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       <dc:date>2012-05-19T14:38:36+02:00</dc:date>
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        <title>Axeos Knowledge Base</title>
        <link>http://www.axeos.eu/wiki/</link>
        <url>http://www.axeos.eu/wiki/lib/images/favicon.ico</url>
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    <item rdf:about="http://www.axeos.eu/wiki/12connect?rev=1313925708&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-08-21T13:21:48+02:00</dc:date>
        <title>12Connect</title>
        <link>http://www.axeos.eu/wiki/12connect?rev=1313925708&amp;do=diff</link>
        <description>12Connect


You can configure your 12Connect SIP trunk as shown below. In this example we have used information:


account number: 65432#

inbound number: 0851234567

authentication: IP


Inbound

Go to menu option Dial plan / Inbound and add the phone number:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/12voip?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>12VoIP</title>
        <link>http://www.axeos.eu/wiki/12voip?rev=1279712682&amp;do=diff</link>
        <description>12VoIP


Logon to the Axeos IP PBX and go to menu option 'Dial Settings' and choose 'Outbound'. Edit or create a new trunk.


Use the login information provided by 12VoIP and complete the settings as shown in the example screenshots below:

 



Once the SIP trunk has been setup, go to menu option 'Dial Settings' and choose 'Inbound' to add the phonenumbers using the following format: 31201234567</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/3starsnet?rev=1334009735&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2012-04-10T00:15:35+02:00</dc:date>
        <title>3StarsNet</title>
        <link>http://www.axeos.eu/wiki/3starsnet?rev=1334009735&amp;do=diff</link>
        <description>3StarsNet


You can configure your 3StarsNet SIP trunk as shown below. In this example we have used information:


inbound number: 035009221

authentication: password


Inbound

Go to menu option Dial plan / Inbound and add the phone number:




Go to menu option Communication / Trunks and add the SIP trunk:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/a1_internet?rev=1281453702&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-08-10T17:21:42+02:00</dc:date>
        <title>A1 Internet</title>
        <link>http://www.axeos.eu/wiki/a1_internet?rev=1281453702&amp;do=diff</link>
        <description>A1 Internet


You will receive the following information from A1 Internet:


	*  Your phone number : for example 03210123456

	*  The (proxy) server: for example 87.233.30.100

	*  The password: for example afT@owjWK



Login to the Axeos PBX, go to menu options 'Dial Settings' and choose 'Outbound'. Create a new outbound trunk.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/aaatel?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>AAATel</title>
        <link>http://www.axeos.eu/wiki/aaatel?rev=1279712682&amp;do=diff</link>
        <description>AAATel


You will receive the following information from AAATel:


	*  Your phone number : for example 31201234567

	*  The (proxy) server: for example sip.aaatel.nl

	*  The username: for example 31201234567

	*  The password: for example bfT@Ecj2K</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/aastra?rev=1305309211&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-05-13T19:53:31+02:00</dc:date>
        <title>Aastra</title>
        <link>http://www.axeos.eu/wiki/aastra?rev=1305309211&amp;do=diff</link>
        <description>Aastra

default admin username: admin

default admin password: 22222</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/access_your_ip_phone_web_gui?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Access your IP Phone with browser</title>
        <link>http://www.axeos.eu/wiki/access_your_ip_phone_web_gui?rev=1279712682&amp;do=diff</link>
        <description>Access your IP Phone with browser


When your IP phone are connected via the LAN/bridge of the AOX and therefore are outside of reach of your desktop then you can do the following if you want to do simple maintenance of your phone. It is adviced to switch off the access by default to minimise risks.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/acd?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Automatic Call Distribution (ACD)</title>
        <link>http://www.axeos.eu/wiki/acd?rev=1279712682&amp;do=diff</link>
        <description>Automatic Call Distribution (ACD)


The ACD feature is available in Axeos PBX release 2.0 and above.


This function helps you to control all aspects of the ACD functionality and can be found as menu option ‘Services / Queues’. ACD functions may cause an extra load on the CPU of the server hardware, which is why (depending on usage requirements) this feature is not always suitable on the smallest AOX model. This feature can enable the PBX determine and communicate the queue position to the calle…</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/allow_dialout?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>allow_dialout</title>
        <link>http://www.axeos.eu/wiki/allow_dialout?rev=1279712682&amp;do=diff</link>
        <description>Er zit nog een leuke extra feature in de 2.0.2 versie. Als je een extensie aanvinkt met de nieuwe optie “allow dialout”. Dank kun je via de voicemail authenticatie de call forward van je toestel remote aanpassen, bijv. door schakelen naar je mobiel als je weet dat je niet meer op kantoor komt.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/atlantic?rev=1280273854&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-28T01:37:34+02:00</dc:date>
        <title>Atlantic Telecom</title>
        <link>http://www.axeos.eu/wiki/atlantic?rev=1280273854&amp;do=diff</link>
        <description>Atlantic Telecom


You will receive the following information from Atlantic Telecom:


	*  Your SIP URL: COMPANY@voip.atlantic.nl
	*  The password: for example afT@owjWK
	*  Your phone number : for example 31201234567


The SIP URL shows the hostname and the username in the following format: username@hostname.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/axeos_sip?rev=1279713605&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:00:05+02:00</dc:date>
        <title>Axeos Demo</title>
        <link>http://www.axeos.eu/wiki/axeos_sip?rev=1279713605&amp;do=diff</link>
        <description>Axeos Demo


The Axeos demo SIP trunk can be configured as following:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/backup_pbx_settings?rev=1280764051&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-08-02T17:47:31+02:00</dc:date>
        <title>Backup PBX Settings</title>
        <link>http://www.axeos.eu/wiki/backup_pbx_settings?rev=1280764051&amp;do=diff</link>
        <description>Backup PBX Settings


Axeos advices all administrators to store a backup of the (v)PBX configuration locally (for example on your PC or Backup device).


