1 Welcome to ayrix

Highlights

The Ayrix is a new communication base integrates latest communication and collaboration technologies. We briefly explain the highlights of our product.

• The design has been revised through the practical use of older systems and its modular structure is based on intuitive user guidance.
• Ayrix has very small footprint on memory and CPU
• Ayrix contains a fully automated upgrading service. That limits your maintenance care extremely
• A simple, integrated licensing system that allows you to order online extensions and track your telephony services.
• The Ayrix is highly configurable by its web interface
• The main menu only contains the most important points for operation and configuration.
• It contains many built-in templates or can simply be duplicated from existing entries.
• No vendor lockin. The Ayrix uses a standard SIP protocol. You can select your SIP terminals from a broad range of suppliers.
• Integrated NAT and STUN support for cloud hosted environments
• The system supports multi-languages

Benefits

• A simplified installation through a disk image enables the installation of the operating system and application in one work step
• The image can be installed on a bare metal or virtual machine of your choice. With a low resource occupation. the Ayrix uses less than 1Gbyte ram and less of 20Gbyte disk space.
• Simple configuration through an intuitive menu overview, while extensive functions are available behind it. The system allows that end users can manage and carry out configuration adjustments themselves.
• Ayrix contains a fully automated upgrading service. That limits your maintenance care extremely. The advantage of Ayrix over its competitors is that the user management and telephone configuration are independent from each other.
• The Ayrix is highly configurable by its build-in web access, configurable for administrator and users, offering to do phone configuration by themselves.
• Security based Intrusion Prevention System (IPS) and Intrusion Detection System (IDS) system as well certificated encryption for web access and phone calls, prevents unwanted access.
• Security by IP black- and white-listing, Geo-Location and Number identification.

IT-Requirements

Recommended System requirements:

An Intel Core server with virtualization software. We recommend VMware, KVM or XEN.

• CPU Family Intel i3 (64 Bit)
• Number of vCPU’s 1
• Memory 1 GB
• Storage 20 GB available storage space,

For higher reliability we recommend a raid Installation, see chapter 16.2).

• Internet connection

For an Ayrix installation you need a virtual server or physical server hardware.

Check out your firewall settings. Open the relevant ports for correct operation. Used ports are listed in the appendix.

Steps to install your own virtual Ayrix

• Go to the website: www.ayrix.ch or www.ayrix.com
• Select on the website your preferred language: German or English
• Navigate to the button” Download”, Click on it, a pop-up window opens.
• This window shows the requirements. Click Download again to get the free ISO disk system installer
• Open your virtualization software. Create a Linux Centos 7 server with minimal requirements or better. Include the Ayrix ISO disk installer and assign a NAT network interface
• Start the virtual machine and follow the installation instructions of Centos. Wait until the configuration screen appears
• Enable/check the general network setting: “Automatically connect to this network when it is available", in “Network & Hostname” settings. Press “Save” and “Done”
• We recommend installing a RAID disk system. For a software RAID installation see chapter 16.2 Raid Configuration
• Set the correct (NTP) Time zone for your country
• Depending on your internet connection and System performance, the installation takes up to 20 minutes. When the installation is almost complete, the Ayrix telephone system will automatically be updated to the latest version over the internet and initialized without any notification on the screen. Be patient

Press “Esc” button to keep track of the loading process*. When finished, the system restarts and the login screen with the IP address of network access appears

• Open a web client at your host and enter:

• The registration and activation page opens

  • If time-out, check your browser security settings: Java Script should be enabled

• Fill in the registration form to register your Ayrix PBX. Remember your initial password

• A confirmation e-mail including license key and username is returned. Check your mailbox

• Now enter or copy your license key in the activation tab. The license validation is limited to 3hrs or 10800 seconds after sending the key request.

• After license activation your PBX is ready to login. Login and password are case sensitive

• Congratulation! Your Ayrix telephone system is operational and ready for telephony configuration

Important license notice:

Depending on your license bundle, some features may be not present or have limited functionality.

*Reseller customers have a predefined registered license bundle. Skip the registration form and continue with the activation tab (see also License).

Already too much? Contact your Ayrix sales representative! We offer you a high grade, fully managed Ayrix-ready-to-go virtual PBX (SaaS) in the Ayrix-cloud.

1. It does not matter static or DHCP due installation. After installation you can change your network settings via the web access without losing license. In menu System – Network – Interface.

2. By pressing the ESC key due the Ayrix download. You should see jobs from /etc/rc.d/rc.local followed up with a number of modules, So you can keep track if the installation is going correctly.

3. (for example, testPC fails on udev kernel Manager -a device Manager– and ends up with a normal screen without any started services > Failure in Bios settings).

Technical Support

We offer charged support services on the released versions of Ayrix PBX.

The support is at level best effort unless Service level agreements are agreed with a support contract with the supplier or distributor. Contact our customer service at support@ayrix.ch.

On beta releases there is no technical support. In case of a failure sent a bug report form via the bug bounty program on the Ayrix website under: www.ayrix.ch

System configuration guidance

The system configuration depends strongly on the customer individual needs. If you are not confident with telephony configuration and its dependencies, we guide you stepwise through a basic configuration setup.

Minimal requirements:

• One (soft-) SIP phone

• A SIP Trunk agreement and setup data from your service provider (TSP).

   

1 Quick Setup Guide

With these 5 STEPS and 2 checks you will get your Ayrix configured and operational.

1. Quick system check
2. Administrator account completion
3. SIP Trunk assignment
4. Direct inbound-Dialing (DiDs)
5. Extension assignment
6. Extension connection
7. Testing connection.

1. Quick system check

In front of the main menu titles icons are present. These Icons will appear in different menus when there is a relation between menu items.

Go to the menu Settings > System > Language and Region.

• Language, check if your preferable language is active

• Time zone check if the NTP clock is set to the right Region to synchronize the local time zone.

• E164 settings, set your country dependent dialing rules. See also chapter E164 Numbering scheme explanation.

   

2. Administrator account completion

Go to the menu Security > User Authorization. > Click on the tab “User”. Select the 1st. line, by default the Administrator. The user data appears in the vertical editor block. Expand the administrator data with the required language, change the administrator password to improve security, and optionally enter his telephone number. Now the application presents itself in the correct language. Leave the other data as is, for the time being. The system indicates with a red line if your password is too weak. You cannot save your entry.

Press “save”.

We strongly advise to add a 2nd administrator entry in case of emergency. Memorize your admin password (-s).

The “Save” action behaves differently depending on the topic you are working on.

3. SIP Trunk assignment

Go to the menu Routing > Trunk > press tab “+ add SIP Trunk”. A pop-up window opens.

Select your country, select your language, and check, if your telephony service provider (TSP) is listed in the template. Select your provider and press "Add".

If your service provider is not listed, enter in the field Service Provider “Default”.

Press. "Add” to assign a new configuration. The window “Configure SIP Trunk” opens.

In the tab basic settings enter a suitable name for your trunk. For example:” Trunk_To_TSP”

Enter in the authentication field, the SIP user ID and password you received from your TSP

Enter the max. number of simultaneous calls. Value “0” means maximum simultaneous calls possible.

The SIP user ID is number. That number does not necessary match with your Caller ID.

Depending on your selection some entries are already filled in. Check and complete the fields according the TSP`s specifications.

Codecs

The codec is the digital conversion of the voice into a data stream. The SIP protocol negotiates the codec to be transmitted between the parties.

Commonly used codecs are: G711a for Europe and Japan, G711u for the USA and G722 for broadband transmission. Read the explanation about codecs in chapter 6.2.2 Codecs

In the window with the basic settings, select "Codecs" in the right vertical configuration area and move the required codecs from the "Available codecs" list to the "Selected codecs" list.

You can rank the codecs by quality. Applicable codecs list should be given by your TSP. If your TSP did not provide you with a list of preferred codecs, move all available codecs to the selected list. The selected codecs are also used for extensions by default.

Tab through the next pages: SIP options, Inbound headers, and Outbound headers. Check and complete the entries according to the configuration data from your TSP. Leave the settings that are not relevant for the time being unchanged.

Press “save” on any page you added or changed entries.

Optionally, you can add a caller ID and regular expression for the called-ID. Regex offers you to change the displayed number of the calling or called subscriber, if this display is not overridden by the extension CID configuration.

Leave these fields empty, for the time being.

Press “save” and press “(<)” to return to the trunk list

In the trunk list you should now note that the activation switch of your trunk is set to "active", and "REGISTERED_ [DAY]" is displayed in the status field.

Congrats again!

If the status shows any other message, check your settings, and correct them. Possibly that your TSP has not yet activated your trunk. Contact your TSP. (see also Trunk Status)

4. Direct inbound Dialing (DiDs)

DiDs addresses your system. DIDs enable incoming calls to be switched through. Depending on the number of trunk lines you have ordered from your TSP, you have received a single number or a number block or number range.

Add the number range exactly as it is specified by your provider, see also Numbering scheme E 614 Explanation.

Select your trunk and press tab DiDs. Assign the number (-block) given by your TSP. and press “save”. Finished!

The "Inbound" window shows the incoming dialing rule list. It shows the relationship between the DID numbers and the destination.

The “Outbound” rules window guides you through a trunk arrangement to the public network.

By assigning the SIP trunk (see step 3. Assigning the SIP trunk), outgoing rules are set autonomously with standard values ​​(= no restrictions). As a result of these standard rules, you can now dial out to the public network without any restrictions.

