually reach the destination (e.g. emergency response team).



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Recording

You can turn on or off call recording for the calls between different IP phones on your network (by default is on), and choose who will be notified about the fact that the call is recorded – and which announcement (one of those defined in the System/Prompts section) will be used.

Field Description Enabled Turn on or off the call recording (for the calls between

extensions) Calling Party Notify the caller before the conversation starts that the

call will be recorded Called Party Notify the recipient of the call Type Which voice prompt should be used for the call

recording announcement



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Music on Hold

This screen defines what should be played if during the call between two extensions, one party was put on hold. Choose one of the pre-defined voice prompts (NO MUSIC means that nothing will be played) and press Apply.



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Voicemail

When an incoming call to some extension is not answered, it will be forwarded to the voicemail so the caller may leave a message.



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Field Description

Extension Extension number Enabled Activate voicemail for this extension Phone Password

Password for accessing voice messages from an IP phone (by dialing voicemail access number). By default, password is set equal to the extension number. If you enter an empty password, system will not ask for the password when calling to the voicemail access number from an IP phone

Web Password Password for accessing voice messages via the web interface. By default, password is set equal to the extension number

E-mail Notification

When a new voice message arrives, send a short notification to the specified email address.

E-mail Forwarding

Send a complete copy of the every new voice message to the specified email address

Queues configuration

Create a queue

Click on the Queues tab. You must create at least one queue, since initially there is no are queues available.



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Fill in the queue parameters and then press Apply button. Field Description

Name Name of the queue Strategy How the incoming call is distributed between the

available extensions. See the Queues chapter for detailed information about the different strategies

Ringing Timeout, sec.

Only applicable for the ringall strategy. How long the incoming call will be waiting to get connected.

You will see a list of the current queues. Note that the queues list can also be used to monitor current calls which go via the particular queue.



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Extensions

After the queue is created, you may edit the list of extensions for it. Press the Extensions button on the left side and then click on the Edit icon for the specific connection.

In order to add an extension to the queue, choose it in the Extension select menu and press Apply.



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Repeat the above step to include all required extensions into the queue, as shown below



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Music on Hold

Here you may control which music the caller will hear while he/she is on hold. You may assign different sound files for every queue.



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Connections configuration

In this section, you should configure how your PBXpress system will be connected to the rest of the telephony world: how it exchanges calls with the other carriers.

Create a new connection

When creating a new connection, you must provide the following parameters:

• Name – unique identifier for the connection • VoIP or PSTN. If this is a connection to telco operator or PBX

which uses one of the PSTN interfaces of PBXpress, choose PSTN.

• The rest of the parameters depend on the connection type (see below)

Creating a new VoIP connection

Avaialable fields:

Field Description SIP Proxy IP IP address of your provider’s SIP proxy. Hostname

cannot be used here to avoid problems with ambiguous domain resolving, since in this case PBXpress must not only resolve IP address when sending an outgoing call to a vendor, but also determine which vendor the call arrives from for the incoming calls based on the remote IP address.

Accounts table (you may enter more than one account per connection)

Account information



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Field Description Reg PBXpress should register to the remote SIP server

using this account. Without the registration, PBXpress may still make outgoing calls, but probably will not be able to receive any incoming calls

Account ID Username for the remote SIP server Password Password for the remote SIP server

Outgoing Caller ID information Field Description

Use Change the outgoing caller Id information. If this checkbox is off, send the same ANI (CLI) phone number and display name as received from the IP phone

Phone # Replace CLI (ANI) for the outgoing calls with the specified value

Display name Replace caller info for the outgoing calls with the specified value

If this connection assumes registration to the remote SIP proxy, after the connection is successfully configured, you may see that there is a green light in the Status column. If some of the accounts are not registered to the SIP server while others are – the blue light will be shown.



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Creating a new PSTN connection

Available fields Board ID of the PSTN interface card installed in the system.

This is required to distinguish between several PSTN cards installed into the PBXpress.

Port Specific port on the card (for instance a single card may have to FXO ports)

Outgoing Caller ID information Field Description

Use Change the outgoing caller Id information. If this checkbox is off, send the same ANI (CLI) phone number and display name as received from the IP phone

Phone # Replace CLI (ANI) for the outgoing calls with the specified value

Display name Replace caller info for the outgoing calls with the specified value

Some of the connection parameters may not be available when creating a new connection and you can only access them when changing existing connection information.