Manual Backup

It is possible to manually download a backup of the configuration files of the PBX.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/basic_cli_menu?rev=1279811583&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-22T17:13:03+02:00</dc:date>
        <title>Basic CLI Menu</title>
        <link>http://www.axeos.eu/wiki/basic_cli_menu?rev=1279811583&amp;do=diff</link>
        <description>Basic CLI Menu


Use an SSH client such as SSH Secure Shell to connect to the PBX on port 22.


Login by using your administrator username and password.


Enter the command: aox_console


Enter the command: menu


For most common commands you can simply enter the commands shown below:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/belcentrale?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Belcentrale</title>
        <link>http://www.axeos.eu/wiki/belcentrale?rev=1279712682&amp;do=diff</link>
        <description>Belcentrale


You will receive the following information from Belcentrale:


	*  Your phone number : for example 31201234567

	*  The (proxy) server: for example pbx12.belcentrale.nl

	*  The password: for example afT@owjWK



Login to the Axeos PBX, go to menu options 'Dial Settings' and choose 'Outbound'. Create a new outbound trunk.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/bizzoncall?rev=1281457322&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-08-10T18:22:02+02:00</dc:date>
        <title>BizzOnCall</title>
        <link>http://www.axeos.eu/wiki/bizzoncall?rev=1281457322&amp;do=diff</link>
        <description>BizzOnCall


BizzOnCall is the SIP service of BizzOnWeb. You will receive the following information from BizzOnWeb:


	*  Your phone number : for example 31851234567

	*  The (proxy) server: for example sip.bizzoncall.nl

	*  The password: for example afT@owjWK</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/breezz_sip?rev=1279713636&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:00:36+02:00</dc:date>
        <title>Breezz</title>
        <link>http://www.axeos.eu/wiki/breezz_sip?rev=1279713636&amp;do=diff</link>
        <description>Breezz


- Use the login information provided by Breezz and complete the settings as shown in the example screenshots below:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/bug_reports?rev=1279719992&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T15:46:32+02:00</dc:date>
        <title>Bug Reports</title>
        <link>http://www.axeos.eu/wiki/bug_reports?rev=1279719992&amp;do=diff</link>
        <description>Bug Reports


Please report any bugs to: 


Please include a detailed description of the bug, including:


	*  The error message displayed

	*  The actions taken shortly before the error occurred

	*  PBX Software Version (see menu 'System' and choose option 'Diagnostics')

	*  Your browser type and version, such as Firefox 3.5.9</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/caller_id_prefix?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Caller ID Prefix</title>
        <link>http://www.axeos.eu/wiki/caller_id_prefix?rev=1279712682&amp;do=diff</link>
        <description>Caller ID Prefix


It is possible to see what number the incoming caller has dialed by using a prefix.


For example, when you have a UK VoIP number and NL VoIP number you are able to instruct the PBX to put 'UK' in front of the callerID of the caller.



1.	Login to the vPBX/PBX

2.	Go to menu option ‘Incoming calls’

3.	select the number from the 'All DID' list and double click on it

4.	Enter the text you will use to identify the call (such as UK) in the ‘CID Prefix’ field

5.	Apply and Save …</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/conference_rooms?rev=1304976614&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-05-09T23:30:14+02:00</dc:date>
        <title>Conference rooms</title>
        <link>http://www.axeos.eu/wiki/conference_rooms?rev=1304976614&amp;do=diff</link>
        <description>Conference rooms


The conferencing room feature is a simple but powerful feature which allows you to create 

To create a new conference room, go to menu option Services, choose Conference room and click on the 'Add' button.





Once you have created your conference room it is accesable from every internal phone.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/deanconnect?rev=1280355357&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-29T00:15:57+02:00</dc:date>
        <title>DeanConnect</title>
        <link>http://www.axeos.eu/wiki/deanconnect?rev=1280355357&amp;do=diff</link>
        <description>DeanConnect


You will receive the following information from DeanConnect:


	*  The (proxy) server: sip.deanconnect.eu
	*  Your username: for example username-u001
	*  The password: for example afT@owjWK
	*  Your phone number : for example 0201234567
	*  Codecs: G711 (alaw), G729, T.38 (fax support)
	*  DTMF: RFC2833
	*  Caller id passing: P-Preferred-Identity</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/debugging_via_de_cli?rev=1302622432&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-04-12T17:33:52+02:00</dc:date>
        <title>Debugging via command line interface (CLI)</title>
        <link>http://www.axeos.eu/wiki/debugging_via_de_cli?rev=1302622432&amp;do=diff</link>
        <description>Debugging via command line interface (CLI)

Debugging system messages


Use any SSH client such as SSH Secure Shell to connect to the PBX on port 22.