5. Extensions assignment

Click on "Extensions" and press the tab key "+ New" at the top right. The configuration panel “Basic settings” opens.

Enter the extension number you want to use, first and last name of the user and his e-mail address for e-mail notification.

Assign a user ID and password. A strong password is already suggested. Check the password by pressing the (eye) "ᴑ".

Add a Calling Identifier (CID) number that is being displayed at the called party.

(Optional) Select the DID number you want to assign for direct inbound dialing,

press” save”. A confirmation message “Data Successfully Saved” is displayed

Go to the Tab Advanced Settings

On the right panel, scroll down to the option: Number of Simultaneous channels

Set the value to 2. This enables your extension with the RTP settings immediately.

Press “Save”

(Optional) Go to tab: Codecs:

These codec settings overwrite the selected of the trunk codec list. Move the preferred codecs from the list of available codecs to the list of selected codecs and click "Save".

Return to the list of assigned extensions. Press "(<)".

6. Extension connection

The configuration for a SIP telephone differs depending on the provider and model. Therefore, we can only explain the extension configuration in general. This configuration is similar to the extension configuration in the Ayrix PBX.

We recommend installing a simple softphone directly on your host for testing. For example: an Estos, Groundwire or Bria.

Read the instruction manual of your SIP Telephone carefully. The telephone should be delivered with a loaded firmware.

Plugin the telephone on a POE switch, or with a connected secondary power supply adapter the telephone becomes operational.

(Almost all) Physical SIP phones are configurable via an IP web console. Check out your manual for easy access.

Go to your phone's website or directly into the phone's menu

• Go to Phone Settings> Administrator Settings>
• Set up the network configuration, depending on the network configuration, set the device to DHCP or give it a static IP address.
• check whether you can ping the phone from your host.

Set-up the Phone settings menu: SIP Server Configuration:

• Assign the DNS name or IP address of the Ayrix PBX

Setting up the line configuration:

• Assign a phone line with a friendly name and the same authentication parameters with user ID and password as the assigned extension in the Ayrix PBX.
• UDP is a frequently used protocol for SIP telephony. With the "Transport" option; Select: "Only UDP".
• Save the configuration and return to the main menu on your phone.
• From the Ayrix menu, open the Extensions menu. and refresh the webpage.
• Check whether the point of your assigned extension has changed from red to green. (= Phone registered).

Your phone is now ready for use. Well done!

7. Testing connection

You have now completed the basic configuration. Test your telephone connection. You should be able to make and receive calls using your assigned trunk.

Resume Ayrix Set-up.

We guided you through the installation and basic configuration of the Ayrix PBX. To keep the setup simple, less relevant fields have not been described.

In the next chapters we fully describe all fields, settings, and functions.

Too much for you? Do not hesitate to call your Ayrix representative.

We offer hosted Ayrix and expert on-site installation service.

Housekeeping: After assigning and saving data of, extensions, queue, ring group or speed. Dialing codes, the (demo) Ayrix does not immediately take over the entered values into the running process. Other functions do.

This might take up to several minutes up to an hour (failure? or dependencies?)

For example, to speed up activating the RTP codec stream on the extensions I edited in the tab advanced settings the channel. By pressing save the RTP codecs where immediately switched on.

2 Dashboard

The dashboard shows an overview of the system status, about:

• memory usage, disk occupation and CPU load of your server
• the number of active calls and call duration
• An overview of the call waiting times of the queues, as well as the selection per queue
• The number of incoming blocked calls by IP addresses in time, caught by the blacklist of the BruteForce Protection module
• the PBX system services, indicated with h a blue (informative) green operational), yellow and red (not operational) status
• Information about the systems configuration, as internal network IP, External (gateway) IP address
• The actual Ayrix PBX software version
• The maximum number of licensed extensions
• The time zone, for calendar usage

It is strongly advised to allow all system users at least read access to the dashboard (see also User Authorization)

For example, the block “CPU load” update line, other fields (information and status block) are not being modified. They become blank, Web is a stateless protocol, just manual a screen refresh overcomes that.

If read/write /delete of the dashboard is disabled, the user immediately gets an error message on opening the web page. That not nice, as you cannot click it away.

3 Extensions

Extension configuration is the main feature of any telephone system. The Extension module allows the user to add and configure Phone terminals on the Ayrix. Each extension connected to the PBX system and has a number.

Extension calls can be made within the company without incurring any costs.

The assigned extensions are listed in the extension window. It has three tabs: “Add”, “Edit”, and “Duplicate”.

You can use the search bar to view a search term to search by first name, last name, or distance. The search term is not between uppercase and lowercase letters.

The extension window provides an overview of the availability of a user in the form of a green dot, or in case of unavailability, a red dot. Further the name and the first name, the e-mail address and the assigned phone number. Under Phone Model the currently registered phones as well as the corresponding IP address and the port can be seen.

Adding a new Extension

To add a new Extension, click the "New"-button to display the “Basic Settings”

• Enter a unique extension number and fill out the required user information
• Enter an Authentication ID and a password, that are required to register a SIP phone or softphone client
• Enter a CID ("Caller ID") for this extension number
• Enter an e-mail address for e-mail messages
• Optional: Add any DID number for inbound calling.
• Click the "Save"-button

CID

The "CID" is the number that will be displayed to the called party. This CID field only accepts numbers.

Other CID entries in the Ayrix with “prefix” field also accept alphabetic characters. With this prefix you can create a name-number display, for example: "HELP800".

DID

The Direct Inbound Dialing (DID) number allows an extension to be reached directly from outside the PBX.

Editing an existing Extension

To edit an existing extension, click on the Line and then click the "Edit"-button to display the configuration or double klick the entry in the list of extensions

• Change the Information you would like to modify
• Once ready click the "Save"-button
• Return to the list panel by pressing (<) in the top right

• Mandatory fields need data entries to save the record. A red line is shown if missing or wrongly written.

   

3.3 Extension features

To add extension features, select an Extension and click the "Edit"-button. There are nine tabs on the top of the screen.

• Calendar
• Voicemail
• Forwarding
• Extension Routing
• Follow me
• Geo CID
• Advanced Settings
• Codecs

Extension features are described in the next paragraphs:

3.3.1 Calendar

The Calendar feature allows users to link to an external calendar to setup a calendar-based calling behavior. That calendar is synchronized.

Supported calendars are:

• Microsoft Exchange Web Service Calendar
• Microsoft Exchange calendar
• CalDAV calendar
• iCal calendar

Destination if appointment

The "destination at appointment" determines the forwarding rule for incoming and internal calls. If there is an appointment entry in the calendar, calls are routed to the destination. If no forwarding is set, an absence tone is given. The Calendar distinct between sub-status 0=Free, 1=Tentative, 2=Busy.

3.3.2 Voicemail

Voicemail is used for recording an audio message when the called party is not available. To do so, the user Interface Voicemail offers to configure the voicemail feature, as well to manage and listen to the recorded messages.

Voice mail messages can be listened by phone or sent by e-mail as a .wav file attachment.

A second way to manage voicemail features is accessing your voicemail box by dialing your own number.

A third method accessing the voicemail box remotely by dialing the destination, wait for the greeting message and dial “*” followed by the pin code.

The Voicemail announcement language is by default English. Set your preferred language in tab “Advanced Settings” – “Language”.

The window contains a configuration panel on the left and the Voicemail administration panel on the right.

The first two folders can be managed on Ayrix in the administration window.

Voicemail list Panel
play download move	Play download delete

Voicemail terminal menu guidance

Dial your own extension number on your phone to listen to your voicemails.

You have two options how to listen your voicemails.

The first option is that you dial the number. The voicemail replies about stored voicemails, after you are guided through a navigation menu to handle them.

The second option is that you dial remotely to the destination, when the voicemail starts you interrupt by pressing “*” and enter the Pin (password).

The terminal menu guidance offers you to record and activate new messages on the phone.

These recorded messages are not! visible on the Ayrix “Voicemail”.web page..

Fig. Call flow Voicemail

3.3.3 Extension Routing

The extension routing module allows you to forward calls in specific states of the extension. The destination can be an alternative destination, a voice mail, an announcement, or just busy tone. Ayrix distinguishes between internal call forwarding and external call forwarding. In this window you configure call forwarding for those specific statuses.

3.3.4 Outbounds

Not only for inbound, but also for outbound routing we can configure extension-based routing rules. For example: A “priority or emergency trunk. Only available for the CEO or chief officer. Whenever the first trunk is occupied or blocked, he can dial out over a 2nd priority trunk.

“Outbounds” extension routing rules have priority over the general Outbound routing rules, (See chapter Outbound).

This Outbound panel is to configure an outbound routing table per Extension.

In that table you can reconfigure a dialing digit numbers that are sent over a trunk by adding or stripping digits from the dialed pattern.

For example: anyone dials a number including country code in the own country and your provider only accepts national numbers in the country. A digit string of “00-41-56-520-xxxx” is sent. On the pattern “4156” we strip the country code for Switzerland “41”. Because of the country settings for national numbers a leading zero is placed. So finally, 056xxxxx is sent., see System Language & Region.

Or in case you want to select on a specific trunk, for Least Cost Routing (LCR) or “In-company” dialing.

Creating outbound rules, based on the dialed digits, you can select a particular trunk due dialing.

For example, on 00-4156-520-xxx (Switzerland) you select Trunk 1, on 00-31-26-311-xxxx you select trunk 2 (Netherlands).

We need to define a trunk access code to distinct between the trunks.