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For PSTN connections, these parameters are:

Field Description Detect Polarity Reversal

Polarity reversal is required to provide a correct answer supervision (detect when the call is actually connected) on the analog lines. This feature requires support from both the phone company and your PSTN equipment, so activate it after you confirmed that it is provided by your telco. Being able to detect a call connect time correctly is required for two things: to be able to display a correct time in CDRs and to be able to start playing “this call is being recorded” announcement for the called party in time. Otherwise the announcement will start playing immediately after the number is dialed, and probably will finish playing before the called party pickup the phone.

Dialing Mode On the analog lines you have an option for tone and pulse dialing (pulse dialing is used on some older telephony systems). Choose the option applicable to your phone line.

Dialing rules

In most cases, the carrier will require a specific numbering format which will not match the one you use on your telephony network. For instance, if you are based in the USA and you place a call to the local telco– you will need to dial the 011 prefix, even if your users never dial it on their IP phone. So PBXpress must convert the number from the format used on your system into the carrier-specific format. This is done by specifying the dialing rules, similar to the ones we configured for our users. Double-click on the corresponding row in the Sample Settings list box to load the sample settings.



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Field Description Your country code

E.164 country prefix, e.g., 1 for USA & Canada or 420 for the Czech Republic

Your area code(s)

Code of the area or city where your phone belongs to. If there is more than one area code which is considered to be local (as for instance, the area codes 604 and 778 in Vancouver, BC) enter all of them, separated by comma

Include area code

If checked, it means that you always dial the number including the area code, even when dialing within your “local” area. For instance, even being within the 604 area code and dialing a local number, you still must dial 604 NNN-NNNN

Prefix for accessing the outside phone network

Dial (if applicable) to get an outside line (e.g., when using some office PBX systems you first must dial 0 and then the actual number)

Prefix for domestic calls

Dial before the phone number when dialing domestic long-distance. (E.g., 1 in the USA & Canada, 0 in many European countries)

International dialing prefix

Dial before the phone number when making an international call. (E.g., 011 in the USA & Canada, 00 in many European countries)

Exceptions Comma-separated list of the special numbers, which will not be treated according to the dialing rules, but rather dialed unchanged. Usually these numbers are emergency numbers (911 in the US, 112 in Europe).



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Out routing

On this screen, you can configure which types of the outgoing calls will be sent via this connection



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Field Description

Use Engage this connection (or the particular account in case if multiple accounts are defined for the connection) into the outgoing call routing.

Preference Positive number (1..99). Allows you to manually adjust routing order when several carriers may terminate the same prefixes.

Prefixes Comma-separated list of prefixes which should be routed via this connection. Each prefix must end with the * (star), otherwise it is considered to be a specific number .See the following examples:

• “1604*” means that call to every number which starts with 1604 (e.g. 1604-123-4567) will be routed through this connection

• “911” means that only calls made to the specific number 911 will be routed via this connection. If you wish to use some connection to route a call to 91118765432 (India, Delhi) – specify “911*” among prefixes for this connection

Backup Use this connection as a backup one, if the other connections fail – even if the prefix for the phone number was not specified for it



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In routing

In routing defines where the incoming call will be delivered which arrives via this connection.

Choose the extension and press Apply. Note that the call may be forwarded to an individual extension as well as to the queue.

You may forward the incoming call to several extensions or queues. PBXpress will try to ring all of them in the order they were specified in the configuration. For example, you may forward the call to “sales” queue and if nobody answers within 45 seconds – transfer the call to the sales department supervisor.



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Hint: when defining a call forwarding to some queue containing several extensions and a ringing policy which is different from ringall (so extensions will be tried sequentially and only one extension will ring at a time) – make sure you specify a timeout large enough to ring all of the extensions.