Login by using your administrator username and password.


	*  command: tail -f syslog
	*  command: dmesg</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/directories?rev=1281015056&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-08-05T15:30:56+02:00</dc:date>
        <title>Central Phonebook/Directory</title>
        <link>http://www.axeos.eu/wiki/directories?rev=1281015056&amp;do=diff</link>
        <description>Central Phonebook/Directory


The PBX has a central phonebook which you can create using menu option Services / Directories.

Create a new directory and add your contacts manually, or upload a .csv file.

The contents of the .csv file should be in the following format:

firstname lastname, phonenumber



Example



Create a file called phonebook.csv

Open the file and enter the following details:



Axeos, 0031204160640

John Simmons, 0123456789

Bob Meldo, 02312345267



Upload the file using t…</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/disa?rev=1306322141&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-05-25T13:15:41+02:00</dc:date>
        <title>DISA</title>
        <link>http://www.axeos.eu/wiki/disa?rev=1306322141&amp;do=diff</link>
        <description>DISA


- To Setup - 

step 1: go to Dial Settings / Inbound and select the number you want to use.

Step 2: edit the number, go to Default action tab and select External Voicemail.

Step 3: click OK and Apply Changes.

Step 4: go to Phones / Extensions and edit the extension you want to use.

Step 5: go to the 'User' tab and ensure 'Allow Dialout' is enabled. Check your voicemail password.

Step 6: click OK and Apply Changes.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/enable_the_firewall?rev=1287094485&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-10-15T00:14:45+02:00</dc:date>
        <title>Firewall</title>
        <link>http://www.axeos.eu/wiki/enable_the_firewall?rev=1287094485&amp;do=diff</link>
        <description>Firewall


The PBX has an integrated software firewall, which helps you to limit access to the platform.


We strongly advice the administrator to make full use of this feature. Contact us if you have doubts on how to configure it correctly.


By factory default the firewall is configured to allow all traffic, which is required to perform the installation.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/enterprisevoip?rev=1315842072&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-09-12T17:41:12+02:00</dc:date>
        <title>Enterprise VoIP</title>
        <link>http://www.axeos.eu/wiki/enterprisevoip?rev=1315842072&amp;do=diff</link>
        <description>Enterprise VoIP


You can configure your Enterprise VoIP SIP trunk as shown below. In this example we have used information:


inbound number: 0201234567

authentication: Password


Inbound

Go to menu option Dial plan / Inbound and add the phone number:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/error_codes?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Error Codes</title>
        <link>http://www.axeos.eu/wiki/error_codes?rev=1279712682&amp;do=diff</link>
        <description>Error Codes


Most errors you will see are related to the GUI, and rarely effect the phone service itself.


It is however important to report any errors, which ensures we can resolve the source of the issue.


Error 3


Error 3 is shown when there is no specific error definition found.

Often error 3 is a browser issue and doesn't impact the phone services

Contact the Axeos support department and report after which action the error appeared.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/fax2email?rev=1280764333&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-08-02T17:52:13+02:00</dc:date>
        <title>Fax2email</title>
        <link>http://www.axeos.eu/wiki/fax2email?rev=1280764333&amp;do=diff</link>
        <description>Fax2email


The Axeos PBX is can receive faxes and delivers them as PDF to your email address.


To configure it you only need to take the following two simple steps:



1. Create softfax extension

Go to menu option 'Phone', 'Extensions', click on 'Add' and select 'Faxmodem extension' to create your softfax extension.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/fax_ata?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>FAX via ATA</title>
        <link>http://www.axeos.eu/wiki/fax_ata?rev=1279712682&amp;do=diff</link>
        <description>FAX via ATA


Traditional FAX machines can be connected to the PBX platform using a FAX converter. This device converts the traditional RJ-11 plug into a RJ-45 plug, which can be connected to the network linking it to the VoIP platform. The FAX number can be transferred (ported) to SIP, making the traditional FAX line redundant.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/feature_requests?rev=1291853427&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-12-09T01:10:27+02:00</dc:date>
        <title>Feature Requests</title>
        <link>http://www.axeos.eu/wiki/feature_requests?rev=1291853427&amp;do=diff</link>
        <description>Feature Requests


Every release new features and functions become available. Most features are the result of requests from our sales channels.


Please contact our sales team if you feel a feature is still missing and would add value to the platform.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/geen_voorloop_0?rev=1279713758&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:02:38+02:00</dc:date>
        <title>Automatic SIP Trunk Selection</title>
        <link>http://www.axeos.eu/wiki/geen_voorloop_0?rev=1279713758&amp;do=diff</link>
        <description>Automatic SIP Trunk Selection


By changing the Dial Plan it is possible for the PBX to add the leading number indicating which SIP trunk is used.