For example, selecting trunk 0 will we need a digit “8”, selecting Trunk 2 we need a digit “9” We add a prefix in front or our Pattern. “4156” we add a prefix “8” and for “3126” we add a prefix “9”.

3.3.5 Follow-me

With the Follow-me function, the user can create a ring group for his own extension (i.e., fixed and mobile numbers). A forwarded call will ring on all these destinations. The forwarding is active on all incoming channels.

The difference between a ring group and Follow me is, when a call is being answered on an internal destination, the originated extension is also occupied. In a ring group the originated number is free. An exaptation is, when the incoming call is answered by a mobile extension, the originated extension is still free and does not go into busy state.

The user can add extensions to the Follow me group by clicking the "New"-button.

You can add external destinations, by selecting from the external number list, and enter the number in the right lower corner of the selecting panel. Enter your external number (including trunk access codes) and press “OK”

Field description

3.3.6 Geo CID

Geo CID allows an Extension user to send a country-specific Caller ID, depending on the E164 country and region settings. The Ayrix is limited to one (“1”) Geo CID table. This Geo CID allowance list can be configured in the menu PBX > Geo CID. This feature is useful where your company has branch offices in different countries or regions and want to serve your customers from headquarters. Per extension, this list can be either enabled or disabled.

Important Notice: Be aware, Ayrix has different! Geographical based functions:

• Geo CID: Send a country/region specific Caller ID (CID)
• Geo Restrictions: Restrict incoming and/or outgoing calls with specific Caller- ID/Calling ID.

3.3.7 Advanced Settings

This feature allows the user to enter additional settings for his extension.

NAT: Explanation

If the caller is in a private subnet behind a NAT gateway, the "Invite" request contains multiple private IP addresses (Listing 2) that cannot be reached by the other side.

The gateway, which carries out the address translation on layer 3 (IP) or on layers 3 and 4 (TCP / UDP, NAPT: Network Address Port Translation) in the packets, is inherently unable to handle IP addresses in higher layers to recognize and modify, but a STUN, -stunning is server is capable to translate these SIP IP addresses from private into public.

The Ayrix can be installed as virtual machine, the Ayrix does have a “build-in” STUN server for this operation. See also System - Network – tab Connectivity.

DTMF Mode

SIP offers multiple methods to transmit Dual Tone Multi Frequency (DTMF) tones to the opposite party during the call. The DTMF Mode selects the protocol of digit- transmission during a call. These DTMF tones can be detected by the IVR or a contact center at the opposite site.

Digits can be transmitted as in-band (tones) in the media stream, that you will hear, or transmitted as Info- or short info element, in the SIP message, (outside the speech channel, you will not hear anything).

*Notice: being interoperable sending DTMF, both sites, the Ayrix as well the SIP terminal needs to be configured correctly.

Example: Basic settings of DTMF on a Grandstream G2515:

See also Section: SIP terminal features

3.3.8 Codecs

To get a transmission of your voice, the telephone needs to convert your voice into a Media stream or RTP.

Due call setup the SIP protocol negotiates about the codec to be used between the parties. The first codec that matches on both sides is chosen. Through the years several codec flavors have been developed. If you assigned a SIP trunk on with a Codec selection, already on beforehand the extensions got already that default codec selection available. However, for every Extension you can create a preferred Codec selection. The codec window shows a “Available Codecs” list and a “Selected Codecs” list. You can move and rank your selection from the available to the selected list.

3.4 SIP terminal features

In conjunction with the Ayrix you will use telephones, more precisely SIP terminals to call or being called. Nowadays SIP terminals have many phone features to make call handling easier. Many functions of the Ayrix are combined with the SIP terminal. We give you a brief overview how to program function keys.

Attention: We describe the functions in general terms. The availability and use of function keys depend on the terminal manufacturer, type and model.

Probably your terminal is so feature rich, that it has much more as the Ayrix offers. We limit our description of these SIP terminal features to the most relevant to assist you to get these functions operational. For a detailed description check your manual and visit the manufacturers website for further explanation.

3.4.1 Function Keys

Many terminals have function keys. We describe in general terms features that can be programmed under a function key.

• Busy Lamp Field & Call Pickup
• Day / Night BLF
• DTMF Mode
• Forward
• Intercom button assignment
• Line registration
• Multiline /channel registration
• Pool & Park + Orbit
• Voicemail indicator

3.4.2 Busy Lamp Field & Call pickup.

Depending on the telephone features a Busy Lamp Field (BLF) can be assigned. BLF is a lamp indicator of another ringing terminal. A call pickup is a 2^(nd) function. A fixed Feature Code for call pickup is *8. On some terminals BLF and Call Pickup are combined, so that you can take over a call of a colleague with a single button.

Example BLF:

Example Pickup/BLF on SNOM715

3.4.3 Day / Night BLF button assignment

Depending on the telephone features a Busy Lamp Field (BLF) can be assigned. A named extension is added due assignment, for example. “dnLamp_1”. That [name] is the reference at sip: [name]@IP_address_or_FQDN on BLF button assignment.

De-/activating by pressing the button switches the Day / Night lamp status at the webpage. The lamp switches from red to green. At the SIP terminal (SNOM715), the status change is announced and the lamp lights-up shortly on (de-) activation.

Example:

Day/ Night assignment on the 2^(nd). (BLF) button on a SNOM715 Telephone set.

<sip:dnLamp_1@demo.ayrix.ch;user=phone>

3.4.4 DTMF mode

About the different Dual Tone Multiple Frequency (DTMF) modes we described already in the chapter Extension-Advanced Settings. (Chapter 3.3.7)

First you configure the correct mode to use the right DTMF mode.

*Notice: Whenever your choice of DTMF tape, In-In-band, RFC2833 or SIP_INFO element, both sites, Phone and PBX, need to accept that mode. Example: Basic settings of DTMF on a SNOM D315: Identity “Number” --> SIP --> DTMF via SIP INFO --> on --> Apply

DTMF Function Key

Second, it’s very handy, if you call a Contact Centre, that leads you through an IVR call flow, by dialing DTMF digits, as is present in the Ayrix, (see chapter IVR) - to create a function key to enable/disable DTMF tone sending.

3.4.5 Forward Button

One of the most wanted features: setting a forwarding, but which code? – A Follow-me button helps you, by pressing a function button, indicating that your incoming calls are being forwarded or not. You could also configure a Day/Night (Chapter 3.4.3) and use this as a “follow me” function.

3.4.6 Intercom button assignment

If you want to user intercom on the Ayrix, the terminals must also support the intercom feature. Check your feature list of your telephones.

Example intercom receiving. SNOM 785 advanced configuration settings for intercom receiving.

Example: line key settings for intercom sending. Intercom: setup or disconnect an intercom connection. Push2Talk: Establish the connection as long the Push2Talk key is pressed.

3.4.7 Line registration

Generally, your phone subscription is registered on a single registrar. Ayrix might just the only registrar of your company. SIP phones are capable subscribe to multiple registrars. The Line registration button helps you to select the number to use.

For example: In a case where 2 users want to use the same terminal but have a different registration:

3.4.8 Multiline / Multi channel registration

Common in the extension assignment is that you create a single subscription on a single line.

On the Ayrix in the tab “Advanced settings” you can extend, “Maximum incoming lines” and/or “maximum simultaneous channels”. See chapter “Extensions -> Advanced settings”.

3.4.9 Voicemail indicator

On the terminal you can enable voicemail indicators. If a new voicemail message has been recorded due your absence, there are two (“2”) indicators, to inform that you received a new message. MWI = Message-waiting Indicator

1. By dial tone: lifting the handset, you get a shuttered dial tone on MWI. Intend for non-display extensions as analogue phones used with a SIP converter
2. By a lamp indicator: A voicemail lamp can be programmed at the terminal to indicate the new message.

Example: SNOM 715, separate voicemail button/lamp. It lights up when new voicemail message is received.

4 Provisioning

4.1 Ayrix Scan & Go App

With the Ayrix Scan & Go App you can profision Snom phones with on touch.

1. Configure the Ayrix PBX and with all extensions.

2. Download and install the Ayrix Scan & Go app from the Apple App Store or Google Play Store. Please note that the device must have a camera.

3. Klick the "App Login QR Code" Button A small window with a QR Code will open. Scan this Code wit the Ayrix Scan & Go App. Your Mobile Device is not coupled with the Ayrix and will show you all the Extensions available on the Ayrix.

4. Scan the Mac address on the packaging of a Snom phone with the Ayrix Scan & Go app and then assign it to an extension on the mobile phone in the Ayrix Scan & Go app.

   

5. You have now successfully connected the phone to the Ayrix.

6. Unpack the phone and connect it to the network. The phone will now automatically get the desired configuration from the Ayrix.

Provisioning is the feature to provision terminals with the latest firmware and configuration files. Normally every terminals-ethernet interface has an unique MAC address, like [08:00:8c:2a:00:ab]. That MAC is manufacturer specific. The firmware and Configuration file is combined in a template. Depending on the availability of the terminal type, templates can be downloaded form a provisioning server or uploaded from a local pc.

With the next menu options “4.1 Template” and “4.2 Phone” we create that relationship between MAC address and firmware template.

To send a provisioning request to the manufacturer, the Ayrix and firewall and/or router need a opened firmware provisioning IP-port. See Appendix: Firewall port list.

4.2 Template

The Template menu offers to create terminal-model specific template. A number of prepared templates can be downloaded or uploaded from the local PC. On the template window the left site three tabs are available, +New, Edit, Duplicate.

