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.

 

  1. 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).

 

Housekeeping: Deselecting the authorizations leads to an immediate blocking of the administrator account, which can only be repaired by direct access to the system’s database.

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

 

  1. 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 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)

 

  1. 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.

 

  1. 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 “(<)”.

 

  1. 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!

 

  1. 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.

 

 

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