Queues and voicemail

What happens if the incoming call was routed to some queue (which has call distribution to three extensions), but no one has answered the call? The call should go to the voicemail, but to which extension? PBXpress will not use voice mailboxes of the extensions, included in the queue (even if the individual extension does have s voice mail enabled). The correct was to configure this is to define in the In routing connection properties that the incoming call shall be first forwarded to the queue and then (if not answered!) to some other extension. This extension should have a voice mail enabled and can be either a real extension (so the phone will ring and there will be a last chance to pick up the call) or just a fake extension, not provisioned on any phone. The voice message left by the caller will be delivered to this extension’s mailbox. See the example configuration below:



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Recording

Recording configuration screen for the connection is basically identical to the one you have seen already in the Extensions section. By setting up parameters here you can define whether calls going through this connection (both incoming and outgoing) should be recorded.

Field Description Enabled Turn on or off the call recording (for the calls between

extensions) Calling Party Notify the caller before the conversation starts that the

call will be recorded Called Party Notify the recipient of the call Type Which voice prompt should be used for the call

recording announcement



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Music on hold

Again, this screen is very similar to the one in the Extensions or Queues section.

For every connection you may choose which music file will be played while the user is placed on hold.

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Using the system After you have completed the steps described in the

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System configuration chapter, you are ready to make calls. There are only few administration tasks you will be responsible for.

Monitoring call records For every call made through PBXpress, an entry is made in the call history. You will be able to see when the call was made, from which phone number (extension) it was originated, what was the destination number and other information. There are two options for the Call History main tab: Recent allows to quickly see the most recent calls made, and Find allows you to perform the advanced search to display specific call records.

Recent

Here all the calls registered in the system are displayed in the reverse chronological order – so the most recent calls are shown on the first page. You can always go back in the history by moving to the next page. You can also listen to the recorded call conversation. The meaning of the icons in the Listen column is:

• - the call was recorded, the recorded file is available on the PBXpress internal storage, so you can listen to the record by clicking on the icon.

• - the call was recorded, but the recorded file is not available on the internal storage anymore, since it was moved to the backup optical disk. In order to listen to the file, click on the icon. You will see a dialog box similar to the one below.

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Locate the optical backup disk with the matching label, insert it into the CD/DVD drive and press ok

• - the call was recorded, the recorded file is available on the backup optical disk which is currently in the drive. Click on the icon in order to listen to the call record

• Empty icon – the call record is not available. It means that the call was unsuccessful or it was not recorded because at the moment the call recording was switched off for this connection.

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Call notes

Each call has a Notes field, where you can put you own identification string for this call. Click on the Add note to add a call description or click on the note text to edit the existing note.

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Find specific calls

You can search calls in the certain date range and in addition, you can filter the result and display only calls from the certain phone number to the certain destination (enter the full phone number instead of the extension).

For the From/To fields you can specify the value either as a date in the format YYYY-MM-DD, or click on the icon to invoke a calendar popup. Also, you may narrow the list of the calls displayed by displaying calls to certain destination only, or calls with the note containing a certain text string.

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Moving old call records to the backup storage

The internal storage of PBXpress can hold a sufficient amount of the recorded conversations, but it is not unlimited. This is why it is recommended to periodically copy the old call records to the optic disk and clean up space on the internal storage for new calls. NOTE: When you move the call record to the optical disk, only the sound file associated with the call is erased on the internal storage. The call record with the information such as call time, origination/destination number and such is still available.

Status

Choose the Call History tab and then click on the Disk Status button.

HDD section

This part is for information purposes only. You can see how much space is available in total for the voice recording and how much of it is used right now.

Optical drive section

You can see the information about the disk currently in the drive, eject it from the drive or load it.

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PBXpress backup info

If the optical disk in the drive contains backup information, brief summary is shown here

Field Description

Volume label Each backup disk is uniquely labeled (disk ID assigned when the disk is written)

Call History This field contains from-to date of the call set, written on this disk. It is possible to copy the call records back to the internal storage using the Restore button. Note that for the purposes of only listening to the voice conversation it is not required. It is sufficient to just place an optical disk in the drive.

Configuration In addition to the call records, it is also possible to store the current PBXpress configuration on the optical disk. This will include configuration settings such as extensions, connections, etc. So in case if your configuration was damaged due to the improper administrator’s actions or if you had performed a factory reset on your PBXpress – you may quickly restore your previous stored configuration.

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Erasing a re-writable optical disk

If the disk is re-writable you can also erase it.