Field description

4.3 Phone

The phone menu offers to create the relationship between created template with the MAC address of your terminal, as written in chapter 4.1.

Field description

5 Callflow

The menu Callflow offers you to configure call routing features. The following features are described:

• IVR
• Queue
• Ring group
• time condition
• conference
• Intercom
• Hold

5.1 IVR

The Interactive Voice Response (IVR) feature allows the PBX to interact with the callers using voice recorded messages and DTMF tones input via the calling device keypad.

5.1.1 Creating a new IVR

To add a new IVR click the "New"-button to display the form

• Enter a name and a timeout
• Optional: Add any DID numbers
• Click the "Save"-button to add the IVR

5.1.2 Drawing the IVR

To draw the IVR click on the "Edit"-button to display the form and go to the "IVR" tab to display the drawing board of the IVR.

• Simply Drag & Drop the elements needed from the list on the left into the drawing grid to add them to the IVR.
• To connect two elements simply click and draw the blue points at the edges of the elements to one another.
• To configure the element, click on the gear-icon to open the settings dialog
• To loop an element, click on the loop-icon. An active loop is shown with a green loop-icon
• To delete an element, click on the red cross-icon

5.1.3 IVR elements

Start

The element "Start" defines the starting point of the IVR. A call routed to this IVR by DID or other processing will start at this point.

PlayFile

The element "PlayFile" plays a selected announcement and then optional routes to the next element.

Get Key

The element "Get Key" expects a key (0-9) as input to route to one or more different elements. Minimum one input key is required to route to the next element. A key is a single digit input from the caller device.

• To add a key, click the gear-icon to open the settings dialog
• Enter a timeout and select an announcement that will be played when the caller reaches this element,
• Select a key from the dropdown at the bottom of the window and optional add a comment
• Click the "Add"-button to add the key-match
• Click "OK" to close the window

Now the element shows one or more keys, depending on the number which has been added, with a blue point at the edge which now can be connected and routed to new elements.

Get String

The element "Get String" expects a number string as input, for example "1234", to route to one or more different elements. Minimum one input string is required to route to the next element. A string is a multiple digit input from the caller device.

• To add a string click the gear-icon to open the settings dialog
• Enter a timeout and select an announcement that will be played when the caller reaches this element,
• Enter a string at the bottom of the window and optional add a comment
• Click the "Add"-button to add the string-match
• Click "OK" to close the window

Now the element shows one or more strings, depending on the number which has been added, with a blue point at the edge which now can be connected and routed to new elements.

Match String

The element "Match String" expects one or more matching number string to route to one element or to a second if the number string does not match. "Yes" for matching and "No" if the input does not match. Minimum one input string is required to route to the next element. A string is a multiple digit input from the caller`s device.

• To add a string click the gear-icon to open the settings dialog
• Enter a timeout and select an announcement that will be played when the caller reaches this element,
• Enter a string at the bottom of the window and optional add a comment
• Click the "Add"-button to add the string-match
• Click "OK" to close the window

Now the element shows one or more strings, depending on the number which has been added and two blue points, one for matching "Yes" and one if the input does not match "No", which now can be connected and routed to new elements.

Destination

The element "Destination" route the call to a defined destination and ends the IVR.

System Record

The element "System Record" lets the caller record a message and ends the IVR.

Adding a System Record

Email

The element "Email" sends a template e-mail and optional routes the call to another element.

5.2 Ring Group

The Ring Group feature allows to create a virtual extension that will in parallel forward the call to all available extensions and phone numbers within the ring group. As soon as the call is answered by one of those extensions, the call legs to all other extensions will be dropped. A listed panel and a vertical configuration panel are shown.

5.3 Queue

Call queueing is a call flow to distribute incoming calls within a group of call agents. The queue feature allows the creation of ringing queues. The Ayrix distinct between static and dynamic agents:

Static agents

Static agents are assigned via web access to the related queue. The agent state of a static agents is that they are immediately logged in to the queue. Agents can be logged on and off with a feature code.

Dynamic agents

Dynamic agents are not assigned to a queue, but they can dynamically login to a queue with feature code “Register Call agent in Queue”, default (see also feature codes). Any user can be in the role of a dynamic agent by dialing the “Register Call agent in Queue” +queue number.

Queued calls can be routed to a next destination or gets busy tone. The Queue panel consists the following tabs.

• Basic Settings
• Agents
• Announcements
• Advanced Settings

The queue page is a listed panel. On the Tabs “Add”, “Edit” “Duplicate”, you can add, - edit a Queue group. By pressing “Duplicate” you simply copy an existing group for further configuration.

5.3.1 Queue Polling strategy

You can choose out of five queue polling strategies for call routing,

5.3.2 Basic settings

The tab basic settings allow you to configure queue basics.

Field description

5.3.3 Agents

The tab Agents allow you to assign extensions who are answering the queued calls.

5.3.4 Announcements

The tab announcements allow you to configure an announcement for the caller and agent.

Field description

5.3.5 Advanced settings

Advanced settings tab is to enable/disable queuing features, Recording- and timer settings for call queuing.

5.4 Conference

The Callflow feature “Conference” creates audio conference rooms. You can create several Conference circuits for internal and external attendees. The maximum number of attendees is limited to maximum of the extensions and trunk lines that are available. When you dial in the pin number is requested.

By selecting a Conference room from the list, you can prepare and send E-mail invitations to attendees. Prepared invitation templates for conferencing are available under: “System”–> “Email Templates”.

The conference window shows you a list panel of the conference rooms. The conference window contains four (“4”) tabs to Add, Edit, Duplicate rooms and Sent invitation mail.

With the search bar you can enter a phrase of an Extension or name to search the correct room. The search entry needs an exact match. It distinct between upper- and lower-case characters.

Pressing tab “add”, a new conference window opens. That window contains further tabs to create a new Conference room. To configure the conference room following tabs are present.

• Basic Settings
• Conference Room Settings
• Recording
• Advanced Settings

5.4.1 Basic Settings

Field description

5.4.1.1 Terminal conference control menu

Conference room settings are set on the tab Conference.

Field description

5.4.2 Advanced settings

Field description

5.4.3 Send Invitation mail

To invite your conference attendees, you can prepare an e-mail to send out. Pressing tab “Send invitation mail opens a window with the prepared template. The selected email is based on your Language selection in the conference Room configuration.

In “System” menu-> “Email” - > tab” E-mail templates” prepared email templates are available. You can add your own email template as well.

xxxxxxxxxx
Get-ExecutionPolicy -List
Output: LocalMachine Unrestricted -> ansonsten umstellen

Set-ExecutionPolicy Unrestricted

5.5 Time Condition

The Time condition menu enables you to prepare calendar-based time conditions, in the call flow for incoming trunk calls.

The Ayrix validates if the time condition is “true” or “false” and executes the action. This time condition is set for inbound calls.

For example: to inform callers with an announcement, for shop opening times, or to forward callers outside opening hours to a security service. Or a meeting event.

The Time Condition validation is a 2^(nd) priority forwarding type, after day/night switch, before 3^(th) individual forwarding for the assigned trunk lines.

5.6 Intercom

The intercom menu is to provide intercom calls, to announce a voice message to a single or multiple phone user simultaneously in a single direction. Intercom is an auto answer and loud speaking function at the receiving extension. To get the Intercom working the telephones must support the intercom feature.

A typical application of the intercom is a chief-secretary or a doctor-nurse setup. The chief/ doctor has a Push2Talk or Intercom button on his Telephone. The secretaries/nurse telephone has the Intercom enabled to answer automatically via the loudspeaker. the chief announces: “next guest/patient please…”. The intercom windows consist of a list panel and a configuration panel.

5.6.1 Intercom button assignment

Intercom must be enabled. The telephones must support the intercom feature.

Check your feature list of your telephones. (SIP terminal features 3.4)

Example intercom receiving. SNOM 785advanced configuration settings for intercom receiving.

Example: line key settings for intercom sending. Intercom: setup or disconnect an intercom connection. Push2Talk: Establish the connection as long the Push2Talk key is pressed.

Important Notice: The following HW types are currently supported:

• Snom, Yealink, Mitel and Polycom

5.7 Hold

Hold is a centralized park function to park calls in the Ayrix. Call parking is a feature that allows a participant in a call to put the other participants on hold while they themselves hang up, and (for example) on another place retrieve the parked call by dialing the park number.

5.7.1 Pool and Park+Orbit

The telephone extension must support the park feature. Here an example for the line key settings. An example of the line key.

6 Day / Night

The Day / Night feature allows to redirect a set of DID numbers to the same specific destination. When a Day /Night template is enabled, in the list the lamp indicator on the web page as well when programmed on the terminal is switched on green.

The Day /Night module has the highest forwarding priority, above individual and group time conditions as well other forwarding’s.

Notice: In case multiple Day /Night templates are enabled, the first active template in the list ranking will be selected.

Switching from day to night status is set manually by enabling the template in the list or set on a telephone with a Busy Lamp Field (BLF) function key button. A listed panel of Day / Night templates and a vertical configuration panel are shown.

Field description

Day/Night BLF button assignment

Depending on the telephone features a Busy Lamp Field (BLF) can be assigned.

A named extension is added due assignment, for example. “dnLamp_1”. That name is the reference at name@IP_address_or_FQDN on BLF button assignment.

De-/activating with the button also switches the Day / Night lamp status at the webpage.