For example, there are various ways to dial a regional number in Amsterdam using trunk 0:


00201234567    - in regional format

00031201234567 - in international format

01234567       - in local format</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/getting_started_handleiding?rev=1279713771&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:02:51+02:00</dc:date>
        <title>AOX-319 Getting Started Guide (DUTCH)</title>
        <link>http://www.axeos.eu/wiki/getting_started_handleiding?rev=1279713771&amp;do=diff</link>
        <description>AOX-319 Getting Started Guide (DUTCH)


De Axeos telefooncentrales zijn voorzien van alle benodigde software, waardoor de configuratie eenvoudig is. De centrale zal voornamelijk beheerd worden door gebruik te maken van de browser en de netwerk verbinding. Echter is het ook mogelijk om via de console verbinding toegang te krijgen tot het systeem.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/gntel?rev=1279713776&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:02:56+02:00</dc:date>
        <title>gnTel</title>
        <link>http://www.axeos.eu/wiki/gntel?rev=1279713776&amp;do=diff</link>
        <description>gnTel


Go to menu option 'Dial Settings' and choose 'Outbound' to create a new SIP trunk.


   


Once the SIP trunk has been setup, go to menu option 'Dial Settings' and choose 'Inbound' to add the phonenumbers.


The numbers should be entered in the following format: 31201234567</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/grandstream?rev=1303331131&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-04-20T22:25:31+02:00</dc:date>
        <title>Grandstream</title>
        <link>http://www.axeos.eu/wiki/grandstream?rev=1303331131&amp;do=diff</link>
        <description>Grandstream


Grandstream IP Phones work well in combination with the Axeos PBX.

We have added the required settings for the Busy Lamp Field (BLF) support in Axeos IP PBX release 3.0.



On the phone enter the fysical phone ID, for example: 200 (don't use 200@IPADDRESS)</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/gsm_gateways?rev=1291941012&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-12-10T01:30:12+02:00</dc:date>
        <title>GSM Gateways</title>
        <link>http://www.axeos.eu/wiki/gsm_gateways?rev=1291941012&amp;do=diff</link>
        <description>GSM Gateways


By using a GSM gateway it is possible to specify mobile traffic to route via the GSM trunk, which often results in significant cost reductions. It is possible to connect GSM gateways (from vendors such as 2N or MCS) to the Axeos IP (v)PBX by taking the following steps:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/index?rev=1279712807&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:46:47+02:00</dc:date>
        <title>Knowledge base for public use – Index</title>
        <link>http://www.axeos.eu/wiki/index?rev=1279712807&amp;do=diff</link>
        <description></description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/installation_issues?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Installation issues</title>
        <link>http://www.axeos.eu/wiki/installation_issues?rev=1279712682&amp;do=diff</link>
        <description>Symptoms


One way traffic : caller can't hear other party


actions to troubleshoot: 


	*  Possibly caused by firewall settings. Check the firewall, ensure any SIP Helper or ALG is disabled
	*  make sure that you have disabled ICE or STUN on your IP phone
	*  Log on to the PBX and disable smart routing on the extension
	*  Log on to the PBX and verify the NAT status on the extension, switch to “yes”</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/ip_phones?rev=1321992415&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-11-22T21:06:55+02:00</dc:date>
        <title>IP Phones and phone usage</title>
        <link>http://www.axeos.eu/wiki/ip_phones?rev=1321992415&amp;do=diff</link>
        <description>IP Phones and phone usage

	*  Aastra
	*  Grandstream
	*  Linksys
	*  Polycom
	*  Siemens
	*  SNOM
	*  XRS

	*  Access your IP Phone with browser</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/isdn_configuration?rev=1285892277&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-10-01T02:17:57+02:00</dc:date>
        <title>ISDN Settings</title>
        <link>http://www.axeos.eu/wiki/isdn_configuration?rev=1285892277&amp;do=diff</link>
        <description>ISDN Settings


The Axeos IP PBX is also available with ISDN BRI or PRI. 

BRI / ISDN2 / NT1


	*  BRI has 2 ISDN channels. You can connect the PBX to NT-1/ISDN
	*  The IP PBX is by default in TE mode
	*  If you connect a single BRI (1 NT1) you should choose the PTMP setting
	*  If you connect a bundled BRI (multiple ISDN2 connections) you should choose the PTP setting
	*  Multiple ISDN2/BRI/NT1 are not always bundled, as they can also work separately from each other
	*  Only when you connect an…</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/linksys_spa922?rev=1279713782&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:03:02+02:00</dc:date>
        <title>Linksys</title>
        <link>http://www.axeos.eu/wiki/linksys_spa922?rev=1279713782&amp;do=diff</link>
        <description>Linksys


Model: SPA922


Follow the steps below to register the IP Phone on the Axeos PBX.

Go with your browser to the IP address of the phone, and click on 'Admin Login'.