Recording a disk

Press the Write disk button on the left side to access this screen. You can specify the date range for the calls to be placed on the disk plus there are two options: copy and move.

• There are two special keywords you can use as a value for the date field. Epoch means “the earliest date possible” and max means “the latest date possible”

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• Copy means that the sound file is not deleted from the PBXpress internal storage, so there will be no changes in the available disk space. This option is convenient if you need to make a special copy of the calls within a certain period (e.g., to do the evaluation of the calls made during the training session)

• Move means that the sound files after being copied to the disk will be deleted from the internal storage. This is a normal routine.

For your convenience, there are sample settings provided: • Copy oldest records – start with the earliest date possible and

copy the call records into the optical disk (as many as it will fit there). The original sound files will stay on the internal storage.

• Move oldest records – same as above, but the sound files will be deleted after optical disk is written. This allows to back up the old call records periodically and free up the space for new calls

• Copy recent records – allows you to make a copy of all calls which were not backed up yet

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Checking voicemail messages This is a separate web interface and in contrary to the PBXpress administrator interface, this one is accessible by your end users. To start, open your web browser and type in the voicemail web interface URL:

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http://N.N.N.N:8000, where N.N.N.N is an IP address or a domain name of your PBXpress box. You will be prompted to enter the username (your extension number) and the password (something assigned to you by the administrator. By default, the password is equal to your extension number).

Messages

After you login, you will see the current messages screen. You can:

• Listen to the messages (click on the Listen icon for the corresponding message)

• Put your own comments for the voice message (click on the Add note text to add a new text description or click on the note text to edit it)

• Erase voice messages (click on the Delete icon next to the message)

Using Find screen you may display only messages: • For the specific time interval (you can either enter dates as year-

month-day or use a calendar popup window) • Only from the specific caller or callers (you do not need to enter

the full phone number – if you will specify just 1866, as shown below – all calls from the phone numbers beginning with 1866 will be displayed)

• Containing certain text string in the note

http://n.n.n.n:8000/

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Properties

There are three types of parameters a user can customize:

Greetings

When somebody reaches your voicemail, he/she will hear an announcement that he/she should leave you a message. Some default prompts are supplied with PBXpress and of course, users may upload their own prompts, recorded in their own voice. To upload a new voice prompt, choose the Properties tab and then the Greetings button on the left side.

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Field Description Name Some name to distinguish this prompt among the

others Type In which situation this prompt should be played? You

can have different prompts played in the following situations:

• Somebody calls an extension, but this IP phone is off-line or not answering

• Somebody calls an extension, but the user is currently on the phone

• Name? Enabled Use this prompt for the specified type. You may have

multiple voice prompts uploaded, but only one of them will be used for the specific announce type (e.g., Unavailable). Active (used) voice prompt is marked with the green icon , unused prompt is marked with the yellow icon)

File Path to the sound file. You may either type the path tp the file manually, or use the Browse button to launch a file open dialogue.

Text Text of the voice prompt. It is highly recommended to put a complete voice prompt here, so then later you can find the required prompt by just quick glance at the table of available prompts, instead of having to listen for every single one of them.

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Passwords

For the improved security the end user may change passwords, assigned to him/her by system administrator. In fact, it is highly recommended that you change your password from the default one (equal to the extension number) immediately upon your first login.

Field Description Phone Password you will use to access your messages from an

IP phone. Since this password must be dialed on your phone’s dial-pad, it must consist only of digits. If you enter an empty password, system will not ask for the password when calling to the voicemail access number from an IP phone

Web Password you will use to access your messages from the web interface

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Email forwarding

You can change how you should be notified about the new voice messages.

Field Description Notification When a new voice message arrives, send a short

notification to the specified email address. Forwarding Send a complete copy of every new voice message to

the specified email address

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Checking voice messages from an IP phone

Dial the voicemail access number. If prompted, enter the voice mailbox password. The following vice menu is available: - 1 Read voicemail messages - 3 Advanced options - 1 Reply - 3 Envelope - 5 Leave a message in another mailbox - * Return to the main menu - 4 Play previous message - 5 Repeat current message - 6 Play next message - 7 Delete/Undelete current message - 8 Forward message to another mailbox - 9 Save message in a folder - * Help; during message playback: Rewind - # Exit; during message playback: Skip forward - 2 Change folders - 0 New - 1 Old - 2 Work - 3 Family - 4 Friends - # Cancel - 3 Advanced options - 5 Leave a message - * Return to the main menu - 0 Mailbox options - 1 Record your unavailable message - 2 Record your busy message - 3 Record your name - 4 Change your password - * Return to the main menu - * Help - # Exit While listening to a recorded voicemail message: Press # to fast forward, or * to rewind by 3sec increments.