Example:

Day/ Night assignment on the 2^(nd). (BLF) button on a SNOM715 Telephone set. See also section: “SIP terminal Features”.

7 Routing

7.1 Trunk

To be able to make phone calls a SIP trunk should be registered on Ayrix.

Additionally, DID numbers have to be defined in order to receive incoming calls.

7.1.1 Adding a new SIP Trunk

To add a new SIP Trunk, click on "Add SIP Trunk" to open the Country/VoIP Provider window.

• Select the country where the VoIP provider operates
• Select the VoIP provider where the SIP Trunk that needs to be registered
• Click the "Add"-button

Now the "Basic Settings" window opens with the new created SIP Trunk.

7.1.2 Configuring the SIP Trunk

In the "Basic Settings" tab some values of the VoIP provider are already set. To complete and save the SIP Trunk data you must add some additional information. A window with Configuration settings panel and Codecs settings panel opens.

• Fill in a Default Caller ID.
• Enter SIP User-ID and the password, according to the providers Information.
• Click "Save"-button

The “Default Caller ID” may look like an inbound number, but in fact it is a recognition number (name it office number) to identify your Trunk by your provider. In the tab DiDs inbound numbers are configured.

The next tabs are shown:

• Basic Settings
• DiDs
• SIP Options
• Inbound headers
• Outbound headers

Basic Settings

This configuration windows contains settings to create and connect a SIP trunk.

Codecs

The right panel shows the Codecs window. The codecs you want to apply for the SIP Trunk has to be selected from the tab "Available Codecs". Various codecs can be selected from the list and moved in the selected Codec list.

The used codec is a grade of the speech quality. The use of a particular codec is negotiated by the SIP protocol due call setup. The first codec that matches on both sides is chosen. In the Selected Codecs you can rank the codecs. Selecting trunk codecs, enables also default codecs for extensions.

Codec list A Codec is the digital translation of the voice to transmit a media RTP or SRTP stream, A description of the available codec usage.

Direct inbound dialing (DiDs).

To connect with the outside parties, Direct inbound dialing (DID) numbers have to be added to the Trunk. These DiDs are the numbers you can dial in to address the destination. Go to the "DIDs" tab. A window with a listed panel and a configuration panel opens.

Press “+New”.

Select in the configuration panel your single number or a number block where the callers can dial in. You can add a single Number or a Number block.

Press “save”.

Your new number block is now listed and available for assignment.

SIP options

In this window can be set the sip options for the Trunk.

Field description

Inbound headers

Due to the different RFC specifications, many signaling options for incoming SIP trunk lines remain. These SIP elements are inbound headers. Headers, or SIP elements that differs from the standard are known by your provider. For each SIP Message element, you can adapt several other SIP identification elements. On the webpage several inbound elements are listed.

We strongly advice to leave the values to default unless a specialist from your provider advice different.

Outbound headers

A variety of outgoing SIP trunk signaling options following different RFCs remains possible. These call source elements depend on specific country and/or provider settings. For each SIP Message element, you can adapt several SIP identification elements. On the webpage several outbound elements are listed that can be changed.

We strongly advice to leave the values to default unless a specialist from your provider advice different. Depending on the type of SIP element there are 2 selectable lists [B&C]

Selectable parameters

7.1.3 Trunk Status

The Trunk list panel, the STATUS field shows you the actual trunk status of the configured trunk(s).

Field description

*Day of the week, MON, TUE, WED, THU, FRI, SAT, SUN

Registration of a trunk may take up a few minutes. Be patient, if the lamp does not immediately switch from Registering into Registered_[DAY] Refresh your website!

7.2 Inbound

Inbound rules route inbound calls to specified destination, for Example Queues, Ring groups, extensions and other endpoints. Inbound rules can be created based on DID numbers.

Adding new Inbound rule

To add a new inbound rule, click on "New" to display the form.

• Enter a name for the Inbound rule
• Choose one or more DID by clicking on "+". If any DID numbers are not available, read the SIP Trunk setup guide
• Optional: Enter one or two SIP Headers
• Optional: Enter a CID Prefix
• Optional: Enter a CID restriction
• Choose one or more destinations by clicking on "+". If no destinations are available, go to the Extensions setup guide
• Click the "Save"-button

Field description

7.3 Outbound

To define how the Ayrix routes outgoing calls, outbounds must be set up. Calls can be routed over different SIP Trunks, based on the dialed number or the length of the number. At least one outbound rule is necessary to make outbound calls on the Ayrix!

7.3.1 Adding a new Outbound rule

To add a new Outbound rule, click the "New"-button to display the form.

• Enter a name and choose one available Trunk sequence
• Optional: Add a Pinset
• Optional: Create a new Trunk sequence
• Click the "Save"-button

Trunk sequences can be created and edited after clicking or either "Create new Trunk sequence" or "Choose existing Trunk sequence".

To move Outbound rules and change their order click on the "Move up/down" Buttons. Outbound rules on the top will be executed before Outbound rules on the bottom.

7.3.2 Defining Sub-rules for an Outbound rule

To define specific rule, click on the tab "Rules" to display the form.

• Click on "Add CalledID rule"
• It is possible to add various sub-rules to the Outbound rules:
• Pattern: Enter a specific Pattern to trigger this Outbound rule, for example - "9*" - all numbers starting with 9 would use this SIP Trunk
• Prefix: Enter a digit which will be added in front of the number before calling out, for Example - "3" - when the caller dials "123456", "3123456" will be called instead
• Strip: Enter a digit for how many numbers should be stripped from the number dialed starting from the left side, for example - "1" - when the caller dials "123456", "23456" will be called instead

Once a "CalledID check" has been added, a "CallerID rule" can be added. Click on the "+" inside the "CalledID check".

• Here a Caller ID can be added
• Choose a Trunk sequence or an outbound rule to which this Caller ID should be routed

All the rules can be moved via "Drag & Drop" to change their order. Rules on the top will be executed before rules on the bottom.

7.4 Emergency Numbers

Emergency Numbers are country specific short dial numbers. Depending on the country selection in Trunk registration, the Ayrix creates a list of the local emergency numbers. These emergency numbers cannot be used for any other functions. You cannot edit or delete the default Emergency Numbers, however adding additional service numbers is possible. Some service numbers are using an unknown numbering plan. They can only be dialed within the country. For example, in Switzerland: the talking watch (161), or REGA -rescue services (1414).

Emergency numbers can you dial directly. These Emergency Numbers overrule all outbound trunk restrictions and chooses the first available free trunk. The format for custom Emergency Numbers is sent with a E164 Unknown” numbering plan.

The windows contain a search list panel and a configuration panel.

8 Records

The Records window offers you to prepare general announcements, as Voicemail Greetings or System recordings.

8.1 Announcements

The Records window offers you to create general announcements for the system and users. Records can be Voicemail Greetings and system records.

The announcement window shows a listed panel of all announcements and a configuration panel. The tabs add, edit, and Duplicate and a search bar, to search a phrase in the Name field, upper/lowercase allowed.

Voicemail Greetings is just a single record.

On “System Record” is a data-record to save announcements for multiple languages. First you create the data-record, then select the source file upload by browsing your local system or Record by Phone, you continue with upload by file or record selecting the extension.

By pressing *Record by phone” a pop-up window appears to select the extension for recording. A guidance leads you through the options.

These announcements can be used for various features.

9 PBX

9.1 Feature Code

The Feature Code list contains codes to enable and disable telephony features.

You can change these codes to a preferable code.

Depending on the function a feature code may contain a start “*” as suffix, meaning that this feature is updated. For example: *56*201

If a feature code function is initialized, then the feature code has only a star as prefix. *21567

There are also read only feature codes, these fixed codes are not listed in the feature code list. See Feature Code for a full overview of read only feature codes in the appendix.

9.2 Transfer Call back

The timeout value in seconds to return the caller to its originator in case of an unsuccessful blind transfer.

9.3 Speed Dialing

Speed Dialing enables you to create a general list of short dialing numbers to dial out extended numbers. This Speed Dialing list is valid for all users. Speed dialing with predefined numbers you can overrule outbound restrictions. For example: you want to call customers in foreign countries, while a general Geo CID restriction is set. Speed dial codes cannot overlap other dialing and outbound destinations.

With Tab

• import CSV, you can upload a list of Speed Dialing codes to the Ayrix
• CSV template, you can download the template list from the Ayrix to your local system

9.4 Music on hold

This MoH feature is to configure all music on hold sound files and radio streams. The window contains 3 tabs: MoH Groups, Sound files and Radio Streams.

9.4.1 MoH Groups

The window displays a list panel and a configuration panel.

In the configuration panel you can add a Sound File or a Radio stream. The files can be uploaded in tab Sound Files. Radio Streams can be added in the tab Radio Streams by adding a file or a stream - a pop-up window of the next tabs opens.

9.4.2 Sound files

Sound files is an upload window to upload file in. *.WAV or *.MP3. format.

9.4.3 Radiostreams

This entry configures live streams, or a radio streams to callers when in Music on Hold (MoH) state.

• Only HTTP-Radiostreams are supported without SSL-Encryption.
• Only MP3-Streams are supported (MPEG-1 / MPEG-2)
• Playlists are not support (M3U / PLS)

9.5 Global VoiceBox

The Global VoiceBox is to assign and manage general voice box recordings that are not assigned to a particular user. The global Voice box is intent for example disconnect states. For example, in a calendar state of an Extension. The Global VoiceBox is a general box. You can assign just one announcement and a voicemailbox.