Now click on 'Advanced' and go to 'Ext 1'



Scroll down to part ‘Proxy and Registration’.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/log_files?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Log Files</title>
        <link>http://www.axeos.eu/wiki/log_files?rev=1279712682&amp;do=diff</link>
        <description>Log Files


Take the following steps to download the error and log files from the PBX:



1. Login to your PBX using your administrator account

2. Go to menu option 'System' and choose 'Diagnostics'

3. Click on 'Generate report' followed by 'Download'

4. Save the logfiles to your desktop and open it (using an unzip tool such as WinZIP)

5. Open the files using your texteditor to view all information</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/modems_firewalls?rev=1284559505&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-09-15T16:05:05+02:00</dc:date>
        <title>Modems &amp; Firewalls</title>
        <link>http://www.axeos.eu/wiki/modems_firewalls?rev=1284559505&amp;do=diff</link>
        <description>Modems &amp; Firewalls

General


It may be required to open TCP port 5060 on your firewall for incoming/outgoing connections. UDP ports 10000 to 20000 also needs to be open.

Cisco


By default Cisco enables SIP ALG (i.e. sip helper to accommodate NAT issues). It makes things worse it must be disabled for a proper functioning PBX behind NAT. The same applies when using hosted.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/mottovoip?rev=1318260945&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-10-10T17:35:45+02:00</dc:date>
        <title>Motto VoIP</title>
        <link>http://www.axeos.eu/wiki/mottovoip?rev=1318260945&amp;do=diff</link>
        <description>Motto VoIP


You will receive the following information from Motto VoIP:


	*  Your phone number : for example 31201234567

	*  The (proxy) server: for example sip.mottovoip.nl (83.143.186.231)

	*  The password: for example sbQBwjWK



Login to the Axeos PBX, go to menu options 'Dial Settings' and choose 'Outbound'. Create a new outbound trunk.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/network_settings?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Network Settings</title>
        <link>http://www.axeos.eu/wiki/network_settings?rev=1279712682&amp;do=diff</link>
        <description>Network Settings


When a new PBX is connected to the LAN (using the first network port on the server) it will request an IP address from the DHCP server on the LAN.


You can verify the IP address in your gateway/router. If you don't have access to the device it is also possible to connect to the PBX using the console cable. Enter the 'menu' and choose 'showip' to verify what IP address has been allocated to the PBX.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/outlook_plug-in?rev=1293034590&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-12-22T17:16:30+02:00</dc:date>
        <title>Outlook Plug-in</title>
        <link>http://www.axeos.eu/wiki/outlook_plug-in?rev=1293034590&amp;do=diff</link>
        <description>Outlook Plug-in


The Outlook Plug-in supports calls from Outlook® and call functionality from other applications.

The Plug-ins are easy to install and very simple to use. The software enables users to initiate calls in various ways from Microsoft Office® applications. This can be achieved by using the contact details on your pc to instigate calls efficiently and accurately, with just two clicks!</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/password_reset?rev=1279719324&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T15:35:24+02:00</dc:date>
        <title>Administrator Password Changes</title>
        <link>http://www.axeos.eu/wiki/password_reset?rev=1279719324&amp;do=diff</link>
        <description>Administrator Password Changes


Logon to the PBX using your browser and the default username and password: admin / admin


Go to menu option 'System', 'User Manager', select the 'admin' user and click on 'Edit'.





Enter your new (strong) administrator password, click on 'Ok' and hit 'Save changes'.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/pbx_mac_addresses?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>PBX MAC Addresses</title>
        <link>http://www.axeos.eu/wiki/pbx_mac_addresses?rev=1279712682&amp;do=diff</link>
        <description>PBX MAC Addresses


Take the following steps if you need to find out what MAC addresses are on the PBX:



1. Login to your PBX using your administrator account

2. Go to menu option 'System' and choose 'Diagnostics'

3. Click on 'Generate report' followed by 'Download'

4. Save the logfiles to your desktop and open it (using an unzip tool such as WinZIP)

5. Open the 'diag' file using your texteditor (such as wordpad/notepad) and scroll to 'Network Interfaces'



You will see all MAC addresses …</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/pbx_user_language?rev=1291759900&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-12-07T23:11:40+02:00</dc:date>
        <title>PBX / User Language</title>
        <link>http://www.axeos.eu/wiki/pbx_user_language?rev=1291759900&amp;do=diff</link>
        <description>PBX / User Language


The PBX has two levels of language settings, at the:


	*  server level

	*  at the user level



The language setting affects voiceprompts such as the voicemail menu.

Avoid mixed language messages by keeping the server level language equal to the user level language.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/phone_codecs?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Codecs</title>
        <link>http://www.axeos.eu/wiki/phone_codecs?rev=1279712682&amp;do=diff</link>
        <description>Codecs


Codecs for Europe: a-law/alaw/malaw/g711a/711a and for USA: u-law/ulaw/mulaw/711u/g711u. Never use the “u” version.


Many phones have by default ulaw selected, this seriously degrades voice quality. As well it uses a large amount of CPU cycles.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/polycom?rev=1302622780&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-04-12T17:39:40+02:00</dc:date>
        <title>Polycom</title>
        <link>http://www.axeos.eu/wiki/polycom?rev=1302622780&amp;do=diff</link>
        <description>Polycom

IP6000


Obtain the IP address of the phone to be able to access the web interface. Press “Menu” on the phone and select “Status”.