Using PBX features from an IP phone Special features such as call transfer require support of the side of the SIP device. While IP phones normally will have dedicated buttons for these functions, on the analog telephone adaptors you will have to use special combinations of the normal dial keys (e.g. #90). Moreover, specific usage details (such as which keys should be pressed and in which sequence) are vendor dependent and may differ from one phone to another. Of course it is very inconvenient for the end user, so PBXpress tries to improve this

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situation by provisioning analog telephone adaptors to use the same keys whenever it is possible. Of course this only applied to the supported SIP devices, which are auto-provisioned by PBXpress.

Put the other party on hold

Sipura Adapters & Cisco ATA

Step 1 Press the flash button on the telephone handset to put the other party on hold Step 2 If necessary, you may now dial some other extension and have a conversation (consultation hold) Step 3 When you wish to resume the first conversation, press the flash button again

Fully unattended transfer (blind transfer)


Step 1 Press the flash button on the telephone handset to put the other party on hold and get a dial tone. Step 2 Press #90 (the transfer service activation code) on your telephone keypad, then enter the phone number to which you want to transfer the other party, and then press #. Step 3 Hang up your phone.

Attended Transfer (consultative transfer)

Cisco ATA

Step 1 Press the flash button on the telephone handset to put the other party on hold and get a dial tone. Step 2 Dial the telephone number to which you would like to transfer the other party. Step 3 You can talk to the other party first and then.... Step 4 Hang up your phone to transfer the other party

Sipura Adapters

Step 1 Press the flash button on the telephone handset to put the other party on hold and get a dial tone. Step 2 Press #91 (the transfer service activation code) on your telephone keypad, then enter the phone number to which you want to transfer the other party, and then press #. Step 3 You can talk to the other party first and then....

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Step 4 Hang up your phone to transfer the other party.

Call Park and Call Pickup


Extension 700 is used to park a call. Step 1 While in a conversation, press the flash button on the telephone handset to put the other party on hold and get a dial tone, then press #90 (the transfer service activation code) and dial 700, press #. Step 2 PBXpress will now announce the parking extension, most probably 701 or 702. Step 3 Now hang up - the caller will be left on hold at the announced extension. Step 4 Walk up to a different phone, dial the announced number (e.g. 701) - the conversation can be continued. If a caller has been parked longer than 45 seconds then PBXpress will again ring the originally dialed extension. Please note that 700 is the extension used for call parking by default, but it may be changed by the PBXpress administrator to whichever value is more appropriate in your case (for instance you wish to use the same extension which was used on your legacy PBX for backward compatibility). The same applied to the call park timeout.

3-way Conference


Step 1 Press the flash button on the telephone handset to put the other party on hold and get a dial tone. Step 2 Dial the telephone number to which you would like to join the conference. Step 3 Wait for the answer and press the flash button again in order to conference the call.

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Are there any laws which regulate call recording?

Generally, it is legal to record any conversation where all the parties to it consent (one party consent if all parties are in a state with corresponding law). The U.S. federal law only requires one-party consent to the recording of a telephone conversation, but explicitly does not protect the taping if it is done for a criminal or tortuous purpose. Many states have similar exceptions You may find more information on this subject from

• EU Telecommunications Data Protection Directive • Infowar.com's Law Section: Computer Privacy Community

www.infowar.com/iwftp/cpd/CPD-Library/, www.infowar.com • State-by-state summaries www.rcfp.org/taping/states.html

If I only connect PBXpress to the LAN, can I still use VoIP services?

Yes, assuming that 1. On your network you are using symmetric NAT, which does not

block outgoing UDP connections from the internal network 2. Remote SIP server supports NAT traversal for signaling and

media traffic PBXpress will be able to communicate with remote SIP servers just fine, so you will be able to make outgoing and receive incoming calls.