The Global VoiceBox is a virtual extension. As there is no extension number specified, you can assign a DiD number to dial-in, so you are able listen to the recorded voicemails. If the Global VoiceBox is selected in a Callflow or used as Extension feature, a caller can leave a voicemail message.

In the window panel you can manage the voicemails in a same way as Extension voicemails.

The Global Voice Box window contain in the left panel the configuration settings, at the right panel the global overview of actual and archived voice mails.

9.6 Geo CID

Geo CID is a feature to create a rule for an Extension to send an individual caller ID (CID) based on E164 country and Region codes. See also Extensions menu - >Geo CID.

In the configuration panel, first the country must be added to the list. After country assignment the Region of that country can be assigned. The region is checked by the incoming digits after the country code and not only limited to area codes.

The window contains a listed panel and a configuration panel. The list panel contains three (“3”) tabs:

• New Region
• Edit
• New Country

Important notice:

In the list panel shows the assigned countries. Only the Country AND Region CIDs are shown in the Extensions menu when the extension has Geo_CID enabled. See Extensions, -> Geo CID, -> “Configure CIDs”

9.7 CTI

CTI is a general term for Computer Telephony Interface. Microsoft developed and released in 1993 the “Telephony Application Programming Interface” (TAPI) to interconnect between PBX and PC. That TAPI protocol became a standard. Other standards for CTI interconnections are CAPI, TSAPI, JTAPI, CSTA, e-CSTA ua-CSTA. LRMS. etc.

9.7.1 TAPI

Telephony Application Programming Interface (TAPI) is a standard interface to connect telephony with a server. A (Windows) TAPI server is responsible to handle the call flows between PBX and client. Microsoft released multi versions. TAPI version 1.x, 2.x and 3.x. TAPI 2.1 and 3.1 are still used today.

There are two types of TAPI interfaces. The 1st party TAPI interface communicates directly with an extension (Single line configuration). The 3rd party TAPI interface handles the telephones via the PBX (multi line configuration).

With the number of options in the TAPI specification, not all features might have been implemented by the manufacturer.

To connect the Ayrix with an TAPI application a TAPI server is required. That server talks to a TAPI driver or TAPI connector.

That tape connector connects the Ayrix IP interface port (5038). See Figure principle TAPI

Ayrix offers to use different TAPI applications. Multiple connections from different IP addresses are possible.

For hosted connections between TAPI clients (s) and Ayrix, a security module restricts unauthorized access on the TAPI port.

Principal TAPi server

The TAPI menu has a listed panel on the left and a configuration panel on the right. The listed panel shows the TAPI server connections. In the configuration panel you can assign a TAPI connection.

TAPI connector or TAPI driver installation

To connect the Ayrix with an TAPI application, or CTI client a TAPI server is required. Since Microsoft Windows XP, every Windows pc can used as TAPI server. Anno 2020 the most TAPI Installer files integrates a complete installation of your TAPI suite, Driver/connector, TAPI server and application. These modules are connected due installation as shown in figure: principle TAPI. Follow the instructions of your TAPI application supplier.

If you want to correct your configuration here is a general! guideline to check/install your TAPI connector/driver on Windows manually.

In Windows, go to Settings –> Control panel à “Phone and Modem options” -> tab: Enhanced, see picture check if your driver is loaded, if not

• Press “Add” to add the TAPI driver, a 2^(nd) window opens. The installed TAPI driver should be visible, select and press “Add” again.
• Back in the Main window press “Configure”, to add/change the connection, lines and extensions.
• Enter the IP address and port number of the Ayrix.
• Add TAPI lines and extensions.

The TAPI installation a server configuration depends on the installation. Refer to your application supplier for a detailed installation description.

10 Fax

10.1 Fax Extensions

Fax Extensions are special extensions used for fax machines. SIP uses a T38 protocol to communicates between SIP fax machines. These extensions negotiate with the opposite party to apply that T38 protocol.

The window has a list panel and a configuration panel.

Field description

10.2 Fax Server

The Fax Server is a Fax-to PDF-converter and send the converted document to an e-mail address. Conversion from email to fax is not possible. The window has a list panel and a configuration panel.

Field description

11 Phone book

The Phonebook is to create add, search, import CSV, telephone book entries. Phonebook entries can be selected between Business-, Mobile-, and Private numbers.

Import of a CSV File is possible. The import adds the CSV entries to the existing database. The 1^(st) line of the CSV is title line and is dropped due import.

To import new entries an empty CSV Template can be exported. See picture.

The window contains a listed panel and a configuration panel.

Listed panel: In the listed panel you can search add, edit, or duplicate. You can sort column contents in an ascending or descending order. and in the column the down selector button you can make columns visible/invisible.

Configuration panel

CSV Template export

Remark:

The full list is imported. Remove the header before importing.

The default column Type is: Business.

12 Security

12.1 Brute Force Protection

One of the major security modules of the Ayrix is the Brute Force Protection module. That module combines an Intrusion detection and an Intrusion and prevention.

The Brute Force detection checks the amount of the failed login attempts and/or the number of telephone registration attempts within a specified time. If the Ayrix receives too many attempts, it will block these Ip addresses to prevent further (unauthorized) access. After the dismiss time it will release these IP addresses.

The Brute Force Protection dismiss IP the address that belongs to the installation host

The Brute Force Protection consist of two tabs Brute Force and IP Black/whitelist

12.1.1 BruteForce

The BruteForce Module checks on telephone registration attempts.as well web access attempts within an elapsed time. Hackers are

Field description

12.1.2 IP Black/whitelist

The IP Black/whitelist is a security mechanism that blocks (blacklisted) traffic directly. The IP Whitelist allows explicit (whitelisted) traffic. The whitelist entries are not checked by the BruteForce mechanism. It can be selected between Web and/or Phone IP ports.

The IP address of the installation host is by default excluded from the blacklist. Text messaging might send via external servers

Field description

12.2 Number Blacklist

The Number_Blacklist is a call flow based on blocking inbound calling, outbound dialing, or both way numbers or a number range. With number registration we can block calls or forward these calls to an alternative destination.

The window consists of a list panel and a configuration panel.

With “Add” you can assign a number.

Anonymous calls “*****” cannot be blacklisted.

12.2 Geo Restrictions

With Geo restrictions you can enable geographical Phone Registering, Webinterface and Country Codes dialing access. The window contains three (“3”) tabs.

By default all regions are switched off (=restricted). This to avoid hacker attacks due Installation. Enable only country and regions where you are register phones, web access, or calling to.

12.3 SSL

Secure Sockets Layer (SSL) is a protocol for securing communication on the Internet. It provides a way to encrypt data before sending it to users. A public key exchange provides a way to send encrypted messages that are decrypted at the opposite site.

When SSL is enabled on a webpage, the URL will have an “https” prefix instead of an “http” prefix. Https uses a different port (see appendix).

SSL certificates are issued through Certificate Authorities (CAs), which are entities entrusted with selling and distributing SSL certificates. CA`s form the backbone of SSL, providing new certificates and verification of existing certificates. To make your web browser with Web encryption of the Ayrix publicly accessible to need to execute the following steps:

1. The first step is to choose a Domain name at a domain registrar, and you choose a CA. Fees may apply for both.
2. The next step is to generate in the tab “Request Certificate” a private key and the certificate signing request (CSR).
3. The generated information needs be sent to the CA,
4. CSRs contain information that will be used in the certificate such as the location of the organization, the domain name, and the email address of the administrator.
5. When the CA verifies the CSR, they will send the certificate along with several additional certificates. These additional certificates are known as intermediate certificates.
6. With the tab “Add Certificates” you install the private and private key, the SSL certificate as well the SSL Bundle
7. Once these certificates are installed, the server is SSL-ready,
8. Reload the webpage to access now the HTTPS://IP_addres_or_FQDN.

Request Certificate

That SSL service generates such a “self-signed” certificate Request (CSR).

The Administrator certificate signer is the 1est administrator registered in the User authorization.

Add Certificate

Most probably, your company have already valid certificates. by an earlier registration. In the Tab “Add Certificates” you can. This certificate data is downloaded to the operating system. Now the webserver has also access to that.

In an upcoming release VoIP security this Certificate generation can be applied for VoIP encryption (see SRTP).

Important security notice: Private keys should not be accessible in normal operation, As the public and private key is displayed on the web page, we strongly advice to restrict the SSL page access only to the IT Administrator(s). That can be restricted in the User Authorization settings.

What is CA bundle? CA bundle is a file that contains root and intermediate certificates. The end-entity certificate along with a CA bundle constitutes the certificate chain

The chain is required to improve compatibility of the certificates with web browsers and other kind of clients so that browsers recognize your certificate, and no security warnings appear.

Comodo may send you a complete CA bundle in a zip file with a *.ca-bundle extension or root and intermediate certificates separately.

12.5 User Authorization

The User Authorization menu offers you to register administrators and users with their authorization for telephony and system usage. Their username must be unique.

Per default the first administrator, “Admin” is present. That Admin is created due the license generation. That Admin cannot be removed.

The user authorization menu forms a two-step configuration. The window “Web User / Group”. contains 2 tabs. “Group” and “User”. In the “Group” tab you create, edit, and delete groups. In the *User “, tab you create, edit and delete users and add/change them to a group.