From that menu select “Network” and then “TCP/IP Configuration Parameters”. You will be shown the IP of the phone. Use your web browser to navigate to this IP.
The interface will be shown. Default user name is Polycom, Password is 456



1. Click on the SIP tab and “Outbound Proxy”



In the “Servers” section, under “Outbound Proxy” configure the fields as …</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/quick_codes?rev=1320423175&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-11-04T17:12:55+02:00</dc:date>
        <title>Quick Codes</title>
        <link>http://www.axeos.eu/wiki/quick_codes?rev=1320423175&amp;do=diff</link>
        <description>Quick Codes


The Quick Codes allow users to easily instigate extension related actions.



Below we show an overview of the different codes available:



*70 – park extension, to park calls

*70 – enable call waiting (do not ring phone when on call)

*71 – disable call waiting

*72 – enable call forward: see below for more info

*73 - disable call forward: see below for more info

*74 allows to set CF for any user

*75 – call pickup

*77 – record a sound

*99 – play recorded sound

*78 – enable…</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/remote_call_forward?rev=1281438482&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-08-10T13:08:02+02:00</dc:date>
        <title>Remote Phone Forward</title>
        <link>http://www.axeos.eu/wiki/remote_call_forward?rev=1281438482&amp;do=diff</link>
        <description>Remote Phone Forward


From version 2.0.2 it is possible to remotely forward your phone extension.

You can connect in inbound number to 'external voicemail'. By entering your voicemail extension and password you get access to the voicemailsystem.



To use this function follow the steps below:

step 1 - link your 'external voicemail' to an inbound number at menu option 'Dial Settings / Inbound'.

step 2 - call the inbound number you've linked to the external voicemail.

step 3 - authenticate us…</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/remote_phone_reboot?rev=1279719658&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T15:40:58+02:00</dc:date>
        <title>Remote Phone Reboot</title>
        <link>http://www.axeos.eu/wiki/remote_phone_reboot?rev=1279719658&amp;do=diff</link>
        <description>Remote Phone Reboot


It is possible to remotely reboot an IP Phone by initiating it on the PBX.

Connect to the PBX using a SSH client and the administrator credentials.


Examples of SSH clients are: 


	*  Secure Shell Client : download
	*  Putty : download</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/secure_remote_axeos_support?rev=1285850796&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-09-30T14:46:36+02:00</dc:date>
        <title>Secure Remote Support</title>
        <link>http://www.axeos.eu/wiki/secure_remote_axeos_support?rev=1285850796&amp;do=diff</link>
        <description>Secure Remote Support


From you support team you may have received a support ticket. Installing this ticket allows the support staff to securely access your PBX.

This feature can only be used in combination with a remote support file provided by Axeos. Please contact us if you require a support file.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/service_provider_platform?rev=1313059319&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-08-11T12:41:59+02:00</dc:date>
        <title>Service Provider Platform (SPP)</title>
        <link>http://www.axeos.eu/wiki/service_provider_platform?rev=1313059319&amp;do=diff</link>
        <description>Service Provider Platform (SPP)


The Service Provider Platform (SPP) is an Operations Support System (OSS) that can easily be linked to your Business/Billing Support System (BSS) for provisioning and/or invoicing. Of course it works perfectly without the use of a BSS bringing you all the great help in managing the operations of your VoIP business. The SPP greatly reduces your OPEX when you are scaling up the Axeos PBX platform and Standalone PBX's. You can easily manage many servers that each h…</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/siemens?rev=1306351785&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-05-25T21:29:45+02:00</dc:date>
        <title>Siemens</title>
        <link>http://www.axeos.eu/wiki/siemens?rev=1306351785&amp;do=diff</link>
        <description>Siemens

Gigaset C470 IP &amp; Gigaset A580 IP


Phone Manual 470 IP: [download]

Phone Manual A580 IP: [download]


This Siemens DECT phone has been tested with the Axeos PBX.


The following settings were used:


 

Step 1: Login to the base station using your pin (default: 0000)</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/sip_doorphones?rev=1324561555&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-12-22T14:45:55+02:00</dc:date>
        <title>SIP Doorphones</title>
        <link>http://www.axeos.eu/wiki/sip_doorphones?rev=1324561555&amp;do=diff</link>
        <description>SIP Doorphones


There are various SIP doorphones available, two we have tested are made by 2N and Robin Telecom.

More information about the 2N doorphone van be found here

More information about Robin Compact SIP V2:

For over 20 years, the Robin Telecom Development door intercom system is known for its robust design, durability, and reliable performance in outdoor applications, in all weather conditions. During the design phase of the successor for the Robin Compact SIP V1, these principles w…</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/sip_providers?rev=1334009693&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2012-04-10T00:14:53+02:00</dc:date>
        <title>SIP Provider Settings</title>
        <link>http://www.axeos.eu/wiki/sip_providers?rev=1334009693&amp;do=diff</link>
        <description>SIP Provider Settings


The Axeos PBX is compatible with most SIP providers. Below are a number of example settings:



	*  12VoIP
	*  12Connect
	*  3StarsNet
	*  A1 Internet
	*  AAATel
	*  Atlantic Telecom
	*  Axeos Demo
	*  Belcentrale
	*  BizzOnCall
	*  DeanConnect
	*  Breezz
	*  Enterprise VoIP
	*  gnTel
	*  Motto VoIP
	*  SpeakUp
	*  Swiftnet
	*  Smart-Tel
	*  Solcon
	*  VoipBuster
	*  VoipQ
	*  VoipR
	*  Voys Telecom
	*  XeloQ</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/smart-tel?rev=1279713792&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:03:12+02:00</dc:date>
        <title>Smart-Tel</title>
        <link>http://www.axeos.eu/wiki/smart-tel?rev=1279713792&amp;do=diff</link>
        <description>Smart-Tel


- Use the login information provided by Smart-Tel and complete the settings as shown in the example screenshots below:



  


As IP Address / Hostname you can use: sip.sipnl.net</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/snom?rev=1322776378&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-12-01T22:52:58+02:00</dc:date>
        <title>SNOM</title>
        <link>http://www.axeos.eu/wiki/snom?rev=1322776378&amp;do=diff</link>
        <description>SNOM

	*  SNOM - How to change ringtone
	*  SNOM - auto provisioning (DUTCH)
	*  Snom M9

firmware update via provisioning


This function is recommended when you have many phones to update. Set it up once and the phones will automatically update on boot.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/snom_300_default_ringtone_aanpassen?rev=1302680077&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-04-13T09:34:37+02:00</dc:date>
        <title>SNOM - How to change ringtone</title>
        <link>http://www.axeos.eu/wiki/snom_300_default_ringtone_aanpassen?rev=1302680077&amp;do=diff</link>
        <description>SNOM - How to change ringtone


Take the following steps to change the default ringtone for a SNOM 300:


1. Login on the IP address of the phone using your browser

2. Go to menu option 'Identity 1' and change ringtone (see screenshot)




TIP: Ringtone 6 is a normal business ringtone</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/snom_dutch?rev=1302680106&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-04-13T09:35:06+02:00</dc:date>
        <title>SNOM - auto provisioning (DUTCH)</title>
        <link>http://www.axeos.eu/wiki/snom_dutch?rev=1302680106&amp;do=diff</link>
        <description>SNOM - auto provisioning (DUTCH)


Login op de PBX beheer omgeving en maak een nieuwe extensie/poort aan. Vul het MAC adres van de telefoon in, en selecteer op het model telefoon zodat de PBX de juiste configuratie zal aanleveren als de telefoon erom vraagt.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/snom_m9?rev=1322776794&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-12-01T22:59:54+02:00</dc:date>
        <title>Snom M9</title>
        <link>http://www.axeos.eu/wiki/snom_m9?rev=1322776794&amp;do=diff</link>
        <description>Snom M9


Each handset can be assigned up to nine SIP accounts (identities). When there is only one
handset per account, that handset will use the same account for incoming and outgoing calls. When
one handset has been assigned to multiple accounts, the same handset will ring whenever a call comes
in on any of the SIP accounts for incoming calls. For outgoing calls, the user will need to select the
Outgoing SIP account by using the “Allow Line Switching” setting (shown below), which allows users…</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/software_version?rev=1291941629&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-12-10T01:40:29+02:00</dc:date>
        <title>Software Version</title>
        <link>http://www.axeos.eu/wiki/software_version?rev=1291941629&amp;do=diff</link>
        <description>Software Version


To verify your (v)PBX software/firmware version, go to menu options: System / Options / Diagnostics</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/solcon_sip?rev=1279713852&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:04:12+02:00</dc:date>
        <title>Solcon</title>
        <link>http://www.axeos.eu/wiki/solcon_sip?rev=1279713852&amp;do=diff</link>
        <description>Solcon


- Use the login information provided by Solcon and complete the settings as shown in the example screenshots below:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/sound_files?rev=1287500486&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-10-19T17:01:26+02:00</dc:date>
        <title>Sound Files</title>
        <link>http://www.axeos.eu/wiki/sound_files?rev=1287500486&amp;do=diff</link>
        <description>Sound Files


The following sound file types are available:


Music On Hold



The Music On Hold files are randomly played on the server. There are two submenu options related to hold music:


- Upload sound

If you want to upload a new sound file, select the .mp3 sound file from your PC using the ‘Browse’ button. Click ‘Ok’ to start uploading the file to the server. 



Please note: The system will remove very high and low tones on the sound file because the speakers of phones will not be able …</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/speakup_sip?rev=1279713856&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:04:16+02:00</dc:date>
        <title>SpeakUp</title>
        <link>http://www.axeos.eu/wiki/speakup_sip?rev=1279713856&amp;do=diff</link>
        <description>SpeakUp


- Use the login information provided by SpeakUp and complete the settings as shown in the example screenshots below:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/start?rev=1324560020&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-12-22T14:20:20+02:00</dc:date>
        <title>Axeos Dealer Knowledge Base</title>
        <link>http://www.axeos.eu/wiki/start?rev=1324560020&amp;do=diff</link>
        <description>[Axeos VoIP PBX Wiki]
Dealers can login and click on the link below to access the secured knowledge base. 


	*  Dealer Area 


Contact our sales team to request your dealer account.