How the employee can use his IP phone while connected to Internet outside of the office network (LAN)?

In order to provide this sort of service PBXpress must utilize the WAN interface, so there will be a public IP address available. The only change required in the IP phone configuration is that the IP phone has to use this IP address as an address of the SIP server. For all IP phones except Cisco

http://www.infowar.com/



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ATA currently this configuration has to be done manually, so the best way is to:

1. Connect IP phone to LAN 2. Provision it using PBXpress configuration tool 3. Consult the information in the popup window which will appear

during the provisioning. You can obtain there more information about the IP phone and how it can be manually configured. Alternatively consult a manual available from the IP phone’s vendor.

4. Manually change the address of the SIP server in the IP phone’s configuration.

5. Now this phone can be plugged into some external network (e.g. behind the user’s home DSL router).

How the user can seamlessly move with his phone between the office network (LAN) and his home network?

If user wishes to have an ability in the evening to take his phone from his work desk, bring it home, plug it there and use (without any configuration changes); and then do the reverse operation in the morning – it is possible if the IP phone supports ability to put two SIP servers (a main one and a backup one). Then the only required change in the IP phone configuration is to keep your PBXpress LAN IP address as the main SIP server address, but add PBXpress WAN IP address as a backup SIP server. See the previous question for instructions how IP phone’s configuration can be altered manually.

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Appendixes Appendix A - Glossary

ANI - Automatic Number Identification. A feature in which a series of digits, either analog or digital, are included in the call, thus identifying the telephone number of the caller. ATA – Analog Telephone Adaptor. CLI - Calling Line Identification. A range of services in which the number of a caller can be accessed by the recipient. DHCP - Dynamic Host Configuration Protocol. A protocol for assigning dynamic IP addresses to devices on a network. In some systems, the device's IP address can even change while it is still connected. Dialing rules – Set of definitions how the phone number shall be presented (for example should it include country code or no). DNS - Domain Name System (or Service), an Internet service that translates domain names to or from IP addresses, which are the actual basis of addresses on the Internet. DSL - Digital Subscriber Line. A technology for bringing higher-bandwidth information to homes and businesses over ordinary copper telephone lines. E1 - Wide-area digital transmission scheme used predominantly in Europe that carries data at a rate of 2.048 Mbps. E1 lines can be leased for private use from common carriers. Extension - an additional telephone set that is connected to the same telephone line. Firmware - Software routines that are permanently written onto read-only memory. FXO - Foreign Exchange Office. An interface that connects to the Public Switched Telephone Network's (PSTN) central office and is the interface

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offered on a standard telephone. Example: RJ-11 connector that allows analog connection to the central office. HDD - Hard Disk Drive IP telephony - IP (Internet Protocol) telephony is a technology that supports voice, data and video transmission via IP-based LANs, WANs, and the Internet. This includes VoIP (Voice over IP). ISDN - Integrated Services Digital Network is a service offered by most telephone carriers for the transmission of voice and data. ISP - Internet Service Provider. A company that provides access to the Internet through modems, ISDN, T1s, etc. LAN - Local Area Network. This may for example be the network connecting together a company's computers. PBX - Private Branch Exchange. A privately-owned telephone switch, often used in large corporations to provide inside telephone connectivity and access to the PSTN. PSTN - Public Switched Telephone Network. Also know as Plain Old Telephone System, this refers to the world's collection of interconnected public telephone networks designed primarily for voice traffic. RADIUS - A client/server protocol and software that enables remote access servers to communicate with a central server to authenticate dial-in users and authorize their access to the requested system or service. RFC - Request For Comments. An RFC is a standard document describing protocols, systems, or procedures used by the Internet community. SIP - Session Initiation Protocol, SIP is an Internet Engineering Task Force (IETF) standard protocol for initiating an interactive user session that involves multimedia elements such as video, voice, chat, gaming, and virtual reality. SMTP - Simple Mail Transfer Protocol – A protocol used to send e-mail on the Internet. SMTP is a set of rules regarding the interaction between a program sending e-mail and a program receiving e-mail. T1 - A type of data connection able to transmit a digital signal at 1.544 megabits per second. T1 lines are often used to link large computer networks, such as those that make up the Internet.