12.5.1 Group

The Group list-up the group names. With the tabs “Add” and “Edit” you can add, edit, and delete group names. As well a search bar entering a phase of a name. to search. Creating different groups enables to set specific permissions, for administrator and user groups. For example: access permissions

Permission In this window you enable or disable the permission per group. An administrator can configure for each group the user permission of all configuration modules. that allows the administrator to limit the module selection in such a way that telephony users can configure the features by themselves.

The listed panel is categorized by a tree of the main menu items and sub-modules.

The permissions are split up in read, write, and delete options. For every Module you can set group permissions.

The Administrators group have all permissions by default.

We strongly advice to leave the category “Dashboard” available for all groups. Restricting Dashboard permissions results in an immediate error message on login.

Users

The Users tab has two panels “Users in group” and “Available users”.

Here you can assign the users membership registration to a group. A user can have multiple memberships. That user has the sum of all Membership permissions.

Field description:

User

Here you can assign system “User” for system administration. The system administration and telephone user configuration are working independently,

The “User” window contains a listed panel and a configuration panel. The Web “User / Group” window contains list panel of the users and a configuration panel.

The listed pane contains three tabs: add edit duplicate, to add a user, to edit or duplicate user entries.

Be aware: in case of edit the user content. The configuration entry is Linked to the password entry. In case of a password change you can only delete and re-assign the user.

Field description

Pinset

With Pinset you can create default Pin-sets for Extension groups. The window shows a list panel and a configuration panel. The pin is used to listen to your recorded voicemails

The Pin for Conference Room access is set separately in the Conference Room configuration.

Configure Pinset Group

Field description

System

Network

The network window centralizes system settings regarding Ip network, routing, and connectivity.

The main window has 3 Tabs: Interface, Route and Connectivity and contains a listed panel and a configuration panel.

Interface

Due installation the Ayrix enables a single network interface. It might that you have multiple ethernet Interfaces. In this window you can and edit and change the network interface settings

of all available interfaces. For example: an IP address change without losing your license.

Field description

Housekeeping: As the administrator is logged in via web access. An IP address change might result in locked administrator access. There for a 2^(nd) admin access is useful. Otherwise, a system reset/restart releases the account.

Route

On the Route part you can assign static routing.to enforce traffic to use a different gateway.

Connectivity

Language, Region & Time

This Language, Region& Time window centralizes three (“3”) general system functions and contains 3 tabs. 1.) “Languages”, 2.) “Time Zone” and 3.) ” E164 Settings”.

The tab “Languages” lists and installs language packages.

The tab “+ install Language packages” to add additional language packs will become operational in an upcoming release.

Field description

The tab Time Zone, you can set the local time of your country as well.

E164

The tab E164 Settings sets your dialing preferences.

This table is also valid for multiple functions in the Ayrix.

These E164 preferences depend on the country and provider.

Field description

E164 Numbering scheme explanation.

Your provider only accepts calls with a correct address presentation in an E164 format.

To address calls, the SIP protocol contains a Uniform Resource Identifier (URI) schema.

A SIP address or URI is written in a similar fashion to an e-mail address: number@sip.inum.net.

The number part of the URI is formatted according an E164-formatted number scheme, issued by the iNUM initiative.

That E164 number scheme is a concatenation of:

International prefix symbol (according E123), country code, area code, company prefix and extension number.

For example: sip: +1-999-123-4567@voip-provider.example.net

In the above example: “+” (E123), “1” (country code, without leading zero`s), “999” (area code), “123” company prefix, “4567” (extension number)

Although the E164 format definition, the iNUM Initiative left some recommendations. Most important are:

The length of all these codes varies per country.

The address format can be International, - with or without leading (E123) notation, National or a private number format.

Resuming, depending on the requirements several combinations are possible, However, incorrectness results that calls are rejected.

E-mail

The E-mail menu is a window to configure e-mail sending on the SMTP server to inform users. The Email module is intent to forward e-mails to a company (internal) mail/exchange server. The E-mail window contains two (“2”) tabs: “Mail Server” and “Email Templates”. The “Mail Server” contains the SMTP server configuration settings.

If you want to receive messages from your Ayrix, for example from your voicemail with its attachment, make sure to configure this module with your SMTP Data.

Mail Server

The mail, or SMPT server is to send, E-mails, for Conference, Extension information, and voicemail attachment to Email recipients. In this window we configure the settings of the SMTP server.

Field description

[TABLE]

E-mail Template

The “Email Templates” window contains a list of predefined e-mail templates for conference invitations. It also contains a configuration panel to create, edit and modify these templates. There are system and custom templates. System templates cannot be removed. For example: templates for a conference invitation.

In the list panel you can see in the column “Modified” if a template was modified.

The content of the templates can be enhanced with XML based key word place holders.

Field description

XML place holders

To create your own template for a conference invitation you can write with XML place holders. In the Callflow menu –> “Conference” - > tab “Send invitation mail”

These place holders are field collected from the Conference Configuration (see chapter 4.41) and the calendar selection in the preparation template.

The following place holders are available:

Field description

Console

TFTP Server

The Trivial File Transfer Protocol (TFTP) and FTP server can be used to upload, store, and download files. This TFTP feature is implemented to offer a central storage for related files and a simple up-and download mechanism for extensions.

Files can be downloaded with assistance of a separate DHCP server. A DHCP request is sent on a start-up, or after a reset of the telephone (-client).

To execute a TFTP file download, you need to set DHCP option 66 (IP address of the Ayrix or FQDN) and option 67 (relevant download file) on the DHCP server.

A manual download of files to your local host is possible with the download button (=HTTP/HTTTPS download). To delete the File, select it from the list, click on the “delete” button in the configuration panel.

For example: Telephone book, Firmware etc.

This TFTP can be used to download for example phonebooks to extensions or a reset file to the extensions.

This window has a listed panel and configuration panel.

The listed panel shows you the uploaded files in the TFTP directory, enables you to search for a particular file.

The tab “Add”, enables in the configuration panel to add new files.

Field description

Notice: TFTP functionality is actual under development. In an upcoming release a full usage of the TFTP server will be available.

Housekeeping: The DHCP is out of scope. However, DHCP option 67 does not necessary mean that only 1 file will be downloaded. This may also contain a place holder, depending on the type of download client (-telephone). They will not only download a file but full directories. Depending the type of DHCP server, it may accept on option 66 also a combination of IP address and file name. So \\IP_adress\file_name.

Backup

The backup feature will become available in one of the upcoming releases

Update

The update feature will become available in one of the upcoming releases

License

On the License window you get an overview of the used licenses. If a license used up, with the green “add” button you can order additional licenses. The additional license will be shown below the screen.

User Access

If the License module is not restricted by the User Authorisation permission generally, the system user has also access to enhance licences.

That access can be also restricted when the licensed module permission is marked as read only.

Fundamentals:

A short interruption of the connection to the license server has no influence on the functionality of the Ayrix. However, after 72 hours a failed license check the system will be deactivated and taken offline to protect against misuse.

The license is committed to the hardware ID of the CPU and IP address. Please consider that the IP address of the server is not changed anymore.

[Picture License Model Overview]

License server (als.ayrix.ch)

• Prices and features are defined and parameterized in the license server

• The features are grouped into 3 categories

  • Instance Features

    • Charge per unit (price x quantity)

  • General Features

    • Charge per user (price x user)

  • Core Features

    • Are included in the license

• Features can be hidden, so that the end-customer does not see all features

• Upgrade packages are configured by the customer on the license server. The seller basically defines which packages the customer can see.

• Upgrade Type

  • Single upgrade

  • Packages upgrade

    • If no packages are defined, the customer cannot upgrade

The license bundle is divided by three (3) different license types:

General features:

These licenses are per created user.in the system.

extension, voicemail, announcement.

Instance licenses

These licenses are for feature functions: They are counted by function, ring group or queue

Core features:

These licenses are needed for general operation features of the PBX. These licenses are included in the license bundle and free of charge.

PBX license replacement:

The PBX license is tied to the system hardware. In case of hardware replacement contact your reseller for a license replacement.

Reboot

In some cases, it is necessary to reboot or shutdown the system. Connections and calls are lost.

Reboot

On a reboot you Reboot the operating system and initialize all Ayrix services. Call waiting queues will be re-initialized. IP address catch with the fail2ban in the BruteForce Protection will be cleared. (Dynamic agents for a Callflow queue will be automatically logged off. They need to logon manually after rebooting the system. The Ayrix PBX returns autonomously to normal operation. Reboot is password protected.

Shutdown

Will shut down and switch off the complete operating system. The shutdown is password protected.

Housekeeping: for example, on an attack, or a lockout

A reboot is just a re-initialization of all Ayrix services.

The BruteForce parameters are stored in the iptables. A reboot does not clear these values.

Troubleshooting

Activity log

Event log

Error log

Help

The Help feature will become available in one of the upcoming releases.

License Error

On an operational Ayrix every 72 hours a license check is executed. The license is based on CPU HW serial, the NIC HW serial number and IP address. A consequence of that mechanism is that an IP address change of the Network Interface card can only be changed via Ayrix web access, menu Network, Tab: Interface.

In a few situations it might happen that the license check fails. The phones remains operational, but web access is locked to the Register/Activate page.

In these cases contact our Customer service Center.

FAQ’s encrypted

Terms

Appendix Firewall

Appendix: Firewall port list

Housekeeping: Calendar = http/https port

License server = http/https port

SSL request= http/https port

SSH port = Access to the server

Specifications

Minimal hardware requirements

• CPU Family Intel Celeron, Intel i3 and Higher (64 Bit Intel processor)
• Number of vCPU’s 1
• Memory 1 GB
• Storage 20 GB available storage space,

we recommend on a on-premises system to apply a (software) RAID disk -see 1.52 Software RAID configuration.