Welcome to the Axeos wiki. On these pages you find information about the Axeos IP PBX VoIP platform and its features and functions.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/swiftnet_sip?rev=1279713861&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:04:21+02:00</dc:date>
        <title>Swiftnet</title>
        <link>http://www.axeos.eu/wiki/swiftnet_sip?rev=1279713861&amp;do=diff</link>
        <description>Swiftnet


- Use the login information provided by Switftnet and complete the settings as shown in the example screenshots below:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/tapi?rev=1291852462&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-12-09T00:54:22+02:00</dc:date>
        <title>TAPI</title>
        <link>http://www.axeos.eu/wiki/tapi?rev=1291852462&amp;do=diff</link>
        <description>TAPI

On the PC / Laptop


The Axeos TAPI can be installed on your Windows PC or laptop. During this process configuration settings will be required.

The following screenshots indicates which settings are required:



In Axeos PBX Software Release 2 the following settings are required:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/time_rules?rev=1279713867&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:04:27+02:00</dc:date>
        <title>Time Rules</title>
        <link>http://www.axeos.eu/wiki/time_rules?rev=1279713867&amp;do=diff</link>
        <description>Time Rules


The time rules enable you to instruct the PBX to route incoming phone traffic based on time and dates.

The time rules show the system when the alternative action should be used. 

For example, this is often used for actions when the office is closed:</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/user_manager?rev=1282061942&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-08-17T18:19:02+02:00</dc:date>
        <title>User Manager</title>
        <link>http://www.axeos.eu/wiki/user_manager?rev=1282061942&amp;do=diff</link>
        <description>User Manager


This feature enables the administrator to manage all user accounts.

There are three levels: 

	*  End User
	*  Local Administrator
	*  Administrator


End User

The End User level allows users to manage settins specific to their account, such as voicemails and call forwarding.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/voicemail_menu?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>Voicemail Menu</title>
        <link>http://www.axeos.eu/wiki/voicemail_menu?rev=1279712682&amp;do=diff</link>
        <description>Voicemail Menu


You can enter *98 on your phone to start the voicemail menu.

* 1 voicemail messages

	*  3 Advanced messaging options 
											*  1 respond  
											*  2 call number back (only internal numbers)
											*  3 state date, time and duration of the message
											*  4 instigate an external call
											*  5 send a message</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/voip_acronyms?rev=1279749128&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T23:52:08+02:00</dc:date>
        <title>VoIP Acronyms</title>
        <link>http://www.axeos.eu/wiki/voip_acronyms?rev=1279749128&amp;do=diff</link>
        <description>VoIP Acronyms


Below is an alphabetized listing of some common VoIP related acronyms:


ACD

Automatic Call Distributor (ACD): Also known as Automated Call Distribution, is a function that distributes incoming calls to (a specific group of) agents. Using this function, the PBX is able to inform incoming calls about queue statistics such as the average waiting time.</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/voipbuster?rev=1279713875&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T14:04:35+02:00</dc:date>
        <title>voipbuster</title>
        <link>http://www.axeos.eu/wiki/voipbuster?rev=1279713875&amp;do=diff</link>
        <description></description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/voipq?rev=1279712682&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-07-21T13:44:42+02:00</dc:date>
        <title>voipq</title>
        <link>http://www.axeos.eu/wiki/voipq?rev=1279712682&amp;do=diff</link>
        <description></description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/voipr?rev=1281973391&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2010-08-16T17:43:11+02:00</dc:date>
        <title>VoipR</title>
        <link>http://www.axeos.eu/wiki/voipr?rev=1281973391&amp;do=diff</link>
        <description>VoipR


Under construction</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/voys?rev=1305298727&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-05-13T16:58:47+02:00</dc:date>
        <title>Voys Telecom</title>
        <link>http://www.axeos.eu/wiki/voys?rev=1305298727&amp;do=diff</link>
        <description>Voys Telecom


You will receive the following information from Voys:


	*  Your phone number : for example 31851234567
	*  The (proxy) server: for example sip.voipgrid.nl
	*  The password: for example afT@owjWK

	*  set register refresh to 100 seconds</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/xeloq_sip?rev=1295613437&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-01-21T13:37:17+02:00</dc:date>
        <title>XeloQ</title>
        <link>http://www.axeos.eu/wiki/xeloq_sip?rev=1295613437&amp;do=diff</link>
        <description>XeloQ


- Use the login information provided by XeloQ and complete the settings as shown in the example screenshots below:

 

Once the SIP trunk has been setup, go to menu option 'Dial Settings' and choose 'Inbound' to add the phonenumbers using the following format: 31201234567</description>
    </item>
    <item rdf:about="http://www.axeos.eu/wiki/xrs?rev=1322221175&amp;do=diff">
        <dc:format>text/html</dc:format>
        <dc:date>2011-11-25T12:39:35+02:00</dc:date>
        <title>XRS</title>
        <link>http://www.axeos.eu/wiki/xrs?rev=1322221175&amp;do=diff</link>
        <description>XRS


Model: XRS5660

TYPE: IP DECT Solution


Please note: before you connect the XRS phone, please update it to the latest firmware: &lt;http://xrs.nl/ondersteuning/firmware&gt;



To configure the XRS5660 model, simply create a new phone on your (v)PBX and follow the steps bellow:</description>
    </item>
</rdf:RDF>