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VoIP - Voice over Internet Protocol; the practice of packet based networks instead of the standard public switched telephone network to send voice data. WAN - Wide Area Network. A network connecting computers within very large areas, such as states, countries, and the world. WAV - WAVeform sound format. Microsoft's format for encoding sound files, which therefore carry the wav extension.

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Appendix B – Supported IP phones and analog phone adaptors

A PBXpress compatible phone or adaptor must satisfy the following requirements:

• It must use SIP protocol. • It should have a G729 codec.

Below you can find a list of phones/adaptors which have been successfully tested with PBXpress.

Sipura IP phone

Manufacturer: http://www.sipura.com Models: SPA-841 (recommended) Firmware upgrade: http://www.sipura.com/support/index.htm Links to order: http://store.voxilla.com Pros:

• Price • Automatically configured by PBXpress • Hands free connector

Cons: • No built-in Ethernet hub/switch

IPTEL TT phoneiptel.org's TT phone

Manufacturer: http://www.iptel.org Product page: http://www.iptel.org/tt/ Firmware upgrade: http://www.iptel.org/tt/fw/ Links to order: Contact [email protected] Pros:

• Price • Integrated Ethernet switch • Automatically configured by PBXpress

Cons: • No hands free connector

http://www.sipura.com/

http://www.sipura.com/support/index.htm

http://store.voxilla.com/

http://www.iptel.org/

http://www.iptel.org/tt/

http://www.iptel.org/tt/fw/

mailto:[email protected]

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Sipura SPA-2000

Manufacturer: http://www.sipura.com Models:

• SPA-1000 • SPA-1001 • SPA-2000 • SPA-2100 (experts only) • SPA-3000

Firmware upgrade: http://www.sipura.com/support/index.htm Links to order: http://store.voxilla.com Pros:

• Price • Automatically configured by PBXpress

Cons: • It is an adaptor and so its features are accessed via flash and *NN

codes • No built-in Ethernet hub/switch

Cisco ATA-186

Manufacturer: http://www.cisco.com Models:

• Cisco ATA 186 • Cisco ATA 188

Firmware upgrade: http://www.cisco.com/cgi-bin/tablebuild.pl/ata186 (password protected, 1-year subscription for $25) Links to order: http://www.loudpacket.com, http://froogle.google.com Pros:

• It is a CISCO • Automatically configured by PBXpress • ATA-188 has a built-in Ethernet switch

Cons: • It is an adaptor and so its features are accessed via flash and *NN

codes • Size and price • There is a charge for firmware upgrades • Difficult to find • Please be aware that it needs SIP software

http://www.sipura.com/

http://www.sipura.com/support/index.htm

http://store.voxilla.com/

http://www.cisco.com/

http://www.cisco.com/cgi-bin/tablebuild.pl/ata186

http://www.cisco.com/cgi-bin/tablebuild.pl/ata186

http://www.loudpacket.com/

http://froogle.google.com/

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Cisco IP Phones

Manufacturer: http://www.cisco.com Models:

• CISCO IP PHONE 7905G • CISCO IP PHONE 7912G

Firmware upgrade: http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 (password protected, 1-year subscription for $25) Links to order: http://www.loudpacket.com, http://froogle.google.comPros:

• It's a CISCO • Automatically configured by PBXpress • Reasonable prices • Optional power over Ethernet • Design fits home and office • Featured are accessed directly

Cons: • There is a charge for firmware upgrades • Difficult upgrade procedure • Difficult to find • Please be aware that it needs SIP software but does not need a call

manager license • The power supply is a separate item • No dedicated hands-free connector

http://www.cisco.com/

http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905

http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905

http://www.loudpacket.com/

http://froogle.google.com/

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Appendix C – Auto-configured IP phones and adaptors

PBXpress is capable to automatically detect and auto-configure the IP phones and analog phone adaptors listed below

Sipura

• SPA-1000 • SPA-1001 • SPA-2000 • SPA-2100 • SPA-3000 • SPA-842

Cisco

• ATA-186 • ATA-188 • 7905G • 7912G

IPtel

• TTphone

PBXpress Admin Guide

Documents

Transcript of PBXpress Admin Guide