• Connection Internet connection

Software RAID configuration

We recommend (physical) RAID disk installation for an extended reliability in operational lifetime. Consequently, a software RAID installation is a manual installation process.

1^(st). Bios settings:

In the Bios settings of the system, you must disable the UEFI settings and choose legacy: The Ayrix boot installer is limited to legacy mode.

Select your boot medium and boot priority, USB, or DVD.

2^(nd) Installation procedure:

Start the centos installation, interrupt* the automatic installation process.in the configuration window, by checking and set in the Configuration the Localization items, “Date & Time,” “Language Support”, “keyboard” and “Network &, Hostname”.

In the configuration go to “Installation destination”, to partition the disks.

• De-select the USB memory stick
• Select both physical disks for installation.
• Select the manual partitioning: “I will configure partitioning”
• Press Done, the Manual partitioning window open

*Notice: The installation starts without interruption automatically on a time out.

We recommend the following three partition sizes:

• /boot 512MB
• /swap 2048MB
• /(root) leave “Desired Capacity” empty and press Add mount point. The installer takes the remaining capacity of your disks.

Other mounting partitions left as is.

• Move to the left side and modify ALL three partitions.
• Change the “Device Type:” from “Standard Partition” to: “RAID”,
• Select Raid Level “RAID1”
• Press “Update Settings” to adapt RAID partitioning.

Label and File System, we leave as is.

• Press “Done”,
• A Summary of Changes window opens check and press “Accept Changes””
• Back in the main Configuration menu press “Begin Installation”.

Hardware specifications

Hardware compliance

Ayrix can be deployed as on-premises system or hosted system. Distributors may deliver a customized package of software only or as on-premises soft-and hardware delivery.

We contribute our distributors with the hardware selection. We verify the hardware manufacturing specifications against EU regulations. Swiss regulations are harmonized with the EU regulation standards.

Environmental operation (EU)

In accordance with:

IEC-EN60068-2-2 Dry heat and

IEC-EN60068-2-38 Temperature/humidity/

ISO7779 Noise

Energy efficiency operation of power supply and PC/server

(EU) 2019/424 PC/Server

(EU) 2019/1782 External power supplies

(EU) 617 /2013 Energy Efficiency operation.

In accordance with IEC-EN62301: Measurement of standby power

Electro Magnetic Compliance

In accordance with:

IEC-EN55032:2017_Emission Class A

IEC-EN55035:2017_Immunity

CE safety regulations.

In accordance

IEC-EN63000 :2016 RoHS compliant.

Software specifications:

Operating system of Ayrix.4.1.0.4 installer. Centos 7.7-1908 Kernel v. 3.10.0-1062.el7

Security upgrades (when available) are updated automatically.

System performance test.

As well on System performance with a traffic benchmark test. For traditional PBX (TDM) the following test sets are defined. On Traffic volumes measured in Erlang

• Extension to Extension
• Incoming traffic to Extension
• Outgoing traffic to trunk
• Busy hour call attempts

Service Grades

Extension to extension traffic: 1% congestion probability

Incoming line to extension: 1% congestion probability

outgoing line to extension: 1% congestion probability

While in TDM you can measure explicit the failed call (attempts), VoIP performance tests is different.

Voice quality has a correlation to traffic load (rising delays) and dropped packets (with UDP).

Test are executed according:

ITU T-REC-P.564-200711

Conformance testing for voice over IP transmission quality assessment models

ETSI TS 103 222-1 V1.2.1 (2018-03)

Speech and multimedia Transmission Quality (STQ); Reference benchmarking, background traffic profiles and KPIs.

Part 1: Reference benchmarking, background traffic profiles and KPIs for VoIP and FoIP in fixed network

The scope of the defined testing procedures is the evaluation of the network access by VoIP and FoIP fixed-network services.

Guaranteed Operating range

Minimal requirements

Deliverables

Warranty

Ayrix is delivered as a full software product as is. There are no warranties on the correct behavior, nor on its consequences of its use of the product, nor you can complain because of a system failure or outage of operating system and/or application.

Service

To grand the Ayrix operation an internet connection is required. The Ayrix checks every 24 hours if an update is available. The Ayrix will be updated automatically as an update package is available. The operating system will be updated if security patches are available. Updating might result to a reduction of operation or a disruption of operation.

Technical Support

We offer charged support services on the released versions of Ayrix PBX.

The support is at level best effort unless Service level agreements are agreed with a support contract with the supplier or distributor. Contact our customer service at support@ayrix.ch.

On beta releases there is no technical support. In case of a failure sent a bug report form via the bug bounty program on the Ayrix website under: www.ayrix.ch

Lifecycle

No lifecycle statement is given at the release of this product.

CentOS

End of lifetime Dates (EOL):

Last updated at 2020-10-24 08:40:14

For a complete specification refer to: https://wiki.centos.org/About/Product. EOL means “End of Life,” or when a version of a product is no longer supported. The definition of ‘no longer supported’ can range from no longer providing security updates, to no longer fixing bugs. It’s important to prepare to upgrade or migrate to newer versions before they become EOL. Fortunately, CentOS version have very long lifecycles, which gives us plenty of time to plan our major version upgrade paths.

Application

Ayrix is a software, developed as closed system. You are hereby granted a limited non-exclusive right to install. The package includes an open-source operating system.

Some modules are forked from open source. See also 16.4 Compliance statement.

Main releases are planned due regular intervals,

Hotfixes do contain only repairs.

Ayrix is available as ISO appliance file.

The ISO template is based on a BIOS based PC model.

Updates of Ayrix will be provided via internet.

Update availability will be checked every 24 hours by the system.

During the Maintenance updates phase, only Security errata and select mission critical bug fixes will be released

Housekeeping: Windows 10/2019 Hyper-V virtualization platform Generation I =Bios)/ Generation II=UEFI.

Compliance statement

Hardware Compliance

Ayris is a software as s service.

If your Ayrix is delivered as hardware, we verified its operation against the hardware standards mentioned in the chapter hardware specification.

Software Compliance

Ayrix, model, number, serial number

This software complies Innovation, science and economic development standards following two conditions: by compliance and copyrights & license rights.

Copyrights and License

Several pieces of free and open-source software have helped Ayrix to where it is today

A few require that we include their license agreements within our product. There for we publish the license terms of these modules.

Ayrix is a closed source project. The binary will be published as an all- inclusive distribution package.

CentOS Linux 7 EULA

CentOS Linux 7 comes with no guarantees or warranties of any sorts, either written or implied. The Distribution is released as GPLv2. Individual packages in the distribution come with their own licenses.

For a full context of Centos legal notices and licensing policy visit: https://centos.org/legal/

Digium Asterisk:

Some modules are forked from Asterisk. Asterisk provides developers with a free, open-source framework that simplifies the process of building communications systems. Digium has continued to sponsor Asterisk as an open source project since 1999, distributing it under the terms of the GNU GPL v2 license. For a full context of asterisk License agreement:

https://store.digium.com/view_agreement.php?id=4 Copyright OpenSSL [YEAR]

Fail2ban:

Fail2Ban is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version.

Fail2Ban is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY, without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details.

Mod-pike:

kamailio5-mod-pike. Kamailio is released under GNU Public License v2 (GPLv2). Beginning with 3.0.0, the application includes parts of code under BSD license that can be used such as individual components. See: https://www.kamailio.org/w/info/

Openssl:

Contents licensed under the terms of the OpenSSL license.

See https://www.openssl.org/source/license.html for details

Maxmind:

https://www.maxmind.com/en/end-user-license-agreement?sign=1

MaxMind End-User License (December 3, 2019)

Copyright of this manual:

Nothing from this manual may be duplicated or distributed without the prior author's permission.

Notes:

1. The license is generated once. As the UUID of the PBX can be always requested,

There is no need to save the license.

2. The number of extensions can be enhanced in the PABX itself. (or by Customer Support), It does not need any license update.

Thanks to

We would like to thank the team of author, co-authors, reviewers, graphic designer, and coordinator, for their contribution to create and completing this manual.

Author: Daniel Maier,. Etienne Stoffels

and our Customer Service

If you have any questions or suggestions regarding this manual, please write to the support team: support@Ayrix.ch

Appendix:

Feature Code list

There are read only feature codes, these fixed codes are not listed in the feature code list.

*8 = general call pickup

*8<number> = individual call pickup

*81 = pickup by rank

*55 = Agent spying: tapping or listening-in on an active party. For ex. Supervisors who checks the business quality of call Centre agents.

“[1..7.9] = enable/disable calendar entries

Icons fontawesome (INTERNAL)

https://fontawesome.com/

	Inbound
	InboundOutbound
	Extension/Fax Extension
	Emergency Numbers
	Security
	Fax
	Ringgroup
	Conferece
	Queue
	Time Condition
	Callflow
	Hold
	Intercom
	Day/Night
	BruteForce Protection
	TFTP Server/Routing
	Routing
	Records
	Announcements
	Network
	IVR
	PBX/Fax Server
	Transfer Callback
	Speed Dialing
	Global VocieBox
	Geo CID
	Feature Code
	Dashboard
	Global VoiceBox
	SSL
	Pinset/User Authorization
	System
	Email
	Geo Restrictions
	Reboot
	Trunk
	Music on Hold
	Phone Book
	Number Blacklist
	Language & Region
	License
	TAPI